[cisco-voip] Mobility Issue
Dane Newman
dane.newman at gmail.com
Wed Jan 16 13:56:10 EST 2013
All
The issue has been resolved I believe (I feel very dumb)....Thank you so
much all for your assistance!
Ryan and Nick was 100% correct my MTP was not working correctly. I found
that it was miss configured and in the default device pool instead of the
correct one. Once I put it in the correct device pools I could hold calls
and everything works correctly now.
On Wed, Jan 16, 2013 at 1:27 PM, Dane Newman <dane.newman at gmail.com> wrote:
> Ryan
>
> Thank you again for responding and sharing your wisdom.
>
> 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.
>
> I then did a search for the IP address of my IP phone 10.1.10.18.
>
> 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?
>
> *With MTP unchecked on the SIP TRUNk*
> *Jan 16 18:33:28.384: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
> Received:
> ACK sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0
> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK4e47f806d
> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
> >;tag=673~d8eefedd-7473-4e00-a4a0-ce8f65d30766-31145612
> To: <sip:17705439047 at 10.1.200.1>;tag=333141D8-1492
> Date: Wed, 16 Jan 2013 18:11:29 GMT
> Call-ID: 24e56400-f61ed51-12-a50010a at 10.1.80.10
> Max-Forwards: 70
> CSeq: 101 ACK
> Allow-Events: presence, kpml
> Content-Type: application/sdp
> Content-Length: 230
> v=0
> o=CiscoSystemsCCM-SIP 673 1 IN IP4 10.1.80.10
> s=SIP Call
> c=IN IP4 10.1.10.18
> b=TIAS:64000
> b=AS:64
> t=0 0
> m=audio 20116 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=ptime:20
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
>
> *With MTP checked on the SIP TRUNk*
>
> *Jan 16 18:39:09.732: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
> Received:
> ACK sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0
> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK60b810f25
> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
> >;tag=723~d8eefedd-7473-4e00-a4a0-ce8f65d30766-31145637
> To: <sip:17705439047 at 10.1.200.1>;tag=3336625C-1CBB
> Date: Wed, 16 Jan 2013 18:17:05 GMT
> Call-ID: ed2aec00-f61eea1-16-a50010a at 10.1.80.10
> Max-Forwards: 70
> CSeq: 101 ACK
> Allow-Events: presence, kpml
> Content-Type: application/sdp
> Content-Length: 230
> v=0
> o=CiscoSystemsCCM-SIP 723 2 IN IP4 10.1.80.10
> s=SIP Call
> c=IN IP4 10.1.10.18
> b=TIAS:64000
> b=AS:64
> t=0 0
> m=audio 16452 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=ptime:20
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
>
>
>
> On Wed, Jan 16, 2013 at 12:59 PM, Ryan Ratliff <rratliff at cisco.com> wrote:
>
>> 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.
>>
>> 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.
>>
>> -Ryan
>>
>> On Jan 16, 2013, at 12:35 PM, Dane Newman <dane.newman at gmail.com> wrote:
>>
>> All
>>
>> Thank you for the infromation you are providing me on this thread. It is
>> a great learning exp for me.
>>
>> 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.
>>
>> 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?
>>
>> I have reset the trunk and rebooted cucm also...
>>
>> Thanks again for the assistance and advice it's much appericated
>>
>> Dane
>>
>>
>>
>> On Wed, Jan 16, 2013 at 12:18 PM, Ryan Ratliff <rratliff at cisco.com>wrote:
>>
>>> Having the MOH servers registered is step 1 of about 10 that have to
>>> happen for MOH to be allocated for the call.
>>> 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?
>>>
>>> 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.
>>>
>>> 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.
>>>
>>> Have you tried resetting the SIP trunk in CUCM yet?
>>>
>>> -Ryan
>>>
>>> On Jan 16, 2013, at 11:26 AM, Dane Newman <dane.newman at gmail.com> wrote:
>>>
>>> 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
>>>
>>> On Wed, Jan 16, 2013 at 10:20 AM, Kenneth Hayes <kennethwhayes at gmail.com
>>> > wrote:
>>>
>>>> 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?
>>>>
>>>> Sent from my iPad
>>>>
>>>> On Jan 16, 2013, at 10:14 AM, Nick Matthews <matthnick at gmail.com>
>>>> wrote:
>>>>
>>>> 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.
>>>>
>>>> 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.
>>>>
>>>> -nick
>>>>
>>>>
>>>> On Tue, Jan 15, 2013 at 8:39 PM, Dane Newman <dane.newman at gmail.com>wrote:
>>>>
>>>>> Nick
>>>>>
>>>>> Thanks for the assistance.
>>>>>
>>>>> 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?
>>>>>
>>>>> On Tue, Jan 15, 2013 at 7:56 PM, Nick Matthews <matthnick at gmail.com>wrote:
>>>>>
>>>>>> 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.
>>>>>>
>>>>>> 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.
>>>>>>
>>>>>> 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.
>>>>>>
>>>>>> 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.
>>>>>>
>>>>>> -nick
>>>>>>
>>>>>>
>>>>>> On Tue, Jan 15, 2013 at 3:42 PM, Kenneth Hayes <
>>>>>> kennethwhayes at gmail.com> wrote:
>>>>>>
>>>>>>> Bind your media and source to your outbound interface on your voice
>>>>>>> service voip.
>>>>>>>
>>>>>>> Sent from my iPhone
>>>>>>>
>>>>>>> On Jan 15, 2013, at 3:39 PM, Dane Newman <dane.newman at gmail.com>
>>>>>>> wrote:
>>>>>>>
>>>>>>> Below is a show run from the router
>>>>>>>
>>>>>>>
>>>>>>> [OK]
>>>>>>> Cisco3825#sh run
>>>>>>> Building configuration...
>>>>>>>
>>>>>>> Current configuration : 10183 bytes
>>>>>>> !
>>>>>>> ! Last configuration change at 20:49:24 UTC Tue Jan 15 2013 by
>>>>>>> dnewman
>>>>>>> version 15.1
>>>>>>> service timestamps debug datetime msec
>>>>>>> service timestamps log datetime msec
>>>>>>> no service password-encryption
>>>>>>> !
>>>>>>> hostname Cisco3825
>>>>>>> !
>>>>>>> boot-start-marker
>>>>>>> boot-end-marker
>>>>>>> !
>>>>>>> !
>>>>>>> !
>>>>>>> aaa new-model
>>>>>>> !
>>>>>>> !
>>>>>>> aaa authentication login default local
>>>>>>> aaa authentication login vpnauth local
>>>>>>> aaa authorization exec default local
>>>>>>> aaa authorization network default local
>>>>>>> aaa authorization network vpnauth local
>>>>>>> !
>>>>>>> !
>>>>>>> !
>>>>>>> !
>>>>>>> !
>>>>>>> aaa session-id common
>>>>>>> !
>>>>>>> no network-clock-participate wic 0
>>>>>>> !
>>>>>>> dot11 syslog
>>>>>>> ip source-route
>>>>>>> !
>>>>>>> ip cef
>>>>>>> !
>>>>>>> !
>>>>>>> !
>>>>>>> !
>>>>>>> ip domain name datasc.local
>>>>>>> ip inspect udp idle-time 1800
>>>>>>> no ipv6 cef
>>>>>>> !
>>>>>>> multilink bundle-name authenticated
>>>>>>> !
>>>>>>> !
>>>>>>> !
>>>>>>> !
>>>>>>> !
>>>>>>> voice-card 0
>>>>>>> dsp services dspfarm
>>>>>>> !
>>>>>>> !
>>>>>>> !
>>>>>>> voice service voip
>>>>>>> ip address trusted list
>>>>>>> ipv4 64.154.41.150 255.255.255.255
>>>>>>> allow-connections sip to sip
>>>>>>> fax protocol pass-through g711ulaw
>>>>>>> sip
>>>>>>> !
>>>>>>> !
>>>>>>> !
>>>>>>> !
>>>>>>> voice translation-rule 1
>>>>>>> rule 1 /6784604564/ /200/
>>>>>>> rule 2 /6784563290/ /100/
>>>>>>> rule 3 /6784563291/ /101/
>>>>>>> rule 4 /6784563292/ /102/
>>>>>>> rule 5 /6784563293/ /103/
>>>>>>> rule 6 /6784563294/ /104/
>>>>>>> rule 7 /6784563295/ /105/
>>>>>>> rule 8 /6784563296/ /106/
>>>>>>> rule 9 /6784563297/ /107/
>>>>>>> rule 10 /6784563298/ /108/
>>>>>>> rule 11 /6784563299/ /109/
>>>>>>> rule 12 /6784604565/ /125/
>>>>>>> !
>>>>>>> !
>>>>>>> voice translation-profile incomingdid
>>>>>>> translate called 1
>>>>>>> !
>>>>>>> !
>>>>>>> crypto pki token default removal timeout 0
>>>>>>> !
>>>>>>> crypto pki trustpoint selfsigned
>>>>>>> enrollment selfsigned
>>>>>>> subject-name CN=connect.datasc.com
>>>>>>> revocation-check crl
>>>>>>> !
>>>>>>> !
>>>>>>> crypto pki certificate chain selfsigned
>>>>>>> certificate self-signed 02
>>>>>>> 30820251 308201BA A0030201 02020102 300D0609 2A864886 F70D0101
>>>>>>> 05050030
>>>>>>> 44311B30 19060355 04031312 636F6E6E 6563742E 64617461 73632E63
>>>>>>> 6F6D3125
>>>>>>> 30230609 2A864886 F70D0109 02161643 6973636F 33383235 2E646174
>>>>>>> 6173632E
>>>>>>> 6C6F6361 6C301E17 0D313231 32323831 39313531 395A170D 32303031
>>>>>>> 30313030
>>>>>>> 30303030 5A304431 1B301906 03550403 1312636F 6E6E6563 742E6461
>>>>>>> 74617363
>>>>>>> 2E636F6D 31253023 06092A86 4886F70D 01090216 16436973 636F3338
>>>>>>> 32352E64
>>>>>>> 61746173 632E6C6F 63616C30 819F300D 06092A86 4886F70D 01010105
>>>>>>> 0003818D
>>>>>>> 00308189 02818100 D9A99B41 8B70C82F 28072967 376E13E8 8F7FC2C2
>>>>>>> 7729B93E
>>>>>>> DDAE31A0 F3613381 78B43E11 5144BE88 DC2FDE14 0035A104 0BBFAEA0
>>>>>>> 9A190598
>>>>>>> 19A124B1 2C4A8EA2 04253BA1 C829EF07 CD0E848D E7AA5269 459449C4
>>>>>>> FABF9CA9
>>>>>>> BC5AF8ED 84FCD99B 260C2B75 57887863 7BB310FB 2C8D1506 FE91FEAC
>>>>>>> 4EDD1712
>>>>>>> A7AFD2C1 BF21C59D 02030100 01A35330 51300F06 03551D13 0101FF04
>>>>>>> 05300301
>>>>>>> 01FF301F 0603551D 23041830 16801475 02C4FB04 4FB3F748 B4012EC5
>>>>>>> 8A571236
>>>>>>> A190CB30 1D060355 1D0E0416 04147502 C4FB044F B3F748B4 012EC58A
>>>>>>> 571236A1
>>>>>>> 90CB300D 06092A86 4886F70D 01010505 00038181 00C2B167 E583F6D8
>>>>>>> 8B742D4F
>>>>>>> 49D27AAD 7EF4E64F 0B5CA5A3 944E8CC7 499A706F AB22283B AE5913A1
>>>>>>> B22BBB20
>>>>>>> E7CF6F9F 41CDD870 1B474E58 9537C1FA 2D93BE4F 4276E9CE 61AE18D3
>>>>>>> EF724BD9
>>>>>>> 33878860 4B3627C0 448C652D 03D4C142 BA3291A3 DDE0C4DD C6ED06C3
>>>>>>> 12E45933
>>>>>>> F1EE5CC2 B5B6CC20 C65AB313 76966F14 AA25CC8D 2A
>>>>>>> quit
>>>>>>> !
>>>>>>> !
>>>>>>> license udi pid CISCO3825 sn FTX1237A1T0
>>>>>>> username xxxxxxx privilege 15 secret xxxxxx
>>>>>>> !
>>>>>>> redundancy
>>>>>>> !
>>>>>>> !
>>>>>>> controller T1 0/0/0
>>>>>>> !
>>>>>>> controller T1 0/0/1
>>>>>>> !
>>>>>>> ip ssh version 2
>>>>>>> !
>>>>>>> !
>>>>>>> crypto isakmp policy 10
>>>>>>> encr aes
>>>>>>> authentication pre-share
>>>>>>> group 2
>>>>>>> crypto isakmp key Recoil90 address 0.0.0.0 0.0.0.0
>>>>>>> crypto isakmp fragmentation
>>>>>>> !
>>>>>>> crypto isakmp client configuration group datasc
>>>>>>> key Recoil90
>>>>>>> dns 4.2.2.2 4.2.2.1
>>>>>>> domain datasc.local
>>>>>>> pool vpnpool
>>>>>>> save-password
>>>>>>> !
>>>>>>> crypto isakmp client configuration group datascsplit
>>>>>>> key Recoil90
>>>>>>> dns 4.2.2.2 4.2.2.1
>>>>>>> domain datasc.local
>>>>>>> pool vpnpool
>>>>>>> acl 101
>>>>>>> save-password
>>>>>>> crypto isakmp profile datasc
>>>>>>> match identity group datasc
>>>>>>> client authentication list vpnauth
>>>>>>> isakmp authorization list vpnauth
>>>>>>> client configuration address respond
>>>>>>> virtual-template 1
>>>>>>> crypto isakmp profile datascsplit
>>>>>>> match identity group datascsplit
>>>>>>> client authentication list vpnauth
>>>>>>> isakmp authorization list vpnauth
>>>>>>> client configuration address respond
>>>>>>> virtual-template 2
>>>>>>> !
