[cisco-voip] Mobility Issue
Dane Newman
dane.newman at gmail.com
Wed Jan 16 13:27:17 EST 2013
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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