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