[cisco-voip] dtmf from cucm to 2821 cube to sip trunk
Dane Newman
dane.newman at gmail.com
Tue Oct 27 15:16:46 EDT 2009
Yes the session progress is receviced by the router
In all my debugs I noticed I have the same thing
*Oct 27 20:25:37.558: //1528/003E40690D00/SIP/Info/sipSPICheckDynPayloadUse:
Dynamic payload(101) could not be reserved.
*Oct 27 20:25:37.558:
//1528/003E40690D00/SIP/Info/sipSPIDoDTMFRelayNegotiation: RTP-NTE DTMF
relay option
*Oct 27 20:25:37.562:
//1528/003E40690D00/SIP/Info/sipSPIDoDTMFRelayNegotiation: Case of full
named event(NE) match in fmtp list of events.
*Oct 27 20:25:37.562:
//-1/xxxxxxxxxxxx/SIP/Info/sip_sdp_get_modem_relay_cap_params: NSE payload
from X-cap = 0
*Oct 27 20:25:37.562:
//1528/003E40690D00/SIP/Info/sip_select_modem_relay_params: X-tmr not
present in SDP. Disable modem relay
Is Dynamic payload(101) could not be reserved telling me I have no dtmf
support?
On Tue, Oct 27, 2009 at 2:56 PM, Ryan Ratliff <rratliff at cisco.com> wrote:
> I doubt that is related to your lack of DTMF but it's most likely the side
> sending the 183 is actually counting 1-16 and printing the 0. The Session
> Progress is received by the router isn't it?
>
> There are only 16 DTMF characters, the 12 on your keypad and 4 hidden ones
> A, B, C, and D.
>
> -Ryan
>
> On Oct 27, 2009, at 2:48 PM, Dane Newman wrote:
>
> The difference I see between the invite and the 183 session progression
> from the telco is
>
> invite
> a=fmtp:101 0-15
>
> session progression
> a=fmtp:101 0-16
>
> Could this miss match in supported digits be what is causing all dtmf not
> to work? How can I make my cisco router support 0-16?
>
> Dane
>
> *Invite*
> **
> **
> v=0
> o=CiscoSystemsSIP-GW-UserAgent 2461 126 IN IP4 173.14.220.57
> s=SIP Call
> c=IN IP4 173.14.220.57
> t=0 0
> m=audio 18770 RTP/AVP 0 101 19
> c=IN IP4 173.14.220.57
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=rtpmap:19 CN/8000
> a=ptime:20
>
>
>
> *session progression*
>
>
> v=0
> o=root 5115 5115 IN IP4 64.34.181.47
> s=session
> c=IN IP4 64.34.181.47
> t=0 0
> m=audio 17646 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
>
> On Tue, Oct 27, 2009 at 2:10 PM, Ryan Ratliff <rratliff at cisco.com> wrote:
>
>> Sorry this part is the actual DTMF:
>>
>> a=rtpmap:101 telephone-event/8000
>>
>> The line you quoted is part of the SDP and references both RTP and DTMF.
>> m=audio 11680 RTP/AVP 0 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>>
>> The fist line means your RTP is on port 11680 and references the a:rtpmap
>> entries for 0 and 101.
>> The second line means your RTP is g.711.
>> The 3rd line is the DTMF with a payload type of 101.
>> The 4th line means it can accept DTMF 0-16
>> The last line is pretty self explanatory (silence suppression disabled).
>>
>> This is a very basic interpretation of the SDP info. RFC 2327 is where
>> you want to go to get into the nitty-gritty details.
>>
>> -Ryan
>>
>> On Oct 27, 2009, at 2:00 PM, Ryan Ratliff wrote:
>>
>> That is RFC2833 DTMF with a payload type of 101.
>>
>> I do know that CUBE cannot do dynamic RFC2833 payload types. It can only
>> send the payloadType defined in the voip dial-peer. So if inbound calls use
>> a different payloadType than outbound calls you will want to update the
>> dial-peers accordingly.
>>
>>
>> -Ryan
>>
>> On Oct 27, 2009, at 12:56 PM, Dane Newman wrote:
>>
>> Well I tried to switch providers just to test it out and now I am getting
>> something back in the 183 but still no dtmf hmm
>>
>> I see they are sending me
>>
>> m=audio 11680 RTP/AVP 0 101
>>
>> How do I interperate that line?
