[cisco-voip] Mobility Issue

[Nick Matthews]

Just for reference - you shouldn't need MTP for this call to work. However,
it's often a way to fix broken call flows.  Most ITSP's also don't run on
Asterisks, which from my experience is used more often as a SMB IP PBX.
>From what I understand it's quite customizable, so their particular
settings for it may be interfering with MoH.

-nick


On Wed, Jan 16, 2013 at 1:56 PM, Dane Newman <dane.newman at gmail.com> wrote:

>
> All
>
> The issue has been resolved I believe (I feel very dumb)....Thank you so
> much all for your assistance!
>
> Ryan and Nick was 100% correct my MTP was not working correctly.  I found
> that it was miss configured and in the default device pool instead of the
> correct one.  Once I put it in the correct device pools I could hold calls
> and everything works correctly now.
> On Wed, Jan 16, 2013 at 1:27 PM, Dane Newman <dane.newman at gmail.com>wrote:
>
>> Ryan
>>
>> Thank you again for responding and sharing your wisdom.
>>
>> I am unsure if I am doing the test to verify your thoughts correctly but
>> I debugged the SIP messages once again.  I placed a call and picked up the
>> call with and without the MTP checkbox checked on the SIP trunk in cucm.
>>
>> I then did a search for the IP address of my IP phone 10.1.10.18.
>>
>> In both debugs with and without it checked I found my IP phone's IP
>> address in the debug.  I believe this might verify your idea that the MTP
>> is not working correctly?
>>
>> *With MTP unchecked on the SIP TRUNk*
>> *Jan 16 18:33:28.384: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>> Received:
>> ACK sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0
>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK4e47f806d
>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>> >;tag=673~d8eefedd-7473-4e00-a4a0-ce8f65d30766-31145612
>> To: <sip:17705439047 at 10.1.200.1>;tag=333141D8-1492
>> Date: Wed, 16 Jan 2013 18:11:29 GMT
>> Call-ID: 24e56400-f61ed51-12-a50010a at 10.1.80.10
>> Max-Forwards: 70
>> CSeq: 101 ACK
>> Allow-Events: presence, kpml
>> Content-Type: application/sdp
>> Content-Length: 230
>> v=0
>> o=CiscoSystemsCCM-SIP 673 1 IN IP4 10.1.80.10
>> s=SIP Call
>> c=IN IP4 10.1.10.18
>> b=TIAS:64000
>> b=AS:64
>> t=0 0
>> m=audio 20116 RTP/AVP 0 101
>> a=rtpmap:0 PCMU/8000
>> a=ptime:20
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>>
>> *With MTP checked on the SIP TRUNk*
>>
>> *Jan 16 18:39:09.732: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>> Received:
>> ACK sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0
>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK60b810f25
>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>> >;tag=723~d8eefedd-7473-4e00-a4a0-ce8f65d30766-31145637
>> To: <sip:17705439047 at 10.1.200.1>;tag=3336625C-1CBB
>> Date: Wed, 16 Jan 2013 18:17:05 GMT
>> Call-ID: ed2aec00-f61eea1-16-a50010a at 10.1.80.10
>> Max-Forwards: 70
>> CSeq: 101 ACK
>> Allow-Events: presence, kpml
>> Content-Type: application/sdp
>> Content-Length: 230
>> v=0
>> o=CiscoSystemsCCM-SIP 723 2 IN IP4 10.1.80.10
>> s=SIP Call
>> c=IN IP4 10.1.10.18
>> b=TIAS:64000
>> b=AS:64
>> t=0 0
>> m=audio 16452 RTP/AVP 0 101
>> a=rtpmap:0 PCMU/8000
>> a=ptime:20
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>>
>>
>>
>> On Wed, Jan 16, 2013 at 12:59 PM, Ryan Ratliff <rratliff at cisco.com>wrote:
>>
>>> What they are doing is interpreting us sending them an Invite with
>>> a=inactive in the SDP as the Cisco phone putting them on hold.  That is a
>>> valid assumption.   What is incorrect (IMO) is them assuming that they need
>>> to generate MOH.  CUCM is the one initiating the hold, it will be the one
>>> to play MOH or not, based upon the way you configure it.   When they
>>> respond to our inactive SDP with one of their own, CUCM sees that as them
>>> putting us on hold.  The end result you see is that in order to get the
>>> call off of hold both sides need to resume it, which isn't happening.
>>>
>>> I still think you need to look at an active call (no hold) on your CUBE
>>> to see where it's sending media to on the internal side.  That IP address
>>> is going to be an MTP (CUCM server, hardware resource) or an IP phone.  If
>>> it's directly to a phone you may as well remove "MTP Required" on the trunk
>>> because you're not actually allocating an MTP.
>>>
>>> -Ryan
>>>
>>> On Jan 16, 2013, at 12:35 PM, Dane Newman <dane.newman at gmail.com> wrote:
>>>
>>> All
>>>
>>> Thank you for the infromation you are providing me on this thread.  It
>>> is a great learning exp for me.
>>>
>>> I just got off the phone with the ITSP and they confirmed the MOH was
>>> coming from them.   They believe if I am typing this correctly  they
>>> (ITSP) claim when I press the hold button I am sending an invite message
>>> and that is resulting in the MOH being played by them.
>>>
>>> I assumed when I pressed the hold key on an external call CUCM would
>>> continue to send the uninterupted audio stream with the MOH mixed in?
>>>
>>> I have reset the trunk and rebooted cucm also...
>>>
>>> Thanks again for the assistance and advice it's much appericated
>>>
>>> Dane
>>>
>>>
>>>
>>> On Wed, Jan 16, 2013 at 12:18 PM, Ryan Ratliff <rratliff at cisco.com>wrote:
>>>
>>>> Having the MOH servers registered is step 1 of about 10 that have to
>>>> happen for MOH to be allocated for the call.
>>>> In the SIP signaling you sent there was no possibility you heard MOH
>>>> from CUCM because the media stream never went back to active after the
>>>> hold.  Can your Asterix play MOH?
>>>>
>>>> You need to look at ccm traces to debug this further.  If you can't
>>>> figure it out, then it's time to call TAC.
>>>>
>>>> You should also take a look at your active call before it's getting put
>>>> on hold.  You've got MTP Required set on the SIP trunk, but if an MTP was
>>>> really getting allocated I don't believe we'd ever set the media inactive
>>>> to the trunk, we'd be telling the MTP about media changes and the trunk
>>>> would just see one media stream to the MTP for the entire call.   At the
>>>> same time if we tried to allocate an MTP but failed, that usually ends up
>>>> disabling supplementary services for the call, which means you never would
>>>> have been allowed to hold in the first place.   It's certainly possible
>>>> that has changed for SIP EO MTPs but for now what is in that signaling
>>>> doesn't jive with what you've sent in your config and description of
>>>> events.
>>>>
>>>> Have you tried resetting the SIP trunk in CUCM yet?
>>>>
>>>> -Ryan
>>>>
>>>> On Jan 16, 2013, at 11:26 AM, Dane Newman <dane.newman at gmail.com>
>>>> wrote:
>>>>
>>>> Yes as per the screen shot the MOH servers are registered.  How do In
>>>> find the audio bit rate?  its just the default moh file I didnt change any
>>>> settings
>>>>
>>>> On Wed, Jan 16, 2013 at 10:20 AM, Kenneth Hayes <
>>>> kennethwhayes at gmail.com> wrote:
>>>>
>>>>> So have you looked in your media resources under music on hold server
>>>>> configurations to make sure it's registered to the right UCM? Also what
>>>>> audio bit rate is your MOH file?
>>>>>
>>>>> Sent from my iPad
>>>>>
>>>>> On Jan 16, 2013, at 10:14 AM, Nick Matthews <matthnick at gmail.com>
>>>>> wrote:
>>>>>
>>>>> I'm not sure at this point, I'll let some of the CUCM experts comment.
>>>>> It's possible during the hold conversation CUCM always sends delayed offer,
>>>>> but I don't have some good traces in front of me to confirm.
>>>>>
>>>>> You can check the original invite CUCM sends to see if there's SDP in
>>>>> that, and it would confirm the MTP is being allocated. If it's sending the
>>>>> INVITE without SDP, your MRG/MRGL or resources are misconfigured or in use.
>>>>>
>>>>> -nick
>>>>>
>>>>>
>>>>> On Tue, Jan 15, 2013 at 8:39 PM, Dane Newman <dane.newman at gmail.com>wrote:
>>>>>
>>>>>> Nick
>>>>>>
>>>>>> Thanks for the assistance.
>>>>>>
>>>>>> This is the first time I am setting up a direct sip connection from
>>>>>> cucm to cube.  I am used to making it an h323 connection.  Attached are
>>>>>> screen shots of my trunk setup.  MTP is checked off I believe already.
>>>>>> Is there a best way to go about troubleshooting cucm to figure out why its
>>>>>> not setting the stream back to active?
>>>>>>
>>>>>> On Tue, Jan 15, 2013 at 7:56 PM, Nick Matthews <matthnick at gmail.com>wrote:
>>>>>>
>>>>>>> It looks like CUCM isn't setting the stream back to active after
>>>>>>> putting it on hold. It sends the re-invite, and the 200 OK from the ITSP
>>>>>>> has the SDP continued with a=inactive.
>>>>>>>
>>>>>>> I don't have some good traces in front of me, but somewhere it needs
>>>>>>> to take that off. I don't think Asterisks is acting incorrectly by
>>>>>>> responding to a delayed offer INVITE that was previously a=inactive with
>>>>>>> a=inactive.
>>>>>>>
>>>>>>> What's also odd is that CUCM is sending the exact same INVITE after
>>>>>>> the first one completes, for both the hold and the resume. The CSeq isn't
>>>>>>> increasing, which I would expect it to.
>>>>>>>
>>>>>>> If you were to check 'MTP' required it may work around the problem,
>>>>>>> but I don't consider inserting MTP's to be a best practice.
>>>>>>>
>>>>>>> -nick
>>>>>>>
>>>>>>>
>>>>>>> On Tue, Jan 15, 2013 at 3:42 PM, Kenneth Hayes <
>>>>>>> kennethwhayes at gmail.com> wrote:
>>>>>>>
>>>>>>>> Bind your media and source to your outbound interface on your voice
>>>>>>>> service voip.
>>>>>>>>
>>>>>>>> Sent from my iPhone
>>>>>>>>
>>>>>>>> On Jan 15, 2013, at 3:39 PM, Dane Newman <dane.newman at gmail.com>
>>>>>>>> wrote:
>>>>>>>>
>>>>>>>> Below is a show run from the router
>>>>>>>>
>>>>>>>>
>>>>>>>> [OK]
>>>>>>>> Cisco3825#sh run
>>>>>>>> Building configuration...
>>>>>>>>
>>>>>>>> Current configuration : 10183 bytes
>>>>>>>> !
>>>>>>>> ! Last configuration change at 20:49:24 UTC Tue Jan 15 2013 by
>>>>>>>> dnewman
>>>>>>>> version 15.1
>>>>>>>> service timestamps debug datetime msec
>>>>>>>> service timestamps log datetime msec
>>>>>>>> no service password-encryption
>>>>>>>> !
>>>>>>>> hostname Cisco3825
>>>>>>>> !
>>>>>>>> boot-start-marker
>>>>>>>> boot-end-marker
>>>>>>>> !
>>>>>>>> !
>>>>>>>> !
>>>>>>>> aaa new-model
>>>>>>>> !
>>>>>>>> !
>>>>>>>> aaa authentication login default local
>>>>>>>> aaa authentication login vpnauth local
>>>>>>>> aaa authorization exec default local
>>>>>>>> aaa authorization network default local
>>>>>>>> aaa authorization network vpnauth local
>>>>>>>> !
>>>>>>>> !
>>>>>>>> !
>>>>>>>> !
>>>>>>>> !
>>>>>>>> aaa session-id common
>>>>>>>> !
>>>>>>>> no network-clock-participate wic 0
>>>>>>>> !
>>>>>>>> dot11 syslog
>>>>>>>> ip source-route
>>>>>>>> !
>>>>>>>> ip cef
>>>>>>>> !
>>>>>>>> !
>>>>>>>> !
>>>>>>>> !
>>>>>>>> ip domain name datasc.local
>>>>>>>> ip inspect udp idle-time 1800
>>>>>>>> no ipv6 cef
>>>>>>>> !
>>>>>>>> multilink bundle-name authenticated
>>>>>>>> !
>>>>>>>> !
>>>>>>>> !
>>>>>>>> !
>>>>>>>> !
>>>>>>>> voice-card 0
>>>>>>>>  dsp services dspfarm
>>>>>>>> !
>>>>>>>> !
>>>>>>>> !
>>>>>>>> voice service voip
>>>>>>>>  ip address trusted list
>>>>>>>>   ipv4 64.154.41.150 255.255.255.255
>>>>>>>>  allow-connections sip to sip
>>>>>>>>  fax protocol pass-through g711ulaw
>>>>>>>>  sip
>>>>>>>> !
>>>>>>>> !
>>>>>>>> !
>>>>>>>> !
>>>>>>>> voice translation-rule 1
>>>>>>>>  rule 1 /6784604564/ /200/
>>>>>>>>  rule 2 /6784563290/ /100/
>>>>>>>>  rule 3 /6784563291/ /101/
>>>>>>>>  rule 4 /6784563292/ /102/
>>>>>>>>  rule 5 /6784563293/ /103/
>>>>>>>>  rule 6 /6784563294/ /104/
>>>>>>>>  rule 7 /6784563295/ /105/
>>>>>>>>  rule 8 /6784563296/ /106/
>>>>>>>>  rule 9 /6784563297/ /107/
>>>>>>>>  rule 10 /6784563298/ /108/
>>>>>>>>  rule 11 /6784563299/ /109/
>>>>>>>>  rule 12 /6784604565/ /125/
>>>>>>>> !
>>>>>>>> !
>>>>>>>> voice translation-profile incomingdid
>>>>>>>>  translate called 1
>>>>>>>> !
>>>>>>>> !
>>>>>>>> crypto pki token default removal timeout 0
>>>>>>>> !
>>>>>>>> crypto pki trustpoint selfsigned
>>>>>>>>  enrollment selfsigned
>>>>>>>>  subject-name CN=connect.datasc.com
>>>>>>>>  revocation-check crl
>>>>>>>> !
>>>>>>>> !
