<div>Ryan</div><div> </div><div>Thank you again for responding and sharing your wisdom.</div><div> </div><div>I am unsure if I am doing the test to verify your thoughts correctly but I debugged the SIP messages once again. I placed a call and picked up the call with and without the MTP checkbox checked on the SIP trunk in cucm.</div>
<div> </div><div>I then did a search for the IP address of my IP phone 10.1.10.18.</div><div> </div><div>In both debugs with and without it checked I found my IP phone's IP address in the debug. I believe this might verify your idea that the MTP is not working correctly?</div>
<div> </div><div><strong>With MTP unchecked on the SIP TRUNk</strong><br>*Jan 16 18:33:28.384: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>Received:<br>ACK sip:17705439047@10.1.200.1:5060;transport=tcp SIP/2.0<br>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK4e47f806d<br>
From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10">sip:6784563290@10.1.80.10</a>>;tag=673~d8eefedd-7473-4e00-a4a0-ce8f65d30766-31145612<br>To: <<a href="mailto:sip%3A17705439047@10.1.200.1">sip:17705439047@10.1.200.1</a>>;tag=333141D8-1492<br>
Date: Wed, 16 Jan 2013 18:11:29 GMT<br>Call-ID: <a href="mailto:24e56400-f61ed51-12-a50010a@10.1.80.10">24e56400-f61ed51-12-a50010a@10.1.80.10</a><br>Max-Forwards: 70<br>CSeq: 101 ACK<br>Allow-Events: presence, kpml<br>Content-Type: application/sdp<br>
Content-Length: 230</div><div>v=0<br>o=CiscoSystemsCCM-SIP 673 1 IN IP4 10.1.80.10<br>s=SIP Call<br>c=IN IP4 10.1.10.18<br>b=TIAS:64000<br>b=AS:64<br>t=0 0<br>m=audio 20116 RTP/AVP 0 101<br>a=rtpmap:0 PCMU/8000<br>a=ptime:20<br>
a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15</div><div> </div><div><strong>With MTP checked on the SIP TRUNk</strong></div><div> </div><div>*Jan 16 18:39:09.732: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>Received:<br>
ACK sip:17705439047@10.1.200.1:5060;transport=tcp SIP/2.0<br>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK60b810f25<br>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10">sip:6784563290@10.1.80.10</a>>;tag=723~d8eefedd-7473-4e00-a4a0-ce8f65d30766-31145637<br>
To: <<a href="mailto:sip%3A17705439047@10.1.200.1">sip:17705439047@10.1.200.1</a>>;tag=3336625C-1CBB<br>Date: Wed, 16 Jan 2013 18:17:05 GMT<br>Call-ID: <a href="mailto:ed2aec00-f61eea1-16-a50010a@10.1.80.10">ed2aec00-f61eea1-16-a50010a@10.1.80.10</a><br>
Max-Forwards: 70<br>CSeq: 101 ACK<br>Allow-Events: presence, kpml<br>Content-Type: application/sdp<br>Content-Length: 230</div><div>v=0<br>o=CiscoSystemsCCM-SIP 723 2 IN IP4 10.1.80.10<br>s=SIP Call<br>c=IN IP4 10.1.10.18<br>
b=TIAS:64000<br>b=AS:64<br>t=0 0<br>m=audio 16452 RTP/AVP 0 101<br>a=rtpmap:0 PCMU/8000<br>a=ptime:20<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15</div><div><br><br> </div><div class="gmail_quote">On Wed, Jan 16, 2013 at 12:59 PM, Ryan Ratliff <span dir="ltr"><<a href="mailto:rratliff@cisco.com" target="_blank">rratliff@cisco.com</a>></span> wrote:<br>
<blockquote style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid" class="gmail_quote"><div style="word-wrap:break-word">What they are doing is interpreting us sending them an Invite with a=inactive in the SDP as the Cisco phone putting them on hold. That is a valid assumption. What is incorrect (IMO) is them assuming that they need to generate MOH. CUCM is the one initiating the hold, it will be the one to play MOH or not, based upon the way you configure it. When they respond to our inactive SDP with one of their own, CUCM sees that as them putting us on hold. The end result you see is that in order to get the call off of hold both sides need to resume it, which isn't happening. <div>
<br></div><div><div>I still think you need to look at an active call (no hold) on your CUBE to see where it's sending media to on the internal side. That IP address is going to be an MTP (CUCM server, hardware resource) or an IP phone. If it's directly to a phone you may as well remove "MTP Required" on the trunk because you're not actually allocating an MTP.</div>
<div><br><div>
<span style="text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;border-collapse:separate">-Ryan</span>
</div>
<br><div><div>On Jan 16, 2013, at 12:35 PM, Dane Newman <<a href="mailto:dane.newman@gmail.com" target="_blank">dane.newman@gmail.com</a>> wrote:</div><br><div>All</div><div><br>Thank you for the infromation you are providing me on this thread. It is a great learning exp for me.</div>
<div> </div><div>I just got off the phone with the ITSP and they confirmed the MOH was coming from them. They believe if I am typing this correctly they (ITSP) claim when I press the hold button I am sending an invite message and that is resulting in the MOH being played by them. </div>
<div> </div><div>I assumed when I pressed the hold key on an external call CUCM would continue to send the uninterupted audio stream with the MOH mixed in?</div><div> </div><div>I have reset the trunk and rebooted cucm also...</div>
<div> </div><div>Thanks again for the assistance and advice it's much appericated</div><div> </div><div>Dane</div><div><br><br> </div><div class="gmail_quote">On Wed, Jan 16, 2013 at 12:18 PM, Ryan Ratliff <span dir="ltr"><<a href="mailto:rratliff@cisco.com" target="_blank">rratliff@cisco.com</a>></span> wrote:<br>
<blockquote style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid" class="gmail_quote"><div style="word-wrap:break-word">Having the MOH servers registered is step 1 of about 10 that have to happen for MOH to be allocated for the call.<div>
In the SIP signaling you sent there was no possibility you heard MOH from CUCM because the media stream never went back to active after the hold. Can your Asterix play MOH? <br><div><br></div><div>You need to look at ccm traces to debug this further. If you can't figure it out, then it's time to call TAC.</div>
<div><br></div><div>You should also take a look at your active call before it's getting put on hold. You've got MTP Required set on the SIP trunk, but if an MTP was really getting allocated I don't believe we'd ever set the media inactive to the trunk, we'd be telling the MTP about media changes and the trunk would just see one media stream to the MTP for the entire call. At the same time if we tried to allocate an MTP but failed, that usually ends up disabling supplementary services for the call, which means you never would have been allowed to hold in the first place. It's certainly possible that has changed for SIP EO MTPs but for now what is in that signaling doesn't jive with what you've sent in your config and description of events. </div>
<div><br></div><div>Have you tried resetting the SIP trunk in CUCM yet?</div><div><br><div>
<span style="text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;border-collapse:separate">-Ryan</span>
</div>
<br><div><div>On Jan 16, 2013, at 11:26 AM, Dane Newman <<a href="mailto:dane.newman@gmail.com" target="_blank">dane.newman@gmail.com</a>> wrote:</div><br>Yes as per the screen shot the MOH servers are registered. How do In find the audio bit rate? its just the default moh file I didnt change any settings<br>
<br><div class="gmail_quote">On Wed, Jan 16, 2013 at 10:20 AM, Kenneth Hayes <span dir="ltr"><<a href="mailto:kennethwhayes@gmail.com" target="_blank">kennethwhayes@gmail.com</a>></span> wrote:<br>
<blockquote style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid" class="gmail_quote"><div dir="auto"><div>So have you looked in your media resources under music on hold server configurations to make sure it's registered to the right UCM? Also what audio bit rate is your MOH file?<br>
<br>Sent from my iPad</div><div><div><br>On Jan 16, 2013, at 10:14 AM, Nick Matthews <<a href="mailto:matthnick@gmail.com" target="_blank">matthnick@gmail.com</a>> wrote:<br><br></div><blockquote type="cite">
<div><div dir="ltr"><div>I'm not sure at this point, I'll let some of the CUCM experts comment. It's possible during the hold conversation CUCM always sends delayed offer, but I don't have some good traces in front of me to confirm. <br>
<br>You can check the original invite CUCM sends to see if there's SDP in that, and it would confirm the MTP is being allocated. If it's sending the INVITE without SDP, your MRG/MRGL or resources are misconfigured or in use.<br>
<br></div>-nick<br></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, Jan 15, 2013 at 8:39 PM, Dane Newman <span dir="ltr"><<a href="mailto:dane.newman@gmail.com" target="_blank">dane.newman@gmail.com</a>></span> wrote:<br>
<blockquote style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid" class="gmail_quote"><div>Nick</div><div><br>Thanks for the assistance.</div><div>
</div><div>This is the first time I am setting up a direct sip connection from cucm to cube. I am used to making it an h323 connection. Attached are screen shots of my trunk setup. MTP is checked off I believe already. Is there a best way to go about troubleshooting cucm to figure out why its not setting the stream back to active?<br>
<br></div><div><div class="gmail_quote">On Tue, Jan 15, 2013 at 7:56 PM, Nick Matthews <span dir="ltr"><<a href="mailto:matthnick@gmail.com" target="_blank">matthnick@gmail.com</a>></span> wrote:<br>
<blockquote style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid" class="gmail_quote">
<div dir="ltr"><div><div><div>It looks like CUCM isn't setting the stream back to active after putting it on hold. It sends the re-invite, and the 200 OK from the ITSP has the SDP continued with a=inactive.<br><br></div>
I don't have some good traces in front of me, but somewhere it needs to take that off. I don't think Asterisks is acting incorrectly by responding to a delayed offer INVITE that was previously a=inactive with a=inactive. <br>
<br></div>What's also odd is that CUCM is sending the exact same INVITE after the first one completes, for both the hold and the resume. The CSeq isn't increasing, which I would expect it to.<br><br></div>If you were to check 'MTP' required it may work around the problem, but I don't consider inserting MTP's to be a best practice.<span><font color="#888888"><br>
<br>-nick<br></font></span></div><div><div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, Jan 15, 2013 at 3:42 PM, Kenneth Hayes <span dir="ltr"><<a href="mailto:kennethwhayes@gmail.com" target="_blank">kennethwhayes@gmail.com</a>></span> wrote:<br>
<blockquote style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid" class="gmail_quote"><div dir="auto"><div>Bind your media and source to your outbound interface on your voice service voip.<br>
<br>Sent from my iPhone</div>
<div>
<div><br>On Jan 15, 2013, at 3:39 PM, Dane Newman <<a href="mailto:dane.newman@gmail.com" target="_blank">dane.newman@gmail.com</a>> wrote:<br><br></div><blockquote type="cite"><div>Below is a show run from the router</div>
<div>
</div><div> </div><div>[OK]<br>Cisco3825#sh run<br>Building configuration...</div><div><br>Current configuration : 10183 bytes<br>!<br>! Last configuration change at 20:49:24 UTC Tue Jan 15 2013 by dnewman<br>
version 15.1<br>service timestamps debug datetime msec<br>service timestamps log datetime msec<br>no service password-encryption<br>!<br>hostname Cisco3825<br>!<br>boot-start-marker<br>boot-end-marker<br>!<br>!<br>!<br>aaa new-model<br>
!<br>!<br>aaa authentication login default local<br>aaa authentication login vpnauth local<br>aaa authorization exec default local<br>aaa authorization network default local<br>aaa authorization network vpnauth local<br>
!<br>
!<br>!<br>!<br>!<br>aaa session-id common<br>!<br>no network-clock-participate wic 0<br>!<br>dot11 syslog<br>ip source-route<br>!<br>ip cef<br>!<br>!<br>!<br>!<br>ip domain name datasc.local<br>ip inspect udp idle-time 1800<br>