>>
>>
>> Received:
>> SIP/2.0 183 Session Progress
>> Via: SIP/2.0/UDP 173.14.220.57:5060
>> ;branch=z9hG4bK749136B;received=173.14.220.57
>> From: <sip:6782282221 at did.voip.les.net<sip%3A6782282221 at did.voip.les.net>
>> >;tag=419FE94-8A1
>> To: <sip:18774675464 at did.voip.les.net<sip%3A18774675464 at did.voip.les.net>
>> >;tag=as5677a12c
>> Call-ID: AF45B372-C25911DE-80DAC992-790F56B7 at 173.14.220.57
>> CSeq: 101 INVITE
>> User-Agent: LES.NET.VoIP
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> Contact: <sip:18774675464 at 64.34.181.47 <sip%3A18774675464 at 64.34.181.47>>
>> Content-Type: application/sdp
>> Content-Length: 214
>> v=0
>> o=root 5115 5115 IN IP4 64.34.181.47
>> s=session
>> c=IN IP4 64.34.181.47
>> t=0 0
>> m=audio 11680 RTP/AVP 0 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> *Oct 27 18:02:12.551: //1345/0008DE602400/SIP/Info/sipSPICheckResponse:
>> INVITE response with no RSEQ - disable IS_REL1XX
>> *Oct 27 18:02:12.551: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentGTD: No
>> GTD found in inbound container
>> *Oct 27 18:02:12.551:
>> //1345/0008DE602400/SIP/Info/sipSPIDoMediaNegotiation: Number of m-lines = 1
>> SIP: Attribute mid, level 1 instance 1 not found.
>> *Oct 27 18:02:12.551:
>> //1345/0008DE602400/SIP/Info/resolve_media_ip_address_to_bind: Media already
>> bound, use existing source_media_ip_addr
>> *Oct 27 18:02:12.551: //1345/0008DE602400/SIP/Media/sipSPISetMediaSrcAddr:
>> Media src addr for stream 1 = 173.14.220.57
>> *Oct 27 18:02:12.551:
>> //1345/0008DE602400/SIP/Info/sipSPIDoAudioNegotiation: Codec (g711ulaw)
>> Negotiation Successful on Static Payload for m-line 1
>> *Oct 27 18:02:12.551:
>> //1345/0008DE602400/SIP/Info/sipSPIDoPtimeNegotiation: No ptime present or
>> multiple ptime attributes that can't be handled
>> *Oct 27 18:02:12.551:
>> //1345/0008DE602400/SIP/Info/sipSPIDoDTMFRelayNegotiation: m-line index 1
>> *Oct 27 18:02:12.551:
>> //1345/0008DE602400/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(101)
>> could not be reserved.
>> *Oct 27 18:02:12.551:
>> //1345/0008DE602400/SIP/Info/sipSPIDoDTMFRelayNegotiation: RTP-NTE DTMF
>> relay option
>> *Oct 27 18:02:12.555:
>> //1345/0008DE602400/SIP/Info/sipSPIDoDTMFRelayNegotiation: Case of full
>> named event(NE) match in fmtp list of events.
>> *Oct 27 18:02:12.555:
>> //-1/xxxxxxxxxxxx/SIP/Info/sip_sdp_get_modem_relay_cap_params: NSE payload
>> from X-cap = 0
>> *Oct 27 18:02:12.555:
>> //1345/0008DE602400/SIP/Info/sip_select_modem_relay_params: X-tmr not
>> present in SDP. Disable modem relay
>> *Oct 27 18:02:12.555:
>> //1345/0008DE602400/SIP/Info/sipSPIGetSDPDirectionAttribute: No direction
>> attribute present or multiple direction attributes that can't be handled for
>> m-line:1 and num-a-lines:0
>> *Oct 27 18:02:12.555:
>> //1345/0008DE602400/SIP/Info/sipSPIDoAudioNegotiation: Codec negotiation
>> successful for media line 1
>> payload_type=0, codec_bytes=160, codec=g711ulaw,
>> dtmf_relay=rtp-nte
>> stream_type=voice+dtmf (1), dest_ip_address=64.34.181.47,
>> dest_port=11680
>> *Oct 27 18:02:12.555:
>> //1345/0008DE602400/SIP/State/sipSPIChangeStreamState: Stream (callid =
>> -1) State changed from (STREAM_DEAD) to (STREAM_ADDING)
>> *Oct 27 18:02:12.555:
>> //1345/0008DE602400/SIP/Media/sipSPIUpdCallWithSdpInfo:
>> Preferred Codec : g711ulaw, bytes :160
>> Preferred DTMF relay : rtp-nte
>> Preferred NTE payload : 101
>> Early Media : No
>> Delayed Media : No
>> Bridge Done : No
>> New Media : No
>> DSP DNLD Reqd : No
>>
>> On Tue, Oct 27, 2009 at 10:47 AM, Nick Matthews <matthnick at gmail.com>wrote:
>>
>>> The 200 OK that you've pasted is confirming the CANCEL that we sent.