>>>>>>>> crypto pki certificate chain selfsigned
>>>>>>>>  certificate self-signed 02
>>>>>>>>   30820251 308201BA A0030201 02020102 300D0609 2A864886 F70D0101
>>>>>>>> 05050030
>>>>>>>>   44311B30 19060355 04031312 636F6E6E 6563742E 64617461 73632E63
>>>>>>>> 6F6D3125
>>>>>>>>   30230609 2A864886 F70D0109 02161643 6973636F 33383235 2E646174
>>>>>>>> 6173632E
>>>>>>>>   6C6F6361 6C301E17 0D313231 32323831 39313531 395A170D 32303031
>>>>>>>> 30313030
>>>>>>>>   30303030 5A304431 1B301906 03550403 1312636F 6E6E6563 742E6461
>>>>>>>> 74617363
>>>>>>>>   2E636F6D 31253023 06092A86 4886F70D 01090216 16436973 636F3338
>>>>>>>> 32352E64
>>>>>>>>   61746173 632E6C6F 63616C30 819F300D 06092A86 4886F70D 01010105
>>>>>>>> 0003818D
>>>>>>>>   00308189 02818100 D9A99B41 8B70C82F 28072967 376E13E8 8F7FC2C2
>>>>>>>> 7729B93E
>>>>>>>>   DDAE31A0 F3613381 78B43E11 5144BE88 DC2FDE14 0035A104 0BBFAEA0
>>>>>>>> 9A190598
>>>>>>>>   19A124B1 2C4A8EA2 04253BA1 C829EF07 CD0E848D E7AA5269 459449C4
>>>>>>>> FABF9CA9
>>>>>>>>   BC5AF8ED 84FCD99B 260C2B75 57887863 7BB310FB 2C8D1506 FE91FEAC
>>>>>>>> 4EDD1712
>>>>>>>>   A7AFD2C1 BF21C59D 02030100 01A35330 51300F06 03551D13 0101FF04
>>>>>>>> 05300301
>>>>>>>>   01FF301F 0603551D 23041830 16801475 02C4FB04 4FB3F748 B4012EC5
>>>>>>>> 8A571236
>>>>>>>>   A190CB30 1D060355 1D0E0416 04147502 C4FB044F B3F748B4 012EC58A
>>>>>>>> 571236A1
>>>>>>>>   90CB300D 06092A86 4886F70D 01010505 00038181 00C2B167 E583F6D8
>>>>>>>> 8B742D4F
>>>>>>>>   49D27AAD 7EF4E64F 0B5CA5A3 944E8CC7 499A706F AB22283B AE5913A1
>>>>>>>> B22BBB20
>>>>>>>>   E7CF6F9F 41CDD870 1B474E58 9537C1FA 2D93BE4F 4276E9CE 61AE18D3
>>>>>>>> EF724BD9
>>>>>>>>   33878860 4B3627C0 448C652D 03D4C142 BA3291A3 DDE0C4DD C6ED06C3
>>>>>>>> 12E45933
>>>>>>>>   F1EE5CC2 B5B6CC20 C65AB313 76966F14 AA25CC8D 2A
>>>>>>>>         quit
>>>>>>>> !
>>>>>>>> !
>>>>>>>> license udi pid CISCO3825 sn FTX1237A1T0
>>>>>>>> username xxxxxxx privilege 15 secret  xxxxxx
>>>>>>>> !
>>>>>>>> redundancy
>>>>>>>> !
>>>>>>>> !
>>>>>>>> controller T1 0/0/0
>>>>>>>> !
>>>>>>>> controller T1 0/0/1
>>>>>>>> !
>>>>>>>> ip ssh version 2
>>>>>>>> !
>>>>>>>> !
>>>>>>>> crypto isakmp policy 10
>>>>>>>>  encr aes
>>>>>>>>  authentication pre-share
>>>>>>>>  group 2
>>>>>>>> crypto isakmp key Recoil90 address 0.0.0.0 0.0.0.0
>>>>>>>> crypto isakmp fragmentation
>>>>>>>> !
>>>>>>>> crypto isakmp client configuration group datasc
>>>>>>>>  key Recoil90
>>>>>>>>  dns 4.2.2.2 4.2.2.1
>>>>>>>>  domain datasc.local
>>>>>>>>  pool vpnpool
>>>>>>>>  save-password
>>>>>>>> !
>>>>>>>> crypto isakmp client configuration group datascsplit
>>>>>>>>  key Recoil90
>>>>>>>>  dns 4.2.2.2 4.2.2.1
>>>>>>>>  domain datasc.local
>>>>>>>>  pool vpnpool
>>>>>>>>  acl 101
>>>>>>>>  save-password
>>>>>>>> crypto isakmp profile datasc
>>>>>>>>    match identity group datasc
>>>>>>>>    client authentication list vpnauth
>>>>>>>>    isakmp authorization list vpnauth
>>>>>>>>    client configuration address respond
>>>>>>>>    virtual-template 1
>>>>>>>> crypto isakmp profile datascsplit
>>>>>>>>    match identity group datascsplit
>>>>>>>>    client authentication list vpnauth
>>>>>>>>    isakmp authorization list vpnauth
>>>>>>>>    client configuration address respond
>>>>>>>>    virtual-template 2
>>>>>>>> !
>>>>>>>> !
>>>>>>>> crypto ipsec transform-set transformset esp-aes
>>>>>>>> crypto ipsec transform-set ezvpntransform esp-aes esp-sha-hmac
>>>>>>>> !
>>>>>>>> crypto ipsec profile VTI
>>>>>>>>  set transform-set ezvpntransform
>>>>>>>>  set isakmp-profile datasc
>>>>>>>> !
>>>>>>>> crypto ipsec profile VTI2
>>>>>>>>  set transform-set ezvpntransform
>>>>>>>>  set isakmp-profile datascsplit
>>>>>>>> !
>>>>>>>> !
>>>>>>>> !
>>>>>>>> !
>>>>>>>> !
>>>>>>>> !
>>>>>>>> !
>>>>>>>> interface Loopback1
>>>>>>>>  ip address 10.1.150.1 255.255.255.0
>>>>>>>>  ip nat inside
>>>>>>>>  ip virtual-reassembly in
>>>>>>>> !
>>>>>>>> interface GigabitEthernet0/0
>>>>>>>>  ip address dhcp
>>>>>>>>  no ip redirects
>>>>>>>>  no ip unreachables
>>>>>>>>  no ip proxy-arp
>>>>>>>>  ip nat outside
>>>>>>>>  ip virtual-reassembly in
>>>>>>>>  duplex auto
>>>>>>>>  speed auto
>>>>>>>>  media-type rj45
>>>>>>>>  hold-queue 240000 in
>>>>>>>> !
>>>>>>>> interface GigabitEthernet0/1
>>>>>>>>  ip address 10.1.200.1 255.255.255.252
>>>>>>>>  ip nat inside
>>>>>>>>  ip virtual-reassembly in
>>>>>>>>  duplex auto
>>>>>>>>  speed auto
>>>>>>>>  media-type rj45
>>>>>>>> !
>>>>>>>> interface Virtual-Template1 type tunnel
>>>>>>>>  ip unnumbered GigabitEthernet0/0
>>>>>>>>  ip nat inside
>>>>>>>>  ip virtual-reassembly in
>>>>>>>>  tunnel source GigabitEthernet0/0
>>>>>>>>  tunnel mode ipsec ipv4
>>>>>>>>  tunnel protection ipsec profile VTI
>>>>>>>> !
>>>>>>>> interface Virtual-Template2 type tunnel
>>>>>>>>  ip unnumbered GigabitEthernet0/0
>>>>>>>>  ip nat inside
>>>>>>>>  ip virtual-reassembly in
>>>>>>>>  tunnel source GigabitEthernet0/0
>>>>>>>>  tunnel mode ipsec ipv4
>>>>>>>>  tunnel protection ipsec profile VTI2
>>>>>>>> !
>>>>>>>> interface Virtual-Template3
>>>>>>>>  ip unnumbered GigabitEthernet0/0
>>>>>>>>  ip nat outside
>>>>>>>>  ip virtual-reassembly in
>>>>>>>>  ip policy route-map anyconnecthop
>>>>>>>> !
>>>>>>>> !
>>>>>>>> router eigrp 1
>>>>>>>>  maximum-paths 1
>>>>>>>>  network 10.0.0.0
>>>>>>>>  redistribute static
>>>>>>>> !
>>>>>>>> ip local pool vpnpool 10.1.100.10 10.1.100.200
>>>>>>>> ip forward-protocol nd
>>>>>>>> ip http server
>>>>>>>> ip http secure-server
>>>>>>>> !
>>>>>>>> !
>>>>>>>> ip nat inside source list NATNETWORKS interface GigabitEthernet0/0
>>>>>>>> overload
>>>>>>>> ip nat inside source static tcp 10.1.50.150 80 interface
>>>>>>>> GigabitEthernet0/0 80
>>>>>>>> ip nat inside source static tcp 10.1.80.100 5001 interface
>>>>>>>> GigabitEthernet0/0 5001
>>>>>>>> ip nat inside source static udp 10.1.80.100 5001 interface
>>>>>>>> GigabitEthernet0/0 5001
>>>>>>>> !
>>>>>>>> ip access-list extended NATNETWORKS
>>>>>>>>  deny   ip 10.0.0.0 0.255.255.255 172.16.0.0 0.15.255.255
>>>>>>>>  deny   ip 10.0.0.0 0.255.255.255 10.0.0.0 0.255.255.255
>>>>>>>>  permit ip 10.0.0.0 0.255.255.255 any
>>>>>>>> ip access-list extended anyconnecthop
>>>>>>>>  deny   ip 10.0.0.0 0.255.255.255 10.0.0.0 0.255.255.255
>>>>>>>>  permit ip 10.0.0.0 0.255.255.255 any
>>>>>>>> !
>>>>>>>> access-list 50 permit 10.0.0.0 0.255.255.255
>>>>>>>> access-list 101 permit ip 10.0.0.0 0.255.255.255 any
>>>>>>>> !
>>>>>>>> !
>>>>>>>> !
>>>>>>>> !
>>>>>>>> route-map anyconnecthop permit 10
>>>>>>>>  match ip address anyconnecthop
>>>>>>>>  set ip next-hop 10.1.150.2
>>>>>>>> !
>>>>>>>> !
>>>>>>>> !
>>>>>>>> !
>>>>>>>> !
>>>>>>>> control-plane
>>>>>>>> !
>>>>>>>> !
>>>>>>>> !
>>>>>>>> !
>>>>>>>> mgcp profile default
>>>>>>>> !
>>>>>>>> !
>>>>>>>> dial-peer voice 100 voip
>>>>>>>>  description Publisher
>>>>>>>>  preference 1
>>>>>>>>  destination-pattern 1..
>>>>>>>>  session protocol sipv2
>>>>>>>>  session target ipv4:10.1.80.10
>>>>>>>>  dtmf-relay rtp-nte
>>>>>>>>  codec g711ulaw
>>>>>>>> !
>>>>>>>> dial-peer voice 101 voip
>>>>>>>>  description Subscriber
>>>>>>>>  preference 2
>>>>>>>>  destination-pattern 1..
>>>>>>>>  session target ipv4:10.1.80.11
>>>>>>>>  dtmf-relay rtp-nte
>>>>>>>>  codec g711ulaw
>>>>>>>> !
>>>>>>>> dial-peer voice 200 voip
>>>>>>>>  description Publisher
>>>>>>>>  preference 1
>>>>>>>>  destination-pattern 2..
>>>>>>>>  progress_ind setup enable 3
>>>>>>>>  progress_ind progress enable 8
>>>>>>>>  session protocol sipv2
>>>>>>>>  session target ipv4:10.1.80.10
>>>>>>>>  dtmf-relay rtp-nte
>>>>>>>>  codec g711ulaw
>>>>>>>> !
>>>>>>>> dial-peer voice 300 voip
>>>>>>>>  description incoming Calldid
>>>>>>>>  translation-profile incoming incomingdid
>>>>>>>>  preference 1
>>>>>>>>  session protocol sipv2
>>>>>>>>  session target sip-server
>>>>>>>>  incoming called-number 678456329.
>>>>>>>>  dtmf-relay rtp-nte
>>>>>>>>  codec g711ulaw
>>>>>>>> !
>>>>>>>> dial-peer voice 301 voip
>>>>>>>>  description incoming Calldid
>>>>>>>>  translation-profile incoming incomingdid
>>>>>>>>  preference 1
>>>>>>>>  session protocol sipv2
>>>>>>>>  session target sip-server
>>>>>>>>  incoming called-number 6784604565
>>>>>>>>  dtmf-relay rtp-nte
>>>>>>>>  codec g711ulaw
>>>>>>>> !
>>>>>>>> dial-peer voice 302 voip
>>>>>>>>  description incoming Calldid
>>>>>>>>  translation-profile incoming incomingdid
>>>>>>>>  preference 1
>>>>>>>>  session protocol sipv2
>>>>>>>>  session target sip-server
>>>>>>>>  incoming called-number 6784604564
>>>>>>>>  dtmf-relay rtp-nte
>>>>>>>>  codec g711ulaw
>>>>>>>> !
>>>>>>>> dial-peer voice 201 voip
>>>>>>>>  description Publisher
>>>>>>>>  preference 2
>>>>>>>>  destination-pattern 2..
>>>>>>>>  progress_ind setup enable 3
>>>>>>>>  progress_ind progress enable 8
>>>>>>>>  session protocol sipv2
>>>>>>>>  session target ipv4:10.1.80.11
>>>>>>>>  dtmf-relay rtp-nte
>>>>>>>>  codec g711ulaw
>>>>>>>> !
>>>>>>>> dial-peer voice 500 voip
>>>>>>>>  description outgoing
>>>>>>>>  preference 1
>>>>>>>>  destination-pattern .T
>>>>>>>>  session protocol sipv2
>>>>>>>>  session target dns:sip.talkinip.net
>>>>>>>>  dtmf-relay rtp-nte
>>>>>>>>  codec g711ulaw
>>>>>>>> !
>>>>>>>> !
>>>>>>>> sip-ua
>>>>>>>>  credentials username xxxxxxxx password 7 xxxxxxx realm
>>>>>>>> sipconnect.ipcomms.net
>>>>>>>>  authentication username xxxxxx password 7 xxxxxxx
>>>>>>>>  authentication username xxxxx password 7 xxxxxxx realm
>>>>>>>> sipconnect.ipcomms.net
>>>>>>>>  set pstn-cause 3 sip-status 486
>>>>>>>>  set pstn-cause 34 sip-status 486
>>>>>>>>  set pstn-cause 47 sip-status 486
>>>>>>>>  registrar dns:sipconnect.ipcomms.net expires 60
>>>>>>>>  sip-server dns:sipconnect.ipcomms.net:5060
>>>>>>>> !
>>>>>>>> !
>>>>>>>> !
>>>>>>>> gatekeeper
>>>>>>>>  shutdown
>>>>>>>> !
>>>>>>>> !
>>>>>>>> call-manager-fallback
>>>>>>>>  max-conferences 8 gain -6
>>>>>>>>  transfer-system full-consult
>>>>>>>>  ip source-address 10.1.200.1 port 2000
>>>>>>>>  max-ephones 20
>>>>>>>>  max-dn 40
>>>>>>>> !
>>>>>>>> !
>>>>>>>> !
>>>>>>>> line con 0
>>>>>>>> line aux 0
>>>>>>>> line vty 0 4
>>>>>>>>  privilege level 15
>>>>>>>>  transport input ssh
>>>>>>>> line vty 5 15
>>>>>>>>  privilege level 15
>>>>>>>>  transport input ssh
>>>>>>>> !
>>>>>>>> scheduler allocate 20000 1000
>>>>>>>> !
>>>>>>>> webvpn gateway gateway_1
>>>>>>>>  ip interface GigabitEthernet0/0 port 443
>>>>>>>>  ssl trustpoint selfsigned
>>>>>>>>  inservice
>>>>>>>>  !
