[cisco-voip] Mobility Issue
Dane Newman
dane.newman at gmail.com
Wed Jan 16 11:26:50 EST 2013
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
>>>>
>>>>
>>>
>>
>
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