no ipv6 cef<br>!<br>multilink bundle-name authenticated<br>!<br>!<br>!<br>!<br>!<br>voice-card 0<br> dsp services dspfarm<br>!<br>!<br>!<br>voice service voip<br> ip address trusted list<br> ipv4 64.154.41.150 255.255.255.255<br>
allow-connections sip to sip<br> fax protocol pass-through g711ulaw<br> sip<br>!<br>!<br>!<br>!<br>voice translation-rule 1<br> rule <a href="tel:1%20%2F6784604564" target="_blank" value="+16784604564">1 /6784604564</a>/ /200/<br>
rule 2 /6784563290/ /100/<br> rule 3 /6784563291/ /101/<br> rule 4 /6784563292/ /102/<br>
rule 5 /6784563293/ /103/<br> rule 6 /6784563294/ /104/<br> rule 7 /6784563295/ /105/<br> rule 8 /6784563296/ /106/<br> rule 9 /6784563297/ /107/<br> rule 10 /6784563298/ /108/<br> rule <a href="tel:11%20%2F6784563299" target="_blank" value="+16784563299">11 /6784563299</a>/ /109/<br>
rule 12 /6784604565/ /125/<br>
!<br>!<br>voice translation-profile incomingdid<br> translate called 1<br>!<br>!<br>crypto pki token default removal timeout 0<br>!<br>crypto pki trustpoint selfsigned<br> enrollment selfsigned<br> subject-name CN=<a href="http://connect.datasc.com/" target="_blank">connect.datasc.com</a><br>
revocation-check crl<br>!<br>!<br>crypto pki certificate chain selfsigned<br> certificate self-signed 02<br> 30820251 308201BA A0030201 02020102 300D0609 2A864886 F70D0101 05050030<br> 44311B30 19060355 04031312 636F6E6E 6563742E 64617461 73632E63 6F6D3125<br>
30230609 2A864886 F70D0109 02161643 6973636F 33383235 2E646174 6173632E<br> 6C6F6361 6C301E17 0D313231 32323831 39313531 395A170D 32303031 30313030<br> 30303030 5A304431 1B301906 03550403 1312636F 6E6E6563 742E6461 74617363<br>
2E636F6D 31253023 06092A86 4886F70D 01090216 16436973 636F3338 32352E64<br> 61746173 632E6C6F 63616C30 819F300D 06092A86 4886F70D 01010105 0003818D<br> 00308189 02818100 D9A99B41 8B70C82F 28072967 376E13E8 8F7FC2C2 7729B93E<br>
DDAE31A0 F3613381 78B43E11 5144BE88 DC2FDE14 0035A104 0BBFAEA0 9A190598<br> 19A124B1 2C4A8EA2 04253BA1 C829EF07 CD0E848D E7AA5269 459449C4 FABF9CA9<br> BC5AF8ED 84FCD99B 260C2B75 57887863 7BB310FB 2C8D1506 FE91FEAC 4EDD1712<br>
A7AFD2C1 BF21C59D 02030100 01A35330 51300F06 03551D13 0101FF04 05300301<br> 01FF301F 0603551D 23041830 16801475 02C4FB04 4FB3F748 B4012EC5 8A571236<br> A190CB30 1D060355 1D0E0416 04147502 C4FB044F B3F748B4 012EC58A 571236A1<br>
90CB300D 06092A86 4886F70D 01010505 00038181 00C2B167 E583F6D8 8B742D4F<br> 49D27AAD 7EF4E64F 0B5CA5A3 944E8CC7 499A706F AB22283B AE5913A1 B22BBB20<br> E7CF6F9F 41CDD870 1B474E58 9537C1FA 2D93BE4F 4276E9CE 61AE18D3 EF724BD9<br>
33878860 4B3627C0 448C652D 03D4C142 BA3291A3 DDE0C4DD C6ED06C3 12E45933<br> F1EE5CC2 B5B6CC20 C65AB313 76966F14 AA25CC8D 2A<br> quit<br>!<br>!<br>license udi pid CISCO3825 sn FTX1237A1T0<br>username xxxxxxx privilege 15 secret xxxxxx<br>
!<br>redundancy<br>!<br>!<br>controller T1 0/0/0<br>!<br>controller T1 0/0/1<br>!<br>ip ssh version 2<br>!<br>!<br>crypto isakmp policy 10<br> encr aes<br> authentication pre-share<br> group 2<br>crypto isakmp key Recoil90 address 0.0.0.0 0.0.0.0<br>
crypto isakmp fragmentation<br>!<br>crypto isakmp client configuration group datasc<br> key Recoil90<br> dns 4.2.2.2 4.2.2.1<br> domain datasc.local<br> pool vpnpool<br> save-password<br>!<br>crypto isakmp client configuration group datascsplit<br>
key Recoil90<br> dns 4.2.2.2 4.2.2.1<br> domain datasc.local<br> pool vpnpool<br> acl 101<br> save-password<br>crypto isakmp profile datasc<br> match identity group datasc<br> client authentication list vpnauth<br> isakmp authorization list vpnauth<br>
client configuration address respond<br> virtual-template 1<br>crypto isakmp profile datascsplit<br> match identity group datascsplit<br> client authentication list vpnauth<br> isakmp authorization list vpnauth<br>
client configuration address respond<br> virtual-template 2<br>!<br>!<br>crypto ipsec transform-set transformset esp-aes<br>crypto ipsec transform-set ezvpntransform esp-aes esp-sha-hmac<br>!<br>crypto ipsec profile VTI<br>
set transform-set ezvpntransform<br> set isakmp-profile datasc<br>!<br>crypto ipsec profile VTI2<br> set transform-set ezvpntransform<br> set isakmp-profile datascsplit<br>!<br>!<br>!<br>!<br>!<br>!<br>!<br>interface Loopback1<br>
ip address 10.1.150.1 255.255.255.0<br> ip nat inside<br> ip virtual-reassembly in<br>!<br>interface GigabitEthernet0/0<br> ip address dhcp<br> no ip redirects<br> no ip unreachables<br> no ip proxy-arp<br> ip nat outside<br>
ip virtual-reassembly in<br> duplex auto<br> speed auto<br> media-type rj45<br> hold-queue 240000 in<br>!<br>interface GigabitEthernet0/1<br> ip address 10.1.200.1 255.255.255.252<br> ip nat inside<br> ip virtual-reassembly in<br>
duplex auto<br> speed auto<br> media-type rj45<br>!<br>interface Virtual-Template1 type tunnel<br> ip unnumbered GigabitEthernet0/0<br> ip nat inside<br> ip virtual-reassembly in<br> tunnel source GigabitEthernet0/0<br>
tunnel mode ipsec ipv4<br>
tunnel protection ipsec profile VTI<br>!<br>interface Virtual-Template2 type tunnel<br> ip unnumbered GigabitEthernet0/0<br> ip nat inside<br> ip virtual-reassembly in<br> tunnel source GigabitEthernet0/0<br> tunnel mode ipsec ipv4<br>
tunnel protection ipsec profile VTI2<br>!<br>interface Virtual-Template3<br> ip unnumbered GigabitEthernet0/0<br> ip nat outside<br> ip virtual-reassembly in<br> ip policy route-map anyconnecthop<br>!<br>!<br>router eigrp 1<br>
maximum-paths 1<br> network 10.0.0.0<br> redistribute static<br>!<br>ip local pool vpnpool 10.1.100.10 10.1.100.200<br>ip forward-protocol nd<br>ip http server<br>ip http secure-server<br>!<br>!<br>ip nat inside source list NATNETWORKS interface GigabitEthernet0/0 overload<br>
ip nat inside source static tcp 10.1.50.150 80 interface GigabitEthernet0/0 80<br>ip nat inside source static tcp 10.1.80.100 5001 interface GigabitEthernet0/0 5001<br>ip nat inside source static udp 10.1.80.100 5001 interface GigabitEthernet0/0 5001<br>
!<br>ip access-list extended NATNETWORKS<br> deny ip 10.0.0.0 0.255.255.255 172.16.0.0 0.15.255.255<br> deny ip 10.0.0.0 0.255.255.255 10.0.0.0 0.255.255.255<br> permit ip 10.0.0.0 0.255.255.255 any<br>ip access-list extended anyconnecthop<br>
deny ip 10.0.0.0 0.255.255.255 10.0.0.0 0.255.255.255<br> permit ip 10.0.0.0 0.255.255.255 any<br>!<br>access-list 50 permit 10.0.0.0 0.255.255.255<br>access-list 101 permit ip 10.0.0.0 0.255.255.255 any<br>!<br>!<br>
!<br>
!<br>route-map anyconnecthop permit 10<br> match ip address anyconnecthop<br> set ip next-hop 10.1.150.2<br>!<br>!<br>!<br>!<br>!<br>control-plane<br>!<br>!<br>!<br>!<br>mgcp profile default<br>!<br>!<br>dial-peer voice 100 voip<br>
description Publisher<br> preference 1<br> destination-pattern 1..<br> session protocol sipv2<br> session target ipv4:10.1.80.10<br> dtmf-relay rtp-nte<br> codec g711ulaw<br>!<br>dial-peer voice 101 voip<br> description Subscriber<br>
preference 2<br> destination-pattern 1..<br> session target ipv4:10.1.80.11<br> dtmf-relay rtp-nte<br> codec g711ulaw<br>!<br>dial-peer voice 200 voip<br> description Publisher<br> preference 1<br> destination-pattern 2..<br>
progress_ind setup enable 3<br> progress_ind progress enable 8<br> session protocol sipv2<br> session target ipv4:10.1.80.10<br> dtmf-relay rtp-nte<br> codec g711ulaw<br>!<br>dial-peer voice 300 voip<br> description incoming Calldid<br>
translation-profile incoming incomingdid<br> preference 1<br> session protocol sipv2<br> session target sip-server<br> incoming called-number 678456329.<br> dtmf-relay rtp-nte<br> codec g711ulaw<br>!<br>dial-peer voice 301 voip<br>
description incoming Calldid<br> translation-profile incoming incomingdid<br> preference 1<br> session protocol sipv2<br> session target sip-server<br> incoming called-number <a href="tel:6784604565" target="_blank" value="+16784604565">6784604565</a><br>
dtmf-relay rtp-nte<br> codec g711ulaw<br>
!<br>dial-peer voice 302 voip<br> description incoming Calldid<br> translation-profile incoming incomingdid<br> preference 1<br> session protocol sipv2<br> session target sip-server<br> incoming called-number <a href="tel:6784604564" target="_blank" value="+16784604564">6784604564</a><br>
dtmf-relay rtp-nte<br> codec g711ulaw<br>!<br>dial-peer voice 201 voip<br> description Publisher<br> preference 2<br> destination-pattern 2..<br> progress_ind setup enable 3<br> progress_ind progress enable 8<br> session protocol sipv2<br>
session target ipv4:10.1.80.11<br> dtmf-relay rtp-nte<br> codec g711ulaw<br>!<br>dial-peer voice 500 voip<br> description outgoing<br> preference 1<br> destination-pattern .T<br> session protocol sipv2<br> session target dns:<a href="http://sip.talkinip.net/" target="_blank">sip.talkinip.net</a><br>
dtmf-relay rtp-nte<br> codec g711ulaw<br>!<br>!<br>sip-ua<br> credentials username xxxxxxxx password 7 xxxxxxx realm <a href="http://sipconnect.ipcomms.net/" target="_blank">sipconnect.ipcomms.net</a><br> authentication username xxxxxx password 7 xxxxxxx<br>
authentication username xxxxx password 7 xxxxxxx realm <a href="http://sipconnect.ipcomms.net/" target="_blank">sipconnect.ipcomms.net</a><br> set pstn-cause 3 sip-status 486<br> set pstn-cause 34 sip-status 486<br> set pstn-cause 47 sip-status 486<br>
registrar dns:<a href="http://sipconnect.ipcomms.net/" target="_blank">sipconnect.ipcomms.net</a> expires 60<br> sip-server dns:<a href="http://sipconnect.ipcomms.net:5060/" target="_blank">sipconnect.ipcomms.net:5060</a><br>
!<br>!<br>!<br>gatekeeper<br>
shutdown<br>!<br>!<br>call-manager-fallback<br> max-conferences 8 gain -6<br> transfer-system full-consult<br> ip source-address 10.1.200.1 port 2000<br> max-ephones 20<br> max-dn 40<br>!<br>!<br>!<br>line con 0<br>line aux 0<br>
line vty 0 4<br> privilege level 15<br> transport input ssh<br>line vty 5 15<br> privilege level 15<br> transport input ssh<br>!<br>scheduler allocate 20000 1000<br>!<br>webvpn gateway gateway_1<br> ip interface GigabitEthernet0/0 port 443<br>
ssl trustpoint selfsigned<br> inservice<br> !<br>webvpn install svc flash:/webvpn/anyconnect-win-<a href="tel:3.1.02026" target="_blank" value="+13102026">3.1.02026</a>-k9.pkg sequence 1<br> !<br>webvpn context datasc<br>
title "DataSC VPN"<br> secondary-color white<br> title-color #CCCC66<br>
text-color black<br> ssl authenticate verify all<br> !<br> url-list "Servers"<br> heading "Server"<br> !<br> !<br> policy group datasc<br> url-list "Servers"<br> functions svc-enabled<br>
svc address-pool "vpnpool" netmask 255.255.255.0<br> svc keep-client-installed<br> svc dns-server primary 4.2.2.2<br> svc dtls<br> virtual-template 3<br> default-group-policy datasc<br> aaa authentication list vpnauth<br>
gateway gateway_1 domain datasc<br> inservice<br>!<br>!<br>webvpn context datascsplit<br> title "DataSC VPN Split"<br> secondary-color white<br> title-color #CCCC66<br> text-color black<br> ssl authenticate verify all<br>
!<br> url-list "Servers"<br> heading "Server"<br> !<br> !<br> policy group datascsplit<br> url-list "Servers"<br> functions svc-enabled<br> svc address-pool "vpnpool" netmask 255.255.255.0<br>
svc split include acl 50<br> svc dns-server primary 4.2.2.2<br> svc dtls<br> default-group-policy datascsplit<br> aaa authentication list vpnauth<br> gateway gateway_1 domain datascsplit<br> inservice<br>!<br>end</div>
<div>Cisco3825#<br><br></div><div class="gmail_quote">On Tue, Jan 15, 2013 at 3:31 PM, Kenneth Hayes <span dir="ltr"><<a href="mailto:kennethwhayes@gmail.com" target="_blank">kennethwhayes@gmail.com</a>></span> wrote:<br>
<blockquote style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid" class="gmail_quote"><div dir="auto"><div>What do your media resources look like?</div>