>>>>>>>> webvpn install svc flash:/webvpn/anyconnect-win-3.1.02026-k9.pkg
>>>>>>>> sequence 1
>>>>>>>>  !
>>>>>>>> webvpn context datasc
>>>>>>>>  title "DataSC VPN"
>>>>>>>>  secondary-color white
>>>>>>>>  title-color #CCCC66
>>>>>>>>  text-color black
>>>>>>>>  ssl authenticate verify all
>>>>>>>>  !
>>>>>>>>  url-list "Servers"
>>>>>>>>    heading "Server"
>>>>>>>>  !
>>>>>>>>  !
>>>>>>>>  policy group datasc
>>>>>>>>    url-list "Servers"
>>>>>>>>    functions svc-enabled
>>>>>>>>    svc address-pool "vpnpool" netmask 255.255.255.0
>>>>>>>>    svc keep-client-installed
>>>>>>>>    svc dns-server primary 4.2.2.2
>>>>>>>>    svc dtls
>>>>>>>>  virtual-template 3
>>>>>>>>  default-group-policy datasc
>>>>>>>>  aaa authentication list vpnauth
>>>>>>>>  gateway gateway_1 domain datasc
>>>>>>>>  inservice
>>>>>>>> !
>>>>>>>> !
>>>>>>>> webvpn context datascsplit
>>>>>>>>  title "DataSC VPN Split"
>>>>>>>>  secondary-color white
>>>>>>>>  title-color #CCCC66
>>>>>>>>  text-color black
>>>>>>>>  ssl authenticate verify all
>>>>>>>>  !
>>>>>>>>  url-list "Servers"
>>>>>>>>    heading "Server"
>>>>>>>>  !
>>>>>>>>  !
>>>>>>>>  policy group datascsplit
>>>>>>>>    url-list "Servers"
>>>>>>>>    functions svc-enabled
>>>>>>>>    svc address-pool "vpnpool" netmask 255.255.255.0
>>>>>>>>    svc split include acl 50
>>>>>>>>    svc dns-server primary 4.2.2.2
>>>>>>>>    svc dtls
>>>>>>>>  default-group-policy datascsplit
>>>>>>>>  aaa authentication list vpnauth
>>>>>>>>  gateway gateway_1 domain datascsplit
>>>>>>>>  inservice
>>>>>>>> !
>>>>>>>> end
>>>>>>>> Cisco3825#
>>>>>>>>
>>>>>>>> On Tue, Jan 15, 2013 at 3:31 PM, Kenneth Hayes <
>>>>>>>> kennethwhayes at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> What do your media resources look like?
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> Also can you show me a copy of your voice service voip config?
>>>>>>>>>
>>>>>>>>> Sent from my iPad
>>>>>>>>>
>>>>>>>>> On Jan 15, 2013, at 3:12 PM, Dane Newman <dane.newman at gmail.com>
>>>>>>>>> wrote:
>>>>>>>>>
>>>>>>>>> Thanks Ryan
>>>>>>>>>
>>>>>>>>> I see I am always getting a 200 ok message after my invites from
>>>>>>>>> the debug
>>>>>>>>>
>>>>>>>>> *Putting a call on HOLD*
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> *Jan 15 20:19:28.086: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>>
>>>>>>>>> Received:
>>>>>>>>>
>>>>>>>>> INVITE sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0
>>>>>>>>>
>>>>>>>>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK2891b89f8fa
>>>>>>>>>
>>>>>>>>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>>>>>>>>
>>>>>>>>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>>>>>>>>
>>>>>>>>> Date: Tue, 15 Jan 2013 19:57:35 GMT
>>>>>>>>>
>>>>>>>>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>>>>>>>>
>>>>>>>>> Supported: timer,resource-priority,replaces
>>>>>>>>>
>>>>>>>>> Min-SE: 1800
>>>>>>>>>
>>>>>>>>> User-Agent: Cisco-CUCM8.6
>>>>>>>>>
>>>>>>>>> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE,
>>>>>>>>> REFER, SUBSCRIBE, NOTIFY
>>>>>>>>>
>>>>>>>>> CSeq: 102 INVITE
>>>>>>>>>
>>>>>>>>> Max-Forwards: 70
>>>>>>>>>
>>>>>>>>> Expires: 180
>>>>>>>>>
>>>>>>>>> Allow-Events: presence
>>>>>>>>>
>>>>>>>>> Supported: X-cisco-srtp-fallback
>>>>>>>>>
>>>>>>>>> Supported: Geolocation
>>>>>>>>>
>>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>>
>>>>>>>>> P-Asserted-Identity: "Dane Newman" <sip:6784563290 at 10.1.80.10>
>>>>>>>>>
>>>>>>>>> Remote-Party-ID: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>>> >;party=calling;screen=yes;privacy=off
>>>>>>>>>
>>>>>>>>> Contact: <sip:6784563290 at 10.1.80.10:5060;transport=tcp>;video
>>>>>>>>>
>>>>>>>>> Content-Type: application/sdp
>>>>>>>>>
>>>>>>>>> Content-Length: 240
>>>>>>>>>
>>>>>>>>> v=0
>>>>>>>>>
>>>>>>>>> o=CiscoSystemsCCM-SIP 7322 3 IN IP4 10.1.80.10
>>>>>>>>>
>>>>>>>>> s=SIP Call
>>>>>>>>>
>>>>>>>>> c=IN IP4 0.0.0.0
>>>>>>>>>
>>>>>>>>> b=TIAS:64000
>>>>>>>>>
>>>>>>>>> b=AS:64
>>>>>>>>>
>>>>>>>>> t=0 0
>>>>>>>>>
>>>>>>>>> m=audio 21476 RTP/AVP 0 101
>>>>>>>>>
>>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>>
>>>>>>>>> a=ptime:20
>>>>>>>>>
>>>>>>>>> a=inactive
>>>>>>>>>
>>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>>>
>>>>>>>>> a=fmtp:101 0-15
>>>>>>>>>
>>>>>>>>> *Jan 15 20:19:28.094: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>>
>>>>>>>>> Sent:
>>>>>>>>>
>>>>>>>>> INVITE sip:17705439047 at 64.154.41.150:5060 SIP/2.0
>>>>>>>>>
>>>>>>>>> Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK691F12E0
>>>>>>>>>
>>>>>>>>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>>>>>>>>> >;tag=2E6BC0B0-2268
>>>>>>>>>
>>>>>>>>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>>>>>>>>
>>>>>>>>> Date: Tue, 15 Jan 2013 20:19:28 GMT
>>>>>>>>>
>>>>>>>>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>>>>>>>>
>>>>>>>>> Supported: 100rel,timer,resource-priority,replaces,sdp-anat
>>>>>>>>>
>>>>>>>>> Min-SE: 1800
>>>>>>>>>
>>>>>>>>> Cisco-Guid: 3257897472-0000065536-0000000035-0173015306
>>>>>>>>>
>>>>>>>>> User-Agent: Cisco-SIPGateway/IOS-12.x
>>>>>>>>>
>>>>>>>>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>>>>>>>>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>>>>>>>>>
>>>>>>>>> CSeq: 103 INVITE
>>>>>>>>>
>>>>>>>>> Max-Forwards: 70
>>>>>>>>>
>>>>>>>>> Timestamp: 1358281168
>>>>>>>>>
>>>>>>>>> Contact: <sip:6784563290 at 98.192.104.214:5060>
>>>>>>>>>
>>>>>>>>> Expires: 180
>>>>>>>>>
>>>>>>>>> Allow-Events: telephone-event
>>>>>>>>>
>>>>>>>>> Authorization: Digest username="6784563290",realm="asterisk",uri="
>>>>>>>>> sip:17705439047 at 64.154.41.150:5060
>>>>>>>>> ",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5
>>>>>>>>>
>>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>>
>>>>>>>>> Content-Type: application/sdp
>>>>>>>>>
>>>>>>>>> Content-Length: 289
>>>>>>>>>
>>>>>>>>> v=0
>>>>>>>>>
>>>>>>>>> o=CiscoSystemsSIP-GW-UserAgent 3168 2739 IN IP4 98.192.104.214
>>>>>>>>>
>>>>>>>>> s=SIP Call
>>>>>>>>>
>>>>>>>>> c=IN IP4 98.192.104.214
>>>>>>>>>
>>>>>>>>> t=0 0
>>>>>>>>>
>>>>>>>>> m=audio 19458 RTP/AVP 0 101 19
>>>>>>>>>
>>>>>>>>> c=IN IP4 98.192.104.214
>>>>>>>>>
>>>>>>>>> a=inactive
>>>>>>>>>
>>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>>
>>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>>>
>>>>>>>>> a=fmtp:101 0-15
>>>>>>>>>
>>>>>>>>> a=rtpmap:19 CN/8000
>>>>>>>>>
>>>>>>>>> a=ptime:20
>>>>>>>>>
>>>>>>>>> *Jan 15 20:19:28.094: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>>
>>>>>>>>> Sent:
>>>>>>>>>
>>>>>>>>> SIP/2.0 100 Trying
>>>>>>>>>
>>>>>>>>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK2891b89f8fa
>>>>>>>>>
>>>>>>>>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>>>>>>>>
>>>>>>>>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>>>>>>>>
>>>>>>>>> Date: Tue, 15 Jan 2013 20:19:28 GMT
>>>>>>>>>
>>>>>>>>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>>>>>>>>
>>>>>>>>> CSeq: 102 INVITE
>>>>>>>>>
>>>>>>>>> Allow-Events: telephone-event
>>>>>>>>>
>>>>>>>>> Server: Cisco-SIPGateway/IOS-12.x
>>>>>>>>>
>>>>>>>>> Content-Length: 0
>>>>>>>>>
>>>>>>>>> *Jan 15 20:19:28.110: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>>
>>>>>>>>> Received:
>>>>>>>>>
>>>>>>>>> SIP/2.0 100 Trying
>>>>>>>>>
>>>>>>>>> Via: SIP/2.0/UDP 98.192.104.214:5060
>>>>>>>>> ;branch=z9hG4bK691F12E0;received=98.192.104.214
>>>>>>>>>
>>>>>>>>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>>>>>>>>> >;tag=2E6BC0B0-2268
>>>>>>>>>
>>>>>>>>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>>>>>>>>
>>>>>>>>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>>>>>>>>
>>>>>>>>> CSeq: 103 INVITE
>>>>>>>>>
>>>>>>>>> Server: Asterisk PBX 1.6.2.13
>>>>>>>>>
>>>>>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
>>>>>>>>> NOTIFY, INFO
>>>>>>>>>
>>>>>>>>> Supported: replaces, timer
>>>>>>>>>
>>>>>>>>> Require: timer
>>>>>>>>>
>>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>>
>>>>>>>>> Contact: <sip:17705439047 at 64.154.41.150>
>>>>>>>>>
>>>>>>>>> Content-Length: 0
>>>>>>>>>
>>>>>>>>> *Jan 15 20:19:28.110: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>>
>>>>>>>>> Received:
>>>>>>>>>
>>>>>>>>> SIP/2.0 200 OK
>>>>>>>>>
>>>>>>>>> Via: SIP/2.0/UDP 98.192.104.214:5060
>>>>>>>>> ;branch=z9hG4bK691F12E0;received=98.192.104.214
>>>>>>>>>
>>>>>>>>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>>>>>>>>> >;tag=2E6BC0B0-2268
>>>>>>>>>
>>>>>>>>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>>>>>>>>
>>>>>>>>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>>>>>>>>
>>>>>>>>> CSeq: 103 INVITE
>>>>>>>>>
>>>>>>>>> Server: Asterisk PBX 1.6.2.13
>>>>>>>>>
>>>>>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
>>>>>>>>> NOTIFY, INFO
>>>>>>>>>
>>>>>>>>> Supported: replaces, timer
>>>>>>>>>
>>>>>>>>> Require: timer
>>>>>>>>>
>>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>>
>>>>>>>>> Contact: <sip:17705439047 at 64.154.41.150>
>>>>>>>>>
>>>>>>>>> Content-Type: application/sdp
>>>>>>>>>
>>>>>>>>> Content-Length: 239
>>>>>>>>>
>>>>>>>>> v=0
>>>>>>>>>
>>>>>>>>> o=root 1685873050 1685873052 IN IP4 64.154.41.150
>>>>>>>>>
>>>>>>>>> s=Asterisk PBX 1.6.2.13
>>>>>>>>>
>>>>>>>>> c=IN IP4 64.154.41.150
>>>>>>>>>
>>>>>>>>> t=0 0
>>>>>>>>>
>>>>>>>>> m=audio 13014 RTP/AVP 0 101
>>>>>>>>>
>>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>>
>>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>>>
>>>>>>>>> a=fmtp:101 0-16
>>>>>>>>>
>>>>>>>>> a=ptime:20
>>>>>>>>>
>>>>>>>>> a=inactive
>>>>>>>>>
>>>>>>>>> *Jan 15 20:19:28.118: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>>
>>>>>>>>> Sent:
>>>>>>>>>
>>>>>>>>> SIP/2.0 200 OK
>>>>>>>>>
>>>>>>>>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK2891b89f8fa
>>>>>>>>>
>>>>>>>>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>>>>>>>>
>>>>>>>>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>>>>>>>>
>>>>>>>>> Date: Tue, 15 Jan 2013 20:19:28 GMT
>>>>>>>>>
>>>>>>>>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>>>>>>>>
>>>>>>>>> CSeq: 102 INVITE
>>>>>>>>>
>>>>>>>>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>>>>>>>>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>>>>>>>>>
>>>>>>>>> Allow-Events: telephone-event
>>>>>>>>>
>>>>>>>>> Remote-Party-ID: <sip:17705439047 at 10.1.200.1
>>>>>>>>> >;party=called;screen=no;privacy=off
>>>>>>>>>
>>>>>>>>> Contact: <sip:17705439047 at 10.1.200.1:5060;transport=tcp>
>>>>>>>>>
>>>>>>>>> Supported: replaces
>>>>>>>>>
>>>>>>>>> Supported: sdp-anat
>>>>>>>>>
>>>>>>>>> Server: Cisco-SIPGateway/IOS-12.x
>>>>>>>>>
>>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>>
>>>>>>>>> Require: timer
>>>>>>>>>
>>>>>>>>> Supported: timer
>>>>>>>>>
>>>>>>>>> Content-Type: application/sdp
>>>>>>>>>
>>>>>>>>> Content-Length: 253
>>>>>>>>>
>>>>>>>>> v=0
>>>>>>>>>
>>>>>>>>> o=CiscoSystemsSIP-GW-UserAgent 4444 5479 IN IP4 10.1.200.1
>>>>>>>>>
>>>>>>>>> s=SIP Call
>>>>>>>>>
>>>>>>>>> c=IN IP4 10.1.200.1
>>>>>>>>>
>>>>>>>>> t=0 0
>>>>>>>>>
>>>>>>>>> m=audio 19514 RTP/AVP 0 101
>>>>>>>>>
>>>>>>>>> c=IN IP4 10.1.200.1
>>>>>>>>>
>>>>>>>>> a=inactive
>>>>>>>>>
>>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>>
>>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>>>
>>>>>>>>> a=fmtp:101 0-16
>>>>>>>>>
>>>>>>>>> a=ptime:20
>>>>>>>>>
>>>>>>>>> *Jan 15 20:19:28.118: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>>
>>>>>>>>> Sent:
>>>>>>>>>
>>>>>>>>> ACK sip:17705439047 at 64.154.41.150:5060 SIP/2.0
>>>>>>>>>
>>>>>>>>> Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK6920266D