>>>>>>> !
>>>>>>> crypto ipsec transform-set transformset esp-aes
>>>>>>> crypto ipsec transform-set ezvpntransform esp-aes esp-sha-hmac
>>>>>>> !
>>>>>>> crypto ipsec profile VTI
>>>>>>> set transform-set ezvpntransform
>>>>>>> set isakmp-profile datasc
>>>>>>> !
>>>>>>> crypto ipsec profile VTI2
>>>>>>> set transform-set ezvpntransform
>>>>>>> set isakmp-profile datascsplit
>>>>>>> !
>>>>>>> !
>>>>>>> !
>>>>>>> !
>>>>>>> !
>>>>>>> !
>>>>>>> !
>>>>>>> interface Loopback1
>>>>>>> ip address 10.1.150.1 255.255.255.0
>>>>>>> ip nat inside
>>>>>>> ip virtual-reassembly in
>>>>>>> !
>>>>>>> interface GigabitEthernet0/0
>>>>>>> ip address dhcp
>>>>>>> no ip redirects
>>>>>>> no ip unreachables
>>>>>>> no ip proxy-arp
>>>>>>> ip nat outside
>>>>>>> ip virtual-reassembly in
>>>>>>> duplex auto
>>>>>>> speed auto
>>>>>>> media-type rj45
>>>>>>> hold-queue 240000 in
>>>>>>> !
>>>>>>> interface GigabitEthernet0/1
>>>>>>> ip address 10.1.200.1 255.255.255.252
>>>>>>> ip nat inside
>>>>>>> ip virtual-reassembly in
>>>>>>> duplex auto
>>>>>>> speed auto
>>>>>>> media-type rj45
>>>>>>> !
>>>>>>> interface Virtual-Template1 type tunnel
>>>>>>> ip unnumbered GigabitEthernet0/0
>>>>>>> ip nat inside
>>>>>>> ip virtual-reassembly in
>>>>>>> tunnel source GigabitEthernet0/0
>>>>>>> tunnel mode ipsec ipv4
>>>>>>> tunnel protection ipsec profile VTI
>>>>>>> !
>>>>>>> interface Virtual-Template2 type tunnel
>>>>>>> ip unnumbered GigabitEthernet0/0
>>>>>>> ip nat inside
>>>>>>> ip virtual-reassembly in
>>>>>>> tunnel source GigabitEthernet0/0
>>>>>>> tunnel mode ipsec ipv4
>>>>>>> tunnel protection ipsec profile VTI2
>>>>>>> !
>>>>>>> interface Virtual-Template3
>>>>>>> ip unnumbered GigabitEthernet0/0
>>>>>>> ip nat outside
>>>>>>> ip virtual-reassembly in
>>>>>>> ip policy route-map anyconnecthop
>>>>>>> !
>>>>>>> !
>>>>>>> router eigrp 1
>>>>>>> maximum-paths 1
>>>>>>> network 10.0.0.0
>>>>>>> redistribute static
>>>>>>> !
>>>>>>> ip local pool vpnpool 10.1.100.10 10.1.100.200
>>>>>>> ip forward-protocol nd
>>>>>>> ip http server
>>>>>>> ip http secure-server
>>>>>>> !
>>>>>>> !
>>>>>>> ip nat inside source list NATNETWORKS interface GigabitEthernet0/0
>>>>>>> overload
>>>>>>> ip nat inside source static tcp 10.1.50.150 80 interface
>>>>>>> GigabitEthernet0/0 80
>>>>>>> ip nat inside source static tcp 10.1.80.100 5001 interface
>>>>>>> GigabitEthernet0/0 5001
>>>>>>> ip nat inside source static udp 10.1.80.100 5001 interface
>>>>>>> GigabitEthernet0/0 5001
>>>>>>> !
>>>>>>> ip access-list extended NATNETWORKS
>>>>>>> deny ip 10.0.0.0 0.255.255.255 172.16.0.0 0.15.255.255
>>>>>>> deny ip 10.0.0.0 0.255.255.255 10.0.0.0 0.255.255.255
>>>>>>> permit ip 10.0.0.0 0.255.255.255 any
>>>>>>> ip access-list extended anyconnecthop
>>>>>>> deny ip 10.0.0.0 0.255.255.255 10.0.0.0 0.255.255.255
>>>>>>> permit ip 10.0.0.0 0.255.255.255 any
>>>>>>> !
>>>>>>> access-list 50 permit 10.0.0.0 0.255.255.255
>>>>>>> access-list 101 permit ip 10.0.0.0 0.255.255.255 any
>>>>>>> !
>>>>>>> !
>>>>>>> !
>>>>>>> !
>>>>>>> route-map anyconnecthop permit 10
>>>>>>> match ip address anyconnecthop
>>>>>>> set ip next-hop 10.1.150.2
>>>>>>> !
>>>>>>> !
>>>>>>> !
>>>>>>> !
>>>>>>> !
>>>>>>> control-plane
>>>>>>> !
>>>>>>> !
>>>>>>> !
>>>>>>> !
>>>>>>> mgcp profile default
>>>>>>> !
>>>>>>> !
>>>>>>> dial-peer voice 100 voip
>>>>>>> description Publisher
>>>>>>> preference 1
>>>>>>> destination-pattern 1..
>>>>>>> session protocol sipv2
>>>>>>> session target ipv4:10.1.80.10
>>>>>>> dtmf-relay rtp-nte
>>>>>>> codec g711ulaw
>>>>>>> !
>>>>>>> dial-peer voice 101 voip
>>>>>>> description Subscriber
>>>>>>> preference 2
>>>>>>> destination-pattern 1..
>>>>>>> session target ipv4:10.1.80.11
>>>>>>> dtmf-relay rtp-nte
>>>>>>> codec g711ulaw
>>>>>>> !
>>>>>>> dial-peer voice 200 voip
>>>>>>> description Publisher
>>>>>>> preference 1
>>>>>>> destination-pattern 2..
>>>>>>> progress_ind setup enable 3
>>>>>>> progress_ind progress enable 8
>>>>>>> session protocol sipv2
>>>>>>> session target ipv4:10.1.80.10
>>>>>>> dtmf-relay rtp-nte
>>>>>>> codec g711ulaw
>>>>>>> !
>>>>>>> dial-peer voice 300 voip
>>>>>>> description incoming Calldid
>>>>>>> translation-profile incoming incomingdid
>>>>>>> preference 1
>>>>>>> session protocol sipv2
>>>>>>> session target sip-server
>>>>>>> incoming called-number 678456329.
>>>>>>> dtmf-relay rtp-nte
>>>>>>> codec g711ulaw
>>>>>>> !
>>>>>>> dial-peer voice 301 voip
>>>>>>> description incoming Calldid
>>>>>>> translation-profile incoming incomingdid
>>>>>>> preference 1
>>>>>>> session protocol sipv2
>>>>>>> session target sip-server
>>>>>>> incoming called-number 6784604565
>>>>>>> dtmf-relay rtp-nte
>>>>>>> codec g711ulaw
>>>>>>> !
>>>>>>> dial-peer voice 302 voip
>>>>>>> description incoming Calldid
>>>>>>> translation-profile incoming incomingdid
>>>>>>> preference 1
>>>>>>> session protocol sipv2
>>>>>>> session target sip-server
>>>>>>> incoming called-number 6784604564
>>>>>>> dtmf-relay rtp-nte
>>>>>>> codec g711ulaw
>>>>>>> !
>>>>>>> dial-peer voice 201 voip
>>>>>>> description Publisher
>>>>>>> preference 2
>>>>>>> destination-pattern 2..
>>>>>>> progress_ind setup enable 3
>>>>>>> progress_ind progress enable 8
>>>>>>> session protocol sipv2
>>>>>>> session target ipv4:10.1.80.11
>>>>>>> dtmf-relay rtp-nte
>>>>>>> codec g711ulaw
>>>>>>> !
>>>>>>> dial-peer voice 500 voip
>>>>>>> description outgoing
>>>>>>> preference 1
>>>>>>> destination-pattern .T
>>>>>>> session protocol sipv2
>>>>>>> session target dns:sip.talkinip.net
>>>>>>> dtmf-relay rtp-nte
>>>>>>> codec g711ulaw
>>>>>>> !
>>>>>>> !
>>>>>>> sip-ua
>>>>>>> credentials username xxxxxxxx password 7 xxxxxxx realm
>>>>>>> sipconnect.ipcomms.net
>>>>>>> authentication username xxxxxx password 7 xxxxxxx
>>>>>>> authentication username xxxxx password 7 xxxxxxx realm
>>>>>>> sipconnect.ipcomms.net
>>>>>>> set pstn-cause 3 sip-status 486
>>>>>>> set pstn-cause 34 sip-status 486
>>>>>>> set pstn-cause 47 sip-status 486
>>>>>>> registrar dns:sipconnect.ipcomms.net expires 60
>>>>>>> sip-server dns:sipconnect.ipcomms.net:5060
>>>>>>> !
>>>>>>> !
>>>>>>> !
>>>>>>> gatekeeper
>>>>>>> shutdown
>>>>>>> !
>>>>>>> !
>>>>>>> call-manager-fallback
>>>>>>> max-conferences 8 gain -6
>>>>>>> transfer-system full-consult
>>>>>>> ip source-address 10.1.200.1 port 2000
>>>>>>> max-ephones 20
>>>>>>> max-dn 40
>>>>>>> !
>>>>>>> !
>>>>>>> !
>>>>>>> line con 0
>>>>>>> line aux 0
>>>>>>> line vty 0 4
>>>>>>> privilege level 15
>>>>>>> transport input ssh
>>>>>>> line vty 5 15
>>>>>>> privilege level 15
>>>>>>> transport input ssh
>>>>>>> !
>>>>>>> scheduler allocate 20000 1000
>>>>>>> !
>>>>>>> webvpn gateway gateway_1
>>>>>>> ip interface GigabitEthernet0/0 port 443
>>>>>>> ssl trustpoint selfsigned
>>>>>>> inservice
>>>>>>> !
>>>>>>> webvpn install svc flash:/webvpn/anyconnect-win-3.1.02026-k9.pkg
>>>>>>> sequence 1
>>>>>>> !
>>>>>>> webvpn context datasc
>>>>>>> title "DataSC VPN"
>>>>>>> secondary-color white
>>>>>>> title-color #CCCC66
>>>>>>> text-color black
>>>>>>> ssl authenticate verify all
>>>>>>> !
>>>>>>> url-list "Servers"
>>>>>>> heading "Server"
>>>>>>> !
>>>>>>> !
>>>>>>> policy group datasc
>>>>>>> url-list "Servers"
>>>>>>> functions svc-enabled
>>>>>>> svc address-pool "vpnpool" netmask 255.255.255.0
>>>>>>> svc keep-client-installed
>>>>>>> svc dns-server primary 4.2.2.2
>>>>>>> svc dtls
>>>>>>> virtual-template 3
>>>>>>> default-group-policy datasc
>>>>>>> aaa authentication list vpnauth
>>>>>>> gateway gateway_1 domain datasc
>>>>>>> inservice
>>>>>>> !
>>>>>>> !
>>>>>>> webvpn context datascsplit
>>>>>>> title "DataSC VPN Split"
>>>>>>> secondary-color white
>>>>>>> title-color #CCCC66
>>>>>>> text-color black
>>>>>>> ssl authenticate verify all
>>>>>>> !
>>>>>>> url-list "Servers"
>>>>>>> heading "Server"
>>>>>>> !
>>>>>>> !
>>>>>>> policy group datascsplit
>>>>>>> url-list "Servers"
>>>>>>> functions svc-enabled
>>>>>>> svc address-pool "vpnpool" netmask 255.255.255.0
>>>>>>> svc split include acl 50
>>>>>>> svc dns-server primary 4.2.2.2
>>>>>>> svc dtls
>>>>>>> default-group-policy datascsplit
>>>>>>> aaa authentication list vpnauth
>>>>>>> gateway gateway_1 domain datascsplit
>>>>>>> inservice
>>>>>>> !
>>>>>>> end
>>>>>>> Cisco3825#
>>>>>>>
>>>>>>> On Tue, Jan 15, 2013 at 3:31 PM, Kenneth Hayes <
>>>>>>> kennethwhayes at gmail.com> wrote:
>>>>>>>
>>>>>>>> What do your media resources look like?
>>>>>>>>
>>>>>>>>
>>>>>>>> Also can you show me a copy of your voice service voip config?