>>> You can tell because in the 200 OK: CSeq: 102 CANCEL. You should see
>>> a 200 OK with the CSeq for 101 INVITE.
>>>
>>> I've seen this for certain IVRs/providers - sometimes they don't
>>> properly terminate a call with a 200 OK. If you were not sending an
>>> SDP in your original INVITE, then you would need the PRACK setting
>>> mentioned. You have two problems, either could fix the problem: They
>>> could advertise DTMF in their 183, or they could send you a 200 OK for
>>> the call. It is assumed you would get DTMF in the 200 OK. It's
>>> common for endpoints that support DTMF to not advertise it in the 183
>>> because you technically shouldn't need DTMF to hear ringback.
>>>
>>> -nick
>>>
>>> On Tue, Oct 27, 2009 at 9:30 AM, Ryan Ratliff <rratliff at cisco.com>
>>> wrote:
>>> > There is no SDP in that 200 OK so I would assume the media info is the
>>> same
>>> > as in the 183 Ringing message. You really need your ITSP to tell you
>>> what
>>> > dtmf method they want you to use on your outbound calls. As Nick said
>>> they
>>> > don't appear to be advertising any dtmf method at all.
>>> > -Ryan
>>> > On Oct 27, 2009, at 8:51 AM, Dane Newman wrote:
>>> > Is the below the ok I should be getting?
>>> >
>>> >
>>> > They did send this with the first debug
>>> >
>>> > Received:
>>> > SIP/2.0 200 OK
>>> > Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK51214CC
>>> > From: <sip:6782282221 at sip.talkinip.net<sip%3A6782282221 at sip.talkinip.net>
>>> >;tag=32DA608-109A
>>> > To: <sip:18774675464 at 64.154.41.200 <sip%3A18774675464 at 64.154.41.200>>
>>> > Call-ID: 9F060E11-C23511DE-8027C992-790F56B7 at 173.14.220.57
>>> > CSeq: 102 CANCEL
>>> > Content-Length: 0
>>> > *Oct 27 13:44:12.828: //922/009B1B501B00/SIP/Info/sipSPICheckResponse:
>>> > non-INVITE response with no RSEQ - do not disable IS_REL1XX
>>> > *Oct 27 13:44:12.828: //922/009B1B501B00/SIP/Info/sipSPIIcpifUpdate:
>>> > CallState: 3 Playout: 0 DiscTime:5333362 ConnTime 0
>>> > *Oct 27 13:44:12.836:
>>> //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads:
>>> > Msg enqueued for SPI with IP addr: [64.154.41.200]:5060
>>> > *Oct 27 13:44:12.840:
>>> > //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event:
>>> > ccsip_spi_get_msg_type returned: 2 for event 1
>>> > *Oct 27 13:44:12.840:
>>> > //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg:
>>> > context=0x00000000
>>> > *Oct 27 13:44:12.840:
>>> //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor:
>>> > Checking Invite Dialog
>>> > *Oct 27 13:44:12.840: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>> >
>>> > This with the 2nd debug
>>> >
>>> > Received:
>>> > SIP/2.0 200 OK
>>> > Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK4A18DE