>>>>>>>>>
>>>>>>>>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>>>>>>>>> >;tag=2E6BC0B0-2268
>>>>>>>>>
>>>>>>>>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>>>>>>>>
>>>>>>>>> Date: Tue, 15 Jan 2013 20:19:28 GMT
>>>>>>>>>
>>>>>>>>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>>>>>>>>
>>>>>>>>> Max-Forwards: 70
>>>>>>>>>
>>>>>>>>> CSeq: 103 ACK
>>>>>>>>>
>>>>>>>>> Authorization: Digest username="6784563290",realm="asterisk",uri="
>>>>>>>>> sip:17705439047 at 64.154.41.150:5060
>>>>>>>>> ",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5
>>>>>>>>>
>>>>>>>>> Allow-Events: telephone-event
>>>>>>>>>
>>>>>>>>> Content-Length: 0
>>>>>>>>>
>>>>>>>>> *Jan 15 20:19:28.122: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>>
>>>>>>>>> Received:
>>>>>>>>>
>>>>>>>>> ACK sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0
>>>>>>>>>
>>>>>>>>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28b4b1305a0
>>>>>>>>>
>>>>>>>>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>>>>>>>>
>>>>>>>>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>>>>>>>>
>>>>>>>>> Date: Tue, 15 Jan 2013 19:57:35 GMT
>>>>>>>>>
>>>>>>>>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>>>>>>>>
>>>>>>>>> Max-Forwards: 70
>>>>>>>>>
>>>>>>>>> CSeq: 102 ACK
>>>>>>>>>
>>>>>>>>> Allow-Events: presence
>>>>>>>>>
>>>>>>>>> Content-Length: 0
>>>>>>>>>
>>>>>>>>> *Jan 15 20:19:28.122: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>>
>>>>>>>>> Received:
>>>>>>>>>
>>>>>>>>> INVITE sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0
>>>>>>>>>
>>>>>>>>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28c43b47e7e
>>>>>>>>>
>>>>>>>>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>>>>>>>>
>>>>>>>>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>>>>>>>>
>>>>>>>>> Date: Tue, 15 Jan 2013 19:57:35 GMT
>>>>>>>>>
>>>>>>>>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>>>>>>>>
>>>>>>>>> Supported: timer,resource-priority,replaces
>>>>>>>>>
>>>>>>>>> Min-SE: 1800
>>>>>>>>>
>>>>>>>>> User-Agent: Cisco-CUCM8.6
>>>>>>>>>
>>>>>>>>> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE,
>>>>>>>>> REFER, SUBSCRIBE, NOTIFY
>>>>>>>>>
>>>>>>>>> CSeq: 103 INVITE
>>>>>>>>>
>>>>>>>>> Max-Forwards: 70
>>>>>>>>>
>>>>>>>>> Expires: 180
>>>>>>>>>
>>>>>>>>> Allow-Events: presence
>>>>>>>>>
>>>>>>>>> Supported: X-cisco-srtp-fallback
>>>>>>>>>
>>>>>>>>> Supported: Geolocation
>>>>>>>>>
>>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>>
>>>>>>>>> P-Asserted-Identity: "Dane Newman" <sip:6784563290 at 10.1.80.10>
>>>>>>>>>
>>>>>>>>> Remote-Party-ID: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>>> >;party=calling;screen=yes;privacy=off
>>>>>>>>>
>>>>>>>>> Contact: <sip:6784563290 at 10.1.80.10:5060;transport=tcp>;video
>>>>>>>>>
>>>>>>>>> Content-Length: 0
>>>>>>>>>
>>>>>>>>> *Jan 15 20:19:28.126: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>>
>>>>>>>>> Sent:
>>>>>>>>>
>>>>>>>>> INVITE sip:17705439047 at 64.154.41.150:5060 SIP/2.0
>>>>>>>>>
>>>>>>>>> Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69211AB3
>>>>>>>>>
>>>>>>>>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>>>>>>>>> >;tag=2E6BC0B0-2268
>>>>>>>>>
>>>>>>>>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>>>>>>>>
>>>>>>>>> Date: Tue, 15 Jan 2013 20:19:28 GMT
>>>>>>>>>
>>>>>>>>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>>>>>>>>
>>>>>>>>> Supported: timer,resource-priority,replaces,sdp-anat
>>>>>>>>>
>>>>>>>>> Min-SE: 1800
>>>>>>>>>
>>>>>>>>> Cisco-Guid: 3257897472-0000065536-0000000035-0173015306
>>>>>>>>>
>>>>>>>>> User-Agent: Cisco-SIPGateway/IOS-12.x
>>>>>>>>>
>>>>>>>>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>>>>>>>>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>>>>>>>>>
>>>>>>>>> CSeq: 104 INVITE
>>>>>>>>>
>>>>>>>>> Max-Forwards: 70
>>>>>>>>>
>>>>>>>>> Timestamp: 1358281168
>>>>>>>>>
>>>>>>>>> Contact: <sip:6784563290 at 98.192.104.214:5060>
>>>>>>>>>
>>>>>>>>> Expires: 180
>>>>>>>>>
>>>>>>>>> Allow-Events: telephone-event
>>>>>>>>>
>>>>>>>>> Authorization: Digest username="6784563290",realm="asterisk",uri="
>>>>>>>>> sip:17705439047 at 64.154.41.150:5060
>>>>>>>>> ",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5
>>>>>>>>>
>>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>>
>>>>>>>>> Content-Length: 0
>>>>>>>>>
>>>>>>>>> *Jan 15 20:19:28.126: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>>
>>>>>>>>> Sent:
>>>>>>>>>
>>>>>>>>> SIP/2.0 100 Trying
>>>>>>>>>
>>>>>>>>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28c43b47e7e
>>>>>>>>>
>>>>>>>>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>>>>>>>>
>>>>>>>>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>>>>>>>>
>>>>>>>>> Date: Tue, 15 Jan 2013 20:19:28 GMT
>>>>>>>>>
>>>>>>>>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>>>>>>>>
>>>>>>>>> CSeq: 103 INVITE
>>>>>>>>>
>>>>>>>>> Allow-Events: telephone-event
>>>>>>>>>
>>>>>>>>> Server: Cisco-SIPGateway/IOS-12.x
>>>>>>>>>
>>>>>>>>> Content-Length: 0
>>>>>>>>>
>>>>>>>>> *Jan 15 20:19:28.146: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>>
>>>>>>>>> Received:
>>>>>>>>>
>>>>>>>>> SIP/2.0 100 Trying
>>>>>>>>>
>>>>>>>>> Via: SIP/2.0/UDP 98.192.104.214:5060
>>>>>>>>> ;branch=z9hG4bK69211AB3;received=98.192.104.214
>>>>>>>>>
>>>>>>>>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>>>>>>>>> >;tag=2E6BC0B0-2268
>>>>>>>>>
>>>>>>>>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>>>>>>>>
>>>>>>>>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>>>>>>>>
>>>>>>>>> CSeq: 104 INVITE
>>>>>>>>>
>>>>>>>>> Server: Asterisk PBX 1.6.2.13
>>>>>>>>>
>>>>>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
>>>>>>>>> NOTIFY, INFO
>>>>>>>>>
>>>>>>>>> Supported: replaces, timer
>>>>>>>>>
>>>>>>>>> Require: timer
>>>>>>>>>
>>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>>
>>>>>>>>> Contact: <sip:17705439047 at 64.154.41.150>
>>>>>>>>>
>>>>>>>>> Content-Length: 0
>>>>>>>>>
>>>>>>>>> *Jan 15 20:19:28.146: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>>
>>>>>>>>> Received:
>>>>>>>>>
>>>>>>>>> SIP/2.0 200 OK
>>>>>>>>>
>>>>>>>>> Via: SIP/2.0/UDP 98.192.104.214:5060
>>>>>>>>> ;branch=z9hG4bK69211AB3;received=98.192.104.214
>>>>>>>>>
>>>>>>>>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>>>>>>>>> >;tag=2E6BC0B0-2268
>>>>>>>>>
>>>>>>>>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>>>>>>>>
>>>>>>>>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>>>>>>>>
>>>>>>>>> CSeq: 104 INVITE
>>>>>>>>>
>>>>>>>>> Server: Asterisk PBX 1.6.2.13
>>>>>>>>>
>>>>>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
>>>>>>>>> NOTIFY, INFO
>>>>>>>>>
>>>>>>>>> Supported: replaces, timer
>>>>>>>>>
>>>>>>>>> Require: timer
>>>>>>>>>
>>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>>
>>>>>>>>> Contact: <sip:17705439047 at 64.154.41.150>
>>>>>>>>>
>>>>>>>>> Content-Type: application/sdp
>>>>>>>>>
>>>>>>>>> Content-Length: 333
>>>>>>>>>
>>>>>>>>> v=0
>>>>>>>>>
>>>>>>>>> o=root 1685873050 1685873053 IN IP4 64.154.41.150
>>>>>>>>>
>>>>>>>>> s=Asterisk PBX 1.6.2.13
>>>>>>>>>
>>>>>>>>> c=IN IP4 64.154.41.150
>>>>>>>>>
>>>>>>>>> t=0 0
>>>>>>>>>
>>>>>>>>> m=audio 13014 RTP/AVP 3 8 0 18 101
>>>>>>>>>
>>>>>>>>> a=rtpmap:3 GSM/8000
>>>>>>>>>
>>>>>>>>> a=rtpmap:8 PCMA/8000
>>>>>>>>>
>>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>>
>>>>>>>>> a=rtpmap:18 G729/8000
>>>>>>>>>
>>>>>>>>> a=fmtp:18 annexb=no
>>>>>>>>>
>>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>>>
>>>>>>>>> a=fmtp:101 0-16
>>>>>>>>>
>>>>>>>>> a=ptime:20
>>>>>>>>>
>>>>>>>>> a=inactive
>>>>>>>>>
>>>>>>>>> *Jan 15 20:19:28.150: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>>
>>>>>>>>> Sent:
>>>>>>>>>
>>>>>>>>> SIP/2.0 200 OK
>>>>>>>>>
>>>>>>>>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28c43b47e7e
>>>>>>>>>
>>>>>>>>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>>>>>>>>
>>>>>>>>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>>>>>>>>
>>>>>>>>> Date: Tue, 15 Jan 2013 20:19:28 GMT
>>>>>>>>>
>>>>>>>>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>>>>>>>>
>>>>>>>>> CSeq: 103 INVITE
>>>>>>>>>
>>>>>>>>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>>>>>>>>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>>>>>>>>>
>>>>>>>>> Allow-Events: telephone-event
>>>>>>>>>
>>>>>>>>> Remote-Party-ID: <sip:17705439047 at 10.1.200.1
>>>>>>>>> >;party=called;screen=no;privacy=off
>>>>>>>>>
>>>>>>>>> Contact: <sip:17705439047 at 10.1.200.1:5060;transport=tcp>
>>>>>>>>>
>>>>>>>>> Supported: replaces
>>>>>>>>>
>>>>>>>>> Supported: sdp-anat
>>>>>>>>>
>>>>>>>>> Server: Cisco-SIPGateway/IOS-12.x
>>>>>>>>>
>>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>>
>>>>>>>>> Require: timer
>>>>>>>>>
>>>>>>>>> Supported: timer
>>>>>>>>>
>>>>>>>>> Content-Type: application/sdp
>>>>>>>>>
>>>>>>>>> Content-Length: 277
>>>>>>>>>
>>>>>>>>> v=0
>>>>>>>>>
>>>>>>>>> o=CiscoSystemsSIP-GW-UserAgent 4444 5480 IN IP4 10.1.200.1
>>>>>>>>>
>>>>>>>>> s=SIP Call
>>>>>>>>>
>>>>>>>>> c=IN IP4 10.1.200.1
>>>>>>>>>
>>>>>>>>> t=0 0
>>>>>>>>>
>>>>>>>>> m=audio 19514 RTP/AVP 0 101 19
>>>>>>>>>
>>>>>>>>> c=IN IP4 10.1.200.1
>>>>>>>>>
>>>>>>>>> a=inactive
>>>>>>>>>
>>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>>
>>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>>>
>>>>>>>>> a=fmtp:101 0-16
>>>>>>>>>
>>>>>>>>> a=rtpmap:19 CN/8000
>>>>>>>>>
>>>>>>>>> a=ptime:20
>>>>>>>>>
>>>>>>>>> *Jan 15 20:19:28.162: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>>
>>>>>>>>> Received:
>>>>>>>>>
>>>>>>>>> ACK sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0
>>>>>>>>>
>>>>>>>>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28d3eadaab3
>>>>>>>>>
>>>>>>>>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>>>>>>>>
>>>>>>>>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>>>>>>>>
>>>>>>>>> Date: Tue, 15 Jan 2013 19:57:35 GMT
>>>>>>>>>
>>>>>>>>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>>>>>>>>
>>>>>>>>> Max-Forwards: 70
>>>>>>>>>
>>>>>>>>> CSeq: 103 ACK
>>>>>>>>>
>>>>>>>>> Allow-Events: presence
>>>>>>>>>
>>>>>>>>> Content-Type: application/sdp
>>>>>>>>>
>>>>>>>>> Content-Length: 209
>>>>>>>>>
>>>>>>>>> v=0
>>>>>>>>>
>>>>>>>>> o=CiscoSystemsCCM-SIP 7322 4 IN IP4 10.1.80.10
>>>>>>>>>
>>>>>>>>> s=SIP Call
>>>>>>>>>
>>>>>>>>> c=IN IP4 0.0.0.0
>>>>>>>>>
>>>>>>>>> b=TIAS:64000
>>>>>>>>>
>>>>>>>>> b=AS:64
>>>>>>>>>
>>>>>>>>> t=0 0
>>>>>>>>>
>>>>>>>>> m=audio 21476 RTP/AVP 0
>>>>>>>>>
>>>>>>>>> a=X-cisco-media:nomedia
>>>>>>>>>
>>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>>
>>>>>>>>> a=ptime:20
>>>>>>>>>
>>>>>>>>> a=inactive
>>>>>>>>>
>>>>>>>>> *Jan 15 20:19:28.170: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>>
>>>>>>>>> Sent:
>>>>>>>>>
>>>>>>>>> ACK sip:17705439047 at 64.154.41.150:5060 SIP/2.0
>>>>>>>>>
>>>>>>>>> Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK692226EA
>>>>>>>>>
>>>>>>>>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>>>>>>>>> >;tag=2E6BC0B0-2268
>>>>>>>>>
>>>>>>>>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>>>>>>>>
>>>>>>>>> Date: Tue, 15 Jan 2013 20:19:28 GMT
>>>>>>>>>
>>>>>>>>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>>>>>>>>
>>>>>>>>> Max-Forwards: 70
>>>>>>>>>
>>>>>>>>> CSeq: 104 ACK
>>>>>>>>>
>>>>>>>>> Authorization: Digest username="6784563290",realm="asterisk",uri="
>>>>>>>>> sip:17705439047 at 64.154.41.150:5060
>>>>>>>>> ",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5
>>>>>>>>>
>>>>>>>>> Allow-Events: telephone-event