<div><br></div><div><br></div><div>Also can you show me a copy of your voice service voip config?<br>
<br>Sent from my iPad</div><div><div><br>On Jan 15, 2013, at 3:12 PM, Dane Newman <<a href="mailto:dane.newman@gmail.com" target="_blank">dane.newman@gmail.com</a>> wrote:<br><br></div><blockquote type="cite">
<div><div>Thanks Ryan</div>
<div> </div><div>I see I am always getting a 200 ok message after my invites from the debug</div><div> </div><div><span lang="EN"><strong>Putting a call on HOLD</strong></span></div><div><div> <br></div><p>*Jan 15 20:19:28.086: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:</p>
<p>Received: </p><p>INVITE sip:<a href="tel:17705439047" target="_blank" value="+17705439047">17705439047</a>@10.1.200.1:5060;transport=tcp SIP/2.0</p><p>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK2891b89f8fa</p><p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10" target="_blank">sip:6784563290@10.1.80.10</a>>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957</p>
<p>To: <<a href="mailto:sip%3A17705439047@10.1.200.1" target="_blank">sip:17705439047@10.1.200.1</a>>;tag=2E6BC6F0-1E22</p><p>Date: Tue, 15 Jan 2013 19:57:35 GMT</p><p>Call-ID: <a href="mailto:c22f9200-f51b49d-c5-a50010a@10.1.80.10" target="_blank">c22f9200-f51b49d-c5-a50010a@10.1.80.10</a></p>
<p>Supported: timer,resource-priority,replaces</p><p>Min-SE: 1800</p><p>User-Agent: Cisco-CUCM8.6</p><p>Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY</p><p>CSeq: 102 INVITE</p><p>
Max-Forwards: 70</p><p>Expires: 180</p><p>Allow-Events: presence</p><p>Supported: X-cisco-srtp-fallback</p><p>Supported: Geolocation</p><p>Session-Expires: 1800;refresher=uas</p><p>P-Asserted-Identity: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10" target="_blank">sip:6784563290@10.1.80.10</a>></p>
<p>Remote-Party-ID: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10" target="_blank">sip:6784563290@10.1.80.10</a>>;party=calling;screen=yes;privacy=off</p><p>Contact: <sip:<a href="tel:6784563290" target="_blank" value="+16784563290">6784563290</a>@10.1.80.10:5060;transport=tcp>;video</p>
<p>Content-Type: application/sdp</p><p>Content-Length: 240</p><p>v=0</p><p>o=CiscoSystemsCCM-SIP 7322 3 IN IP4 10.1.80.10</p><p>s=SIP Call</p><p>c=IN IP4 0.0.0.0</p><p>b=TIAS:64000</p><p>b=AS:64</p><p>t=0 0</p><p>m=audio 21476 RTP/AVP 0 101</p>
<p>a=rtpmap:0 PCMU/8000</p><p>a=ptime:20</p><p>a=inactive</p><p>a=rtpmap:101 telephone-event/8000</p><p>a=fmtp:101 0-15</p><p>*Jan 15 20:19:28.094: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p><p>Sent: </p><p>INVITE <a href="http://sip:17705439047@64.154.41.150:5060/" target="_blank">sip:17705439047@64.154.41.150:5060</a> SIP/2.0</p>
<p>Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK691F12E0</p><p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@sipconnect.ipcomms.net" target="_blank">sip:6784563290@sipconnect.ipcomms.net</a>>;tag=2E6BC0B0-2268</p>
<p>To: <<a href="mailto:sip%3A17705439047@sip.talkinip.net" target="_blank">sip:17705439047@sip.talkinip.net</a>>;tag=as7c5ff82e</p><p>Date: Tue, 15 Jan 2013 20:19:28 GMT</p><p>Call-ID: <a href="mailto:A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214" target="_blank">A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214</a></p>
<p>Supported: 100rel,timer,resource-priority,replaces,sdp-anat</p><p>Min-SE: 1800</p><p>Cisco-Guid: 3257897472-0000065536-0000000035-0173015306</p><p>User-Agent: Cisco-SIPGateway/IOS-12.x</p><p>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER</p>
<p>CSeq: 103 INVITE</p><p>Max-Forwards: 70</p><p>Timestamp: 1358281168</p><p>Contact: <<a href="http://sip:6784563290@98.192.104.214:5060/" target="_blank">sip:6784563290@98.192.104.214:5060</a>></p><p>Expires: 180</p>
<p>Allow-Events: telephone-event</p><p>Authorization: Digest username="<a href="tel:6784563290" target="_blank" value="+16784563290">6784563290</a>",realm="asterisk",uri="<a href="http://sip:17705439047@64.154.41.150:5060/" target="_blank">sip:17705439047@64.154.41.150:5060</a>",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5</p>
<p>Session-Expires: 1800;refresher=uas</p><p>Content-Type: application/sdp</p><p>Content-Length: 289</p><p>v=0</p><p>o=CiscoSystemsSIP-GW-UserAgent 3168 2739 IN IP4 98.192.104.214</p><p>s=SIP Call</p><p>c=IN IP4 98.192.104.214</p>
<p>t=0 0</p><p>m=audio 19458 RTP/AVP 0 101 19</p><p>c=IN IP4 98.192.104.214</p><p>a=inactive</p><p>a=rtpmap:0 PCMU/8000</p><p>a=rtpmap:101 telephone-event/8000</p><p>a=fmtp:101 0-15</p><p>a=rtpmap:19 CN/8000</p><p>a=ptime:20</p>
<p>*Jan 15 20:19:28.094: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p><p>Sent: </p><p>SIP/2.0 100 Trying</p><p>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK2891b89f8fa</p><p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10" target="_blank">sip:6784563290@10.1.80.10</a>>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957</p>
<p>To: <<a href="mailto:sip%3A17705439047@10.1.200.1" target="_blank">sip:17705439047@10.1.200.1</a>>;tag=2E6BC6F0-1E22</p><p>Date: Tue, 15 Jan 2013 20:19:28 GMT</p><p>Call-ID: <a href="mailto:c22f9200-f51b49d-c5-a50010a@10.1.80.10" target="_blank">c22f9200-f51b49d-c5-a50010a@10.1.80.10</a></p>
<p>CSeq: 102 INVITE</p><p>Allow-Events: telephone-event</p><p>Server: Cisco-SIPGateway/IOS-12.x</p><p>Content-Length: 0</p><div> <br></div><p>*Jan 15 20:19:28.110: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p><p>Received: </p>
<p>SIP/2.0 100 Trying</p><p>Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK691F12E0;received=98.192.104.214</p><p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@sipconnect.ipcomms.net" target="_blank">sip:6784563290@sipconnect.ipcomms.net</a>>;tag=2E6BC0B0-2268</p>
<p>To: <<a href="mailto:sip%3A17705439047@sip.talkinip.net" target="_blank">sip:17705439047@sip.talkinip.net</a>>;tag=as7c5ff82e</p><p>Call-ID: <a href="mailto:A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214" target="_blank">A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214</a></p>
<p>CSeq: 103 INVITE</p><p>Server: Asterisk PBX 1.6.2.13</p><p>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO</p><p>Supported: replaces, timer</p><p>Require: timer</p><p>Session-Expires: 1800;refresher=uas</p>
<p>Contact: <<a href="mailto:sip%3A17705439047@64.154.41.150" target="_blank">sip:17705439047@64.154.41.150</a>></p><p>Content-Length: 0</p><div> <br></div><p>*Jan 15 20:19:28.110: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p>
<p>Received: </p><p>SIP/2.0 200 OK</p><p>Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK691F12E0;received=98.192.104.214</p><p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@sipconnect.ipcomms.net" target="_blank">sip:6784563290@sipconnect.ipcomms.net</a>>;tag=2E6BC0B0-2268</p>
<p>To: <<a href="mailto:sip%3A17705439047@sip.talkinip.net" target="_blank">sip:17705439047@sip.talkinip.net</a>>;tag=as7c5ff82e</p><p>Call-ID: <a href="mailto:A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214" target="_blank">A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214</a></p>
<p>CSeq: 103 INVITE</p><p>Server: Asterisk PBX 1.6.2.13</p><p>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO</p><p>Supported: replaces, timer</p><p>Require: timer</p><p>Session-Expires: 1800;refresher=uas</p>
<p>Contact: <<a href="mailto:sip%3A17705439047@64.154.41.150" target="_blank">sip:17705439047@64.154.41.150</a>></p><p>Content-Type: application/sdp</p><p>Content-Length: 239</p><p>v=0</p><p>o=root 1685873050 1685873052 IN IP4 <a href="tel:64.154.41.150" target="_blank" value="+16415441150">64.154.41.150</a></p>
<p>s=Asterisk PBX 1.6.2.13</p><p>c=IN IP4 <a href="tel:64.154.41.150" target="_blank" value="+16415441150">64.154.41.150</a></p><p>t=0 0</p><p>m=audio 13014 RTP/AVP 0 101</p><p>a=rtpmap:0 PCMU/8000</p><p>a=rtpmap:101 telephone-event/8000</p>
<p>a=fmtp:101 0-16</p><p>a=ptime:20</p><p>a=inactive</p><p>*Jan 15 20:19:28.118: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p><p>Sent: </p><p>SIP/2.0 200 OK</p><p>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK2891b89f8fa</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10" target="_blank">sip:6784563290@10.1.80.10</a>>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957</p><p>To: <<a href="mailto:sip%3A17705439047@10.1.200.1" target="_blank">sip:17705439047@10.1.200.1</a>>;tag=2E6BC6F0-1E22</p>
<p>Date: Tue, 15 Jan 2013 20:19:28 GMT</p><p>Call-ID: <a href="mailto:c22f9200-f51b49d-c5-a50010a@10.1.80.10" target="_blank">c22f9200-f51b49d-c5-a50010a@10.1.80.10</a></p><p>CSeq: 102 INVITE</p><p>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER</p>
<p>Allow-Events: telephone-event</p><p>Remote-Party-ID: <<a href="mailto:sip%3A17705439047@10.1.200.1" target="_blank">sip:17705439047@10.1.200.1</a>>;party=called;screen=no;privacy=off</p><p>Contact: <sip:<a href="tel:17705439047" target="_blank" value="+17705439047">17705439047</a>@10.1.200.1:5060;transport=tcp></p>
<p>Supported: replaces</p><p>Supported: sdp-anat</p><p>Server: Cisco-SIPGateway/IOS-12.x</p><p>Session-Expires: 1800;refresher=uas</p><p>Require: timer</p><p>Supported: timer</p><p>Content-Type: application/sdp</p><p>Content-Length: 253</p>
<p>v=0</p><p>o=CiscoSystemsSIP-GW-UserAgent 4444 5479 IN IP4 10.1.200.1</p><p>s=SIP Call</p><p>c=IN IP4 10.1.200.1</p><p>t=0 0</p><p>m=audio 19514 RTP/AVP 0 101</p><p>c=IN IP4 10.1.200.1</p><p>a=inactive</p><p>a=rtpmap:0 PCMU/8000</p>
<p>a=rtpmap:101 telephone-event/8000</p><p>a=fmtp:101 0-16</p><p>a=ptime:20</p><p>*Jan 15 20:19:28.118: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p><p>Sent: </p><p>ACK <a href="http://sip:17705439047@64.154.41.150:5060/" target="_blank">sip:17705439047@64.154.41.150:5060</a> SIP/2.0</p>
<p>Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK6920266D</p><p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@sipconnect.ipcomms.net" target="_blank">sip:6784563290@sipconnect.ipcomms.net</a>>;tag=2E6BC0B0-2268</p>
<p>To: <<a href="mailto:sip%3A17705439047@sip.talkinip.net" target="_blank">sip:17705439047@sip.talkinip.net</a>>;tag=as7c5ff82e</p><p>Date: Tue, 15 Jan 2013 20:19:28 GMT</p><p>Call-ID: <a href="mailto:A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214" target="_blank">A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214</a></p>
<p>Max-Forwards: 70</p><p>CSeq: 103 ACK</p><p>Authorization: Digest username="<a href="tel:6784563290" target="_blank" value="+16784563290">6784563290</a>",realm="asterisk",uri="<a href="http://sip:17705439047@64.154.41.150:5060/" target="_blank">sip:17705439047@64.154.41.150:5060</a>",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5</p>
<p>Allow-Events: telephone-event</p><p>Content-Length: 0</p><div> <br></div><p>*Jan 15 20:19:28.122: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:</p><p>Received: </p><p>ACK sip:<a href="tel:17705439047" target="_blank" value="+17705439047">17705439047</a>@10.1.200.1:5060;transport=tcp SIP/2.0</p>
<p>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28b4b1305a0</p><p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10" target="_blank">sip:6784563290@10.1.80.10</a>>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957</p>
<p>To: <<a href="mailto:sip%3A17705439047@10.1.200.1" target="_blank">sip:17705439047@10.1.200.1</a>>;tag=2E6BC6F0-1E22</p><p>Date: Tue, 15 Jan 2013 19:57:35 GMT</p><p>Call-ID: <a href="mailto:c22f9200-f51b49d-c5-a50010a@10.1.80.10" target="_blank">c22f9200-f51b49d-c5-a50010a@10.1.80.10</a></p>
<p>Max-Forwards: 70</p><p>CSeq: 102 ACK</p><p>Allow-Events: presence</p><p>Content-Length: 0</p><div> <br></div><p>*Jan 15 20:19:28.122: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:</p><p>Received: </p><p>INVITE sip:<a href="tel:17705439047" target="_blank" value="+17705439047">17705439047</a>@10.1.200.1:5060;transport=tcp SIP/2.0</p>