>>>>>>>>
>>>>>>>> Sent from my iPad
>>>>>>>>
>>>>>>>> On Jan 15, 2013, at 3:12 PM, Dane Newman <dane.newman at gmail.com>
>>>>>>>> wrote:
>>>>>>>>
>>>>>>>> Thanks Ryan
>>>>>>>>
>>>>>>>> I see I am always getting a 200 ok message after my invites from
>>>>>>>> the debug
>>>>>>>>
>>>>>>>> *Putting a call on HOLD*
>>>>>>>>
>>>>>>>>
>>>>>>>> *Jan 15 20:19:28.086: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>
>>>>>>>> Received:
>>>>>>>>
>>>>>>>> INVITE sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0
>>>>>>>>
>>>>>>>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK2891b89f8fa
>>>>>>>>
>>>>>>>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>>>>>>>
>>>>>>>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>>>>>>>
>>>>>>>> Date: Tue, 15 Jan 2013 19:57:35 GMT
>>>>>>>>
>>>>>>>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>>>>>>>
>>>>>>>> Supported: timer,resource-priority,replaces
>>>>>>>>
>>>>>>>> Min-SE: 1800
>>>>>>>>
>>>>>>>> User-Agent: Cisco-CUCM8.6
>>>>>>>>
>>>>>>>> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE,
>>>>>>>> REFER, SUBSCRIBE, NOTIFY
>>>>>>>>
>>>>>>>> CSeq: 102 INVITE
>>>>>>>>
>>>>>>>> Max-Forwards: 70
>>>>>>>>
>>>>>>>> Expires: 180
>>>>>>>>
>>>>>>>> Allow-Events: presence
>>>>>>>>
>>>>>>>> Supported: X-cisco-srtp-fallback
>>>>>>>>
>>>>>>>> Supported: Geolocation
>>>>>>>>
>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>
>>>>>>>> P-Asserted-Identity: "Dane Newman" <sip:6784563290 at 10.1.80.10>
>>>>>>>>
>>>>>>>> Remote-Party-ID: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>> >;party=calling;screen=yes;privacy=off
>>>>>>>>
>>>>>>>> Contact: <sip:6784563290 at 10.1.80.10:5060;transport=tcp>;video
>>>>>>>>
>>>>>>>> Content-Type: application/sdp
>>>>>>>>
>>>>>>>> Content-Length: 240
>>>>>>>>
>>>>>>>> v=0
>>>>>>>>
>>>>>>>> o=CiscoSystemsCCM-SIP 7322 3 IN IP4 10.1.80.10
>>>>>>>>
>>>>>>>> s=SIP Call
>>>>>>>>
>>>>>>>> c=IN IP4 0.0.0.0
>>>>>>>>
>>>>>>>> b=TIAS:64000
>>>>>>>>
>>>>>>>> b=AS:64
>>>>>>>>
>>>>>>>> t=0 0
>>>>>>>>
>>>>>>>> m=audio 21476 RTP/AVP 0 101
>>>>>>>>
>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>
>>>>>>>> a=ptime:20
>>>>>>>>
>>>>>>>> a=inactive
>>>>>>>>
>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>>
>>>>>>>> a=fmtp:101 0-15
>>>>>>>>
>>>>>>>> *Jan 15 20:19:28.094: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>
>>>>>>>> Sent:
>>>>>>>>
>>>>>>>> INVITE sip:17705439047 at 64.154.41.150:5060 SIP/2.0
>>>>>>>>
>>>>>>>> Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK691F12E0
>>>>>>>>
>>>>>>>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>>>>>>>> >;tag=2E6BC0B0-2268
>>>>>>>>
>>>>>>>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>>>>>>>
>>>>>>>> Date: Tue, 15 Jan 2013 20:19:28 GMT
>>>>>>>>
>>>>>>>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>>>>>>>
>>>>>>>> Supported: 100rel,timer,resource-priority,replaces,sdp-anat
>>>>>>>>
>>>>>>>> Min-SE: 1800
>>>>>>>>
>>>>>>>> Cisco-Guid: 3257897472-0000065536-0000000035-0173015306
>>>>>>>>
>>>>>>>> User-Agent: Cisco-SIPGateway/IOS-12.x
>>>>>>>>
>>>>>>>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>>>>>>>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>>>>>>>>
>>>>>>>> CSeq: 103 INVITE
>>>>>>>>
>>>>>>>> Max-Forwards: 70
>>>>>>>>
>>>>>>>> Timestamp: 1358281168
>>>>>>>>
>>>>>>>> Contact: <sip:6784563290 at 98.192.104.214:5060>
>>>>>>>>
>>>>>>>> Expires: 180
>>>>>>>>
>>>>>>>> Allow-Events: telephone-event
>>>>>>>>
>>>>>>>> Authorization: Digest username="6784563290",realm="asterisk",uri="
>>>>>>>> sip:17705439047 at 64.154.41.150:5060
>>>>>>>> ",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5
>>>>>>>>
>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>
>>>>>>>> Content-Type: application/sdp
>>>>>>>>
>>>>>>>> Content-Length: 289
>>>>>>>>
>>>>>>>> v=0
>>>>>>>>
>>>>>>>> o=CiscoSystemsSIP-GW-UserAgent 3168 2739 IN IP4 98.192.104.214
>>>>>>>>
>>>>>>>> s=SIP Call
>>>>>>>>
>>>>>>>> c=IN IP4 98.192.104.214
>>>>>>>>
>>>>>>>> t=0 0
>>>>>>>>
>>>>>>>> m=audio 19458 RTP/AVP 0 101 19
>>>>>>>>
>>>>>>>> c=IN IP4 98.192.104.214
>>>>>>>>
>>>>>>>> a=inactive
>>>>>>>>
>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>
>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>>
>>>>>>>> a=fmtp:101 0-15
>>>>>>>>
>>>>>>>> a=rtpmap:19 CN/8000
>>>>>>>>
>>>>>>>> a=ptime:20
>>>>>>>>
>>>>>>>> *Jan 15 20:19:28.094: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>
>>>>>>>> Sent:
>>>>>>>>
>>>>>>>> SIP/2.0 100 Trying
>>>>>>>>
>>>>>>>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK2891b89f8fa
>>>>>>>>
>>>>>>>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>>>>>>>
>>>>>>>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>>>>>>>
>>>>>>>> Date: Tue, 15 Jan 2013 20:19:28 GMT
>>>>>>>>
>>>>>>>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>>>>>>>
>>>>>>>> CSeq: 102 INVITE
>>>>>>>>
>>>>>>>> Allow-Events: telephone-event
>>>>>>>>
>>>>>>>> Server: Cisco-SIPGateway/IOS-12.x
>>>>>>>>
>>>>>>>> Content-Length: 0
>>>>>>>>
>>>>>>>> *Jan 15 20:19:28.110: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>
>>>>>>>> Received:
>>>>>>>>
>>>>>>>> SIP/2.0 100 Trying
>>>>>>>>
>>>>>>>> Via: SIP/2.0/UDP 98.192.104.214:5060
>>>>>>>> ;branch=z9hG4bK691F12E0;received=98.192.104.214
>>>>>>>>
>>>>>>>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>>>>>>>> >;tag=2E6BC0B0-2268
>>>>>>>>
>>>>>>>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>>>>>>>
>>>>>>>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>>>>>>>
>>>>>>>> CSeq: 103 INVITE
>>>>>>>>
>>>>>>>> Server: Asterisk PBX 1.6.2.13
>>>>>>>>
>>>>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>>>>>>>> INFO
>>>>>>>>
>>>>>>>> Supported: replaces, timer
>>>>>>>>
>>>>>>>> Require: timer
>>>>>>>>
>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>
>>>>>>>> Contact: <sip:17705439047 at 64.154.41.150>
>>>>>>>>
>>>>>>>> Content-Length: 0
>>>>>>>>
>>>>>>>> *Jan 15 20:19:28.110: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>
>>>>>>>> Received:
>>>>>>>>
>>>>>>>> SIP/2.0 200 OK
>>>>>>>>
>>>>>>>> Via: SIP/2.0/UDP 98.192.104.214:5060
>>>>>>>> ;branch=z9hG4bK691F12E0;received=98.192.104.214
>>>>>>>>
>>>>>>>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>>>>>>>> >;tag=2E6BC0B0-2268
>>>>>>>>
>>>>>>>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>>>>>>>
>>>>>>>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>>>>>>>
>>>>>>>> CSeq: 103 INVITE
>>>>>>>>
>>>>>>>> Server: Asterisk PBX 1.6.2.13
>>>>>>>>
>>>>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>>>>>>>> INFO
>>>>>>>>
>>>>>>>> Supported: replaces, timer
>>>>>>>>
>>>>>>>> Require: timer
>>>>>>>>
>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>
>>>>>>>> Contact: <sip:17705439047 at 64.154.41.150>
>>>>>>>>
>>>>>>>> Content-Type: application/sdp
>>>>>>>>
>>>>>>>> Content-Length: 239
>>>>>>>>
>>>>>>>> v=0
>>>>>>>>
>>>>>>>> o=root 1685873050 1685873052 IN IP4 64.154.41.150
>>>>>>>>
>>>>>>>> s=Asterisk PBX 1.6.2.13
>>>>>>>>
>>>>>>>> c=IN IP4 64.154.41.150
>>>>>>>>
>>>>>>>> t=0 0
>>>>>>>>
>>>>>>>> m=audio 13014 RTP/AVP 0 101
>>>>>>>>
>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>
>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>>
>>>>>>>> a=fmtp:101 0-16
>>>>>>>>
>>>>>>>> a=ptime:20
>>>>>>>>
>>>>>>>> a=inactive
>>>>>>>>
>>>>>>>> *Jan 15 20:19:28.118: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>
>>>>>>>> Sent:
>>>>>>>>
>>>>>>>> SIP/2.0 200 OK
>>>>>>>>
>>>>>>>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK2891b89f8fa
>>>>>>>>
>>>>>>>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>>>>>>>
>>>>>>>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>>>>>>>
>>>>>>>> Date: Tue, 15 Jan 2013 20:19:28 GMT
>>>>>>>>
>>>>>>>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>>>>>>>
>>>>>>>> CSeq: 102 INVITE
>>>>>>>>
>>>>>>>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>>>>>>>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>>>>>>>>
>>>>>>>> Allow-Events: telephone-event
>>>>>>>>
>>>>>>>> Remote-Party-ID: <sip:17705439047 at 10.1.200.1
>>>>>>>> >;party=called;screen=no;privacy=off
>>>>>>>>
>>>>>>>> Contact: <sip:17705439047 at 10.1.200.1:5060;transport=tcp>
>>>>>>>>
>>>>>>>> Supported: replaces
>>>>>>>>
>>>>>>>> Supported: sdp-anat
>>>>>>>>
>>>>>>>> Server: Cisco-SIPGateway/IOS-12.x
>>>>>>>>
>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>
>>>>>>>> Require: timer
>>>>>>>>
>>>>>>>> Supported: timer
>>>>>>>>
>>>>>>>> Content-Type: application/sdp
>>>>>>>>
>>>>>>>> Content-Length: 253
>>>>>>>>
>>>>>>>> v=0
>>>>>>>>
>>>>>>>> o=CiscoSystemsSIP-GW-UserAgent 4444 5479 IN IP4 10.1.200.1
>>>>>>>>
>>>>>>>> s=SIP Call
>>>>>>>>
>>>>>>>> c=IN IP4 10.1.200.1
>>>>>>>>
>>>>>>>> t=0 0
>>>>>>>>
>>>>>>>> m=audio 19514 RTP/AVP 0 101
>>>>>>>>
>>>>>>>> c=IN IP4 10.1.200.1
>>>>>>>>
>>>>>>>> a=inactive
>>>>>>>>
>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>
>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>>
>>>>>>>> a=fmtp:101 0-16
>>>>>>>>
>>>>>>>> a=ptime:20
>>>>>>>>
>>>>>>>> *Jan 15 20:19:28.118: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>
>>>>>>>> Sent:
>>>>>>>>
>>>>>>>> ACK sip:17705439047 at 64.154.41.150:5060 SIP/2.0
>>>>>>>>
>>>>>>>> Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK6920266D
>>>>>>>>
>>>>>>>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>>>>>>>> >;tag=2E6BC0B0-2268
>>>>>>>>
>>>>>>>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>>>>>>>
>>>>>>>> Date: Tue, 15 Jan 2013 20:19:28 GMT
>>>>>>>>
>>>>>>>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>>>>>>>
>>>>>>>> Max-Forwards: 70
>>>>>>>>
>>>>>>>> CSeq: 103 ACK
>>>>>>>>
>>>>>>>> Authorization: Digest username="6784563290",realm="asterisk",uri="
>>>>>>>> sip:17705439047 at 64.154.41.150:5060
>>>>>>>> ",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5
>>>>>>>>
>>>>>>>> Allow-Events: telephone-event
>>>>>>>>
>>>>>>>> Content-Length: 0
>>>>>>>>
>>>>>>>> *Jan 15 20:19:28.122: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>
>>>>>>>> Received:
>>>>>>>>
>>>>>>>> ACK sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0
>>>>>>>>
>>>>>>>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28b4b1305a0
>>>>>>>>
>>>>>>>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>>>>>>>
>>>>>>>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>>>>>>>
>>>>>>>> Date: Tue, 15 Jan 2013 19:57:35 GMT
>>>>>>>>
>>>>>>>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>>>>>>>
>>>>>>>> Max-Forwards: 70
>>>>>>>>
>>>>>>>> CSeq: 102 ACK
>>>>>>>>
>>>>>>>> Allow-Events: presence
>>>>>>>>
>>>>>>>> Content-Length: 0
>>>>>>>>