>>> > From: <sip:6782282221 at sip.talkinip.net<sip%3A6782282221 at sip.talkinip.net>
>>> >;tag=2EDA9C8-25D6
>>> > To: <sip:18774675464 at 64.154.41.200 <sip%3A18774675464 at 64.154.41.200>>
>>> > Call-ID: DB9895B8-C22B11DE-801EC992-790F56B7 at 173.14.220.57
>>> > CSeq: 102 CANCEL
>>> > Content-Length: 0
>>> > *Oct 27 12:34:15.900: //846/8094E28C1800/SIP/Info/sipSPICheckResponse:
>>> > non-INVITE response with no RSEQ - do not disable IS_REL1XX
>>> > *Oct 27 12:34:15.900: //846/8094E28C1800/SIP/Info/sipSPIIcpifUpdate:
>>> > CallState: 3 Playout: 0 DiscTime:4913670 ConnTime 0
>>> > *Oct 27 12:34:15.912:
>>> //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads:
>>> > Msg enqueued for SPI with IP addr: [64.154.41.200]:5060
>>> > *Oct 27 12:34:15.912:
>>> > //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event:
>>> > ccsip_spi_get_msg_type returned: 2 for event 1
>>> > *Oct 27 12:34:15.912:
>>> > //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg:
>>> > context=0x00000000
>>> > *Oct 27 12:34:15.912:
>>> //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor:
>>> > Checking Invite Dialog
>>> > *Oct 27 12:34:15.912: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>> > Received:
>>> > SIP/2.0 487 Request Terminated
>>> > To: <sip:18774675464 at 64.154.41.200 <sip%3A18774675464 at 64.154.41.200>
>>> >;tag=3465630735-938664
>>> > From: <sip:6782282221 at sip.talkinip.net<sip%3A6782282221 at sip.talkinip.net>
>>> >;tag=2EDA9C8-25D6
>>> > Contact: <sip:18774675464 at 64.154.41.200:5060>
>>> > Call-ID: DB9895B8-C22B11DE-801EC992-790F56B7 at 173.14.220.57
>>> > CSeq: 102 INVITE
>>> > Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK4A18DE
>>> > Content-Length: 0
>>> >
>>> > On Tue, Oct 27, 2009 at 8:43 AM, Nick Matthews <matthnick at gmail.com>
>>> wrote:
>>> >>
>>> >> In the 183 Session Progress they're not advertising DTMF:
>>> >>
>>> >> m=audio 45846 RTP/AVP 0
>>> >>
>>> >> There should be a 100 or 101 there. Although, 183 is just ringback.
>>> >> You would want to pick up on the other side and they should send a 200
>>> >> OK with a new SDP. If the other side did pick up, you need to tell
>>> >> the provider that they need to send a 200 OK, because they're not.
>>> >>
>>> >>
>>> >> -nick
>>> >>
>>> >> On Tue, Oct 27, 2009 at 7:36 AM, Dane Newman <dane.newman at gmail.com>
>>> >> wrote:
>>> >> > Nick
>>> >> >
>>> >> > I removed voice-class sip asymmetric payload dtmf and added in the
>>> >> > other
>>> >> > line
>>> >> >
>>> >> > Just to state incoming dtmf works but not outbound the ITSP has told
>>> me
>>> >> > they
>>> >> > are using two different sip servers/vendors for processing inbound
>>> and
>>> >> > outbound
>>> >> > How does this translate into what I should sent the following too?