<p>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28c43b47e7e</p><p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10" target="_blank">sip:6784563290@10.1.80.10</a>>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957</p>
<p>To: <<a href="mailto:sip%3A17705439047@10.1.200.1" target="_blank">sip:17705439047@10.1.200.1</a>>;tag=2E6BC6F0-1E22</p><p>Date: Tue, 15 Jan 2013 19:57:35 GMT</p><p>Call-ID: <a href="mailto:c22f9200-f51b49d-c5-a50010a@10.1.80.10" target="_blank">c22f9200-f51b49d-c5-a50010a@10.1.80.10</a></p>
<p>Supported: timer,resource-priority,replaces</p><p>Min-SE: 1800</p><p>User-Agent: Cisco-CUCM8.6</p><p>Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY</p><p>CSeq: 103 INVITE</p><p>
Max-Forwards: 70</p><p>Expires: 180</p><p>Allow-Events: presence</p><p>Supported: X-cisco-srtp-fallback</p><p>Supported: Geolocation</p><p>Session-Expires: 1800;refresher=uas</p><p>P-Asserted-Identity: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10" target="_blank">sip:6784563290@10.1.80.10</a>></p>
<p>Remote-Party-ID: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10" target="_blank">sip:6784563290@10.1.80.10</a>>;party=calling;screen=yes;privacy=off</p><p>Contact: <sip:<a href="tel:6784563290" target="_blank" value="+16784563290">6784563290</a>@10.1.80.10:5060;transport=tcp>;video</p>
<p>Content-Length: 0</p><div> <br></div><p>*Jan 15 20:19:28.126: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p><p>Sent: </p><p>INVITE <a href="http://sip:17705439047@64.154.41.150:5060/" target="_blank">sip:17705439047@64.154.41.150:5060</a> SIP/2.0</p>
<p>Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69211AB3</p><p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@sipconnect.ipcomms.net" target="_blank">sip:6784563290@sipconnect.ipcomms.net</a>>;tag=2E6BC0B0-2268</p>
<p>To: <<a href="mailto:sip%3A17705439047@sip.talkinip.net" target="_blank">sip:17705439047@sip.talkinip.net</a>>;tag=as7c5ff82e</p><p>Date: Tue, 15 Jan 2013 20:19:28 GMT</p><p>Call-ID: <a href="mailto:A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214" target="_blank">A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214</a></p>
<p>Supported: timer,resource-priority,replaces,sdp-anat</p><p>Min-SE: 1800</p><p>Cisco-Guid: 3257897472-0000065536-0000000035-0173015306</p><p>User-Agent: Cisco-SIPGateway/IOS-12.x</p><p>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER</p>
<p>CSeq: 104 INVITE</p><p>Max-Forwards: 70</p><p>Timestamp: 1358281168</p><p>Contact: <<a href="http://sip:6784563290@98.192.104.214:5060/" target="_blank">sip:6784563290@98.192.104.214:5060</a>></p><p>Expires: 180</p>
<p>Allow-Events: telephone-event</p><p>Authorization: Digest username="<a href="tel:6784563290" target="_blank" value="+16784563290">6784563290</a>",realm="asterisk",uri="<a href="http://sip:17705439047@64.154.41.150:5060/" target="_blank">sip:17705439047@64.154.41.150:5060</a>",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5</p>
<p>Session-Expires: 1800;refresher=uas</p><p>Content-Length: 0</p><div> <br></div><p>*Jan 15 20:19:28.126: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p><p>Sent: </p><p>SIP/2.0 100 Trying</p><p>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28c43b47e7e</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10" target="_blank">sip:6784563290@10.1.80.10</a>>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957</p><p>To: <<a href="mailto:sip%3A17705439047@10.1.200.1" target="_blank">sip:17705439047@10.1.200.1</a>>;tag=2E6BC6F0-1E22</p>
<p>Date: Tue, 15 Jan 2013 20:19:28 GMT</p><p>Call-ID: <a href="mailto:c22f9200-f51b49d-c5-a50010a@10.1.80.10" target="_blank">c22f9200-f51b49d-c5-a50010a@10.1.80.10</a></p><p>CSeq: 103 INVITE</p><p>Allow-Events: telephone-event</p>
<p>Server: Cisco-SIPGateway/IOS-12.x</p><p>Content-Length: 0</p><div> <br></div><p>*Jan 15 20:19:28.146: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p><p>Received: </p><p>SIP/2.0 100 Trying</p><p>Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69211AB3;received=98.192.104.214</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@sipconnect.ipcomms.net" target="_blank">sip:6784563290@sipconnect.ipcomms.net</a>>;tag=2E6BC0B0-2268</p><p>To: <<a href="mailto:sip%3A17705439047@sip.talkinip.net" target="_blank">sip:17705439047@sip.talkinip.net</a>>;tag=as7c5ff82e</p>
<p>Call-ID: <a href="mailto:A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214" target="_blank">A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214</a></p><p>CSeq: 104 INVITE</p><p>Server: Asterisk PBX 1.6.2.13</p><p>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO</p>
<p>Supported: replaces, timer</p><p>Require: timer</p><p>Session-Expires: 1800;refresher=uas</p><p>Contact: <<a href="mailto:sip%3A17705439047@64.154.41.150" target="_blank">sip:17705439047@64.154.41.150</a>></p><p>
Content-Length: 0</p><div> <br></div><p>*Jan 15 20:19:28.146: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p><p>Received: </p><p>SIP/2.0 200 OK</p><p>Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69211AB3;received=98.192.104.214</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@sipconnect.ipcomms.net" target="_blank">sip:6784563290@sipconnect.ipcomms.net</a>>;tag=2E6BC0B0-2268</p><p>To: <<a href="mailto:sip%3A17705439047@sip.talkinip.net" target="_blank">sip:17705439047@sip.talkinip.net</a>>;tag=as7c5ff82e</p>
<p>Call-ID: <a href="mailto:A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214" target="_blank">A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214</a></p><p>CSeq: 104 INVITE</p><p>Server: Asterisk PBX 1.6.2.13</p><p>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO</p>
<p>Supported: replaces, timer</p><p>Require: timer</p><p>Session-Expires: 1800;refresher=uas</p><p>Contact: <<a href="mailto:sip%3A17705439047@64.154.41.150" target="_blank">sip:17705439047@64.154.41.150</a>></p><p>
Content-Type: application/sdp</p><p>Content-Length: 333</p><p>v=0</p><p>o=root 1685873050 1685873053 IN IP4 <a href="tel:64.154.41.150" target="_blank" value="+16415441150">64.154.41.150</a></p><p>s=Asterisk PBX 1.6.2.13</p>
<p>c=IN IP4 <a href="tel:64.154.41.150" target="_blank" value="+16415441150">64.154.41.150</a></p><p>t=0 0</p><p>m=audio 13014 RTP/AVP 3 8 0 18 101</p><p>a=rtpmap:3 GSM/8000</p><p>a=rtpmap:8 PCMA/8000</p><p>a=rtpmap:0 PCMU/8000</p>
<p>a=rtpmap:18 G729/8000</p><p>a=fmtp:18 annexb=no</p><p>a=rtpmap:101 telephone-event/8000</p><p>a=fmtp:101 0-16</p><p>a=ptime:20</p><p>a=inactive</p><p>*Jan 15 20:19:28.150: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p>
<p>Sent: </p><p>SIP/2.0 200 OK</p><p>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28c43b47e7e</p><p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10" target="_blank">sip:6784563290@10.1.80.10</a>>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957</p>
<p>To: <<a href="mailto:sip%3A17705439047@10.1.200.1" target="_blank">sip:17705439047@10.1.200.1</a>>;tag=2E6BC6F0-1E22</p><p>Date: Tue, 15 Jan 2013 20:19:28 GMT</p><p>Call-ID: <a href="mailto:c22f9200-f51b49d-c5-a50010a@10.1.80.10" target="_blank">c22f9200-f51b49d-c5-a50010a@10.1.80.10</a></p>
<p>CSeq: 103 INVITE</p><p>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER</p><p>Allow-Events: telephone-event</p><p>Remote-Party-ID: <<a href="mailto:sip%3A17705439047@10.1.200.1" target="_blank">sip:17705439047@10.1.200.1</a>>;party=called;screen=no;privacy=off</p>
<p>Contact: <sip:<a href="tel:17705439047" target="_blank" value="+17705439047">17705439047</a>@10.1.200.1:5060;transport=tcp></p><p>Supported: replaces</p><p>Supported: sdp-anat</p><p>Server: Cisco-SIPGateway/IOS-12.x</p>
<p>Session-Expires: 1800;refresher=uas</p><p>Require: timer</p><p>Supported: timer</p><p>Content-Type: application/sdp</p><p>Content-Length: 277</p><p>v=0</p><p>o=CiscoSystemsSIP-GW-UserAgent 4444 5480 IN IP4 10.1.200.1</p>
<p>s=SIP Call</p><p>c=IN IP4 10.1.200.1</p><p>t=0 0</p><p>m=audio 19514 RTP/AVP 0 101 19</p><p>c=IN IP4 10.1.200.1</p><p>a=inactive</p><p>a=rtpmap:0 PCMU/8000</p><p>a=rtpmap:101 telephone-event/8000</p><p>a=fmtp:101 0-16</p>
<p>a=rtpmap:19 CN/8000</p><p>a=ptime:20</p><p>*Jan 15 20:19:28.162: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:</p><p>Received: </p><p>ACK sip:<a href="tel:17705439047" target="_blank" value="+17705439047">17705439047</a>@10.1.200.1:5060;transport=tcp SIP/2.0</p>
<p>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28d3eadaab3</p><p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10" target="_blank">sip:6784563290@10.1.80.10</a>>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957</p>
<p>To: <<a href="mailto:sip%3A17705439047@10.1.200.1" target="_blank">sip:17705439047@10.1.200.1</a>>;tag=2E6BC6F0-1E22</p><p>Date: Tue, 15 Jan 2013 19:57:35 GMT</p><p>Call-ID: <a href="mailto:c22f9200-f51b49d-c5-a50010a@10.1.80.10" target="_blank">c22f9200-f51b49d-c5-a50010a@10.1.80.10</a></p>
<p>Max-Forwards: 70</p><p>CSeq: 103 ACK</p><p>Allow-Events: presence</p><p>Content-Type: application/sdp</p><p>Content-Length: 209</p><p>v=0</p><p>o=CiscoSystemsCCM-SIP 7322 4 IN IP4 10.1.80.10</p><p>s=SIP Call</p><p>c=IN IP4 0.0.0.0</p>
<p>b=TIAS:64000</p><p>b=AS:64</p><p>t=0 0</p><p>m=audio 21476 RTP/AVP 0</p><p>a=X-cisco-media:nomedia</p><p>a=rtpmap:0 PCMU/8000</p><p>a=ptime:20</p><p>a=inactive</p><p>*Jan 15 20:19:28.170: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p>
<p>Sent: </p><p>ACK <a href="http://sip:17705439047@64.154.41.150:5060/" target="_blank">sip:17705439047@64.154.41.150:5060</a> SIP/2.0</p><p>Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK692226EA</p><p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@sipconnect.ipcomms.net" target="_blank">sip:6784563290@sipconnect.ipcomms.net</a>>;tag=2E6BC0B0-2268</p>
<p>To: <<a href="mailto:sip%3A17705439047@sip.talkinip.net" target="_blank">sip:17705439047@sip.talkinip.net</a>>;tag=as7c5ff82e</p><p>Date: Tue, 15 Jan 2013 20:19:28 GMT</p><p>Call-ID: <a href="mailto:A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214" target="_blank">A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214</a></p>
<p>Max-Forwards: 70</p><p>CSeq: 104 ACK</p><p>Authorization: Digest username="<a href="tel:6784563290" target="_blank" value="+16784563290">6784563290</a>",realm="asterisk",uri="<a href="http://sip:17705439047@64.154.41.150:5060/" target="_blank">sip:17705439047@64.154.41.150:5060</a>",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5</p>
<p>Allow-Events: telephone-event</p><p>Content-Type: application/sdp</p><p>Content-Length: 251</p><p>v=0</p><p>o=CiscoSystemsSIP-GW-UserAgent 3168 2740 IN IP4 98.192.104.214</p><p>s=SIP Call</p><p>c=IN IP4 0.0.0.0</p><p>t=0 0</p>
<p>m=audio 19458 RTP/AVP 0 101</p><p>c=IN IP4 0.0.0.0</p><p>a=inactive</p><p>a=rtpmap:0 PCMU/8000</p><p>a=rtpmap:101 telephone-event/8000</p><p>a=fmtp:101 0-16</p><p>a=ptime:20</p><div> <br></div><p><strong>Unholding the call the MOH continues on the previously held caller while the user hears nothing</strong></p>
<p><strong></strong> </p><div> <br></div><p>*Jan 15 20:19:35.166: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:</p><p>Received: </p><p>INVITE sip:<a href="tel:17705439047" target="_blank" value="+17705439047">17705439047</a>@10.1.200.1:5060;transport=tcp SIP/2.0</p>
<p>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28e6157f41c</p><p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10" target="_blank">sip:6784563290@10.1.80.10</a>>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957</p>