>>>>>>>> *Jan 15 20:19:28.122: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>
>>>>>>>> Received:
>>>>>>>>
>>>>>>>> INVITE sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0
>>>>>>>>
>>>>>>>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28c43b47e7e
>>>>>>>>
>>>>>>>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>>>>>>>
>>>>>>>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>>>>>>>
>>>>>>>> Date: Tue, 15 Jan 2013 19:57:35 GMT
>>>>>>>>
>>>>>>>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>>>>>>>
>>>>>>>> Supported: timer,resource-priority,replaces
>>>>>>>>
>>>>>>>> Min-SE: 1800
>>>>>>>>
>>>>>>>> User-Agent: Cisco-CUCM8.6
>>>>>>>>
>>>>>>>> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE,
>>>>>>>> REFER, SUBSCRIBE, NOTIFY
>>>>>>>>
>>>>>>>> CSeq: 103 INVITE
>>>>>>>>
>>>>>>>> Max-Forwards: 70
>>>>>>>>
>>>>>>>> Expires: 180
>>>>>>>>
>>>>>>>> Allow-Events: presence
>>>>>>>>
>>>>>>>> Supported: X-cisco-srtp-fallback
>>>>>>>>
>>>>>>>> Supported: Geolocation
>>>>>>>>
>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>
>>>>>>>> P-Asserted-Identity: "Dane Newman" <sip:6784563290 at 10.1.80.10>
>>>>>>>>
>>>>>>>> Remote-Party-ID: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>> >;party=calling;screen=yes;privacy=off
>>>>>>>>
>>>>>>>> Contact: <sip:6784563290 at 10.1.80.10:5060;transport=tcp>;video
>>>>>>>>
>>>>>>>> Content-Length: 0
>>>>>>>>
>>>>>>>> *Jan 15 20:19:28.126: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>
>>>>>>>> Sent:
>>>>>>>>
>>>>>>>> INVITE sip:17705439047 at 64.154.41.150:5060 SIP/2.0
>>>>>>>>
>>>>>>>> Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69211AB3
>>>>>>>>
>>>>>>>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>>>>>>>> >;tag=2E6BC0B0-2268
>>>>>>>>
>>>>>>>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>>>>>>>
>>>>>>>> Date: Tue, 15 Jan 2013 20:19:28 GMT
>>>>>>>>
>>>>>>>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>>>>>>>
>>>>>>>> Supported: timer,resource-priority,replaces,sdp-anat
>>>>>>>>
>>>>>>>> Min-SE: 1800
>>>>>>>>
>>>>>>>> Cisco-Guid: 3257897472-0000065536-0000000035-0173015306
>>>>>>>>
>>>>>>>> User-Agent: Cisco-SIPGateway/IOS-12.x
>>>>>>>>
>>>>>>>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>>>>>>>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>>>>>>>>
>>>>>>>> CSeq: 104 INVITE
>>>>>>>>
>>>>>>>> Max-Forwards: 70
>>>>>>>>
>>>>>>>> Timestamp: 1358281168
>>>>>>>>
>>>>>>>> Contact: <sip:6784563290 at 98.192.104.214:5060>
>>>>>>>>
>>>>>>>> Expires: 180
>>>>>>>>
>>>>>>>> Allow-Events: telephone-event
>>>>>>>>
>>>>>>>> Authorization: Digest username="6784563290",realm="asterisk",uri="
>>>>>>>> sip:17705439047 at 64.154.41.150:5060
>>>>>>>> ",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5
>>>>>>>>
>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>
>>>>>>>> Content-Length: 0
>>>>>>>>
>>>>>>>> *Jan 15 20:19:28.126: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>
>>>>>>>> Sent:
>>>>>>>>
>>>>>>>> SIP/2.0 100 Trying
>>>>>>>>
>>>>>>>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28c43b47e7e
>>>>>>>>
>>>>>>>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>>>>>>>
>>>>>>>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>>>>>>>
>>>>>>>> Date: Tue, 15 Jan 2013 20:19:28 GMT
>>>>>>>>
>>>>>>>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>>>>>>>
>>>>>>>> CSeq: 103 INVITE
>>>>>>>>
>>>>>>>> Allow-Events: telephone-event
>>>>>>>>
>>>>>>>> Server: Cisco-SIPGateway/IOS-12.x
>>>>>>>>
>>>>>>>> Content-Length: 0
>>>>>>>>
>>>>>>>> *Jan 15 20:19:28.146: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>
>>>>>>>> Received:
>>>>>>>>
>>>>>>>> SIP/2.0 100 Trying
>>>>>>>>
>>>>>>>> Via: SIP/2.0/UDP 98.192.104.214:5060
>>>>>>>> ;branch=z9hG4bK69211AB3;received=98.192.104.214
>>>>>>>>
>>>>>>>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>>>>>>>> >;tag=2E6BC0B0-2268
>>>>>>>>
>>>>>>>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>>>>>>>
>>>>>>>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>>>>>>>
>>>>>>>> CSeq: 104 INVITE
>>>>>>>>
>>>>>>>> Server: Asterisk PBX 1.6.2.13
>>>>>>>>
>>>>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>>>>>>>> INFO
>>>>>>>>
>>>>>>>> Supported: replaces, timer
>>>>>>>>
>>>>>>>> Require: timer
>>>>>>>>
>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>
>>>>>>>> Contact: <sip:17705439047 at 64.154.41.150>
>>>>>>>>
>>>>>>>> Content-Length: 0
>>>>>>>>
>>>>>>>> *Jan 15 20:19:28.146: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>
>>>>>>>> Received:
>>>>>>>>
>>>>>>>> SIP/2.0 200 OK
>>>>>>>>
>>>>>>>> Via: SIP/2.0/UDP 98.192.104.214:5060
>>>>>>>> ;branch=z9hG4bK69211AB3;received=98.192.104.214
>>>>>>>>
>>>>>>>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>>>>>>>> >;tag=2E6BC0B0-2268
>>>>>>>>
>>>>>>>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>>>>>>>
>>>>>>>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>>>>>>>
>>>>>>>> CSeq: 104 INVITE
>>>>>>>>
>>>>>>>> Server: Asterisk PBX 1.6.2.13
>>>>>>>>
>>>>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>>>>>>>> INFO
>>>>>>>>
>>>>>>>> Supported: replaces, timer
>>>>>>>>
>>>>>>>> Require: timer
>>>>>>>>
>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>
>>>>>>>> Contact: <sip:17705439047 at 64.154.41.150>
>>>>>>>>
>>>>>>>> Content-Type: application/sdp
>>>>>>>>
>>>>>>>> Content-Length: 333
>>>>>>>>
>>>>>>>> v=0
>>>>>>>>
>>>>>>>> o=root 1685873050 1685873053 IN IP4 64.154.41.150
>>>>>>>>
>>>>>>>> s=Asterisk PBX 1.6.2.13
>>>>>>>>
>>>>>>>> c=IN IP4 64.154.41.150
>>>>>>>>
>>>>>>>> t=0 0
>>>>>>>>
>>>>>>>> m=audio 13014 RTP/AVP 3 8 0 18 101
>>>>>>>>
>>>>>>>> a=rtpmap:3 GSM/8000
>>>>>>>>
>>>>>>>> a=rtpmap:8 PCMA/8000
>>>>>>>>
>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>
>>>>>>>> a=rtpmap:18 G729/8000
>>>>>>>>
>>>>>>>> a=fmtp:18 annexb=no
>>>>>>>>
>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>>
>>>>>>>> a=fmtp:101 0-16
>>>>>>>>
>>>>>>>> a=ptime:20
>>>>>>>>
>>>>>>>> a=inactive
>>>>>>>>
>>>>>>>> *Jan 15 20:19:28.150: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>
>>>>>>>> Sent:
>>>>>>>>
>>>>>>>> SIP/2.0 200 OK
>>>>>>>>
>>>>>>>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28c43b47e7e
>>>>>>>>
>>>>>>>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>>>>>>>
>>>>>>>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>>>>>>>
>>>>>>>> Date: Tue, 15 Jan 2013 20:19:28 GMT
>>>>>>>>
>>>>>>>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>>>>>>>
>>>>>>>> CSeq: 103 INVITE
>>>>>>>>
>>>>>>>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>>>>>>>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>>>>>>>>
>>>>>>>> Allow-Events: telephone-event
>>>>>>>>
>>>>>>>> Remote-Party-ID: <sip:17705439047 at 10.1.200.1
>>>>>>>> >;party=called;screen=no;privacy=off
>>>>>>>>
>>>>>>>> Contact: <sip:17705439047 at 10.1.200.1:5060;transport=tcp>
>>>>>>>>
>>>>>>>> Supported: replaces
>>>>>>>>
>>>>>>>> Supported: sdp-anat
>>>>>>>>
>>>>>>>> Server: Cisco-SIPGateway/IOS-12.x
>>>>>>>>
>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>
>>>>>>>> Require: timer
>>>>>>>>
>>>>>>>> Supported: timer
>>>>>>>>
>>>>>>>> Content-Type: application/sdp
>>>>>>>>
>>>>>>>> Content-Length: 277
>>>>>>>>
>>>>>>>> v=0
>>>>>>>>
>>>>>>>> o=CiscoSystemsSIP-GW-UserAgent 4444 5480 IN IP4 10.1.200.1
>>>>>>>>
>>>>>>>> s=SIP Call
>>>>>>>>
>>>>>>>> c=IN IP4 10.1.200.1
>>>>>>>>
>>>>>>>> t=0 0
>>>>>>>>
>>>>>>>> m=audio 19514 RTP/AVP 0 101 19
>>>>>>>>
>>>>>>>> c=IN IP4 10.1.200.1
>>>>>>>>
>>>>>>>> a=inactive
>>>>>>>>
>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>
>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>>
>>>>>>>> a=fmtp:101 0-16
>>>>>>>>
>>>>>>>> a=rtpmap:19 CN/8000
>>>>>>>>
>>>>>>>> a=ptime:20
>>>>>>>>
>>>>>>>> *Jan 15 20:19:28.162: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>
>>>>>>>> Received:
>>>>>>>>
>>>>>>>> ACK sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0
>>>>>>>>
>>>>>>>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28d3eadaab3
>>>>>>>>
>>>>>>>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>>>>>>>
>>>>>>>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>>>>>>>
>>>>>>>> Date: Tue, 15 Jan 2013 19:57:35 GMT
>>>>>>>>
>>>>>>>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>>>>>>>
>>>>>>>> Max-Forwards: 70
>>>>>>>>
>>>>>>>> CSeq: 103 ACK
>>>>>>>>
>>>>>>>> Allow-Events: presence
>>>>>>>>
>>>>>>>> Content-Type: application/sdp
>>>>>>>>
>>>>>>>> Content-Length: 209
>>>>>>>>
>>>>>>>> v=0
>>>>>>>>
>>>>>>>> o=CiscoSystemsCCM-SIP 7322 4 IN IP4 10.1.80.10
>>>>>>>>
>>>>>>>> s=SIP Call
>>>>>>>>
>>>>>>>> c=IN IP4 0.0.0.0
>>>>>>>>
>>>>>>>> b=TIAS:64000
>>>>>>>>
>>>>>>>> b=AS:64
>>>>>>>>
>>>>>>>> t=0 0
>>>>>>>>
>>>>>>>> m=audio 21476 RTP/AVP 0
>>>>>>>>
>>>>>>>> a=X-cisco-media:nomedia
>>>>>>>>
>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>
>>>>>>>> a=ptime:20
>>>>>>>>
>>>>>>>> a=inactive
>>>>>>>>
>>>>>>>> *Jan 15 20:19:28.170: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>
>>>>>>>> Sent:
>>>>>>>>
>>>>>>>> ACK sip:17705439047 at 64.154.41.150:5060 SIP/2.0
>>>>>>>>
>>>>>>>> Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK692226EA
>>>>>>>>
>>>>>>>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>>>>>>>> >;tag=2E6BC0B0-2268
>>>>>>>>
>>>>>>>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>>>>>>>
>>>>>>>> Date: Tue, 15 Jan 2013 20:19:28 GMT
>>>>>>>>
>>>>>>>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>>>>>>>
>>>>>>>> Max-Forwards: 70
>>>>>>>>
>>>>>>>> CSeq: 104 ACK
>>>>>>>>
>>>>>>>> Authorization: Digest username="6784563290",realm="asterisk",uri="
>>>>>>>> sip:17705439047 at 64.154.41.150:5060
>>>>>>>> ",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5
>>>>>>>>
>>>>>>>> Allow-Events: telephone-event
>>>>>>>>
>>>>>>>> Content-Type: application/sdp
>>>>>>>>
>>>>>>>> Content-Length: 251
>>>>>>>>
>>>>>>>> v=0
>>>>>>>>
>>>>>>>> o=CiscoSystemsSIP-GW-UserAgent 3168 2740 IN IP4 98.192.104.214
>>>>>>>>
>>>>>>>> s=SIP Call
>>>>>>>>
>>>>>>>> c=IN IP4 0.0.0.0
>>>>>>>>
>>>>>>>> t=0 0
>>>>>>>>
>>>>>>>> m=audio 19458 RTP/AVP 0 101
>>>>>>>>
>>>>>>>> c=IN IP4 0.0.0.0
>>>>>>>>
>>>>>>>> a=inactive
>>>>>>>>
>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>
>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>>
>>>>>>>> a=fmtp:101 0-16
>>>>>>>>
>>>>>>>> a=ptime:20
>>>>>>>>
>>>>>>>>
>>>>>>>> *Unholding the call the MOH continues on the previously held
>>>>>>>> caller while the user hears nothing*
>>>>>>>>
>>>>>>>> **
>>>>>>>>
>>>>>>>>
>>>>>>>> *Jan 15 20:19:35.166: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>
>>>>>>>> Received:
>>>>>>>>
>>>>>>>> INVITE sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0
>>>>>>>>
>>>>>>>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28e6157f41c
>>>>>>>>
>>>>>>>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>>>>>>>
>>>>>>>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>>>>>>>
>>>>>>>> Date: Tue, 15 Jan 2013 19:57:42 GMT
>>>>>>>>
>>>>>>>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>>>>>>>
>>>>>>>> Supported: timer,resource-priority,replaces
>>>>>>>>
>>>>>>>> Min-SE: 1800
>>>>>>>>
>>>>>>>> User-Agent: Cisco-CUCM8.6
>>>>>>>>
>>>>>>>> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE,
>>>>>>>> REFER, SUBSCRIBE, NOTIFY
>>>>>>>>
>>>>>>>> CSeq: 104 INVITE
>>>>>>>>
>>>>>>>> Max-Forwards: 70
>>>>>>>>
>>>>>>>> Expires: 180
>>>>>>>>
>>>>>>>> Allow-Events: presence
>>>>>>>>
>>>>>>>> Supported: X-cisco-srtp-fallback
>>>>>>>>
>>>>>>>> Supported: Geolocation
>>>>>>>>
>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>
>>>>>>>> P-Asserted-Identity: "Dane Newman" <sip:6784563290 at 10.1.80.10>
>>>>>>>>
>>>>>>>> Remote-Party-ID: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>> >;party=calling;screen=yes;privacy=off
>>>>>>>>
>>>>>>>> Contact: <sip:6784563290 at 10.1.80.10:5060
>>>>>>>> ;transport=tcp>;video;audio;video
>>>>>>>>
>>>>>>>> Content-Length: 0
>>>>>>>>
>>>>>>>> *Jan 15 20:19:35.170: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>
>>>>>>>> Sent:
>>>>>>>>
>>>>>>>> INVITE sip:17705439047 at 64.154.41.150:5060 SIP/2.0
>>>>>>>>
>>>>>>>> Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69232672
>>>>>>>>
>>>>>>>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>>>>>>>> >;tag=2E6BC0B0-2268
>>>>>>>>
>>>>>>>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>>>>>>>
>>>>>>>> Date: Tue, 15 Jan 2013 20:19:35 GMT
>>>>>>>>
>>>>>>>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>>>>>>>
>>>>>>>> Supported: timer,resource-priority,replaces,sdp-anat
>>>>>>>>
>>>>>>>> Min-SE: 1800
>>>>>>>>
>>>>>>>> Cisco-Guid: 3257897472-0000065536-0000000035-0173015306
>>>>>>>>
>>>>>>>> User-Agent: Cisco-SIPGateway/IOS-12.x
>>>>>>>>
>>>>>>>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>>>>>>>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>>>>>>>>
>>>>>>>> CSeq: 105 INVITE
>>>>>>>>
>>>>>>>> Max-Forwards: 70
>>>>>>>>
>>>>>>>> Timestamp: 1358281175
>>>>>>>>
>>>>>>>> Contact: <sip:6784563290 at 98.192.104.214:5060>
>>>>>>>>
>>>>>>>> Expires: 180
>>>>>>>>
>>>>>>>> Allow-Events: telephone-event
>>>>>>>>
>>>>>>>> Authorization: Digest username="6784563290",realm="asterisk",uri="
>>>>>>>> sip:17705439047 at 64.154.41.150:5060
>>>>>>>> ",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5
>>>>>>>>
>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>
>>>>>>>> Content-Length: 0
>>>>>>>>
>>>>>>>> *Jan 15 20:19:35.190: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>
>>>>>>>> Sent:
>>>>>>>>
>>>>>>>> SIP/2.0 100 Trying
>>>>>>>>
>>>>>>>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28e6157f41c
>>>>>>>>
>>>>>>>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>>>>>>>
>>>>>>>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>>>>>>>
>>>>>>>> Date: Tue, 15 Jan 2013 20:19:35 GMT
>>>>>>>>
>>>>>>>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>>>>>>>
>>>>>>>> CSeq: 104 INVITE
>>>>>>>>
>>>>>>>> Allow-Events: telephone-event
>>>>>>>>
>>>>>>>> Server: Cisco-SIPGateway/IOS-12.x
>>>>>>>>
>>>>>>>> Content-Length: 0
>>>>>>>>
>>>>>>>> *Jan 15 20:19:35.194: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>
>>>>>>>> Received:
>>>>>>>>
>>>>>>>> SIP/2.0 100 Trying
>>>>>>>>
>>>>>>>> Via: SIP/2.0/UDP 98.192.104.214:5060
>>>>>>>> ;branch=z9hG4bK69232672;received=98.192.104.214
>>>>>>>>
>>>>>>>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>>>>>>>> >;tag=2E6BC0B0-2268
>>>>>>>>
>>>>>>>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>>>>>>>
>>>>>>>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>>>>>>>
>>>>>>>> CSeq: 105 INVITE
>>>>>>>>
>>>>>>>> Server: Asterisk PBX 1.6.2.13
>>>>>>>>
>>>>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>>>>>>>> INFO
>>>>>>>>
>>>>>>>> Supported: replaces, timer
>>>>>>>>
>>>>>>>> Require: timer
>>>>>>>>
>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>
>>>>>>>> Contact: <sip:17705439047 at 64.154.41.150>
>>>>>>>>
>>>>>>>> Content-Length: 0
>>>>>>>>
>>>>>>>> *Jan 15 20:19:35.194: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>
>>>>>>>> Received:
>>>>>>>>
>>>>>>>> SIP/2.0 200 OK
>>>>>>>>
>>>>>>>> Via: SIP/2.0/UDP 98.192.104.214:5060
>>>>>>>> ;branch=z9hG4bK69232672;received=98.192.104.214
>>>>>>>>
>>>>>>>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>>>>>>>> >;tag=2E6BC0B0-2268
>>>>>>>>
>>>>>>>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>>>>>>>
>>>>>>>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>>>>>>>
>>>>>>>> CSeq: 105 INVITE
>>>>>>>>
>>>>>>>> Server: Asterisk PBX 1.6.2.13
>>>>>>>>
>>>>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>>>>>>>> INFO
>>>>>>>>
>>>>>>>> Supported: replaces, timer
>>>>>>>>
>>>>>>>> Require: timer
>>>>>>>>
>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>
>>>>>>>> Contact: <sip:17705439047 at 64.154.41.150>
>>>>>>>>
>>>>>>>> Content-Type: application/sdp
>>>>>>>>
>>>>>>>> Content-Length: 333
>>>>>>>>
>>>>>>>> v=0
>>>>>>>>
>>>>>>>> o=root 1685873050 1685873054 IN IP4 64.154.41.150
>>>>>>>>
>>>>>>>> s=Asterisk PBX 1.6.2.13
>>>>>>>>
>>>>>>>> c=IN IP4 64.154.41.150
>>>>>>>>
>>>>>>>> t=0 0
>>>>>>>>
>>>>>>>> m=audio 13014 RTP/AVP 3 8 0 18 101
>>>>>>>>
>>>>>>>> a=rtpmap:3 GSM/8000
>>>>>>>>
>>>>>>>> a=rtpmap:8 PCMA/8000
>>>>>>>>
>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>
>>>>>>>> a=rtpmap:18 G729/8000
>>>>>>>>
>>>>>>>> a=fmtp:18 annexb=no
>>>>>>>>
>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>>
>>>>>>>> a=fmtp:101 0-16
>>>>>>>>
>>>>>>>> a=ptime:20
>>>>>>>>
>>>>>>>> a=inactive
>>>>>>>>
>>>>>>>> *Jan 15 20:19:35.198: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>
>>>>>>>> Sent:
>>>>>>>>
>>>>>>>> SIP/2.0 200 OK
>>>>>>>>
>>>>>>>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28e6157f41c
>>>>>>>>
>>>>>>>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>>>>>>>
>>>>>>>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>>>>>>>
>>>>>>>> Date: Tue, 15 Jan 2013 20:19:35 GMT
>>>>>>>>
>>>>>>>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>>>>>>>
>>>>>>>> CSeq: 104 INVITE
>>>>>>>>
>>>>>>>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>>>>>>>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>>>>>>>>
>>>>>>>> Allow-Events: telephone-event
>>>>>>>>
>>>>>>>> Remote-Party-ID: <sip:17705439047 at 10.1.200.1
>>>>>>>> >;party=called;screen=no;privacy=off
>>>>>>>>
>>>>>>>> Contact: <sip:17705439047 at 10.1.200.1:5060;transport=tcp>
>>>>>>>>
>>>>>>>> Supported: replaces
>>>>>>>>
>>>>>>>> Supported: sdp-anat
>>>>>>>>
>>>>>>>> Server: Cisco-SIPGateway/IOS-12.x
>>>>>>>>
>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>
>>>>>>>> Require: timer
>>>>>>>>
>>>>>>>> Supported: timer
>>>>>>>>
>>>>>>>> Content-Type: application/sdp
>>>>>>>>
>>>>>>>> Content-Length: 277
>>>>>>>>
>>>>>>>> v=0
>>>>>>>>
>>>>>>>> o=CiscoSystemsSIP-GW-UserAgent 4444 5481 IN IP4 10.1.200.1
>>>>>>>>
>>>>>>>> s=SIP Call
>>>>>>>>
>>>>>>>> c=IN IP4 10.1.200.1
>>>>>>>>
>>>>>>>> t=0 0
>>>>>>>>
>>>>>>>> m=audio 19514 RTP/AVP 0 101 19
>>>>>>>>
>>>>>>>> c=IN IP4 10.1.200.1
>>>>>>>>
>>>>>>>> a=inactive
>>>>>>>>
>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>
>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>>
>>>>>>>> a=fmtp:101 0-16
>>>>>>>>
>>>>>>>> a=rtpmap:19 CN/8000
>>>>>>>>
>>>>>>>> a=ptime:20
>>>>>>>>
>>>>>>>> *Jan 15 20:19:35.206: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>
>>>>>>>> Received:
>>>>>>>>
>>>>>>>> ACK sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0
>>>>>>>>
>>>>>>>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28f6dca6616
>>>>>>>>
>>>>>>>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>>>>>>>
>>>>>>>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>>>>>>>
>>>>>>>> Date: Tue, 15 Jan 2013 19:57:42 GMT
>>>>>>>>
>>>>>>>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>>>>>>>
>>>>>>>> Max-Forwards: 70
>>>>>>>>
>>>>>>>> CSeq: 104 ACK
>>>>>>>>
>>>>>>>> Allow-Events: presence, kpml
>>>>>>>>
>>>>>>>> Content-Type: application/sdp
>>>>>>>>
>>>>>>>> Content-Length: 243
>>>>>>>>
>>>>>>>> v=0
>>>>>>>>
>>>>>>>> o=CiscoSystemsCCM-SIP 7322 5 IN IP4 10.1.80.10
>>>>>>>>
>>>>>>>> s=SIP Call
>>>>>>>>
>>>>>>>> c=IN IP4 10.1.10.18
>>>>>>>>
>>>>>>>> b=TIAS:64000
>>>>>>>>
>>>>>>>> b=AS:64
>>>>>>>>
>>>>>>>> t=0 0
>>>>>>>>
>>>>>>>> m=audio 21476 RTP/AVP 0 101
>>>>>>>>
>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>
>>>>>>>> a=ptime:20
>>>>>>>>
>>>>>>>> a=inactive
>>>>>>>>
>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>>
>>>>>>>> a=fmtp:101 0-15
>>>>>>>>
>>>>>>>> *Jan 15 20:19:35.210: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>
>>>>>>>> Sent:
>>>>>>>>
>>>>>>>> ACK sip:17705439047 at 64.154.41.150:5060 SIP/2.0
>>>>>>>>
>>>>>>>> Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69246AB
>>>>>>>>
>>>>>>>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>>>>>>>> >;tag=2E6BC0B0-2268
>>>>>>>>
>>>>>>>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>>>>>>>
>>>>>>>> Date: Tue, 15 Jan 2013 20:19:35 GMT
>>>>>>>>
>>>>>>>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>>>>>>>
>>>>>>>> Max-Forwards: 70
>>>>>>>>
>>>>>>>> CSeq: 105 ACK
>>>>>>>>
>>>>>>>> Authorization: Digest username="6784563290",realm="asterisk",uri="
>>>>>>>> sip:17705439047 at 64.154.41.150:5060
>>>>>>>> ",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5
>>>>>>>>
>>>>>>>> Allow-Events: telephone-event
>>>>>>>>
>>>>>>>> Content-Type: application/sdp
>>>>>>>>
>>>>>>>> Content-Length: 265
>>>>>>>>
>>>>>>>> v=0
>>>>>>>>
>>>>>>>> o=CiscoSystemsSIP-GW-UserAgent 3168 2741 IN IP4 98.192.104.214
>>>>>>>>
>>>>>>>> s=SIP Call
>>>>>>>>
>>>>>>>> c=IN IP4 98.192.104.214
>>>>>>>>
>>>>>>>> t=0 0
>>>>>>>>
>>>>>>>> m=audio 19458 RTP/AVP 0 101
>>>>>>>>
>>>>>>>> c=IN IP4 98.192.104.214
>>>>>>>>
>>>>>>>> a=inactive
>>>>>>>>
>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>
>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>>
>>>>>>>> a=fmtp:101 0-16
>>>>>>>>
>>>>>>>> a=ptime:20
>>>>>>>>
>>>>>>>> Cisco3825#
>>>>>>>>
>>>>>>>> Cisco3825#
>>>>>>>>
>>>>>>>>
>>>>>>>> Cisco3825#
>>>>>>>>
>>>>>>>>
>>>>>>>> INVITE sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0
>>>>>>>>
>>>>>>>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28e6157f41c
>>>>>>>>
