[cisco-voip] Mobility Issue
Dane Newman
dane.newman at gmail.com
Tue Jan 15 20:39:09 EST 2013
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
>>
>>
>
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