>>>>>>>>>
>>>>>>>>> Content-Type: application/sdp
>>>>>>>>>
>>>>>>>>> Content-Length: 251
>>>>>>>>>
>>>>>>>>> v=0
>>>>>>>>>
>>>>>>>>> o=CiscoSystemsSIP-GW-UserAgent 3168 2740 IN IP4 98.192.104.214
>>>>>>>>>
>>>>>>>>> s=SIP Call
>>>>>>>>>
>>>>>>>>> c=IN IP4 0.0.0.0
>>>>>>>>>
>>>>>>>>> t=0 0
>>>>>>>>>
>>>>>>>>> m=audio 19458 RTP/AVP 0 101
>>>>>>>>>
>>>>>>>>> c=IN IP4 0.0.0.0
>>>>>>>>>
>>>>>>>>> a=inactive
>>>>>>>>>
>>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>>
>>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>>>
>>>>>>>>> a=fmtp:101 0-16
>>>>>>>>>
>>>>>>>>> a=ptime:20
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> *Unholding the call the MOH continues on the previously held
>>>>>>>>> caller while the user hears nothing*
>>>>>>>>>
>>>>>>>>> **
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> *Jan 15 20:19:35.166: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>>
>>>>>>>>> Received:
>>>>>>>>>
>>>>>>>>> INVITE sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0
>>>>>>>>>
>>>>>>>>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28e6157f41c
>>>>>>>>>
>>>>>>>>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>>>>>>>>
>>>>>>>>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>>>>>>>>
>>>>>>>>> Date: Tue, 15 Jan 2013 19:57:42 GMT
>>>>>>>>>
>>>>>>>>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>>>>>>>>
>>>>>>>>> Supported: timer,resource-priority,replaces
>>>>>>>>>
>>>>>>>>> Min-SE: 1800
>>>>>>>>>
>>>>>>>>> User-Agent: Cisco-CUCM8.6
>>>>>>>>>
>>>>>>>>> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE,
>>>>>>>>> REFER, SUBSCRIBE, NOTIFY
>>>>>>>>>
>>>>>>>>> CSeq: 104 INVITE
>>>>>>>>>
>>>>>>>>> Max-Forwards: 70
>>>>>>>>>
>>>>>>>>> Expires: 180
>>>>>>>>>
>>>>>>>>> Allow-Events: presence
>>>>>>>>>
>>>>>>>>> Supported: X-cisco-srtp-fallback
>>>>>>>>>
>>>>>>>>> Supported: Geolocation
>>>>>>>>>
>>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>>
>>>>>>>>> P-Asserted-Identity: "Dane Newman" <sip:6784563290 at 10.1.80.10>
>>>>>>>>>
>>>>>>>>> Remote-Party-ID: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>>> >;party=calling;screen=yes;privacy=off
>>>>>>>>>
>>>>>>>>> Contact: <sip:6784563290 at 10.1.80.10:5060
>>>>>>>>> ;transport=tcp>;video;audio;video
>>>>>>>>>
>>>>>>>>> Content-Length: 0
>>>>>>>>>
>>>>>>>>> *Jan 15 20:19:35.170: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>>
>>>>>>>>> Sent:
>>>>>>>>>
>>>>>>>>> INVITE sip:17705439047 at 64.154.41.150:5060 SIP/2.0
>>>>>>>>>
>>>>>>>>> Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69232672
>>>>>>>>>
>>>>>>>>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>>>>>>>>> >;tag=2E6BC0B0-2268
>>>>>>>>>
>>>>>>>>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>>>>>>>>
>>>>>>>>> Date: Tue, 15 Jan 2013 20:19:35 GMT
>>>>>>>>>
>>>>>>>>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>>>>>>>>
>>>>>>>>> Supported: timer,resource-priority,replaces,sdp-anat
>>>>>>>>>
>>>>>>>>> Min-SE: 1800
>>>>>>>>>
>>>>>>>>> Cisco-Guid: 3257897472-0000065536-0000000035-0173015306
>>>>>>>>>
>>>>>>>>> User-Agent: Cisco-SIPGateway/IOS-12.x
>>>>>>>>>
>>>>>>>>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>>>>>>>>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>>>>>>>>>
>>>>>>>>> CSeq: 105 INVITE
>>>>>>>>>
>>>>>>>>> Max-Forwards: 70
>>>>>>>>>
>>>>>>>>> Timestamp: 1358281175
>>>>>>>>>
>>>>>>>>> Contact: <sip:6784563290 at 98.192.104.214:5060>
>>>>>>>>>
>>>>>>>>> Expires: 180
>>>>>>>>>
>>>>>>>>> Allow-Events: telephone-event
>>>>>>>>>
>>>>>>>>> Authorization: Digest username="6784563290",realm="asterisk",uri="
>>>>>>>>> sip:17705439047 at 64.154.41.150:5060
>>>>>>>>> ",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5
>>>>>>>>>
>>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>>
>>>>>>>>> Content-Length: 0
>>>>>>>>>
>>>>>>>>> *Jan 15 20:19:35.190: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>>
>>>>>>>>> Sent:
>>>>>>>>>
>>>>>>>>> SIP/2.0 100 Trying
>>>>>>>>>
>>>>>>>>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28e6157f41c
>>>>>>>>>
>>>>>>>>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>>>>>>>>
>>>>>>>>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>>>>>>>>
>>>>>>>>> Date: Tue, 15 Jan 2013 20:19:35 GMT
>>>>>>>>>
>>>>>>>>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>>>>>>>>
>>>>>>>>> CSeq: 104 INVITE
>>>>>>>>>
>>>>>>>>> Allow-Events: telephone-event
>>>>>>>>>
>>>>>>>>> Server: Cisco-SIPGateway/IOS-12.x
>>>>>>>>>
>>>>>>>>> Content-Length: 0
>>>>>>>>>
>>>>>>>>> *Jan 15 20:19:35.194: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>>
>>>>>>>>> Received:
>>>>>>>>>
>>>>>>>>> SIP/2.0 100 Trying
>>>>>>>>>
>>>>>>>>> Via: SIP/2.0/UDP 98.192.104.214:5060
>>>>>>>>> ;branch=z9hG4bK69232672;received=98.192.104.214
>>>>>>>>>
>>>>>>>>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>>>>>>>>> >;tag=2E6BC0B0-2268
>>>>>>>>>
>>>>>>>>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>>>>>>>>
>>>>>>>>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>>>>>>>>
>>>>>>>>> CSeq: 105 INVITE
>>>>>>>>>
>>>>>>>>> Server: Asterisk PBX 1.6.2.13
>>>>>>>>>
>>>>>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
>>>>>>>>> NOTIFY, INFO
>>>>>>>>>
>>>>>>>>> Supported: replaces, timer
>>>>>>>>>
>>>>>>>>> Require: timer
>>>>>>>>>
>>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>>
>>>>>>>>> Contact: <sip:17705439047 at 64.154.41.150>
>>>>>>>>>
>>>>>>>>> Content-Length: 0
>>>>>>>>>
>>>>>>>>> *Jan 15 20:19:35.194: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>>
>>>>>>>>> Received:
>>>>>>>>>
>>>>>>>>> SIP/2.0 200 OK
>>>>>>>>>
>>>>>>>>> Via: SIP/2.0/UDP 98.192.104.214:5060
>>>>>>>>> ;branch=z9hG4bK69232672;received=98.192.104.214
>>>>>>>>>
>>>>>>>>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>>>>>>>>> >;tag=2E6BC0B0-2268
>>>>>>>>>
>>>>>>>>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>>>>>>>>
>>>>>>>>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>>>>>>>>
>>>>>>>>> CSeq: 105 INVITE
>>>>>>>>>
>>>>>>>>> Server: Asterisk PBX 1.6.2.13
>>>>>>>>>
>>>>>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
>>>>>>>>> NOTIFY, INFO
>>>>>>>>>
>>>>>>>>> Supported: replaces, timer
>>>>>>>>>
>>>>>>>>> Require: timer
>>>>>>>>>
>>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>>
>>>>>>>>> Contact: <sip:17705439047 at 64.154.41.150>
>>>>>>>>>
>>>>>>>>> Content-Type: application/sdp
>>>>>>>>>
>>>>>>>>> Content-Length: 333
>>>>>>>>>
>>>>>>>>> v=0
>>>>>>>>>
>>>>>>>>> o=root 1685873050 1685873054 IN IP4 64.154.41.150
>>>>>>>>>
>>>>>>>>> s=Asterisk PBX 1.6.2.13
>>>>>>>>>
>>>>>>>>> c=IN IP4 64.154.41.150
>>>>>>>>>
>>>>>>>>> t=0 0
>>>>>>>>>
>>>>>>>>> m=audio 13014 RTP/AVP 3 8 0 18 101
>>>>>>>>>
>>>>>>>>> a=rtpmap:3 GSM/8000
>>>>>>>>>
>>>>>>>>> a=rtpmap:8 PCMA/8000
>>>>>>>>>
>>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>>
>>>>>>>>> a=rtpmap:18 G729/8000
>>>>>>>>>
>>>>>>>>> a=fmtp:18 annexb=no
>>>>>>>>>
>>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>>>
>>>>>>>>> a=fmtp:101 0-16
>>>>>>>>>
>>>>>>>>> a=ptime:20
>>>>>>>>>
>>>>>>>>> a=inactive
>>>>>>>>>
>>>>>>>>> *Jan 15 20:19:35.198: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>>
>>>>>>>>> Sent:
>>>>>>>>>
>>>>>>>>> SIP/2.0 200 OK
>>>>>>>>>
>>>>>>>>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28e6157f41c
>>>>>>>>>
>>>>>>>>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>>>>>>>>
>>>>>>>>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>>>>>>>>
>>>>>>>>> Date: Tue, 15 Jan 2013 20:19:35 GMT
>>>>>>>>>
>>>>>>>>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>>>>>>>>
>>>>>>>>> CSeq: 104 INVITE
>>>>>>>>>
>>>>>>>>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>>>>>>>>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>>>>>>>>>
>>>>>>>>> Allow-Events: telephone-event
>>>>>>>>>
>>>>>>>>> Remote-Party-ID: <sip:17705439047 at 10.1.200.1
>>>>>>>>> >;party=called;screen=no;privacy=off
>>>>>>>>>
>>>>>>>>> Contact: <sip:17705439047 at 10.1.200.1:5060;transport=tcp>
>>>>>>>>>
>>>>>>>>> Supported: replaces
>>>>>>>>>
>>>>>>>>> Supported: sdp-anat
>>>>>>>>>
>>>>>>>>> Server: Cisco-SIPGateway/IOS-12.x
>>>>>>>>>
>>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>>
>>>>>>>>> Require: timer
>>>>>>>>>
>>>>>>>>> Supported: timer
>>>>>>>>>
>>>>>>>>> Content-Type: application/sdp
>>>>>>>>>
>>>>>>>>> Content-Length: 277
>>>>>>>>>
>>>>>>>>> v=0
>>>>>>>>>
>>>>>>>>> o=CiscoSystemsSIP-GW-UserAgent 4444 5481 IN IP4 10.1.200.1
>>>>>>>>>
>>>>>>>>> s=SIP Call
>>>>>>>>>
>>>>>>>>> c=IN IP4 10.1.200.1
>>>>>>>>>
>>>>>>>>> t=0 0
>>>>>>>>>
>>>>>>>>> m=audio 19514 RTP/AVP 0 101 19
>>>>>>>>>
>>>>>>>>> c=IN IP4 10.1.200.1
>>>>>>>>>
>>>>>>>>> a=inactive
>>>>>>>>>
>>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>>
>>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>>>
>>>>>>>>> a=fmtp:101 0-16
>>>>>>>>>
>>>>>>>>> a=rtpmap:19 CN/8000
>>>>>>>>>
>>>>>>>>> a=ptime:20
>>>>>>>>>
>>>>>>>>> *Jan 15 20:19:35.206: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>>
>>>>>>>>> Received:
>>>>>>>>>
>>>>>>>>> ACK sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0
>>>>>>>>>
>>>>>>>>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28f6dca6616
>>>>>>>>>
>>>>>>>>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>>>>>>>>
>>>>>>>>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>>>>>>>>
>>>>>>>>> Date: Tue, 15 Jan 2013 19:57:42 GMT
>>>>>>>>>
>>>>>>>>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>>>>>>>>
>>>>>>>>> Max-Forwards: 70
>>>>>>>>>
>>>>>>>>> CSeq: 104 ACK
>>>>>>>>>
>>>>>>>>> Allow-Events: presence, kpml
>>>>>>>>>
>>>>>>>>> Content-Type: application/sdp
>>>>>>>>>
>>>>>>>>> Content-Length: 243
>>>>>>>>>
>>>>>>>>> v=0
>>>>>>>>>
>>>>>>>>> o=CiscoSystemsCCM-SIP 7322 5 IN IP4 10.1.80.10
>>>>>>>>>
>>>>>>>>> s=SIP Call
>>>>>>>>>
>>>>>>>>> c=IN IP4 10.1.10.18
>>>>>>>>>
>>>>>>>>> b=TIAS:64000
>>>>>>>>>
>>>>>>>>> b=AS:64
>>>>>>>>>
>>>>>>>>> t=0 0
>>>>>>>>>
>>>>>>>>> m=audio 21476 RTP/AVP 0 101
>>>>>>>>>
>>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>>
>>>>>>>>> a=ptime:20
>>>>>>>>>
>>>>>>>>> a=inactive
>>>>>>>>>
>>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>>>
>>>>>>>>> a=fmtp:101 0-15
>>>>>>>>>
>>>>>>>>> *Jan 15 20:19:35.210: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>>
>>>>>>>>> Sent:
>>>>>>>>>
>>>>>>>>> ACK sip:17705439047 at 64.154.41.150:5060 SIP/2.0
>>>>>>>>>
>>>>>>>>> Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69246AB
>>>>>>>>>
>>>>>>>>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>>>>>>>>> >;tag=2E6BC0B0-2268
>>>>>>>>>
>>>>>>>>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>>>>>>>>
>>>>>>>>> Date: Tue, 15 Jan 2013 20:19:35 GMT
>>>>>>>>>
>>>>>>>>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>>>>>>>>
>>>>>>>>> Max-Forwards: 70
>>>>>>>>>
>>>>>>>>> CSeq: 105 ACK
>>>>>>>>>
>>>>>>>>> Authorization: Digest username="6784563290",realm="asterisk",uri="
>>>>>>>>> sip:17705439047 at 64.154.41.150:5060
>>>>>>>>> ",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5
>>>>>>>>>
>>>>>>>>> Allow-Events: telephone-event
>>>>>>>>>
>>>>>>>>> Content-Type: application/sdp
>>>>>>>>>
>>>>>>>>> Content-Length: 265
>>>>>>>>>
>>>>>>>>> v=0
>>>>>>>>>
>>>>>>>>> o=CiscoSystemsSIP-GW-UserAgent 3168 2741 IN IP4 98.192.104.214
>>>>>>>>>
>>>>>>>>> s=SIP Call
>>>>>>>>>
>>>>>>>>> c=IN IP4 98.192.104.214
>>>>>>>>>
>>>>>>>>> t=0 0
>>>>>>>>>
>>>>>>>>> m=audio 19458 RTP/AVP 0 101
>>>>>>>>>
>>>>>>>>> c=IN IP4 98.192.104.214
>>>>>>>>>
>>>>>>>>> a=inactive
>>>>>>>>>
>>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>>