>>> >> >
>>> >> > rtp payload-type nse
>>> >> > rtp payload-type nte
>>> >> >
>>> >> > In the debug trhe following where set
>>> >> >
>>> >> > rtp payload-type nse 101
>>> >> > rtp payload-type nte 100
>>> >> >
>>> >> > In the debug of ccsip If I am looking at it correctly I see me
>>> sending
>>> >> > this
>>> >> >
>>> >> > *Oct 27 12:34:09.128:
>>> >> > //846/8094E28C1800/SIP/Media/sipSPIAddSDPMediaPayload:
>>> >> > Preferred method of dtmf relay is: 6, with payload: 100
>>> >> > *Oct 27 12:34:09.128:
>>> >> > //846/8094E28C1800/SIP/Info/sipSPIAddSDPPayloadAttributes:
>>> >> > max_event 15
>>> >> >
>>> >> > and
>>> >> >
>>> >> >
>>> >> > *Oct 27 12:34:10.836:
>>> >> > //-1/xxxxxxxxxxxx/SIP/Info/sip_sdp_get_modem_relay_cap_params: NSE
>>> >> > payload
>>> >> > from X-cap = 0
>>> >> > *Oct 27 12:34:10.836:
>>> >> > //846/8094E28C1800/SIP/Info/sip_select_modem_relay_params: X-tmr not
>>> >> > present
>>> >> > in SDP. Disable modem relay
>>> >> >
>>> >> >
>>> >> > Sent:
>>> >> > INVITE sip:18774675464 at 64.154.41.200:5060 SIP/2.0
>>> >> > Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK4A01ECD
>>> >> > Remote-Party-ID:
>>> >> > <sip:6782282221 at 173.14.220.57 <sip%3A6782282221 at 173.14.220.57>
>>> >;party=calling;screen=yes;privacy=off
>>> >> > From: <sip:6782282221 at sip.talkinip.net<sip%3A6782282221 at sip.talkinip.net>
>>> >;tag=2EDA9C8-25D6
>>> >> > To: <sip:18774675464 at 64.154.41.200<sip%3A18774675464 at 64.154.41.200>
>>> >
>>> >> > Date: Tue, 27 Oct 2009 12:34:09 GMT
>>> >> > Call-ID: DB9895B8-C22B11DE-801EC992-790F56B7 at 173.14.220.57
>>> >> > Supported: 100rel,timer,resource-priority,replaces,sdp-anat
>>> >> > Min-SE: 1800
>>> >> > Cisco-Guid: 2157240972-3604177326-402682881-167847941
>>> >> > User-Agent: Cisco-SIPGateway/IOS-12.x
>>> >> > Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>>> >> > SUBSCRIBE,
>>> >> > NOTIFY, INFO, REGISTER
>>> >> > CSeq: 101 INVITE
>>> >> > Max-Forwards: 70
>>> >> > Timestamp: 1256646849
>>> >> > Contact: <sip:6782282221 at 173.14.220.57:5060>
>>> >> > Expires: 180
>>> >> > Allow-Events: telephone-event
>>> >> > Content-Type: application/sdp
>>> >> > Content-Disposition: session;handling=required
>>> >> > Content-Length: 250
>>> >> > v=0
>>> >> > o=CiscoSystemsSIP-GW-UserAgent 7043 4703 IN IP4 173.14.220.57
>>> >> > s=SIP Call
>>> >> > c=IN IP4 173.14.220.57
>>> >> > t=0 0
>>> >> > m=audio 16462 RTP/AVP 0 100
>>> >> > c=IN IP4 173.14.220.57
>>> >> > a=rtpmap:0 PCMU/8000
>>> >> > a=rtpmap:100 telephone-event/8000
>>> >> > a=fmtp:100 0-15
>>> >> > a=ptime:20
>>> >> >
>>> >> >
>>> >> > Then when I do a search for fmtp again further down I see
>>> >> >
>>> >> > Sent:
>>> >> > INVITE sip:18774675464 at 64.154.41.200:5060 SIP/2.0
>>> >> > Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK4A18DE
>>> >> > Remote-Party-ID:
>>> >> > <sip:6782282221 at 173.14.220.57 <sip%3A6782282221 at 173.14.220.57>
>>> >;party=calling;screen=yes;privacy=off
>>> >> > From: <sip:6782282221 at sip.talkinip.net<sip%3A6782282221 at sip.talkinip.net>
>>> >;tag=2EDA9C8-25D6
>>> >> > To: <sip:18774675464 at 64.154.41.200<sip%3A18774675464 at 64.154.41.200>
>>> >
>>> >> > Date: Tue, 27 Oct 2009 12:34:09 GMT
>>> >> > Call-ID: DB9895B8-C22B11DE-801EC992-790F56B7 at 173.14.220.57
>>> >> > Supported: 100rel,timer,resource-priority,replaces,sdp-anat
>>> >> > Min-SE: 1800
>>> >> > Cisco-Guid: 2157240972-3604177326-402682881-167847941
>>> >> > User-Agent: Cisco-SIPGateway/IOS-12.x
>>> >> > Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>>> >> > SUBSCRIBE,
>>> >> > NOTIFY, INFO, REGISTER
>>> >> > CSeq: 102 INVITE
>>> >> > Max-Forwards: 70
>>> >> > Timestamp: 1256646849
>>> >> > Contact: <sip:6782282221 at 173.14.220.57:5060>
>>> >> > Expires: 180
>>> >> > Allow-Events: telephone-event
>>> >> > Proxy-Authorization: Digest
>>> >> >
>>> >> > username="1648245954",realm="64.154.41.110",uri="
>>> sip:18774675464 at 64.154.41.200:5060