<p>To: <<a href="mailto:sip%3A17705439047@10.1.200.1" target="_blank">sip:17705439047@10.1.200.1</a>>;tag=2E6BC6F0-1E22</p><p>Date: Tue, 15 Jan 2013 19:57:42 GMT</p><p>Call-ID: <a href="mailto:c22f9200-f51b49d-c5-a50010a@10.1.80.10" target="_blank">c22f9200-f51b49d-c5-a50010a@10.1.80.10</a></p>
<p>Supported: timer,resource-priority,replaces</p><p>Min-SE: 1800</p><p>User-Agent: Cisco-CUCM8.6</p><p>Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY</p><p>CSeq: 104 INVITE</p><p>
Max-Forwards: 70</p><p>Expires: 180</p><p>Allow-Events: presence</p><p>Supported: X-cisco-srtp-fallback</p><p>Supported: Geolocation</p><p>Session-Expires: 1800;refresher=uas</p><p>P-Asserted-Identity: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10" target="_blank">sip:6784563290@10.1.80.10</a>></p>
<p>Remote-Party-ID: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10" target="_blank">sip:6784563290@10.1.80.10</a>>;party=calling;screen=yes;privacy=off</p><p>Contact: <sip:<a href="tel:6784563290" target="_blank" value="+16784563290">6784563290</a>@10.1.80.10:5060;transport=tcp>;video;audio;video</p>
<p>Content-Length: 0</p><div> <br></div><p>*Jan 15 20:19:35.170: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p><p>Sent: </p><p>INVITE <a href="http://sip:17705439047@64.154.41.150:5060/" target="_blank">sip:17705439047@64.154.41.150:5060</a> SIP/2.0</p>
<p>Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69232672</p><p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@sipconnect.ipcomms.net" target="_blank">sip:6784563290@sipconnect.ipcomms.net</a>>;tag=2E6BC0B0-2268</p>
<p>To: <<a href="mailto:sip%3A17705439047@sip.talkinip.net" target="_blank">sip:17705439047@sip.talkinip.net</a>>;tag=as7c5ff82e</p><p>Date: Tue, 15 Jan 2013 20:19:35 GMT</p><p>Call-ID: <a href="mailto:A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214" target="_blank">A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214</a></p>
<p>Supported: timer,resource-priority,replaces,sdp-anat</p><p>Min-SE: 1800</p><p>Cisco-Guid: 3257897472-0000065536-0000000035-0173015306</p><p>User-Agent: Cisco-SIPGateway/IOS-12.x</p><p>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER</p>
<p>CSeq: 105 INVITE</p><p>Max-Forwards: 70</p><p>Timestamp: 1358281175</p><p>Contact: <<a href="http://sip:6784563290@98.192.104.214:5060/" target="_blank">sip:6784563290@98.192.104.214:5060</a>></p><p>Expires: 180</p>
<p>Allow-Events: telephone-event</p><p>Authorization: Digest username="<a href="tel:6784563290" target="_blank" value="+16784563290">6784563290</a>",realm="asterisk",uri="<a href="http://sip:17705439047@64.154.41.150:5060/" target="_blank">sip:17705439047@64.154.41.150:5060</a>",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5</p>
<p>Session-Expires: 1800;refresher=uas</p><p>Content-Length: 0</p><div> <br></div><p>*Jan 15 20:19:35.190: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p><p>Sent: </p><p>SIP/2.0 100 Trying</p><p>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28e6157f41c</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10" target="_blank">sip:6784563290@10.1.80.10</a>>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957</p><p>To: <<a href="mailto:sip%3A17705439047@10.1.200.1" target="_blank">sip:17705439047@10.1.200.1</a>>;tag=2E6BC6F0-1E22</p>
<p>Date: Tue, 15 Jan 2013 20:19:35 GMT</p><p>Call-ID: <a href="mailto:c22f9200-f51b49d-c5-a50010a@10.1.80.10" target="_blank">c22f9200-f51b49d-c5-a50010a@10.1.80.10</a></p><p>CSeq: 104 INVITE</p><p>Allow-Events: telephone-event</p>
<p>Server: Cisco-SIPGateway/IOS-12.x</p><p>Content-Length: 0</p><div> <br></div><p>*Jan 15 20:19:35.194: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p><p>Received: </p><p>SIP/2.0 100 Trying</p><p>Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69232672;received=98.192.104.214</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@sipconnect.ipcomms.net" target="_blank">sip:6784563290@sipconnect.ipcomms.net</a>>;tag=2E6BC0B0-2268</p><p>To: <<a href="mailto:sip%3A17705439047@sip.talkinip.net" target="_blank">sip:17705439047@sip.talkinip.net</a>>;tag=as7c5ff82e</p>
<p>Call-ID: <a href="mailto:A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214" target="_blank">A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214</a></p><p>CSeq: 105 INVITE</p><p>Server: Asterisk PBX 1.6.2.13</p><p>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO</p>
<p>Supported: replaces, timer</p><p>Require: timer</p><p>Session-Expires: 1800;refresher=uas</p><p>Contact: <<a href="mailto:sip%3A17705439047@64.154.41.150" target="_blank">sip:17705439047@64.154.41.150</a>></p><p>
Content-Length: 0</p><div> <br></div><p>*Jan 15 20:19:35.194: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p><p>Received: </p><p>SIP/2.0 200 OK</p><p>Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69232672;received=98.192.104.214</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@sipconnect.ipcomms.net" target="_blank">sip:6784563290@sipconnect.ipcomms.net</a>>;tag=2E6BC0B0-2268</p><p>To: <<a href="mailto:sip%3A17705439047@sip.talkinip.net" target="_blank">sip:17705439047@sip.talkinip.net</a>>;tag=as7c5ff82e</p>
<p>Call-ID: <a href="mailto:A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214" target="_blank">A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214</a></p><p>CSeq: 105 INVITE</p><p>Server: Asterisk PBX 1.6.2.13</p><p>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO</p>
<p>Supported: replaces, timer</p><p>Require: timer</p><p>Session-Expires: 1800;refresher=uas</p><p>Contact: <<a href="mailto:sip%3A17705439047@64.154.41.150" target="_blank">sip:17705439047@64.154.41.150</a>></p><p>
Content-Type: application/sdp</p><p>Content-Length: 333</p><p>v=0</p><p>o=root 1685873050 1685873054 IN IP4 <a href="tel:64.154.41.150" target="_blank" value="+16415441150">64.154.41.150</a></p><p>s=Asterisk PBX 1.6.2.13</p>
<p>c=IN IP4 <a href="tel:64.154.41.150" target="_blank" value="+16415441150">64.154.41.150</a></p><p>t=0 0</p><p>m=audio 13014 RTP/AVP 3 8 0 18 101</p><p>a=rtpmap:3 GSM/8000</p><p>a=rtpmap:8 PCMA/8000</p><p>a=rtpmap:0 PCMU/8000</p>
<p>a=rtpmap:18 G729/8000</p><p>a=fmtp:18 annexb=no</p><p>a=rtpmap:101 telephone-event/8000</p><p>a=fmtp:101 0-16</p><p>a=ptime:20</p><p>a=inactive</p><p>*Jan 15 20:19:35.198: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p>
<p>Sent: </p><p>SIP/2.0 200 OK</p><p>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28e6157f41c</p><p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10" target="_blank">sip:6784563290@10.1.80.10</a>>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957</p>
<p>To: <<a href="mailto:sip%3A17705439047@10.1.200.1" target="_blank">sip:17705439047@10.1.200.1</a>>;tag=2E6BC6F0-1E22</p><p>Date: Tue, 15 Jan 2013 20:19:35 GMT</p><p>Call-ID: <a href="mailto:c22f9200-f51b49d-c5-a50010a@10.1.80.10" target="_blank">c22f9200-f51b49d-c5-a50010a@10.1.80.10</a></p>
<p>CSeq: 104 INVITE</p><p>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER</p><p>Allow-Events: telephone-event</p><p>Remote-Party-ID: <<a href="mailto:sip%3A17705439047@10.1.200.1" target="_blank">sip:17705439047@10.1.200.1</a>>;party=called;screen=no;privacy=off</p>
<p>Contact: <sip:<a href="tel:17705439047" target="_blank" value="+17705439047">17705439047</a>@10.1.200.1:5060;transport=tcp></p><p>Supported: replaces</p><p>Supported: sdp-anat</p><p>Server: Cisco-SIPGateway/IOS-12.x</p>
<p>Session-Expires: 1800;refresher=uas</p><p>Require: timer</p><p>Supported: timer</p><p>Content-Type: application/sdp</p><p>Content-Length: 277</p><p>v=0</p><p>o=CiscoSystemsSIP-GW-UserAgent 4444 5481 IN IP4 10.1.200.1</p>
<p>s=SIP Call</p><p>c=IN IP4 10.1.200.1</p><p>t=0 0</p><p>m=audio 19514 RTP/AVP 0 101 19</p><p>c=IN IP4 10.1.200.1</p><p>a=inactive</p><p>a=rtpmap:0 PCMU/8000</p><p>a=rtpmap:101 telephone-event/8000</p><p>a=fmtp:101 0-16</p>
<p>a=rtpmap:19 CN/8000</p><p>a=ptime:20</p><p>*Jan 15 20:19:35.206: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:</p><p>Received: </p><p>ACK sip:<a href="tel:17705439047" target="_blank" value="+17705439047">17705439047</a>@10.1.200.1:5060;transport=tcp SIP/2.0</p>
<p>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28f6dca6616</p><p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10" target="_blank">sip:6784563290@10.1.80.10</a>>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957</p>
<p>To: <<a href="mailto:sip%3A17705439047@10.1.200.1" target="_blank">sip:17705439047@10.1.200.1</a>>;tag=2E6BC6F0-1E22</p><p>Date: Tue, 15 Jan 2013 19:57:42 GMT</p><p>Call-ID: <a href="mailto:c22f9200-f51b49d-c5-a50010a@10.1.80.10" target="_blank">c22f9200-f51b49d-c5-a50010a@10.1.80.10</a></p>
<p>Max-Forwards: 70</p><p>CSeq: 104 ACK</p><p>Allow-Events: presence, kpml</p><p>Content-Type: application/sdp</p><p>Content-Length: 243</p><p>v=0</p><p>o=CiscoSystemsCCM-SIP 7322 5 IN IP4 10.1.80.10</p><p>s=SIP Call</p><p>
c=IN IP4 10.1.10.18</p><p>b=TIAS:64000</p><p>b=AS:64</p><p>t=0 0</p><p>m=audio 21476 RTP/AVP 0 101</p><p>a=rtpmap:0 PCMU/8000</p><p>a=ptime:20</p><p>a=inactive</p><p>a=rtpmap:101 telephone-event/8000</p><p>a=fmtp:101 0-15</p>
<p>*Jan 15 20:19:35.210: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p><p>Sent: </p><p>ACK <a href="http://sip:17705439047@64.154.41.150:5060/" target="_blank">sip:17705439047@64.154.41.150:5060</a> SIP/2.0</p><p>Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69246AB</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@sipconnect.ipcomms.net" target="_blank">sip:6784563290@sipconnect.ipcomms.net</a>>;tag=2E6BC0B0-2268</p><p>To: <<a href="mailto:sip%3A17705439047@sip.talkinip.net" target="_blank">sip:17705439047@sip.talkinip.net</a>>;tag=as7c5ff82e</p>
<p>Date: Tue, 15 Jan 2013 20:19:35 GMT</p><p>Call-ID: <a href="mailto:A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214" target="_blank">A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214</a></p><p>Max-Forwards: 70</p><p>
CSeq: 105 ACK</p><p>Authorization: Digest username="<a href="tel:6784563290" target="_blank" value="+16784563290">6784563290</a>",realm="asterisk",uri="<a href="http://sip:17705439047@64.154.41.150:5060/" target="_blank">sip:17705439047@64.154.41.150:5060</a>",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5</p>
<p>Allow-Events: telephone-event</p><p>Content-Type: application/sdp</p><p>Content-Length: 265</p><p>v=0</p><p>o=CiscoSystemsSIP-GW-UserAgent 3168 2741 IN IP4 98.192.104.214</p><p>s=SIP Call</p><p>c=IN IP4 98.192.104.214</p>
<p>t=0 0</p><p>m=audio 19458 RTP/AVP 0 101</p><p>c=IN IP4 98.192.104.214</p><p>a=inactive</p><p>a=rtpmap:0 PCMU/8000</p><p>a=rtpmap:101 telephone-event/8000</p><p>a=fmtp:101 0-16</p><p>a=ptime:20</p><p>Cisco3825#</p><p>Cisco3825#</p>
<div> <br></div><p>Cisco3825#</p><span lang="EN"></span></div><div><div> <br></div><p>INVITE sip:<a href="tel:17705439047" target="_blank" value="+17705439047">17705439047</a>@10.1.200.1:5060;transport=tcp SIP/2.0</p><p>
Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28e6157f41c</p><p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10" target="_blank">sip:6784563290@10.1.80.10</a>>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957</p>
<p>To: <<a href="mailto:sip%3A17705439047@10.1.200.1" target="_blank">sip:17705439047@10.1.200.1</a>>;tag=2E6BC6F0-1E22</p><p>Date: Tue, 15 Jan 2013 19:57:42 GMT</p><p>Call-ID: <a href="mailto:c22f9200-f51b49d-c5-a50010a@10.1.80.10" target="_blank">c22f9200-f51b49d-c5-a50010a@10.1.80.10</a></p>
<p>Supported: timer,resource-priority,replaces</p><p>Min-SE: 1800</p><p>User-Agent: Cisco-CUCM8.6</p><p>Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY</p><p>CSeq: 104 INVITE</p><p>