>>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>>>
>>>>>>>>> a=fmtp:101 0-16
>>>>>>>>>
>>>>>>>>> a=ptime:20
>>>>>>>>>
>>>>>>>>> Cisco3825#
>>>>>>>>>
>>>>>>>>> Cisco3825#
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> Cisco3825#
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> INVITE sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0
>>>>>>>>>
>>>>>>>>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28e6157f41c
>>>>>>>>>
>>>>>>>>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>>>>>>>>
>>>>>>>>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>>>>>>>>
>>>>>>>>> Date: Tue, 15 Jan 2013 19:57:42 GMT
>>>>>>>>>
>>>>>>>>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>>>>>>>>
>>>>>>>>> Supported: timer,resource-priority,replaces
>>>>>>>>>
>>>>>>>>> Min-SE: 1800
>>>>>>>>>
>>>>>>>>> User-Agent: Cisco-CUCM8.6
>>>>>>>>>
>>>>>>>>> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE,
>>>>>>>>> REFER, SUBSCRIBE, NOTIFY
>>>>>>>>>
>>>>>>>>> CSeq: 104 INVITE
>>>>>>>>>
>>>>>>>>> Max-Forwards: 70
>>>>>>>>>
>>>>>>>>> Expires: 180
>>>>>>>>>
>>>>>>>>> Allow-Events: presence
>>>>>>>>>
>>>>>>>>> Supported: X-cisco-srtp-fallback
>>>>>>>>>
>>>>>>>>> Supported: Geolocation
>>>>>>>>>
>>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>>
>>>>>>>>> P-Asserted-Identity: "Dane Newman" <sip:6784563290 at 10.1.80.10>
>>>>>>>>>
>>>>>>>>> Remote-Party-ID: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>>> >;party=calling;screen=yes;privacy=off
>>>>>>>>>
>>>>>>>>> Contact: <sip:6784563290 at 10.1.80.10:5060
>>>>>>>>> ;transport=tcp>;video;audio;video
>>>>>>>>>
>>>>>>>>> Content-Length: 0
>>>>>>>>>
>>>>>>>>> *Jan 15 20:19:35.170: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>>
>>>>>>>>> Sent:
>>>>>>>>>
>>>>>>>>> INVITE sip:17705439047 at 64.154.41.150:5060 SIP/2.0
>>>>>>>>>
>>>>>>>>> Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69232672
>>>>>>>>>
>>>>>>>>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>>>>>>>>> >;tag=2E6BC0B0-2268
>>>>>>>>>
>>>>>>>>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>>>>>>>>
>>>>>>>>> Date: Tue, 15 Jan 2013 20:19:35 GMT
>>>>>>>>>
>>>>>>>>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>>>>>>>>
>>>>>>>>> Supported: timer,resource-priority,replaces,sdp-anat
>>>>>>>>>
>>>>>>>>> Min-SE: 1800
>>>>>>>>>
>>>>>>>>> Cisco-Guid: 3257897472-0000065536-0000000035-0173015306
>>>>>>>>>
>>>>>>>>> User-Agent: Cisco-SIPGateway/IOS-12.x
>>>>>>>>>
>>>>>>>>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>>>>>>>>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>>>>>>>>>
>>>>>>>>> CSeq: 105 INVITE
>>>>>>>>>
>>>>>>>>> Max-Forwards: 70
>>>>>>>>>
>>>>>>>>> Timestamp: 1358281175
>>>>>>>>>
>>>>>>>>> Contact: <sip:6784563290 at 98.192.104.214:5060>
>>>>>>>>>
>>>>>>>>> Expires: 180
>>>>>>>>>
>>>>>>>>> Allow-Events: telephone-event
>>>>>>>>>
>>>>>>>>> Authorization: Digest username="6784563290",realm="asterisk",uri="
>>>>>>>>> sip:17705439047 at 64.154.41.150:5060
>>>>>>>>> ",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5
>>>>>>>>>
>>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>>
>>>>>>>>> Content-Length: 0
>>>>>>>>>
>>>>>>>>> *Jan 15 20:19:35.190: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>>
>>>>>>>>> Sent:
>>>>>>>>>
>>>>>>>>> SIP/2.0 100 Trying
>>>>>>>>>
>>>>>>>>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28e6157f41c
>>>>>>>>>
>>>>>>>>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>>>>>>>>
>>>>>>>>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>>>>>>>>
>>>>>>>>> Date: Tue, 15 Jan 2013 20:19:35 GMT
>>>>>>>>>
>>>>>>>>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>>>>>>>>
>>>>>>>>> CSeq: 104 INVITE
>>>>>>>>>
>>>>>>>>> Allow-Events: telephone-event
>>>>>>>>>
>>>>>>>>> Server: Cisco-SIPGateway/IOS-12.x
>>>>>>>>>
>>>>>>>>> Content-Length: 0
>>>>>>>>>
>>>>>>>>> *Jan 15 20:19:35.194: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>>
>>>>>>>>> Received:
>>>>>>>>>
>>>>>>>>> SIP/2.0 100 Trying
>>>>>>>>>
>>>>>>>>> Via: SIP/2.0/UDP 98.192.104.214:5060
>>>>>>>>> ;branch=z9hG4bK69232672;received=98.192.104.214
>>>>>>>>>
>>>>>>>>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>>>>>>>>> >;tag=2E6BC0B0-2268
>>>>>>>>>
>>>>>>>>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>>>>>>>>
>>>>>>>>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>>>>>>>>
>>>>>>>>> CSeq: 105 INVITE
>>>>>>>>>
>>>>>>>>> Server: Asterisk PBX 1.6.2.13
>>>>>>>>>
>>>>>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
>>>>>>>>> NOTIFY, INFO
>>>>>>>>>
>>>>>>>>> Supported: replaces, timer
>>>>>>>>>
>>>>>>>>> Require: timer
>>>>>>>>>
>>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>>
>>>>>>>>> Contact: <sip:17705439047 at 64.154.41.150>
>>>>>>>>>
>>>>>>>>> Content-Length: 0
>>>>>>>>>
>>>>>>>>> *Jan 15 20:19:35.194: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>>
>>>>>>>>> Received:
>>>>>>>>>
>>>>>>>>> SIP/2.0 200 OK
>>>>>>>>>
>>>>>>>>> Via: SIP/2.0/UDP 98.192.104.214:5060
>>>>>>>>> ;branch=z9hG4bK69232672;received=98.192.104.214
>>>>>>>>>
>>>>>>>>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>>>>>>>>> >;tag=2E6BC0B0-2268
>>>>>>>>>
>>>>>>>>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>>>>>>>>
>>>>>>>>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>>>>>>>>
>>>>>>>>> CSeq: 105 INVITE
>>>>>>>>>
>>>>>>>>> Server: Asterisk PBX 1.6.2.13
>>>>>>>>>
>>>>>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
>>>>>>>>> NOTIFY, INFO
>>>>>>>>>
>>>>>>>>> Supported: replaces, timer
>>>>>>>>>
>>>>>>>>> Require: timer
>>>>>>>>>
>>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>>
>>>>>>>>> Contact: <sip:17705439047 at 64.154.41.150>
>>>>>>>>>
>>>>>>>>> Content-Type: application/sdp
>>>>>>>>>
>>>>>>>>> Content-Length: 333
>>>>>>>>>
>>>>>>>>> v=0
>>>>>>>>>
>>>>>>>>> o=root 1685873050 1685873054 IN IP4 64.154.41.150
>>>>>>>>>
>>>>>>>>> s=Asterisk PBX 1.6.2.13
>>>>>>>>>
>>>>>>>>> c=IN IP4 64.154.41.150
>>>>>>>>>
>>>>>>>>> t=0 0
>>>>>>>>>
>>>>>>>>> m=audio 13014 RTP/AVP 3 8 0 18 101
>>>>>>>>>
>>>>>>>>> a=rtpmap:3 GSM/8000
>>>>>>>>>
>>>>>>>>> a=rtpmap:8 PCMA/8000
>>>>>>>>>
>>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>>
>>>>>>>>> a=rtpmap:18 G729/8000
>>>>>>>>>
>>>>>>>>> a=fmtp:18 annexb=no
>>>>>>>>>
>>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>>>
>>>>>>>>> a=fmtp:101 0-16
>>>>>>>>>
>>>>>>>>> a=ptime:20
>>>>>>>>>
>>>>>>>>> a=inactive
>>>>>>>>>
>>>>>>>>> *Jan 15 20:19:35.198: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>>
>>>>>>>>> Sent:
>>>>>>>>>
>>>>>>>>> SIP/2.0 200 OK
>>>>>>>>>
>>>>>>>>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28e6157f41c
>>>>>>>>>
>>>>>>>>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>>>>>>>>
>>>>>>>>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>>>>>>>>
>>>>>>>>> Date: Tue, 15 Jan 2013 20:19:35 GMT
>>>>>>>>>
>>>>>>>>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>>>>>>>>
>>>>>>>>> CSeq: 104 INVITE
>>>>>>>>>
>>>>>>>>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>>>>>>>>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>>>>>>>>>
>>>>>>>>> Allow-Events: telephone-event
>>>>>>>>>
>>>>>>>>> Remote-Party-ID: <sip:17705439047 at 10.1.200.1
>>>>>>>>> >;party=called;screen=no;privacy=off
>>>>>>>>>
>>>>>>>>> Contact: <sip:17705439047 at 10.1.200.1:5060;transport=tcp>
>>>>>>>>>
>>>>>>>>> Supported: replaces
>>>>>>>>>
>>>>>>>>> Supported: sdp-anat
>>>>>>>>>
>>>>>>>>> Server: Cisco-SIPGateway/IOS-12.x
>>>>>>>>>
>>>>>>>>> Session-Expires: 1800;refresher=uas
>>>>>>>>>
>>>>>>>>> Require: timer
>>>>>>>>>
>>>>>>>>> Supported: timer
>>>>>>>>>
>>>>>>>>> Content-Type: application/sdp
>>>>>>>>>
>>>>>>>>> Content-Length: 277
>>>>>>>>>
>>>>>>>>> v=0
>>>>>>>>>
>>>>>>>>> o=CiscoSystemsSIP-GW-UserAgent 4444 5481 IN IP4 10.1.200.1
>>>>>>>>>
>>>>>>>>> s=SIP Call
>>>>>>>>>
>>>>>>>>> c=IN IP4 10.1.200.1
>>>>>>>>>
>>>>>>>>> t=0 0
>>>>>>>>>
>>>>>>>>> m=audio 19514 RTP/AVP 0 101 19
>>>>>>>>>
>>>>>>>>> c=IN IP4 10.1.200.1
>>>>>>>>>
>>>>>>>>> a=inactive
>>>>>>>>>
>>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>>
>>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>>>
>>>>>>>>> a=fmtp:101 0-16
>>>>>>>>>
>>>>>>>>> a=rtpmap:19 CN/8000
>>>>>>>>>
>>>>>>>>> a=ptime:20
>>>>>>>>>
>>>>>>>>> *Jan 15 20:19:35.206: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>>
>>>>>>>>> Received:
>>>>>>>>>
>>>>>>>>> ACK sip:17705439047 at 10.1.200.1:5060;transport=tcp SIP/2.0
>>>>>>>>>
>>>>>>>>> Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28f6dca6616
>>>>>>>>>
>>>>>>>>> From: "Dane Newman" <sip:6784563290 at 10.1.80.10
>>>>>>>>> >;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957
>>>>>>>>>
>>>>>>>>> To: <sip:17705439047 at 10.1.200.1>;tag=2E6BC6F0-1E22
>>>>>>>>>
>>>>>>>>> Date: Tue, 15 Jan 2013 19:57:42 GMT
>>>>>>>>>
>>>>>>>>> Call-ID: c22f9200-f51b49d-c5-a50010a at 10.1.80.10
>>>>>>>>>
>>>>>>>>> Max-Forwards: 70
>>>>>>>>>
>>>>>>>>> CSeq: 104 ACK
>>>>>>>>>
>>>>>>>>> Allow-Events: presence, kpml
>>>>>>>>>
>>>>>>>>> Content-Type: application/sdp
>>>>>>>>>
>>>>>>>>> Content-Length: 243
>>>>>>>>>
>>>>>>>>> v=0
>>>>>>>>>
>>>>>>>>> o=CiscoSystemsCCM-SIP 7322 5 IN IP4 10.1.80.10
>>>>>>>>>
>>>>>>>>> s=SIP Call
>>>>>>>>>
>>>>>>>>> c=IN IP4 10.1.10.18
>>>>>>>>>
>>>>>>>>> b=TIAS:64000
>>>>>>>>>
>>>>>>>>> b=AS:64
>>>>>>>>>
>>>>>>>>> t=0 0
>>>>>>>>>
>>>>>>>>> m=audio 21476 RTP/AVP 0 101
>>>>>>>>>
>>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>>
>>>>>>>>> a=ptime:20
>>>>>>>>>
>>>>>>>>> a=inactive
>>>>>>>>>
>>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>>>
>>>>>>>>> a=fmtp:101 0-15
>>>>>>>>>
>>>>>>>>> *Jan 15 20:19:35.210: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:
>>>>>>>>>
>>>>>>>>> Sent:
>>>>>>>>>
>>>>>>>>> ACK sip:17705439047 at 64.154.41.150:5060 SIP/2.0
>>>>>>>>>
>>>>>>>>> Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69246AB
>>>>>>>>>
>>>>>>>>> From: "Dane Newman" <sip:6784563290 at sipconnect.ipcomms.net
>>>>>>>>> >;tag=2E6BC0B0-2268
>>>>>>>>>
>>>>>>>>> To: <sip:17705439047 at sip.talkinip.net>;tag=as7c5ff82e
>>>>>>>>>
>>>>>>>>> Date: Tue, 15 Jan 2013 20:19:35 GMT
>>>>>>>>>
>>>>>>>>> Call-ID: A750328E-5E8711E2-8DC8A5CA-19425D52 at 98.192.104.214
>>>>>>>>>
>>>>>>>>> Max-Forwards: 70
>>>>>>>>>
>>>>>>>>> CSeq: 105 ACK
>>>>>>>>>
>>>>>>>>> Authorization: Digest username="6784563290",realm="asterisk",uri="
>>>>>>>>> sip:17705439047 at 64.154.41.150:5060
>>>>>>>>> ",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5
>>>>>>>>>
>>>>>>>>> Allow-Events: telephone-event
>>>>>>>>>
>>>>>>>>> Content-Type: application/sdp
>>>>>>>>>
>>>>>>>>> Content-Length: 265
>>>>>>>>>
>>>>>>>>> v=0
>>>>>>>>>
>>>>>>>>> o=CiscoSystemsSIP-GW-UserAgent 3168 2741 IN IP4 98.192.104.214
>>>>>>>>>
>>>>>>>>> s=SIP Call
>>>>>>>>>
>>>>>>>>> c=IN IP4 98.192.104.214
>>>>>>>>>
>>>>>>>>> t=0 0
>>>>>>>>>
>>>>>>>>> m=audio 19458 RTP/AVP 0 101
>>>>>>>>>
>>>>>>>>> c=IN IP4 98.192.104.214
>>>>>>>>>
>>>>>>>>> a=inactive
>>>>>>>>>
>>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>>
>>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>>>
>>>>>>>>> a=fmtp:101 0-16
>>>>>>>>>
>>>>>>>>> a=ptime:20
>>>>>>>>>
>>>>>>>>> Cisco3825#
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Tue, Jan 15, 2013 at 2:28 PM, Ryan Ratliff <rratliff at cisco.com>wrote:
>>>>>>>>>
>>>>>>>>>> ccsip message is what I'd go with just to see the signaling with
>>>>>>>>>> no other stuff.  Depending on what that shows and what your gateway is
>>>>>>>>>> doing to the signals you may need to expand from there.