>>> ",response="ab63d4755ff4182631ad2db0f9ed0e44",nonce="12901115532:303fa5d884d6d0a5a0328a838545395b",algorithm=MD5
>>> >> > Content-Type: application/sdp
>>> >> > Content-Disposition: session;handling=required
>>> >> > Content-Length: 250
>>> >> > v=0
>>> >> > o=CiscoSystemsSIP-GW-UserAgent 7043 4703 IN IP4 173.14.220.57
>>> >> > s=SIP Call
>>> >> > c=IN IP4 173.14.220.57
>>> >> > t=0 0
>>> >> > m=audio 16462 RTP/AVP 0 100
>>> >> > c=IN IP4 173.14.220.57
>>> >> > a=rtpmap:0 PCMU/8000
>>> >> > a=rtpmap:100 telephone-event/8000
>>> >> > a=fmtp:100 0-15
>>> >> > a=ptime:20
>>> >> > *Oct 27 12:34:09.332:
>>> >> > //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads:
>>> >> > Msg enqueued for SPI with IP addr: [64.154.41.200]:5060
>>> >> > *Oct 27 12:34:09.332:
>>> >> > //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event:
>>> >> > ccsip_spi_get_msg_type returned: 2 for event 1
>>> >> > *Oct 27 12:34:09.332:
>>> >> > //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg:
>>> >> > context=0x00000000
>>> >> > *Oct 27 12:34:09.332:
>>> >> > //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor:
>>> >> > Checking Invite Dialog
>>> >> > *Oct 27 12:34:09.332: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>> >> > Received:
>>> >> > SIP/2.0 100 Trying
>>> >> > Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK4A18DE
>>> >> > From: <sip:6782282221 at sip.talkinip.net<sip%3A6782282221 at sip.talkinip.net>
>>> >;tag=2EDA9C8-25D6
>>> >> > To: <sip:18774675464 at 64.154.41.200<sip%3A18774675464 at 64.154.41.200>
>>> >
>>> >> > Call-ID: DB9895B8-C22B11DE-801EC992-790F56B7 at 173.14.220.57
>>> >> > CSeq: 102 INVITE
>>> >> > Content-Length: 0
>>> >> > *Oct 27 12:34:09.332:
>>> //846/8094E28C1800/SIP/Info/sipSPICheckResponse:
>>> >> > INVITE response with no RSEQ - disable IS_REL1XX
>>> >> > *Oct 27 12:34:09.332:
>>> //846/8094E28C1800/SIP/State/sipSPIChangeState:
>>> >> > 0x4A357FCC : State change from (STATE_SENT_INVITE, SUBSTATE_NONE)
>>> to
>>> >> > (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)
>>> >> > *Oct 27 12:34:10.832:
>>> >> > //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads:
>>> >> > Msg enqueued for SPI with IP addr: [64.154.41.200]:5060
>>> >> > *Oct 27 12:34:10.832:
>>> >> > //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event:
>>> >> > ccsip_spi_get_msg_type returned: 2 for event 1
>>> >> > *Oct 27 12:34:10.832:
>>> >> > //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg:
>>> >> > context=0x00000000
>>> >> > *Oct 27 12:34:10.836:
>>> >> > //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor:
>>> >> > Checking Invite Dialog
>>> >> > *Oct 27 12:34:10.836: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>> >> > Received:
>>> >> > SIP/2.0 183 Session Progress
>>> >> > To: <sip:18774675464 at 64.154.41.200<sip%3A18774675464 at 64.154.41.200>
>>> >;tag=3465630735-938664
>>> >> > From: <sip:6782282221 at sip.talkinip.net<sip%3A6782282221 at sip.talkinip.net>
>>> >;tag=2EDA9C8-25D6
>>> >> > Contact: <sip:18774675464 at 64.154.41.200:5060>
>>> >> > Call-ID: DB9895B8-C22B11DE-801EC992-790F56B7 at 173.14.220.57
>>> >> > CSeq: 102 INVITE
>>> >> > Content-Type: application/sdp
>>> >> > Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK4A18DE
>>> >> > Content-Length: 146
>>> >> > v=0
>>> >> > o=msx71 490 6110 IN IP4 64.154.41.200
>>> >> > s=sip call
>>> >> > c=IN IP4 64.154.41.101
>>> >> > t=0 0
>>> >> > m=audio 45846 RTP/AVP 0
>>> >> > a=ptime:20
>>> >> > a=rtpmap:0 PCMU/8000
>>> >> > *Oct 27 12:34:10.836:
>>> //846/8094E28C1800/SIP/Info/sipSPICheckResponse:
>>> >> > INVITE response with no RSEQ - disable IS_REL1XX
>>> >> > *Oct 27 12:34:10.836:
>>> //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentGTD: No
>>> >> > GTD
>>> >> > found in inbound container
>>> >> > *Oct 27 12:34:10.836:
>>> >> > //846/8094E28C1800/SIP/Info/sipSPIDoMediaNegotiation:
>>> >> > Number of m-lines = 1
>>> >> > SIP: Attribute mid, level 1 instance 1 not found.