Max-Forwards: 70</p><p>Expires: 180</p><p>Allow-Events: presence</p><p>Supported: X-cisco-srtp-fallback</p><p>Supported: Geolocation</p><p>Session-Expires: 1800;refresher=uas</p><p>P-Asserted-Identity: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10" target="_blank">sip:6784563290@10.1.80.10</a>></p>
<p>Remote-Party-ID: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10" target="_blank">sip:6784563290@10.1.80.10</a>>;party=calling;screen=yes;privacy=off</p><p>Contact: <sip:<a href="tel:6784563290" target="_blank" value="+16784563290">6784563290</a>@10.1.80.10:5060;transport=tcp>;video;audio;video</p>
<p>Content-Length: 0</p><div> <br></div><p>*Jan 15 20:19:35.170: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p><p>Sent: </p><p>INVITE <a href="http://sip:17705439047@64.154.41.150:5060/" target="_blank">sip:17705439047@64.154.41.150:5060</a> SIP/2.0</p>
<p>Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69232672</p><p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@sipconnect.ipcomms.net" target="_blank">sip:6784563290@sipconnect.ipcomms.net</a>>;tag=2E6BC0B0-2268</p>
<p>To: <<a href="mailto:sip%3A17705439047@sip.talkinip.net" target="_blank">sip:17705439047@sip.talkinip.net</a>>;tag=as7c5ff82e</p><p>Date: Tue, 15 Jan 2013 20:19:35 GMT</p><p>Call-ID: <a href="mailto:A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214" target="_blank">A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214</a></p>
<p>Supported: timer,resource-priority,replaces,sdp-anat</p><p>Min-SE: 1800</p><p>Cisco-Guid: 3257897472-0000065536-0000000035-0173015306</p><p>User-Agent: Cisco-SIPGateway/IOS-12.x</p><p>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER</p>
<p>CSeq: 105 INVITE</p><p>Max-Forwards: 70</p><p>Timestamp: 1358281175</p><p>Contact: <<a href="http://sip:6784563290@98.192.104.214:5060/" target="_blank">sip:6784563290@98.192.104.214:5060</a>></p><p>Expires: 180</p>
<p>Allow-Events: telephone-event</p><p>Authorization: Digest username="<a href="tel:6784563290" target="_blank" value="+16784563290">6784563290</a>",realm="asterisk",uri="<a href="http://sip:17705439047@64.154.41.150:5060/" target="_blank">sip:17705439047@64.154.41.150:5060</a>",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5</p>
<p>Session-Expires: 1800;refresher=uas</p><p>Content-Length: 0</p><div> <br></div><p>*Jan 15 20:19:35.190: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p><p>Sent: </p><p>SIP/2.0 100 Trying</p><p>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28e6157f41c</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10" target="_blank">sip:6784563290@10.1.80.10</a>>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957</p><p>To: <<a href="mailto:sip%3A17705439047@10.1.200.1" target="_blank">sip:17705439047@10.1.200.1</a>>;tag=2E6BC6F0-1E22</p>
<p>Date: Tue, 15 Jan 2013 20:19:35 GMT</p><p>Call-ID: <a href="mailto:c22f9200-f51b49d-c5-a50010a@10.1.80.10" target="_blank">c22f9200-f51b49d-c5-a50010a@10.1.80.10</a></p><p>CSeq: 104 INVITE</p><p>Allow-Events: telephone-event</p>
<p>Server: Cisco-SIPGateway/IOS-12.x</p><p>Content-Length: 0</p><div> <br></div><p>*Jan 15 20:19:35.194: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p><p>Received: </p><p>SIP/2.0 100 Trying</p><p>Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69232672;received=98.192.104.214</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@sipconnect.ipcomms.net" target="_blank">sip:6784563290@sipconnect.ipcomms.net</a>>;tag=2E6BC0B0-2268</p><p>To: <<a href="mailto:sip%3A17705439047@sip.talkinip.net" target="_blank">sip:17705439047@sip.talkinip.net</a>>;tag=as7c5ff82e</p>
<p>Call-ID: <a href="mailto:A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214" target="_blank">A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214</a></p><p>CSeq: 105 INVITE</p><p>Server: Asterisk PBX 1.6.2.13</p><p>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO</p>
<p>Supported: replaces, timer</p><p>Require: timer</p><p>Session-Expires: 1800;refresher=uas</p><p>Contact: <<a href="mailto:sip%3A17705439047@64.154.41.150" target="_blank">sip:17705439047@64.154.41.150</a>></p><p>
Content-Length: 0</p><div> <br></div><p>*Jan 15 20:19:35.194: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p><p>Received: </p><p>SIP/2.0 200 OK</p><p>Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69232672;received=98.192.104.214</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@sipconnect.ipcomms.net" target="_blank">sip:6784563290@sipconnect.ipcomms.net</a>>;tag=2E6BC0B0-2268</p><p>To: <<a href="mailto:sip%3A17705439047@sip.talkinip.net" target="_blank">sip:17705439047@sip.talkinip.net</a>>;tag=as7c5ff82e</p>
<p>Call-ID: <a href="mailto:A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214" target="_blank">A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214</a></p><p>CSeq: 105 INVITE</p><p>Server: Asterisk PBX 1.6.2.13</p><p>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO</p>
<p>Supported: replaces, timer</p><p>Require: timer</p><p>Session-Expires: 1800;refresher=uas</p><p>Contact: <<a href="mailto:sip%3A17705439047@64.154.41.150" target="_blank">sip:17705439047@64.154.41.150</a>></p><p>
Content-Type: application/sdp</p><p>Content-Length: 333</p><p>v=0</p><p>o=root 1685873050 1685873054 IN IP4 <a href="tel:64.154.41.150" target="_blank" value="+16415441150">64.154.41.150</a></p><p>s=Asterisk PBX 1.6.2.13</p>
<p>c=IN IP4 <a href="tel:64.154.41.150" target="_blank" value="+16415441150">64.154.41.150</a></p><p>t=0 0</p><p>m=audio 13014 RTP/AVP 3 8 0 18 101</p><p>a=rtpmap:3 GSM/8000</p><p>a=rtpmap:8 PCMA/8000</p><p>a=rtpmap:0 PCMU/8000</p>
<p>a=rtpmap:18 G729/8000</p><p>a=fmtp:18 annexb=no</p><p>a=rtpmap:101 telephone-event/8000</p><p>a=fmtp:101 0-16</p><p>a=ptime:20</p><p>a=inactive</p><p>*Jan 15 20:19:35.198: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p>
<p>Sent: </p><p>SIP/2.0 200 OK</p><p>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28e6157f41c</p><p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10" target="_blank">sip:6784563290@10.1.80.10</a>>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957</p>
<p>To: <<a href="mailto:sip%3A17705439047@10.1.200.1" target="_blank">sip:17705439047@10.1.200.1</a>>;tag=2E6BC6F0-1E22</p><p>Date: Tue, 15 Jan 2013 20:19:35 GMT</p><p>Call-ID: <a href="mailto:c22f9200-f51b49d-c5-a50010a@10.1.80.10" target="_blank">c22f9200-f51b49d-c5-a50010a@10.1.80.10</a></p>
<p>CSeq: 104 INVITE</p><p>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER</p><p>Allow-Events: telephone-event</p><p>Remote-Party-ID: <<a href="mailto:sip%3A17705439047@10.1.200.1" target="_blank">sip:17705439047@10.1.200.1</a>>;party=called;screen=no;privacy=off</p>
<p>Contact: <sip:<a href="tel:17705439047" target="_blank" value="+17705439047">17705439047</a>@10.1.200.1:5060;transport=tcp></p><p>Supported: replaces</p><p>Supported: sdp-anat</p><p>Server: Cisco-SIPGateway/IOS-12.x</p>
<p>Session-Expires: 1800;refresher=uas</p><p>Require: timer</p><p>Supported: timer</p><p>Content-Type: application/sdp</p><p>Content-Length: 277</p><p>v=0</p><p>o=CiscoSystemsSIP-GW-UserAgent 4444 5481 IN IP4 10.1.200.1</p>
<p>s=SIP Call</p><p>c=IN IP4 10.1.200.1</p><p>t=0 0</p><p>m=audio 19514 RTP/AVP 0 101 19</p><p>c=IN IP4 10.1.200.1</p><p>a=inactive</p><p>a=rtpmap:0 PCMU/8000</p><p>a=rtpmap:101 telephone-event/8000</p><p>a=fmtp:101 0-16</p>
<p>a=rtpmap:19 CN/8000</p><p>a=ptime:20</p><p>*Jan 15 20:19:35.206: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:</p><p>Received: </p><p>ACK sip:<a href="tel:17705439047" target="_blank" value="+17705439047">17705439047</a>@10.1.200.1:5060;transport=tcp SIP/2.0</p>
<p>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28f6dca6616</p><p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10" target="_blank">sip:6784563290@10.1.80.10</a>>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957</p>
<p>To: <<a href="mailto:sip%3A17705439047@10.1.200.1" target="_blank">sip:17705439047@10.1.200.1</a>>;tag=2E6BC6F0-1E22</p><p>Date: Tue, 15 Jan 2013 19:57:42 GMT</p><p>Call-ID: <a href="mailto:c22f9200-f51b49d-c5-a50010a@10.1.80.10" target="_blank">c22f9200-f51b49d-c5-a50010a@10.1.80.10</a></p>
<p>Max-Forwards: 70</p><p>CSeq: 104 ACK</p><p>Allow-Events: presence, kpml</p><p>Content-Type: application/sdp</p><p>Content-Length: 243</p><p>v=0</p><p>o=CiscoSystemsCCM-SIP 7322 5 IN IP4 10.1.80.10</p><p>s=SIP Call</p><p>
c=IN IP4 10.1.10.18</p><p>b=TIAS:64000</p><p>b=AS:64</p><p>t=0 0</p><p>m=audio 21476 RTP/AVP 0 101</p><p>a=rtpmap:0 PCMU/8000</p><p>a=ptime:20</p><p>a=inactive</p><p>a=rtpmap:101 telephone-event/8000</p><p>a=fmtp:101 0-15</p>
<p>*Jan 15 20:19:35.210: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p><p>Sent: </p><p>ACK <a href="http://sip:17705439047@64.154.41.150:5060/" target="_blank">sip:17705439047@64.154.41.150:5060</a> SIP/2.0</p><p>Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69246AB</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@sipconnect.ipcomms.net" target="_blank">sip:6784563290@sipconnect.ipcomms.net</a>>;tag=2E6BC0B0-2268</p><p>To: <<a href="mailto:sip%3A17705439047@sip.talkinip.net" target="_blank">sip:17705439047@sip.talkinip.net</a>>;tag=as7c5ff82e</p>
<p>Date: Tue, 15 Jan 2013 20:19:35 GMT</p><p>Call-ID: <a href="mailto:A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214" target="_blank">A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214</a></p><p>Max-Forwards: 70</p><p>
CSeq: 105 ACK</p><p>Authorization: Digest username="<a href="tel:6784563290" target="_blank" value="+16784563290">6784563290</a>",realm="asterisk",uri="<a href="http://sip:17705439047@64.154.41.150:5060/" target="_blank">sip:17705439047@64.154.41.150:5060</a>",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5</p>
<p>Allow-Events: telephone-event</p><p>Content-Type: application/sdp</p><p>Content-Length: 265</p><p>v=0</p><p>o=CiscoSystemsSIP-GW-UserAgent 3168 2741 IN IP4 98.192.104.214</p><p>s=SIP Call</p><p>c=IN IP4 98.192.104.214</p>
<p>t=0 0</p><p>m=audio 19458 RTP/AVP 0 101</p><p>c=IN IP4 98.192.104.214</p><p>a=inactive</p><p>a=rtpmap:0 PCMU/8000</p><p>a=rtpmap:101 telephone-event/8000</p><p>a=fmtp:101 0-16</p><p>a=ptime:20</p><p>Cisco3825#</p><br><br>
</div><div class="gmail_quote">On Tue, Jan 15, 2013 at 2:28 PM, Ryan Ratliff <span dir="ltr"><<a href="mailto:rratliff@cisco.com" target="_blank">rratliff@cisco.com</a>></span> wrote:<br><blockquote style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid" class="gmail_quote">
<div style="word-wrap:break-word">ccsip message is what I'd go with just to see the signaling with no other stuff. Depending on what that shows and what your gateway is doing to the signals you may need to expand from there.<div>
<span><font color="#888888"><br><div>
<span style="text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;border-collapse:separate">-Ryan</span>
</div></font></span><div>
<br><div><div>On Jan 15, 2013, at 2:11 PM, Dane Newman <<a href="mailto:dane.newman@gmail.com" target="_blank">dane.newman@gmail.com</a>> wrote:</div><br><div>Ryan</div><div> </div><div>What is the proper debug to use to caputre the useful information?</div>
<div> </div><div>Dane</div><div><br><br> </div><div class="gmail_quote">On Tue, Jan 15, 2013 at 12:42 PM, Ryan Ratliff <span dir="ltr"><<a href="mailto:rratliff@cisco.com" target="_blank">rratliff@cisco.com</a>></span> wrote:<br>
<blockquote style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid" class="gmail_quote"><div style="word-wrap:break-word">Without sip messages I can't get any clues from that.<div>
<span><font color="#888888"><br><div>
<span style="text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;border-collapse:separate">-Ryan</span>
</div></font></span><div>