>>>>>>>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>>>>>>>
>>>>>>>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>>>>>>>
>>>>>>>> Date: Tue, 15 Jan 2013 19:57:42 GMT
>>>>>>>>
>>>>>>>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>>>>>>>
>>>>>>>> Supported: timer,resource-priority,replaces
>>>>>>>>
>>>>>>>> Min-SE: 1800
>>>>>>>>
>>>>>>>> User-Agent: Cisco-CUCM8.6
>>>>>>>>
>>>>>>>> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE,
>>>>>>>> REFER, SUBSCRIBE, NOTIFY
>>>>>>>>
>>>>>>>> CSeq: 104 INVITE
>>>>>>>>
>>>>>>>> Max-Forwards: 70
>>>>>>>>
>>>>>>>> Expires: 180
>>>>>>>>
>>>>>>>> Allow-Events: presence
>>>>>>>>
>>>>>>>> Supported: X-cisco-srtp-fallback
>>>>>>>>
>>>>>>>> Supported: Geolocation
>>>>>>>>
>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>
>>>>>>>> P-Asserted-Identity: "Dane Newman" <sip:6784563290 at 10.1.80.10>
>>>>>>>>
>>>>>>>> Remote-Party-ID: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>> >;party=calling;screen=yes;privacy=off
>>>>>>>>
>>>>>>>> Contact: <sip:6784563290 at 10.1.80.10:5060
>>>>>>>> ;transport=tcp>;video;audio;video
>>>>>>>>
>>>>>>>> Content-Length: 0
>>>>>>>>
>>>>>>>> *Jan 15 20:19:35.170: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>
>>>>>>>> Sent:
>>>>>>>>
>>>>>>>> INVITE sip:17705439047 at 64.154.41.150:5060 SIP/2.0
>>>>>>>>
>>>>>>>> Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69232672
>>>>>>>>
>>>>>>>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>>>>>>>> >;tag=2E6BC0B0-2268
>>>>>>>>
>>>>>>>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>>>>>>>
>>>>>>>> Date: Tue, 15 Jan 2013 20:19:35 GMT
>>>>>>>>
>>>>>>>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>>>>>>>
>>>>>>>> Supported: timer,resource-priority,replaces,sdp-anat
>>>>>>>>
>>>>>>>> Min-SE: 1800
>>>>>>>>
>>>>>>>> Cisco-Guid: 3257897472-0000065536-0000000035-0173015306
>>>>>>>>
>>>>>>>> User-Agent: Cisco-SIPGateway/IOS-12.x
>>>>>>>>
>>>>>>>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>>>>>>>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>>>>>>>>
>>>>>>>> CSeq: 105 INVITE
>>>>>>>>
>>>>>>>> Max-Forwards: 70
>>>>>>>>
>>>>>>>> Timestamp: 1358281175
>>>>>>>>
>>>>>>>> Contact: <sip:6784563290 at 98.192.104.214:5060>
>>>>>>>>
>>>>>>>> Expires: 180
>>>>>>>>
>>>>>>>> Allow-Events: telephone-event
>>>>>>>>
>>>>>>>> Authorization: Digest username="6784563290",realm="asterisk",uri="
>>>>>>>> sip:17705439047 at 64.154.41.150:5060
>>>>>>>> ",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5
>>>>>>>>
>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>
>>>>>>>> Content-Length: 0
>>>>>>>>
>>>>>>>> *Jan 15 20:19:35.190: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>
>>>>>>>> Sent:
>>>>>>>>
>>>>>>>> SIP/2.0 100 Trying
>>>>>>>>
>>>>>>>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28e6157f41c
>>>>>>>>
>>>>>>>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>>>>>>>
>>>>>>>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>>>>>>>
>>>>>>>> Date: Tue, 15 Jan 2013 20:19:35 GMT
>>>>>>>>
>>>>>>>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>>>>>>>
>>>>>>>> CSeq: 104 INVITE
>>>>>>>>
>>>>>>>> Allow-Events: telephone-event
>>>>>>>>
>>>>>>>> Server: Cisco-SIPGateway/IOS-12.x
>>>>>>>>
>>>>>>>> Content-Length: 0
>>>>>>>>
>>>>>>>> *Jan 15 20:19:35.194: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>
>>>>>>>> Received:
>>>>>>>>
>>>>>>>> SIP/2.0 100 Trying
>>>>>>>>
>>>>>>>> Via: SIP/2.0/UDP 98.192.104.214:5060
>>>>>>>> ;branch=z9hG4bK69232672;received=98.192.104.214
>>>>>>>>
>>>>>>>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>>>>>>>> >;tag=2E6BC0B0-2268
>>>>>>>>
>>>>>>>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>>>>>>>
>>>>>>>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>>>>>>>
>>>>>>>> CSeq: 105 INVITE
>>>>>>>>
>>>>>>>> Server: Asterisk PBX 1.6.2.13
>>>>>>>>
>>>>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>>>>>>>> INFO
>>>>>>>>
>>>>>>>> Supported: replaces, timer
>>>>>>>>
>>>>>>>> Require: timer
>>>>>>>>
>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>
>>>>>>>> Contact: <sip:17705439047 at 64.154.41.150>
>>>>>>>>
>>>>>>>> Content-Length: 0
>>>>>>>>
>>>>>>>> *Jan 15 20:19:35.194: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>
>>>>>>>> Received:
>>>>>>>>
>>>>>>>> SIP/2.0 200 OK
>>>>>>>>
>>>>>>>> Via: SIP/2.0/UDP 98.192.104.214:5060
>>>>>>>> ;branch=z9hG4bK69232672;received=98.192.104.214
>>>>>>>>
>>>>>>>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>>>>>>>> >;tag=2E6BC0B0-2268
>>>>>>>>
>>>>>>>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>>>>>>>
>>>>>>>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>>>>>>>
>>>>>>>> CSeq: 105 INVITE
>>>>>>>>
>>>>>>>> Server: Asterisk PBX 1.6.2.13
>>>>>>>>
>>>>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>>>>>>>> INFO
>>>>>>>>
>>>>>>>> Supported: replaces, timer
>>>>>>>>
>>>>>>>> Require: timer
>>>>>>>>
>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>
>>>>>>>> Contact: <sip:17705439047 at 64.154.41.150>
>>>>>>>>
>>>>>>>> Content-Type: application/sdp
>>>>>>>>
>>>>>>>> Content-Length: 333
>>>>>>>>
>>>>>>>> v=0
>>>>>>>>
>>>>>>>> o=root 1685873050 1685873054 IN IP4 64.154.41.150
>>>>>>>>
>>>>>>>> s=Asterisk PBX 1.6.2.13
>>>>>>>>
>>>>>>>> c=IN IP4 64.154.41.150
>>>>>>>>
>>>>>>>> t=0 0
>>>>>>>>
>>>>>>>> m=audio 13014 RTP/AVP 3 8 0 18 101
>>>>>>>>
>>>>>>>> a=rtpmap:3 GSM/8000
>>>>>>>>
>>>>>>>> a=rtpmap:8 PCMA/8000
>>>>>>>>
>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>
>>>>>>>> a=rtpmap:18 G729/8000
>>>>>>>>
>>>>>>>> a=fmtp:18 annexb=no
>>>>>>>>
>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>>
>>>>>>>> a=fmtp:101 0-16
>>>>>>>>
>>>>>>>> a=ptime:20
>>>>>>>>
>>>>>>>> a=inactive
>>>>>>>>
>>>>>>>> *Jan 15 20:19:35.198: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>
>>>>>>>> Sent:
>>>>>>>>
>>>>>>>> SIP/2.0 200 OK
>>>>>>>>
>>>>>>>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28e6157f41c
>>>>>>>>
>>>>>>>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>>>>>>>
>>>>>>>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>>>>>>>
>>>>>>>> Date: Tue, 15 Jan 2013 20:19:35 GMT
>>>>>>>>
>>>>>>>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>>>>>>>
>>>>>>>> CSeq: 104 INVITE
>>>>>>>>
>>>>>>>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>>>>>>>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>>>>>>>>
>>>>>>>> Allow-Events: telephone-event
>>>>>>>>
>>>>>>>> Remote-Party-ID: <sip:17705439047 at 10.1.200.1
>>>>>>>> >;party=called;screen=no;privacy=off
>>>>>>>>
>>>>>>>> Contact: <sip:17705439047 at 10.1.200.1:5060;transport=tcp>
>>>>>>>>
>>>>>>>> Supported: replaces
>>>>>>>>
>>>>>>>> Supported: sdp-anat
>>>>>>>>
>>>>>>>> Server: Cisco-SIPGateway/IOS-12.x
>>>>>>>>
>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>
>>>>>>>> Require: timer
>>>>>>>>
>>>>>>>> Supported: timer
>>>>>>>>
>>>>>>>> Content-Type: application/sdp
>>>>>>>>
>>>>>>>> Content-Length: 277
>>>>>>>>
>>>>>>>> v=0
>>>>>>>>
>>>>>>>> o=CiscoSystemsSIP-GW-UserAgent 4444 5481 IN IP4 10.1.200.1
>>>>>>>>
>>>>>>>> s=SIP Call
>>>>>>>>
>>>>>>>> c=IN IP4 10.1.200.1
>>>>>>>>
>>>>>>>> t=0 0
>>>>>>>>
>>>>>>>> m=audio 19514 RTP/AVP 0 101 19
>>>>>>>>
>>>>>>>> c=IN IP4 10.1.200.1
>>>>>>>>
>>>>>>>> a=inactive
>>>>>>>>
>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>
>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>>
>>>>>>>> a=fmtp:101 0-16
>>>>>>>>
>>>>>>>> a=rtpmap:19 CN/8000
>>>>>>>>
>>>>>>>> a=ptime:20
>>>>>>>>
>>>>>>>> *Jan 15 20:19:35.206: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>
>>>>>>>> Received:
>>>>>>>>
>>>>>>>> ACK sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0
>>>>>>>>
>>>>>>>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28f6dca6616
>>>>>>>>
>>>>>>>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>>>>>>>
>>>>>>>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>>>>>>>
>>>>>>>> Date: Tue, 15 Jan 2013 19:57:42 GMT
>>>>>>>>
>>>>>>>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>>>>>>>
>>>>>>>> Max-Forwards: 70
>>>>>>>>
>>>>>>>> CSeq: 104 ACK
>>>>>>>>
>>>>>>>> Allow-Events: presence, kpml
>>>>>>>>
>>>>>>>> Content-Type: application/sdp
>>>>>>>>
>>>>>>>> Content-Length: 243
>>>>>>>>
>>>>>>>> v=0
>>>>>>>>
>>>>>>>> o=CiscoSystemsCCM-SIP 7322 5 IN IP4 10.1.80.10
>>>>>>>>
>>>>>>>> s=SIP Call
>>>>>>>>
>>>>>>>> c=IN IP4 10.1.10.18
>>>>>>>>
>>>>>>>> b=TIAS:64000
>>>>>>>>
>>>>>>>> b=AS:64
>>>>>>>>
>>>>>>>> t=0 0
>>>>>>>>
>>>>>>>> m=audio 21476 RTP/AVP 0 101
>>>>>>>>
>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>
>>>>>>>> a=ptime:20
>>>>>>>>
>>>>>>>> a=inactive
>>>>>>>>
>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>>
>>>>>>>> a=fmtp:101 0-15
>>>>>>>>
>>>>>>>> *Jan 15 20:19:35.210: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>
>>>>>>>> Sent:
>>>>>>>>
>>>>>>>> ACK sip:17705439047 at 64.154.41.150:5060 SIP/2.0
>>>>>>>>
>>>>>>>> Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69246AB
>>>>>>>>
>>>>>>>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>>>>>>>> >;tag=2E6BC0B0-2268
>>>>>>>>
>>>>>>>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>>>>>>>
>>>>>>>> Date: Tue, 15 Jan 2013 20:19:35 GMT
>>>>>>>>
>>>>>>>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>>>>>>>
>>>>>>>> Max-Forwards: 70
>>>>>>>>
>>>>>>>> CSeq: 105 ACK
>>>>>>>>
>>>>>>>> Authorization: Digest username="6784563290",realm="asterisk",uri="
>>>>>>>> sip:17705439047 at 64.154.41.150:5060
>>>>>>>> ",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5
>>>>>>>>
>>>>>>>> Allow-Events: telephone-event
>>>>>>>>
>>>>>>>> Content-Type: application/sdp
>>>>>>>>
>>>>>>>> Content-Length: 265
>>>>>>>>
>>>>>>>> v=0
>>>>>>>>
>>>>>>>> o=CiscoSystemsSIP-GW-UserAgent 3168 2741 IN IP4 98.192.104.214
>>>>>>>>
>>>>>>>> s=SIP Call
>>>>>>>>
>>>>>>>> c=IN IP4 98.192.104.214
>>>>>>>>
>>>>>>>> t=0 0
>>>>>>>>
>>>>>>>> m=audio 19458 RTP/AVP 0 101
>>>>>>>>
>>>>>>>> c=IN IP4 98.192.104.214
>>>>>>>>
>>>>>>>> a=inactive
>>>>>>>>
>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>
>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>>
>>>>>>>> a=fmtp:101 0-16
>>>>>>>>
>>>>>>>> a=ptime:20
>>>>>>>>
>>>>>>>> Cisco3825#
>>>>>>>>
>>>>>>>>
>>>>>>>> On Tue, Jan 15, 2013 at 2:28 PM, Ryan Ratliff <rratliff at cisco.com>wrote:
>>>>>>>>
>>>>>>>>> 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.
>>>>>>>>>
>>>>>>>>> -Ryan
>>>>>>>>>
>>>>>>>>> On Jan 15, 2013, at 2:11 PM, Dane Newman <dane.newman at gmail.com>
>>>>>>>>> wrote:
>>>>>>>>>
>>>>>>>>> Ryan
>>>>>>>>>
>>>>>>>>> What is the proper debug to use to caputre the useful information?