>>>>>>>>>>
>>>>>>>>>> -Ryan
>>>>>>>>>>
>>>>>>>>>> On Jan 15, 2013, at 2:11 PM, Dane Newman <dane.newman at gmail.com>
>>>>>>>>>> wrote:
>>>>>>>>>>
>>>>>>>>>> Ryan
>>>>>>>>>>
>>>>>>>>>> What is the proper debug to use to caputre the useful information?
>>>>>>>>>>
>>>>>>>>>> Dane
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On Tue, Jan 15, 2013 at 12:42 PM, Ryan Ratliff <
>>>>>>>>>> rratliff at cisco.com> wrote:
>>>>>>>>>>
>>>>>>>>>>> Without sip messages I can't get any clues from that.
>>>>>>>>>>>
>>>>>>>>>>> -Ryan
>>>>>>>>>>>
>>>>>>>>>>> On Jan 15, 2013, at 12:35 PM, Dane Newman <dane.newman at gmail.com>
>>>>>>>>>>> wrote:
>>>>>>>>>>>
>>>>>>>>>>> Thanks Ryan for the input
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> *On the call when I hold the call the following debug pops
>>>>>>>>>>> out....*
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> *Jan 15 17:56:05.246:
>>>>>>>>>>> //13939/922252E78D73/SIP/Error/ccsip_api_request_offer: Unable to add
>>>>>>>>>>> passthru hdrs to
>>>>>>>>>>>                                container
>>>>>>>>>>> SIP: Attribute mid, level 1 instance 1 not found.
>>>>>>>>>>> SIP: (13938) Group (a= group line) attribute, level 65535
>>>>>>>>>>> instance 1 not found.
>>>>>>>>>>> *Jan 15 17:56:05.274:
>>>>>>>>>>> //13938/922252E78D73/SIP/Error/ccsip_api_response_answer: Unable to add
>>>>>>>>>>>                                            passthru headers to
>>>>>>>>>>> container
>>>>>>>>>>> SIP: Attribute mid, level 1 instance 1 not found.
>>>>>>>>>>> SIP: (13939) Group (a= group line) attribute, level 65535
>>>>>>>>>>> instance 1 not found.
>>>>>>>>>>> *Jan 15 17:56:05.286:
>>>>>>>>>>> //13939/922252E78D73/SIP/Error/ccsip_api_request_offer: Unable to add
>>>>>>>>>>> passthru hdrs to
>>>>>>>>>>>                                container
>>>>>>>>>>> *Jan 15 17:56:05.302:
>>>>>>>>>>> //13938/922252E78D73/SIP/Error/ccsip_api_response_answer: Unable to add
>>>>>>>>>>>                                            passthru headers to
>>>>>>>>>>> container
>>>>>>>>>>> SIP: Attribute mid, level 1 instance 1 not found.
>>>>>>>>>>> SIP: (13939) Group (a= group line) attribute, level 65535
>>>>>>>>>>> instance 1 not found.
>>>>>>>>>>> SIP: Attribute mid, level 1 instance 1 not found.
>>>>>>>>>>> *Jan 15 17:56:05.322:
>>>>>>>>>>> //13939/922252E78D73/SIP/Error/sipSPIProcessAckMedia: Could not modify QoS
>>>>>>>>>>> params for midcall INVITE
>>>>>>>>>>>
>>>>>>>>>>> *After I try to unhold the call the following debug comes
>>>>>>>>>>> out....*
>>>>>>>>>>> **
>>>>>>>>>>>
>>>>>>>>>>> *Jan 15 17:56:18.874:
>>>>>>>>>>> //13939/922252E78D73/SIP/Error/ccsip_api_request_offer: Unable to add
>>>>>>>>>>> passthru hdrs to
>>>>>>>>>>>                                container
>>>>>>>>>>> *Jan 15 17:56:18.894:
>>>>>>>>>>> //13938/922252E78D73/SIP/Error/ccsip_api_response_answer: Unable to add
>>>>>>>>>>>                                            passthru headers to
>>>>>>>>>>> container
>>>>>>>>>>> SIP: Attribute mid, level 1 instance 1 not found.
>>>>>>>>>>> SIP: (13939) Group (a= group line) attribute, level 65535
>>>>>>>>>>> instance 1 not found.
>>>>>>>>>>> SIP: Attribute mid, level 1 instance 1 not found.
>>>>>>>>>>> *Jan 15 17:56:18.906:
>>>>>>>>>>> //13939/922252E78D73/SIP/Error/sipSPIProcessAckMedia: Could not modify QoS
>>>>>>>>>>> params for midcall INVITE
>>>>>>>>>>> Cisco3825#
>>>>>>>>>>> Cisco3825#
>>>>>>>>>>> Cisco3825#
>>>>>>>>>>>
>>>>>>>>>>> On Tue, Jan 15, 2013 at 9:42 AM, Ryan Ratliff <
>>>>>>>>>>> rratliff at cisco.com> wrote:
>>>>>>>>>>>
>>>>>>>>>>>> Given you have an ITSP it's most likely the initial hold that's
>>>>>>>>>>>> failing, which is only manifesting when you try to resume it.  If you
>>>>>>>>>>>> haven't noticed already  this is also very likely causing transfers to fail.
>>>>>>>>>>>>
>>>>>>>>>>>> Take a look at the SIP signaling for a call.   I believe the
>>>>>>>>>>>> most common cause to this is the ITSP not handling our transition from
>>>>>>>>>>>> active->inactive->sendonly->active from hold to MOH to resume.   The
>>>>>>>>>>>> "Duplex Streaming Enabled" parameter is there just for this type of problem.
>>>>>>>>>>>>
>>>>>>>>>>>> -Ryan
>>>>>>>>>>>>
>>>>>>>>>>>> On Jan 14, 2013, at 6:40 PM, Dane Newman <dane.newman at gmail.com>
>>>>>>>>>>>> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>> *Hello Kenneth*
>>>>>>>>>>>> **
>>>>>>>>>>>> *I have restarted both CUCM servers so this should have
>>>>>>>>>>>> restarted the services when the utils system restart happened*
>>>>>>>>>>>> **
>>>>>>>>>>>>
>>>>>>>>>>>> *on my router I see I am using g711 from the debug *
>>>>>>>>>>>> **
>>>>>>>>>>>> *I ran a debug voip ccapi inout *
>>>>>>>>>>>> **
>>>>>>>>>>>> *I connected a call calling from an external number to a DiD
>>>>>>>>>>>> inside of my system.  Once the call was connected I put the call on hold
>>>>>>>>>>>> and the following debug came out..the music on hold played for the external
>>>>>>>>>>>> caller*
>>>>>>>>>>>>
>>>>>>>>>>>> *Jan 14 23:47:40.779:
>>>>>>>>>>>> //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:
>>>>>>>>>>>>    Stop Tone On Digit=FALSE, Tone=Null,
>>>>>>>>>>>>    Tone Direction=Sum Network, Params=0x0, Call Id=12741
>>>>>>>>>>>> *Jan 14 23:47:40.783:
>>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
>>>>>>>>>>>>    Destination Interface=0xC05A65AC, Destination Call Id=12741,
>>>>>>>>>>>> Source Call Id=12742, Xmit Function=0x64204BAC
>>>>>>>>>>>> *Jan 14 23:47:40.783:
>>>>>>>>>>>> //12741/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
>>>>>>>>>>>> *Jan 14 23:47:40.783: cc_api_get_xcode_stream : 4702
>>>>>>>>>>>> *Jan 14 23:47:40.783:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>>>>>>>>>    Destination Interface=0xC05A65AC, Destination Call Id=12741,
>>>>>>>>>>>> Source Call Id=12742,
>>>>>>>>>>>>    Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
>>>>>>>>>>>>    Modem=0x0, Codec Bytes=20, Signal Type=2)
>>>>>>>>>>>> *Jan 14 23:47:40.783:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>>>>>>>>>    Caps(Playout Mode=1, Playout Initial=60(ms), Playout
>>>>>>>>>>>> Min=40(ms),
>>>>>>>>>>>>    Playout Max=1000(ms), Fax Nom=300(ms))
>>>>>>>>>>>> *Jan 14 23:47:40.783:
>>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>>>>>>>>>    Destination Interface=0xC05A65AC, Destination Call Id=12742,
>>>>>>>>>>>> Source Call Id=12741,
>>>>>>>>>>>>    Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1),
>>>>>>>>>>>> Vad=ON(0x2),
>>>>>>>>>>>>    Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num
>>>>>>>>>>>> Start=1046)
>>>>>>>>>>>> *Jan 14 23:47:40.783:
>>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>>>>>>>>>    Destination Interface=0xC05A65AC, Destination Call Id=12742,
>>>>>>>>>>>> Source Call Id=12741,
>>>>>>>>>>>>    Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1),
>>>>>>>>>>>> Vad=ON(0x2),
>>>>>>>>>>>>    Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num
>>>>>>>>>>>> Start=1046)
>>>>>>>>>>>> *Jan 14 23:47:40.783:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>>>    Event=170, Call Id=12742
>>>>>>>>>>>> *Jan 14 23:47:40.783:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>>>    Event Is Sent To Conferenced SPI(s) Directly
>>>>>>>>>>>> *Jan 14 23:47:40.783:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_call_feature:
>>>>>>>>>>>>    Feature Type=50, Interface=0xC05A65AC, Call Id=12742
>>>>>>>>>>>> *Jan 14 23:47:40.811:
>>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>>>>>>>>>    Destination Interface=0xC05A65AC, Destination Call Id=12742,
>>>>>>>>>>>> Source Call Id=12741,
>>>>>>>>>>>>    Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
>>>>>>>>>>>>    Modem=0x0, Codec Bytes=20, Signal Type=2)
>>>>>>>>>>>> *Jan 14 23:47:40.811:
>>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>>>>>>>>>    Caps(Playout Mode=1, Playout Initial=60(ms), Playout
>>>>>>>>>>>> Min=40(ms),
>>>>>>>>>>>>    Playout Max=1000(ms), Fax Nom=300(ms))
>>>>>>>>>>>> *Jan 14 23:47:40.811:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>>>>>>>>>    Destination Interface=0xC05A65AC, Destination Call Id=12741,
>>>>>>>>>>>> Source Call Id=12742,
>>>>>>>>>>>>    Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1),
>>>>>>>>>>>> Vad=ON(0x2),
>>>>>>>>>>>>    Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num
>>>>>>>>>>>> Start=1516)
>>>>>>>>>>>> *Jan 14 23:47:40.811:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>>>>>>>>>    Destination Interface=0xC05A65AC, Destination Call Id=12741,
>>>>>>>>>>>> Source Call Id=12742,
>>>>>>>>>>>>    Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1),
>>>>>>>>>>>> Vad=ON(0x2),
>>>>>>>>>>>>    Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num
>>>>>>>>>>>> Start=1516)
>>>>>>>>>>>> *Jan 14 23:47:40.811:
>>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>>>    Event=171, Call Id=12741
>>>>>>>>>>>> *Jan 14 23:47:40.811:
>>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>>>    Event Is Sent To Conferenced SPI(s) Directly
>>>>>>>>>>>> *Jan 14 23:47:40.815:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_call_facility:
>>>>>>>>>>>>    Interface=0xC05A65AC, Call Id=12742
>>>>>>>>>>>> *Jan 14 23:47:40.819:
>>>>>>>>>>>> //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:
>>>>>>>>>>>>    Stop Tone On Digit=FALSE, Tone=Null,
>>>>>>>>>>>>    Tone Direction=Sum Network, Params=0x0, Call Id=12741
>>>>>>>>>>>> *Jan 14 23:47:40.819:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>>>    Event=96, Call Id=12742
>>>>>>>>>>>> *Jan 14 23:47:40.819:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>>>    Event Is Sent To Conferenced SPI(s) Directly
>>>>>>>>>>>> *Jan 14 23:47:40.839:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
>>>>>>>>>>>>    Destination Interface=0xC05A65AC, Destination Call Id=12742,
>>>>>>>>>>>> Source Call Id=12741, Xmit Function=0x64204BAC
>>>>>>>>>>>> *Jan 14 23:47:40.839:
>>>>>>>>>>>> //12742/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
>>>>>>>>>>>> *Jan 14 23:47:40.839: cc_api_get_xcode_stream : 4702
>>>>>>>>>>>> *Jan 14 23:47:40.843:
>>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>>>>>>>>>    Destination Interface=0xC05A65AC, Destination Call Id=12742,
>>>>>>>>>>>> Source Call Id=12741,
>>>>>>>>>>>>    Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
>>>>>>>>>>>>    Modem=0x0, Codec Bytes=20, Signal Type=2)
>>>>>>>>>>>> *Jan 14 23:47:40.843:
>>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>>>>>>>>>    Caps(Playout Mode=1, Playout Initial=60(ms), Playout
>>>>>>>>>>>> Min=40(ms),
>>>>>>>>>>>>    Playout Max=1000(ms), Fax Nom=300(ms))
>>>>>>>>>>>> *Jan 14 23:47:40.843:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>>>>>>>>>    Destination Interface=0xC05A65AC, Destination Call Id=12741,
>>>>>>>>>>>> Source Call Id=12742,
>>>>>>>>>>>>    Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1),
>>>>>>>>>>>> Vad=ON(0x2),
>>>>>>>>>>>>    Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num
>>>>>>>>>>>> Start=1516)
>>>>>>>>>>>> *Jan 14 23:47:40.843:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>>>>>>>>>    Destination Interface=0xC05A65AC, Destination Call Id=12741,
>>>>>>>>>>>> Source Call Id=12742,
>>>>>>>>>>>>    Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1),
>>>>>>>>>>>> Vad=ON(0x2),
>>>>>>>>>>>>    Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num
>>>>>>>>>>>> Start=1516)
>>>>>>>>>>>> *Jan 14 23:47:40.843:
>>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>>>    Event=170, Call Id=12741
>>>>>>>>>>>> *Jan 14 23:47:40.843:
>>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>>>    Event Is Sent To Conferenced SPI(s) Directly
>>>>>>>>>>>> *Jan 14 23:47:40.859:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>>>>>>>>>    Destination Interface=0xC05A65AC, Destination Call Id=12741,
>>>>>>>>>>>> Source Call Id=12742,
>>>>>>>>>>>>    Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
>>>>>>>>>>>>    Modem=0x0, Codec Bytes=20, Signal Type=2)
>>>>>>>>>>>> *Jan 14 23:47:40.863:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>>>>>>>>>    Caps(Playout Mode=1, Playout Initial=60(ms), Playout
>>>>>>>>>>>> Min=40(ms),
>>>>>>>>>>>>    Playout Max=1000(ms), Fax Nom=300(ms))
>>>>>>>>>>>> *Jan 14 23:47:40.863:
>>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>>>>>>>>>    Destination Interface=0xC05A65AC, Destination Call Id=12742,
>>>>>>>>>>>> Source Call Id=12741,
>>>>>>>>>>>>    Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1),
>>>>>>>>>>>> Vad=ON(0x2),
>>>>>>>>>>>>    Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num
>>>>>>>>>>>> Start=3996)
>>>>>>>>>>>> *Jan 14 23:47:40.863:
>>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>>>>>>>>>    Destination Interface=0xC05A65AC, Destination Call Id=12742,
>>>>>>>>>>>> Source Call Id=12741,
>>>>>>>>>>>>    Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1),
>>>>>>>>>>>> Vad=ON(0x2),
>>>>>>>>>>>>    Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num