<br><div><div>On Jan 15, 2013, at 12:35 PM, Dane Newman <<a href="mailto:dane.newman@gmail.com" target="_blank">dane.newman@gmail.com</a>> wrote:</div><br><div>Thanks Ryan for the input</div><div> </div><div> </div>
<div><strong>On the call when I hold the call the following debug pops out....</strong></div><div> </div><div><br>*Jan 15 17:56:05.246: //13939/922252E78D73/SIP/Error/ccsip_api_request_offer: Unable to add passthru hdrs to<br>
container<br>SIP: Attribute mid, level 1 instance 1 not found.<br>SIP: (13938) Group (a= group line) attribute, level 65535 instance 1 not found.<br>*Jan 15 17:56:05.274: //13938/922252E78D73/SIP/Error/ccsip_api_response_answer: Unable to add<br>
passthru headers to container<br>SIP: Attribute mid, level 1 instance 1 not found.<br>SIP: (13939) Group (a= group line) attribute, level 65535 instance 1 not found.<br>*Jan 15 17:56:05.286: //13939/922252E78D73/SIP/Error/ccsip_api_request_offer: Unable to add passthru hdrs to<br>
container<br>*Jan 15 17:56:05.302: //13938/922252E78D73/SIP/Error/ccsip_api_response_answer: Unable to add<br> passthru headers to container<br>SIP: Attribute mid, level 1 instance 1 not found.<br>
SIP: (13939) Group (a= group line) attribute, level 65535 instance 1 not found.<br>SIP: Attribute mid, level 1 instance 1 not found.<br>*Jan 15 17:56:05.322: //13939/922252E78D73/SIP/Error/sipSPIProcessAckMedia: Could not modify QoS params for midcall INVITE<br>
</div><div> </div><div><strong>After I try to unhold the call the following debug comes out....</strong></div><div><strong></strong> </div><div><br>*Jan 15 17:56:18.874: //13939/922252E78D73/SIP/Error/ccsip_api_request_offer: Unable to add passthru hdrs to<br>
container<br>*Jan 15 17:56:18.894: //13938/922252E78D73/SIP/Error/ccsip_api_response_answer: Unable to add<br> passthru headers to container<br>SIP: Attribute mid, level 1 instance 1 not found.<br>
SIP: (13939) Group (a= group line) attribute, level 65535 instance 1 not found.<br>SIP: Attribute mid, level 1 instance 1 not found.<br>*Jan 15 17:56:18.906: //13939/922252E78D73/SIP/Error/sipSPIProcessAckMedia: Could not modify QoS params for midcall INVITE<br>
Cisco3825#<br>Cisco3825#<br>Cisco3825#<br><br></div><div class="gmail_quote">On Tue, Jan 15, 2013 at 9:42 AM, Ryan Ratliff <span dir="ltr"><<a href="mailto:rratliff@cisco.com" target="_blank">rratliff@cisco.com</a>></span> wrote:<br>
<blockquote style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid" class="gmail_quote"><div style="word-wrap:break-word">Given you have an ITSP it's most likely the initial hold that's failing, which is only manifesting when you try to resume it. If you haven't noticed already this is also very likely causing transfers to fail.<div>
<br></div><div>Take a look at the SIP signaling for a call. I believe the most common cause to this is the ITSP not handling our transition from active->inactive->sendonly->active from hold to MOH to resume. The "Duplex Streaming Enabled" parameter is there just for this type of problem.</div>
<div><span><font color="#888888"><br><div>
<span style="text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;border-collapse:separate">-Ryan</span>
</div></font></span><div>
<br><div><div>On Jan 14, 2013, at 6:40 PM, Dane Newman <<a href="mailto:dane.newman@gmail.com" target="_blank">dane.newman@gmail.com</a>> wrote:</div><br><div><strong>Hello Kenneth</strong></div><div><strong></strong> </div>
<div><strong>I have restarted both CUCM servers so this should have restarted the services when the utils system restart happened</strong></div><div><strong></strong> </div>
<div><br><strong>on my router I see I am using g711 from the debug </strong></div><div><strong></strong> </div><div><strong>I ran a debug voip ccapi inout </strong></div><div><strong></strong> </div><div><strong>I connected a call calling from an external number to a DiD inside of my system. Once the call was connected I put the call on hold and the following debug came out..the music on hold played for the external caller</strong></div>
<div> </div><div>*Jan 14 23:47:40.779: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:<br> Stop Tone On Digit=FALSE, Tone=Null,<br> Tone Direction=Sum Network, Params=0x0, Call Id=12741<br>*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:<br>
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call Id=12742, Xmit Function=0x64204BAC<br>*Jan 14 23:47:40.783: //12741/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:</div><div>*Jan 14 23:47:40.783: cc_api_get_xcode_stream : 4702<br>
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:<br> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call Id=12742,<br> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,<br> Modem=0x0, Codec Bytes=20, Signal Type=2)<br>
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:<br> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),<br> Playout Max=1000(ms), Fax Nom=300(ms))<br>*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:<br>
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call Id=12741,<br> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),<br> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)<br>
*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:<br> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call Id=12741,<br> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),<br>
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)<br>*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_event_indication:<br> Event=170, Call Id=12742<br>*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_event_indication:<br>
Event Is Sent To Conferenced SPI(s) Directly<br>*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_call_feature:<br> Feature Type=50, Interface=0xC05A65AC, Call Id=12742<br>*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:<br>
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call Id=12741,<br> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,<br> Modem=0x0, Codec Bytes=20, Signal Type=2)<br>*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:<br>
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),<br> Playout Max=1000(ms), Fax Nom=300(ms))<br>*Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:<br> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call Id=12742,<br>
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),<br> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)<br>*Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:<br> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call Id=12742,<br>
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),<br> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)<br>*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_event_indication:<br>
Event=171, Call Id=12741<br>*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_event_indication:<br> Event Is Sent To Conferenced SPI(s) Directly<br>*Jan 14 23:47:40.815: //12742/9906C1828C05/CCAPI/cc_api_call_facility:<br>
Interface=0xC05A65AC, Call Id=12742<br>*Jan 14 23:47:40.819: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:<br> Stop Tone On Digit=FALSE, Tone=Null,<br> Tone Direction=Sum Network, Params=0x0, Call Id=12741<br>*Jan 14 23:47:40.819: //12742/9906C1828C05/CCAPI/cc_api_event_indication:<br>
Event=96, Call Id=12742<br>*Jan 14 23:47:40.819: //12742/9906C1828C05/CCAPI/cc_api_event_indication:<br> Event Is Sent To Conferenced SPI(s) Directly<br>*Jan 14 23:47:40.839: //12742/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:<br>
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call Id=12741, Xmit Function=0x64204BAC<br>*Jan 14 23:47:40.839: //12742/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:</div><div>*Jan 14 23:47:40.839: cc_api_get_xcode_stream : 4702<br>
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:<br> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call Id=12741,<br> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,<br> Modem=0x0, Codec Bytes=20, Signal Type=2)<br>
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:<br> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),<br> Playout Max=1000(ms), Fax Nom=300(ms))<br>*Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:<br>
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call Id=12742,<br> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),<br> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)<br>
*Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:<br> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call Id=12742,<br> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),<br>
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)<br>*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_event_indication:<br> Event=170, Call Id=12741<br>*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_event_indication:<br>
Event Is Sent To Conferenced SPI(s) Directly<br>*Jan 14 23:47:40.859: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:<br> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call Id=12742,<br> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,<br>
Modem=0x0, Codec Bytes=20, Signal Type=2)<br>*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:<br> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),<br> Playout Max=1000(ms), Fax Nom=300(ms))<br>
*Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:<br> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call Id=12741,<br> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),<br>
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)<br>*Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:<br> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call Id=12741,<br>
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),<br> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)<br>*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_event_indication:<br>
Event=171, Call Id=12742<br>*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_event_indication:<br> Event Is Sent To Conferenced SPI(s) Directly<br>*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_call_facility:<br>
Interface=0xC05A65AC, Call Id=12742<br>Cisco3825#<br>Cisco3825#<br>Cisco3825#</div><div> </div><div> </div><div><strong>I then after that took off the hold and the following debug came out. The call on the PSDN side still played the hold music while there was no voice on the deskphone side.</strong></div>
<div> </div><div>*Jan 14 23:47:40.779: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:<br> Stop Tone On Digit=FALSE, Tone=Null,<br> Tone Direction=Sum Network, Params=0x0, Call Id=12741<br>*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:<br>
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call Id=12742, Xmit Function=0x64204BAC<br>*Jan 14 23:47:40.783: //12741/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:</div><div>*Jan 14 23:47:40.783: cc_api_get_xcode_stream : 4702<br>
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:<br> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call Id=12742,<br> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,<br> Modem=0x0, Codec Bytes=20, Signal Type=2)<br>
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:<br> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),<br> Playout Max=1000(ms), Fax Nom=300(ms))<br>*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:<br>
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call Id=12741,<br> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),<br> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)<br>
*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:<br> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call Id=12741,<br> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),<br>