>>>>>>>>>
>>>>>>>>> Dane
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Tue, Jan 15, 2013 at 12:42 PM, Ryan Ratliff <rratliff at cisco.com
>>>>>>>>> > wrote:
>>>>>>>>>
>>>>>>>>>> Without sip messages I can't get any clues from that.
>>>>>>>>>>
>>>>>>>>>> -Ryan
>>>>>>>>>>
>>>>>>>>>> On Jan 15, 2013, at 12:35 PM, Dane Newman <dane.newman at gmail.com>
>>>>>>>>>> wrote:
>>>>>>>>>>
>>>>>>>>>> Thanks Ryan for the input
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> *On the call when I hold the call the following debug pops
>>>>>>>>>> out....*
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> *Jan 15 17:56:05.246:
>>>>>>>>>> //13939/922252E78D73/SIP/Error/ccsip_api_request_offer: Unable to add
>>>>>>>>>> passthru hdrs to
>>>>>>>>>> container
>>>>>>>>>> SIP: Attribute mid, level 1 instance 1 not found.
>>>>>>>>>> SIP: (13938) Group (a= group line) attribute, level 65535
>>>>>>>>>> instance 1 not found.
>>>>>>>>>> *Jan 15 17:56:05.274:
>>>>>>>>>> //13938/922252E78D73/SIP/Error/ccsip_api_response_answer: Unable to add
>>>>>>>>>> passthru headers to
>>>>>>>>>> container
>>>>>>>>>> SIP: Attribute mid, level 1 instance 1 not found.
>>>>>>>>>> SIP: (13939) Group (a= group line) attribute, level 65535
>>>>>>>>>> instance 1 not found.
>>>>>>>>>> *Jan 15 17:56:05.286:
>>>>>>>>>> //13939/922252E78D73/SIP/Error/ccsip_api_request_offer: Unable to add
>>>>>>>>>> passthru hdrs to
>>>>>>>>>> container
>>>>>>>>>> *Jan 15 17:56:05.302:
>>>>>>>>>> //13938/922252E78D73/SIP/Error/ccsip_api_response_answer: Unable to add
>>>>>>>>>> passthru headers to
>>>>>>>>>> container
>>>>>>>>>> SIP: Attribute mid, level 1 instance 1 not found.
>>>>>>>>>> SIP: (13939) Group (a= group line) attribute, level 65535
>>>>>>>>>> instance 1 not found.
>>>>>>>>>> SIP: Attribute mid, level 1 instance 1 not found.
>>>>>>>>>> *Jan 15 17:56:05.322:
>>>>>>>>>> //13939/922252E78D73/SIP/Error/sipSPIProcessAckMedia: Could not modify QoS
>>>>>>>>>> params for midcall INVITE
>>>>>>>>>>
>>>>>>>>>> *After I try to unhold the call the following debug comes out....
>>>>>>>>>> *
>>>>>>>>>> **
>>>>>>>>>>
>>>>>>>>>> *Jan 15 17:56:18.874:
>>>>>>>>>> //13939/922252E78D73/SIP/Error/ccsip_api_request_offer: Unable to add
>>>>>>>>>> passthru hdrs to
>>>>>>>>>> container
>>>>>>>>>> *Jan 15 17:56:18.894:
>>>>>>>>>> //13938/922252E78D73/SIP/Error/ccsip_api_response_answer: Unable to add
>>>>>>>>>> passthru headers to
>>>>>>>>>> container
>>>>>>>>>> SIP: Attribute mid, level 1 instance 1 not found.
>>>>>>>>>> SIP: (13939) Group (a= group line) attribute, level 65535
>>>>>>>>>> instance 1 not found.
>>>>>>>>>> SIP: Attribute mid, level 1 instance 1 not found.
>>>>>>>>>> *Jan 15 17:56:18.906:
>>>>>>>>>> //13939/922252E78D73/SIP/Error/sipSPIProcessAckMedia: Could not modify QoS
>>>>>>>>>> params for midcall INVITE
>>>>>>>>>> Cisco3825#
>>>>>>>>>> Cisco3825#
>>>>>>>>>> Cisco3825#
>>>>>>>>>>
>>>>>>>>>> On Tue, Jan 15, 2013 at 9:42 AM, Ryan Ratliff <rratliff at cisco.com
>>>>>>>>>> > wrote:
>>>>>>>>>>
>>>>>>>>>>> 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.
>>>>>>>>>>>
>>>>>>>>>>> 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.
>>>>>>>>>>>
>>>>>>>>>>> -Ryan
>>>>>>>>>>>
>>>>>>>>>>> On Jan 14, 2013, at 6:40 PM, Dane Newman <dane.newman at gmail.com>
>>>>>>>>>>> wrote:
>>>>>>>>>>>
>>>>>>>>>>> *Hello Kenneth*
>>>>>>>>>>> **
>>>>>>>>>>> *I have restarted both CUCM servers so this should have
>>>>>>>>>>> restarted the services when the utils system restart happened*
>>>>>>>>>>> **
>>>>>>>>>>>
>>>>>>>>>>> *on my router I see I am using g711 from the debug *
>>>>>>>>>>> **
>>>>>>>>>>> *I ran a debug voip ccapi inout *
>>>>>>>>>>> **
>>>>>>>>>>> *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*
>>>>>>>>>>>
>>>>>>>>>>> *Jan 14 23:47:40.779:
>>>>>>>>>>> //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:
>>>>>>>>>>> Stop Tone On Digit=FALSE, Tone=Null,
>>>>>>>>>>> Tone Direction=Sum Network, Params=0x0, Call Id=12741
>>>>>>>>>>> *Jan 14 23:47:40.783:
>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
>>>>>>>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12741,
>>>>>>>>>>> Source Call Id=12742, Xmit Function=0x64204BAC
>>>>>>>>>>> *Jan 14 23:47:40.783:
>>>>>>>>>>> //12741/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
>>>>>>>>>>> *Jan 14 23:47:40.783: cc_api_get_xcode_stream : 4702
>>>>>>>>>>> *Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>>>>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12741,
>>>>>>>>>>> Source Call Id=12742,
>>>>>>>>>>> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
>>>>>>>>>>> Modem=0x0, Codec Bytes=20, Signal Type=2)
>>>>>>>>>>> *Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>>>>>>>> Caps(Playout Mode=1, Playout Initial=60(ms), Playout
>>>>>>>>>>> Min=40(ms),
>>>>>>>>>>> Playout Max=1000(ms), Fax Nom=300(ms))
>>>>>>>>>>> *Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>>>>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12742,
>>>>>>>>>>> Source Call Id=12741,
>>>>>>>>>>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1),
>>>>>>>>>>> Vad=ON(0x2),
>>>>>>>>>>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num
>>>>>>>>>>> Start=1046)
>>>>>>>>>>> *Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>>>>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12742,
>>>>>>>>>>> Source Call Id=12741,
>>>>>>>>>>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1),
>>>>>>>>>>> Vad=ON(0x2),
>>>>>>>>>>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num
>>>>>>>>>>> Start=1046)
>>>>>>>>>>> *Jan 14 23:47:40.783:
>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>> Event=170, Call Id=12742
>>>>>>>>>>> *Jan 14 23:47:40.783:
>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>> Event Is Sent To Conferenced SPI(s) Directly
>>>>>>>>>>> *Jan 14 23:47:40.783:
>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_call_feature:
>>>>>>>>>>> Feature Type=50, Interface=0xC05A65AC, Call Id=12742
>>>>>>>>>>> *Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>>>>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12742,
>>>>>>>>>>> Source Call Id=12741,
>>>>>>>>>>> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
>>>>>>>>>>> Modem=0x0, Codec Bytes=20, Signal Type=2)
>>>>>>>>>>> *Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>>>>>>>> Caps(Playout Mode=1, Playout Initial=60(ms), Playout
>>>>>>>>>>> Min=40(ms),
>>>>>>>>>>> Playout Max=1000(ms), Fax Nom=300(ms))
>>>>>>>>>>> *Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>>>>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12741,
>>>>>>>>>>> Source Call Id=12742,
>>>>>>>>>>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1),
>>>>>>>>>>> Vad=ON(0x2),
>>>>>>>>>>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num
>>>>>>>>>>> Start=1516)
>>>>>>>>>>> *Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>>>>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12741,
>>>>>>>>>>> Source Call Id=12742,
>>>>>>>>>>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1),
>>>>>>>>>>> Vad=ON(0x2),
>>>>>>>>>>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num
>>>>>>>>>>> Start=1516)
>>>>>>>>>>> *Jan 14 23:47:40.811:
>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>> Event=171, Call Id=12741
>>>>>>>>>>> *Jan 14 23:47:40.811:
>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>> Event Is Sent To Conferenced SPI(s) Directly
>>>>>>>>>>> *Jan 14 23:47:40.815:
>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_call_facility:
>>>>>>>>>>> Interface=0xC05A65AC, Call Id=12742
>>>>>>>>>>> *Jan 14 23:47:40.819:
>>>>>>>>>>> //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:
>>>>>>>>>>> Stop Tone On Digit=FALSE, Tone=Null,
>>>>>>>>>>> Tone Direction=Sum Network, Params=0x0, Call Id=12741
>>>>>>>>>>> *Jan 14 23:47:40.819:
>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>> Event=96, Call Id=12742
>>>>>>>>>>> *Jan 14 23:47:40.819:
>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>> Event Is Sent To Conferenced SPI(s) Directly
>>>>>>>>>>> *Jan 14 23:47:40.839:
>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
>>>>>>>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12742,
>>>>>>>>>>> Source Call Id=12741, Xmit Function=0x64204BAC
>>>>>>>>>>> *Jan 14 23:47:40.839:
>>>>>>>>>>> //12742/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
>>>>>>>>>>> *Jan 14 23:47:40.839: cc_api_get_xcode_stream : 4702
>>>>>>>>>>> *Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>>>>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12742,
>>>>>>>>>>> Source Call Id=12741,
>>>>>>>>>>> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
>>>>>>>>>>> Modem=0x0, Codec Bytes=20, Signal Type=2)
>>>>>>>>>>> *Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>>>>>>>> Caps(Playout Mode=1, Playout Initial=60(ms), Playout
>>>>>>>>>>> Min=40(ms),
>>>>>>>>>>> Playout Max=1000(ms), Fax Nom=300(ms))
>>>>>>>>>>> *Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>>>>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12741,
>>>>>>>>>>> Source Call Id=12742,
>>>>>>>>>>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1),
>>>>>>>>>>> Vad=ON(0x2),
>>>>>>>>>>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num
>>>>>>>>>>> Start=1516)
>>>>>>>>>>> *Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>>>>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12741,
>>>>>>>>>>> Source Call Id=12742,
>>>>>>>>>>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1),
>>>>>>>>>>> Vad=ON(0x2),
>>>>>>>>>>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num
>>>>>>>>>>> Start=1516)
>>>>>>>>>>> *Jan 14 23:47:40.843:
>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>> Event=170, Call Id=12741
>>>>>>>>>>> *Jan 14 23:47:40.843:
>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>> Event Is Sent To Conferenced SPI(s) Directly
>>>>>>>>>>> *Jan 14 23:47:40.859: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>>>>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12741,
>>>>>>>>>>> Source Call Id=12742,
>>>>>>>>>>> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
>>>>>>>>>>> Modem=0x0, Codec Bytes=20, Signal Type=2)
>>>>>>>>>>> *Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>>>>>>>> Caps(Playout Mode=1, Playout Initial=60(ms), Playout
>>>>>>>>>>> Min=40(ms),
>>>>>>>>>>> Playout Max=1000(ms), Fax Nom=300(ms))
>>>>>>>>>>> *Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>>>>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12742,
>>>>>>>>>>> Source Call Id=12741,
>>>>>>>>>>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1),
>>>>>>>>>>> Vad=ON(0x2),
>>>>>>>>>>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num
>>>>>>>>>>> Start=3996)
>>>>>>>>>>> *Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>>>>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12742,
>>>>>>>>>>> Source Call Id=12741,
>>>>>>>>>>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1),
>>>>>>>>>>> Vad=ON(0x2),
>>>>>>>>>>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num
>>>>>>>>>>> Start=3996)
>>>>>>>>>>> *Jan 14 23:47:40.863:
>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>> Event=171, Call Id=12742
>>>>>>>>>>> *Jan 14 23:47:40.863:
>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>> Event Is Sent To Conferenced SPI(s) Directly
>>>>>>>>>>> *Jan 14 23:47:40.863:
>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_call_facility:
>>>>>>>>>>> Interface=0xC05A65AC, Call Id=12742
>>>>>>>>>>> Cisco3825#
>>>>>>>>>>> Cisco3825#
>>>>>>>>>>> Cisco3825#
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> *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.*
>>>>>>>>>>>
>>>>>>>>>>> *Jan 14 23:47:40.779:
>>>>>>>>>>> //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:
>>>>>>>>>>> Stop Tone On Digit=FALSE, Tone=Null,
>>>>>>>>>>> Tone Direction=Sum Network, Params=0x0, Call Id=12741
>>>>>>>>>>> *Jan 14 23:47:40.783:
>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
>>>>>>>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12741,
>>>>>>>>>>> Source Call Id=12742, Xmit Function=0x64204BAC
>>>>>>>>>>> *Jan 14 23:47:40.783:
>>>>>>>>>>> //12741/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
>>>>>>>>>>> *Jan 14 23:47:40.783: cc_api_get_xcode_stream : 4702
>>>>>>>>>>> *Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>>>>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12741,
>>>>>>>>>>> Source Call Id=12742,
>>>>>>>>>>> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
>>>>>>>>>>> Modem=0x0, Codec Bytes=20, Signal Type=2)
>>>>>>>>>>> *Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>>>>>>>> Caps(Playout Mode=1, Playout Initial=60(ms), Playout
>>>>>>>>>>> Min=40(ms),
>>>>>>>>>>> Playout Max=1000(ms), Fax Nom=300(ms))
>>>>>>>>>>> *Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>>>>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12742,
>>>>>>>>>>> Source Call Id=12741,
>>>>>>>>>>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1),
>>>>>>>>>>> Vad=ON(0x2),
>>>>>>>>>>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num
>>>>>>>>>>> Start=1046)
>>>>>>>>>>> *Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>>>>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12742,
>>>>>>>>>>> Source Call Id=12741,
>>>>>>>>>>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1),
>>>>>>>>>>> Vad=ON(0x2),
>>>>>>>>>>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num
>>>>>>>>>>> Start=1046)
>>>>>>>>>>> *Jan 14 23:47:40.783:
>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>> Event=170, Call Id=12742
>>>>>>>>>>> *Jan 14 23:47:40.783:
>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>> Event Is Sent To Conferenced SPI(s) Directly
>>>>>>>>>>> *Jan 14 23:47:40.783:
>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_call_feature:
>>>>>>>>>>> Feature Type=50, Interface=0xC05A65AC, Call Id=12742
>>>>>>>>>>> *Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>>>>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12742,
>>>>>>>>>>> Source Call Id=12741,
>>>>>>>>>>> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
>>>>>>>>>>> Modem=0x0, Codec Bytes=20, Signal Type=2)
>>>>>>>>>>> *Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>>>>>>>> Caps(Playout Mode=1, Playout Initial=60(ms), Playout
>>>>>>>>>>> Min=40(ms),
>>>>>>>>>>> Playout Max=1000(ms), Fax Nom=300(ms))
>>>>>>>>>>> *Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>>>>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12741,
>>>>>>>>>>> Source Call Id=12742,
>>>>>>>>>>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1),
>>>>>>>>>>> Vad=ON(0x2),
>>>>>>>>>>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num
>>>>>>>>>>> Start=1516)
>>>>>>>>>>> *Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>>>>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12741,
>>>>>>>>>>> Source Call Id=12742,
>>>>>>>>>>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1),
>>>>>>>>>>> Vad=ON(0x2),
>>>>>>>>>>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num
>>>>>>>>>>> Start=1516)
>>>>>>>>>>> *Jan 14 23:47:40.811:
>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>> Event=171, Call Id=12741
>>>>>>>>>>> *Jan 14 23:47:40.811:
>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>> Event Is Sent To Conferenced SPI(s) Directly
>>>>>>>>>>> *Jan 14 23:47:40.815:
>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_call_facility:
>>>>>>>>>>> Interface=0xC05A65AC, Call Id=12742
>>>>>>>>>>> *Jan 14 23:47:40.819:
>>>>>>>>>>> //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:
>>>>>>>>>>> Stop Tone On Digit=FALSE, Tone=Null,
>>>>>>>>>>> Tone Direction=Sum Network, Params=0x0, Call Id=12741
>>>>>>>>>>> *Jan 14 23:47:40.819:
>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>> Event=96, Call Id=12742
>>>>>>>>>>> *Jan 14 23:47:40.819:
>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>> Event Is Sent To Conferenced SPI(s) Directly
>>>>>>>>>>> *Jan 14 23:47:40.839:
>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
>>>>>>>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12742,
>>>>>>>>>>> Source Call Id=12741, Xmit Function=0x64204BAC
>>>>>>>>>>> *Jan 14 23:47:40.839:
>>>>>>>>>>> //12742/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
>>>>>>>>>>> *Jan 14 23:47:40.839: cc_api_get_xcode_stream : 4702
>>>>>>>>>>> *Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>>>>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12742,
>>>>>>>>>>> Source Call Id=12741,
>>>>>>>>>>> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
>>>>>>>>>>> Modem=0x0, Codec Bytes=20, Signal Type=2)
>>>>>>>>>>> *Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>>>>>>>> Caps(Playout Mode=1, Playout Initial=60(ms), Playout
>>>>>>>>>>> Min=40(ms),
>>>>>>>>>>> Playout Max=1000(ms), Fax Nom=300(ms))
>>>>>>>>>>> *Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>>>>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12741,
>>>>>>>>>>> Source Call Id=12742,
>>>>>>>>>>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1),
>>>>>>>>>>> Vad=ON(0x2),
>>>>>>>>>>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num
>>>>>>>>>>> Start=1516)
>>>>>>>>>>> *Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>>>>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12741,
>>>>>>>>>>> Source Call Id=12742,
>>>>>>>>>>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1),
>>>>>>>>>>> Vad=ON(0x2),
>>>>>>>>>>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num
>>>>>>>>>>> Start=1516)
>>>>>>>>>>> *Jan 14 23:47:40.843:
>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>> Event=170, Call Id=12741
>>>>>>>>>>> *Jan 14 23:47:40.843:
>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>> Event Is Sent To Conferenced SPI(s) Directly
>>>>>>>>>>> *Jan 14 23:47:40.859: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>>>>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12741,
>>>>>>>>>>> Source Call Id=12742,
>>>>>>>>>>> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
>>>>>>>>>>> Modem=0x0, Codec Bytes=20, Signal Type=2)
>>>>>>>>>>> *Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>>>>>>>> Caps(Playout Mode=1, Playout Initial=60(ms), Playout
>>>>>>>>>>> Min=40(ms),
>>>>>>>>>>> Playout Max=1000(ms), Fax Nom=300(ms))
>>>>>>>>>>> *Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>>>>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12742,
>>>>>>>>>>> Source Call Id=12741,
>>>>>>>>>>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1),
>>>>>>>>>>> Vad=ON(0x2),
>>>>>>>>>>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num
>>>>>>>>>>> Start=3996)
>>>>>>>>>>> *Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>>>>>>>> Destination Interface=0xC05A65AC, Destination Call Id=12742,
>>>>>>>>>>> Source Call Id=12741,
>>>>>>>>>>> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1),
>>>>>>>>>>> Vad=ON(0x2),
>>>>>>>>>>> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num
>>>>>>>>>>> Start=3996)
>>>>>>>>>>> *Jan 14 23:47:40.863:
>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>> Event=171, Call Id=12742
>>>>>>>>>>> *Jan 14 23:47:40.863:
>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>> Event Is Sent To Conferenced SPI(s) Directly
>>>>>>>>>>> *Jan 14 23:47:40.863:
>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_call_facility:
>>>>>>>>>>> Interface=0xC05A65AC, Call Id=12742
>>>>>>>>>>> Cisco3825#
>>>>>>>>>>> Cisco3825#
>>>>>>>>>>> Cisco3825#
>>>>>>>>>>>
>>>>>>>>>>> On Mon, Jan 14, 2013 at 6:20 PM, Kenneth Hayes <
>>>>>>>>>>> kennethwhayes at gmail.com> wrote:
>>>>>>>>>>>
>>>>>>>>>>>> Have you also restarted the Cisco IP Media Services?
>>>>>>>>>>>>
>>>>>>>>>>>> Sent from my iPhone
>>>>>>>>>>>>
>>>>>>>>>>>> On Jan 14, 2013, at 6:12 PM, Dane Newman <dane.newman at gmail.com>
>>>>>>>>>>>> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>> My ITSP will only support 711ulaw for me currently I believe.
>>>>>>>>>>>> They hard coded it with me when I was initially setting it up.
>>>>>>>>>>>>
>>>>>>>>>>>> Do you think this could be a codec issue? How would I go about
>>>>>>>>>>>> identifying if it is?
>>>>>>>>>>>>
>>>>>>>>>>>> Dane
>>>>>>>>>>>>
>>>>>>>>>>>> On Mon, Jan 14, 2013 at 6:09 PM, Kenneth Hayes <
>>>>>>>>>>>> kennethwhayes at gmail.com> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>> Have you tried different audio codecs?
>>>>>>>>>>>>>
>>>>>>>>>>>>> Sent from my iPhone
>>>>>>>>>>>>>
>>>>>>>>>>>>> On Jan 14, 2013, at 6:06 PM, Dane Newman <
>>>>>>>>>>>>> dane.newman at gmail.com> wrote:
>>>>>>>>>>>>>
>>>>>>>>>>>>> Ryan (sorry I forgot to reply to all)
>>>>>>>>>>>>>
>>>>>>>>>>>>> Thanks for the Reply
>>>>>>>>>>>>> Oddly enough we are.
>>>>>>>>>>>>> This probably has something to do with MOH in general?
>>>>>>>>>>>>>
>>>>>>>>>>>>> Internally when I user puts another user on hold everything
>>>>>>>>>>>>> works. No MOH plays and they can hold and unhold the call just fine.
>>>>>>>>>>>>> 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.
>>>>>>>>>>>>>
>>>>>>>>>>>>> On Mon, Jan 14, 2013 at 5:25 PM, Ryan Ratliff <
>>>>>>>>>>>>> rratliff at cisco.com> wrote:
>>>>>>>>>>>>>
>>>>>>>>>>>>>> Do you get similar behavior if you just hold and resume the
>>>>>>>>>>>>>> call outside SNR features?
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> -Ryan
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> On Jan 14, 2013, at 4:18 PM, Dane Newman <
>>>>>>>>>>>>>> dane.newman at gmail.com> wrote:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Using keyboard-interactive authentication.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Password:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Cisco3825#
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Cisco3825#sh ver
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Cisco IOS Software, 3800 Software
>>>>>>>>>>>>>> (C3825-ADVENTERPRISEK9_IVS_LI-M), Version 15.1
>>>>>>>>>>>>>> (4)M5, RELEASE SOFTWARE (fc1)
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Technical Support: http://www.cisco.com/techsupport
>>>>>>>>>>>>>> Copyright (c) 1986-2012 by Cisco Systems, Inc.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Compiled Tue 04-Sep-12 17:25 by prod_rel_team
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> ROM: System Bootstrap, Version 12.4(13r)T10, RELEASE SOFTWARE
>>>>>>>>>>>>>> (fc1)
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Cisco3825 uptime is 1 week, 1 day, 1 hour, 38 minutes
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> System returned to ROM by power-on
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> System image file is
>>>>>>>>>>>>>> "flash:c3825-adventerprisek9_ivs_li-mz.151-4.M5.bin"
>>>>>>>>>>>>>> Last reload type: Normal Reload
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> This product contains cryptographic features and is subject
>>>>>>>>>>>>>> to United
>>>>>>>>>>>>>> States and local country laws governing import, export,
>>>>>>>>>>>>>> transfer and
>>>>>>>>>>>>>> use. Delivery of Cisco cryptographic products does not imply
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> third-party authority to import, export, distribute or use
>>>>>>>>>>>>>> encryption.
>>>>>>>>>>>>>> Importers, exporters, distributors and users are responsible
>>>>>>>>>>>>>> for
>>>>>>>>>>>>>> compliance with U.S. and local country laws. By using this
>>>>>>>>>>>>>> product you
>>>>>>>>>>>>>> agree to comply with applicable laws and regulations. If you
>>>>>>>>>>>>>> are unable
>>>>>>>>>>>>>> to comply with U.S. and local laws, return this product
>>>>>>>>>>>>>> immediately.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> A summary of U.S. laws governing Cisco cryptographic products
>>>>>>>>>>>>>> may be found at:
>>>>>>>>>>>>>> http://www.cisco.com/wwl/export/crypto/tool/stqrg.html
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> If you require further assistance please contact us by
>>>>>>>>>>>>>> sending email to
>>>>>>>>>>>>>> export at cisco.com.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Cisco 3825 (revision 1.2) with 1011712K/36864K bytes of
>>>>>>>>>>>>>> memory.
>>>>>>>>>>>>>> Processor board ID FTX1237A1T0
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> 2 Gigabit Ethernet interfaces
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> 2 Channelized T1/PRI ports
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> 1 Virtual Private Network (VPN) Module
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> DRAM configuration is 64 bits wide with parity enabled.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> 479K bytes of NVRAM.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> 500472K bytes of ATA System CompactFlash (Read/Write)
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> License Info:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> License UDI:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> -------------------------------------------------
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Device# PID SN
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Sent from my mobile device
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> On Jan 14, 2013, at 4:11 PM, Kenneth Hayes <
>>>>>>>>>>>>>> kennethwhayes at gmail.com> wrote:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> What version of code are you running on the CUBE?
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Sent from my iPhone
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> On Jan 14, 2013, at 3:43 PM, Dane Newman <
>>>>>>>>>>>>>> dane.newman at gmail.com> wrote:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Hello
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> 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.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> 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.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> My topology is as follows..
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> PSDN --> SIP TRUNK FROM ITSP --> 3825 CUBE --->CUCM
>>>>>>>>>>>>>> -->DESKPHOHE
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Calls are sent back out the SIP trunk to the ITSP when using
>>>>>>>>>>>>>> mobile connect/snr.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Does anyone have any ideas how I can make 2 way voice happen
>>>>>>>>>>>>>> instead of the hold music when the calls are picked up?
>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>> cisco-voip mailing list
>>>>>>>>>>>>>> cisco-voip at puck.nether.net
>>>>>>>>>>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>> cisco-voip mailing list
>>>>>>>>>>>>>> cisco-voip at puck.nether.net
>>>>>>>>>>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> cisco-voip mailing list
>>>>>>> cisco-voip at puck.nether.net
>>>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>
>>> <moh.jpg>_______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>>
>>
>>
>
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