>>>>>>>>>>>> Start=3996)
>>>>>>>>>>>> *Jan 14 23:47:40.863:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>>>    Event=171, Call Id=12742
>>>>>>>>>>>> *Jan 14 23:47:40.863:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>>>    Event Is Sent To Conferenced SPI(s) Directly
>>>>>>>>>>>> *Jan 14 23:47:40.863:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_call_facility:
>>>>>>>>>>>>    Interface=0xC05A65AC, Call Id=12742
>>>>>>>>>>>> Cisco3825#
>>>>>>>>>>>> Cisco3825#
>>>>>>>>>>>> Cisco3825#
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> *I then after that took off the hold and the following debug
>>>>>>>>>>>> came out.  The call on the PSDN side still played the hold music while
>>>>>>>>>>>> there was no voice on the deskphone side.*
>>>>>>>>>>>>
>>>>>>>>>>>> *Jan 14 23:47:40.779:
>>>>>>>>>>>> //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:
>>>>>>>>>>>>    Stop Tone On Digit=FALSE, Tone=Null,
>>>>>>>>>>>>    Tone Direction=Sum Network, Params=0x0, Call Id=12741
>>>>>>>>>>>> *Jan 14 23:47:40.783:
>>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
>>>>>>>>>>>>    Destination Interface=0xC05A65AC, Destination Call Id=12741,
>>>>>>>>>>>> Source Call Id=12742, Xmit Function=0x64204BAC
>>>>>>>>>>>> *Jan 14 23:47:40.783:
>>>>>>>>>>>> //12741/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
>>>>>>>>>>>> *Jan 14 23:47:40.783: cc_api_get_xcode_stream : 4702
>>>>>>>>>>>> *Jan 14 23:47:40.783:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>>>>>>>>>    Destination Interface=0xC05A65AC, Destination Call Id=12741,
>>>>>>>>>>>> Source Call Id=12742,
>>>>>>>>>>>>    Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
>>>>>>>>>>>>    Modem=0x0, Codec Bytes=20, Signal Type=2)
>>>>>>>>>>>> *Jan 14 23:47:40.783:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>>>>>>>>>    Caps(Playout Mode=1, Playout Initial=60(ms), Playout
>>>>>>>>>>>> Min=40(ms),
>>>>>>>>>>>>    Playout Max=1000(ms), Fax Nom=300(ms))
>>>>>>>>>>>> *Jan 14 23:47:40.783:
>>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>>>>>>>>>    Destination Interface=0xC05A65AC, Destination Call Id=12742,
>>>>>>>>>>>> Source Call Id=12741,
>>>>>>>>>>>>    Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1),
>>>>>>>>>>>> Vad=ON(0x2),
>>>>>>>>>>>>    Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num
>>>>>>>>>>>> Start=1046)
>>>>>>>>>>>> *Jan 14 23:47:40.783:
>>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>>>>>>>>>    Destination Interface=0xC05A65AC, Destination Call Id=12742,
>>>>>>>>>>>> Source Call Id=12741,
>>>>>>>>>>>>    Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1),
>>>>>>>>>>>> Vad=ON(0x2),
>>>>>>>>>>>>    Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num
>>>>>>>>>>>> Start=1046)
>>>>>>>>>>>> *Jan 14 23:47:40.783:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>>>    Event=170, Call Id=12742
>>>>>>>>>>>> *Jan 14 23:47:40.783:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>>>    Event Is Sent To Conferenced SPI(s) Directly
>>>>>>>>>>>> *Jan 14 23:47:40.783:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_call_feature:
>>>>>>>>>>>>    Feature Type=50, Interface=0xC05A65AC, Call Id=12742
>>>>>>>>>>>> *Jan 14 23:47:40.811:
>>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>>>>>>>>>    Destination Interface=0xC05A65AC, Destination Call Id=12742,
>>>>>>>>>>>> Source Call Id=12741,
>>>>>>>>>>>>    Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
>>>>>>>>>>>>    Modem=0x0, Codec Bytes=20, Signal Type=2)
>>>>>>>>>>>> *Jan 14 23:47:40.811:
>>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>>>>>>>>>    Caps(Playout Mode=1, Playout Initial=60(ms), Playout
>>>>>>>>>>>> Min=40(ms),
>>>>>>>>>>>>    Playout Max=1000(ms), Fax Nom=300(ms))
>>>>>>>>>>>> *Jan 14 23:47:40.811:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>>>>>>>>>    Destination Interface=0xC05A65AC, Destination Call Id=12741,
>>>>>>>>>>>> Source Call Id=12742,
>>>>>>>>>>>>    Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1),
>>>>>>>>>>>> Vad=ON(0x2),
>>>>>>>>>>>>    Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num
>>>>>>>>>>>> Start=1516)
>>>>>>>>>>>> *Jan 14 23:47:40.811:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>>>>>>>>>    Destination Interface=0xC05A65AC, Destination Call Id=12741,
>>>>>>>>>>>> Source Call Id=12742,
>>>>>>>>>>>>    Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1),
>>>>>>>>>>>> Vad=ON(0x2),
>>>>>>>>>>>>    Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num
>>>>>>>>>>>> Start=1516)
>>>>>>>>>>>> *Jan 14 23:47:40.811:
>>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>>>    Event=171, Call Id=12741
>>>>>>>>>>>> *Jan 14 23:47:40.811:
>>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>>>    Event Is Sent To Conferenced SPI(s) Directly
>>>>>>>>>>>> *Jan 14 23:47:40.815:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_call_facility:
>>>>>>>>>>>>    Interface=0xC05A65AC, Call Id=12742
>>>>>>>>>>>> *Jan 14 23:47:40.819:
>>>>>>>>>>>> //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:
>>>>>>>>>>>>    Stop Tone On Digit=FALSE, Tone=Null,
>>>>>>>>>>>>    Tone Direction=Sum Network, Params=0x0, Call Id=12741
>>>>>>>>>>>> *Jan 14 23:47:40.819:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>>>    Event=96, Call Id=12742
>>>>>>>>>>>> *Jan 14 23:47:40.819:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>>>    Event Is Sent To Conferenced SPI(s) Directly
>>>>>>>>>>>> *Jan 14 23:47:40.839:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:
>>>>>>>>>>>>    Destination Interface=0xC05A65AC, Destination Call Id=12742,
>>>>>>>>>>>> Source Call Id=12741, Xmit Function=0x64204BAC
>>>>>>>>>>>> *Jan 14 23:47:40.839:
>>>>>>>>>>>> //12742/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
>>>>>>>>>>>> *Jan 14 23:47:40.839: cc_api_get_xcode_stream : 4702
>>>>>>>>>>>> *Jan 14 23:47:40.843:
>>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>>>>>>>>>    Destination Interface=0xC05A65AC, Destination Call Id=12742,
>>>>>>>>>>>> Source Call Id=12741,
>>>>>>>>>>>>    Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
>>>>>>>>>>>>    Modem=0x0, Codec Bytes=20, Signal Type=2)
>>>>>>>>>>>> *Jan 14 23:47:40.843:
>>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>>>>>>>>>    Caps(Playout Mode=1, Playout Initial=60(ms), Playout
>>>>>>>>>>>> Min=40(ms),
>>>>>>>>>>>>    Playout Max=1000(ms), Fax Nom=300(ms))
>>>>>>>>>>>> *Jan 14 23:47:40.843:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>>>>>>>>>    Destination Interface=0xC05A65AC, Destination Call Id=12741,
>>>>>>>>>>>> Source Call Id=12742,
>>>>>>>>>>>>    Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1),
>>>>>>>>>>>> Vad=ON(0x2),
>>>>>>>>>>>>    Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num
>>>>>>>>>>>> Start=1516)
>>>>>>>>>>>> *Jan 14 23:47:40.843:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>>>>>>>>>    Destination Interface=0xC05A65AC, Destination Call Id=12741,
>>>>>>>>>>>> Source Call Id=12742,
>>>>>>>>>>>>    Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1),
>>>>>>>>>>>> Vad=ON(0x2),
>>>>>>>>>>>>    Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num
>>>>>>>>>>>> Start=1516)
>>>>>>>>>>>> *Jan 14 23:47:40.843:
>>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>>>    Event=170, Call Id=12741
>>>>>>>>>>>> *Jan 14 23:47:40.843:
>>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>>>    Event Is Sent To Conferenced SPI(s) Directly
>>>>>>>>>>>> *Jan 14 23:47:40.859:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>>>>>>>>>    Destination Interface=0xC05A65AC, Destination Call Id=12741,
>>>>>>>>>>>> Source Call Id=12742,
>>>>>>>>>>>>    Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,
>>>>>>>>>>>>    Modem=0x0, Codec Bytes=20, Signal Type=2)
>>>>>>>>>>>> *Jan 14 23:47:40.863:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_caps_ind:
>>>>>>>>>>>>    Caps(Playout Mode=1, Playout Initial=60(ms), Playout
>>>>>>>>>>>> Min=40(ms),
>>>>>>>>>>>>    Playout Max=1000(ms), Fax Nom=300(ms))
>>>>>>>>>>>> *Jan 14 23:47:40.863:
>>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>>>>>>>>>    Destination Interface=0xC05A65AC, Destination Call Id=12742,
>>>>>>>>>>>> Source Call Id=12741,
>>>>>>>>>>>>    Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1),
>>>>>>>>>>>> Vad=ON(0x2),
>>>>>>>>>>>>    Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num
>>>>>>>>>>>> Start=3996)
>>>>>>>>>>>> *Jan 14 23:47:40.863:
>>>>>>>>>>>> //12741/9906C1828C05/CCAPI/cc_api_caps_ack:
>>>>>>>>>>>>    Destination Interface=0xC05A65AC, Destination Call Id=12742,
>>>>>>>>>>>> Source Call Id=12741,
>>>>>>>>>>>>    Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1),
>>>>>>>>>>>> Vad=ON(0x2),
>>>>>>>>>>>>    Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num
>>>>>>>>>>>> Start=3996)
>>>>>>>>>>>> *Jan 14 23:47:40.863:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>>>    Event=171, Call Id=12742
>>>>>>>>>>>> *Jan 14 23:47:40.863:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_event_indication:
>>>>>>>>>>>>    Event Is Sent To Conferenced SPI(s) Directly
>>>>>>>>>>>> *Jan 14 23:47:40.863:
>>>>>>>>>>>> //12742/9906C1828C05/CCAPI/cc_api_call_facility:
>>>>>>>>>>>>    Interface=0xC05A65AC, Call Id=12742
>>>>>>>>>>>> Cisco3825#
>>>>>>>>>>>> Cisco3825#
>>>>>>>>>>>> Cisco3825#
>>>>>>>>>>>>
>>>>>>>>>>>> On Mon, Jan 14, 2013 at 6:20 PM, Kenneth Hayes <
>>>>>>>>>>>> kennethwhayes at gmail.com> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>> Have you also restarted the Cisco IP Media Services?
>>>>>>>>>>>>>
>>>>>>>>>>>>> Sent from my iPhone
>>>>>>>>>>>>>
>>>>>>>>>>>>> On Jan 14, 2013, at 6:12 PM, Dane Newman <
>>>>>>>>>>>>> dane.newman at gmail.com> wrote:
>>>>>>>>>>>>>
>>>>>>>>>>>>> My ITSP will only support 711ulaw for me currently I believe.
>>>>>>>>>>>>> They hard coded it with me when I was initially setting it up.
>>>>>>>>>>>>>
>>>>>>>>>>>>> Do you think this could be a codec issue?  How would I go
>>>>>>>>>>>>> about identifying if it is?
>>>>>>>>>>>>>
>>>>>>>>>>>>> Dane
>>>>>>>>>>>>>
>>>>>>>>>>>>> On Mon, Jan 14, 2013 at 6:09 PM, Kenneth Hayes <
>>>>>>>>>>>>> kennethwhayes at gmail.com> wrote:
>>>>>>>>>>>>>
>>>>>>>>>>>>>> Have you tried different audio codecs?
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Sent from my iPhone
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> On Jan 14, 2013, at 6:06 PM, Dane Newman <
>>>>>>>>>>>>>> dane.newman at gmail.com> wrote:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Ryan (sorry I forgot to reply to all)
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Thanks for the Reply
>>>>>>>>>>>>>> Oddly enough we are.
>>>>>>>>>>>>>> This probably has something to do with MOH in general?
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Internally when I user puts another user on hold everything
>>>>>>>>>>>>>> works. No MOH plays and they can hold and unhold the call just fine.
>>>>>>>>>>>>>>  I tested calling from an external number. Once I put the
>>>>>>>>>>>>>> external caller on hold the MOH played but I was unable to resume the call.
>>>>>>>>>>>>>> When I hit resume on the deskphone the MOH still played to the external
>>>>>>>>>>>>>> caller and there was no sound on the deskphone.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> On Mon, Jan 14, 2013 at 5:25 PM, Ryan Ratliff <
>>>>>>>>>>>>>> rratliff at cisco.com> wrote:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Do you get similar behavior if you just hold and resume the
>>>>>>>>>>>>>>> call outside SNR features?
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> -Ryan
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> On Jan 14, 2013, at 4:18 PM, Dane Newman <
>>>>>>>>>>>>>>> dane.newman at gmail.com> wrote:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Using keyboard-interactive authentication.
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Password:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Cisco3825#
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Cisco3825#sh ver
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Cisco IOS Software, 3800 Software
>>>>>>>>>>>>>>> (C3825-ADVENTERPRISEK9_IVS_LI-M), Version 15.1
>>>>>>>>>>>>>>> (4)M5, RELEASE SOFTWARE (fc1)
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Technical Support: http://www.cisco.com/techsupport
>>>>>>>>>>>>>>> Copyright (c) 1986-2012 by Cisco Systems, Inc.
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Compiled Tue 04-Sep-12 17:25 by prod_rel_team
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> ROM: System Bootstrap, Version 12.4(13r)T10, RELEASE
>>>>>>>>>>>>>>> SOFTWARE (fc1)
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Cisco3825 uptime is 1 week, 1 day, 1 hour,
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>  ...
>
> [Message clipped]
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20130116/49391d36/attachment.html>