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)<br>*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_event_indication:<br> Event=170, Call Id=12742<br>*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_event_indication:<br>
Event Is Sent To Conferenced SPI(s) Directly<br>*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_call_feature:<br> Feature Type=50, Interface=0xC05A65AC, Call Id=12742<br>*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:<br>
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call Id=12741,<br> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,<br> Modem=0x0, Codec Bytes=20, Signal Type=2)<br>*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:<br>
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),<br> Playout Max=1000(ms), Fax Nom=300(ms))<br>*Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:<br> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call Id=12742,<br>
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),<br> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)<br>*Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:<br> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call Id=12742,<br>
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),<br> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)<br>*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_event_indication:<br>
Event=171, Call Id=12741<br>*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_event_indication:<br> Event Is Sent To Conferenced SPI(s) Directly<br>*Jan 14 23:47:40.815: //12742/9906C1828C05/CCAPI/cc_api_call_facility:<br>
Interface=0xC05A65AC, Call Id=12742<br>*Jan 14 23:47:40.819: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:<br> Stop Tone On Digit=FALSE, Tone=Null,<br> Tone Direction=Sum Network, Params=0x0, Call Id=12741<br>*Jan 14 23:47:40.819: //12742/9906C1828C05/CCAPI/cc_api_event_indication:<br>
Event=96, Call Id=12742<br>*Jan 14 23:47:40.819: //12742/9906C1828C05/CCAPI/cc_api_event_indication:<br> Event Is Sent To Conferenced SPI(s) Directly<br>*Jan 14 23:47:40.839: //12742/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:<br>
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call Id=12741, Xmit Function=0x64204BAC<br>*Jan 14 23:47:40.839: //12742/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:</div><div>*Jan 14 23:47:40.839: cc_api_get_xcode_stream : 4702<br>
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:<br> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call Id=12741,<br> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,<br> Modem=0x0, Codec Bytes=20, Signal Type=2)<br>
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:<br> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),<br> Playout Max=1000(ms), Fax Nom=300(ms))<br>*Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:<br>
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call Id=12742,<br> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),<br> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)<br>
*Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:<br> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call Id=12742,<br> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),<br>
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)<br>*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_event_indication:<br> Event=170, Call Id=12741<br>*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_event_indication:<br>
Event Is Sent To Conferenced SPI(s) Directly<br>*Jan 14 23:47:40.859: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:<br> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call Id=12742,<br> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,<br>
Modem=0x0, Codec Bytes=20, Signal Type=2)<br>*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:<br> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),<br> Playout Max=1000(ms), Fax Nom=300(ms))<br>
*Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:<br> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call Id=12741,<br> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),<br>
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)<br>*Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:<br> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call Id=12741,<br>
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),<br> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)<br>*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_event_indication:<br>
Event=171, Call Id=12742<br>*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_event_indication:<br> Event Is Sent To Conferenced SPI(s) Directly<br>*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_call_facility:<br>
Interface=0xC05A65AC, Call Id=12742<br>Cisco3825#<br>Cisco3825#<br>Cisco3825#</div><div> </div><div class="gmail_quote">On Mon, Jan 14, 2013 at 6:20 PM, Kenneth Hayes <span dir="ltr"><<a href="mailto:kennethwhayes@gmail.com" target="_blank">kennethwhayes@gmail.com</a>></span> wrote:<br>
<blockquote style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid" class="gmail_quote"><div dir="auto"><div>Have you also restarted the Cisco IP Media Services?<br>
<br>Sent from my iPhone</div><div><div><br>On Jan 14, 2013, at 6:12 PM, Dane Newman <<a href="mailto:dane.newman@gmail.com" target="_blank">dane.newman@gmail.com</a>> wrote:<br>
<br></div><blockquote type="cite"><div>My ITSP will only support 711ulaw for me currently I believe. They hard coded it with me when I was initially setting it up.</div><div> </div><div>Do you think this could be a codec issue? How would I go about identifying if it is?</div>
<div> </div><div>Dane<br><br></div><div class="gmail_quote">On Mon, Jan 14, 2013 at 6:09 PM, Kenneth Hayes <span dir="ltr"><<a href="mailto:kennethwhayes@gmail.com" target="_blank">kennethwhayes@gmail.com</a>></span> wrote:<br>
<blockquote style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid" class="gmail_quote"><div dir="auto"><div>Have you tried different audio codecs?<br>
<br>Sent from my iPhone</div><div><div><br>On Jan 14, 2013, at 6:06 PM, Dane Newman <<a href="mailto:dane.newman@gmail.com" target="_blank">dane.newman@gmail.com</a>> wrote:<br>
<br></div><blockquote type="cite"><div>Ryan (sorry I forgot to reply to all)</div><div><br>Thanks for the Reply</div><div> </div><div>Oddly enough we are.</div><div> </div><div>This probably has something to do with MOH in general?</div>
<div><br>Internally when I user puts another user on hold everything works. No MOH plays and they can hold and unhold the call just fine.</div>
<div> </div><div>I tested calling from an external number. Once I put the external caller on hold the MOH played but I was unable to resume the call. When I hit resume on the deskphone the MOH still played to the external caller and there was no sound on the deskphone.<br>
<br></div><div class="gmail_quote">On Mon, Jan 14, 2013 at 5:25 PM, Ryan Ratliff <span dir="ltr"><<a href="mailto:rratliff@cisco.com" target="_blank">rratliff@cisco.com</a>></span> wrote:<br><blockquote style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid" class="gmail_quote">
<div style="word-wrap:break-word">Do you get similar behavior if you just hold and resume the call outside SNR features? <div><span><font color="#888888"><br><div>
<span style="text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;border-collapse:separate">-Ryan</span>
</div></font></span><div>
<br><div><div>On Jan 14, 2013, at 4:18 PM, Dane Newman <<a href="mailto:dane.newman@gmail.com" target="_blank">dane.newman@gmail.com</a>> wrote:</div><br><div>Using keyboard-interactive authentication. <br>
Password: <br><br>Cisco3825# <br>Cisco3825#sh ver <br>
Cisco IOS Software, 3800 Software (C3825-ADVENTERPRISEK9_IVS_LI-M), Version 15.1<br>(4)M5, RELEASE SOFTWARE (fc1) <br>Technical Support: <a href="http://www.cisco.com/techsupport" target="_blank">http://www.cisco.com/techsupport</a> <br>
Copyright (c) 1986-2012 by Cisco Systems, Inc. <br>Compiled Tue 04-Sep-12 17:25 by prod_rel_team <br><br>ROM: System Bootstrap, Version 12.4(13r)T10, RELEASE SOFTWARE (fc1) <br>
<br>Cisco3825 uptime is 1 week, 1 day, 1 hour, 38 minutes <br>System returned to ROM by power-on <br>System image file is "flash:c3825-adventerprisek9_ivs_li-mz.151-4.M5.bin" <br>
Last reload type: Normal Reload <br><br><br>This product contains cryptographic features and is subject to United <br>States and local country laws governing import, export, transfer and <br>
use. Delivery of Cisco cryptographic products does not imply <br>third-party authority to import, export, distribute or use encryption. <br>Importers, exporters, distributors and users are responsible for <br>
compliance with U.S. and local country laws. By using this product you <br>agree to comply with applicable laws and regulations. If you are unable <br>to comply with U.S. and local laws, return this product immediately. <br>
<br>A summary of U.S. laws governing Cisco cryptographic products may be found at: <br><a href="http://www.cisco.com/wwl/export/crypto/tool/stqrg.html" target="_blank">http://www.cisco.com/wwl/export/crypto/tool/stqrg.html</a> <br>
<br>If you require further assistance please contact us by sending email to <br><a href="mailto:export@cisco.com" target="_blank">export@cisco.com</a>. <br>
<br>Cisco 3825 (revision 1.2) with 1011712K/36864K bytes of memory. <br>Processor board ID FTX1237A1T0 <br>2 Gigabit Ethernet interfaces <br>
2 Channelized T1/PRI ports <br>1 Virtual Private Network (VPN) Module <br>DRAM configuration is 64 bits wide with parity enabled. <br>
479K bytes of NVRAM. <br>500472K bytes of ATA System CompactFlash (Read/Write) <br><br><br>License Info: <br>
<br>License UDI: <br><br>------------------------------------------------- <br>Device# PID SN <br>
<br>Sent from my mobile device<br><br>On Jan 14, 2013, at 4:11 PM, Kenneth Hayes <<a href="mailto:kennethwhayes@gmail.com" target="_blank">kennethwhayes@gmail.com</a>> wrote:<br><br><blockquote type="cite">What version of code are you running on the CUBE?<br>
<br>Sent from my iPhone<br><br>On Jan 14, 2013, at 3:43 PM, Dane Newman <<a href="mailto:dane.newman@gmail.com" target="_blank">dane.newman@gmail.com</a>> wrote:<br><br><blockquote type="cite">Hello<br><br>I have an issue when users are connected to a call and hit the mobility soft key button on 9971 phones when a call is active, the phone system rings on the mobile number configured in the system. When they pick up the the mobile number it just plays what sounds like hold music on both ends of the call (I believe this music is coming from cucm but I haven't heard it before) instead of providing 2 way voice.<br>
<br>In another senario with what I believe is the same issue. If a user picks up on there cell phone first (using single number reach) opposed to the deskphone the call is connected with 2 way voice and no issues exist. If the user then hangs up his cell phone with the intent to take the call on his deskphone the calling party starts hearing the hold music. Once the user picks up the call on his deskphone he hears nothing but the calling party is still hearing the hold music. It is not working as intended where 2 way voice happens once the user hangs up his mobile phone and picks up on his deskphone 2 way voice should happen.<br>
<br>My topology is as follows..<br><br><br>PSDN --> SIP TRUNK FROM ITSP --> 3825 CUBE --->CUCM -->DESKPHOHE<br><br>Calls are sent back out the SIP trunk to the ITSP when using mobile connect/snr.<br><br>Does anyone have any ideas how I can make 2 way voice happen instead of the hold music when the calls are picked up?<br>
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