>>> >> > *Oct 27 12:34:10.836:
>>> >> > //846/8094E28C1800/SIP/Info/resolve_media_ip_address_to_bind: Media
>>> >> > already
>>> >> > bound, use existing source_media_ip_addr
>>> >> > *Oct 27 12:34:10.836:
>>> >> > //846/8094E28C1800/SIP/Media/sipSPISetMediaSrcAddr:
>>> >> > Media src addr for stream 1 = 173.14.220.57
>>> >> > *Oct 27 12:34:10.836:
>>> >> > //846/8094E28C1800/SIP/Info/sipSPIDoAudioNegotiation:
>>> >> > Codec (g711ulaw) Negotiation Successful on Static Payload for m-line
>>> 1
>>> >> > *Oct 27 12:34:10.836:
>>> >> > //846/8094E28C1800/SIP/Info/sipSPIDoPtimeNegotiation:
>>> >> > One ptime attribute found - value:20
>>> >> > *Oct 27 12:34:10.836:
>>> >> > //-1/xxxxxxxxxxxx/SIP/Info/convert_ptime_to_codec_bytes: Values
>>> :Codec:
>>> >> > g711ulaw ptime :20, codecbytes: 160
>>> >> > *Oct 27 12:34:10.836:
>>> >> > //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values
>>> :Codec:
>>> >> > g711ulaw codecbytes :160, ptime: 20
>>> >> > *Oct 27 12:34:10.836:
>>> >> > //846/8094E28C1800/SIP/Media/sipSPIDoPtimeNegotiation:
>>> >> > Offered ptime:20, Negotiated ptime:20 Negotiated codec bytes: 160
>>> for
>>> >> > codec
>>> >> > g711ulaw
>>> >> > *Oct 27 12:34:10.836:
>>> >> > //846/8094E28C1800/SIP/Info/sipSPIDoDTMFRelayNegotiation: m-line
>>> index 1
>>> >> > *Oct 27 12:34:10.836:
>>> >> > //846/8094E28C1800/SIP/Info/sipSPICheckDynPayloadUse:
>>> >> > Dynamic payload(100) could not be reserved.
>>> >> > *Oct 27 12:34:10.836:
>>> >> > //846/8094E28C1800/SIP/Info/sipSPIDoDTMFRelayNegotiation: Case of
>>> full
>>> >> > named
>>> >> > event(NE) match in fmtp list of events.
>>> >> > *Oct 27 12:34:10.836:
>>> >> > //-1/xxxxxxxxxxxx/SIP/Info/sip_sdp_get_modem_relay_cap_params: NSE
>>> >> > payload
>>> >> > from X-cap = 0
>>> >> > *Oct 27 12:34:10.836:
>>> >> > //846/8094E28C1800/SIP/Info/sip_select_modem_relay_params: X-tmr not
>>> >> > present
>>> >> > in SDP. Disable modem relay
>>> >> > *Oct 27 12:34:10.836:
>>> >> > //846/8094E28C1800/SIP/Info/sipSPIGetSDPDirectionAttribute: No
>>> direction
>>> >> > attribute present or multiple direction attributes that can't be
>>> handled
>>> >> > for
>>> >> > m-line:1 and num-a-lines:0
>>> >> > *Oct 27 12:34:10.836:
>>> >> > //846/8094E28C1800/SIP/Info/sipSPIDoAudioNegotiation:
>>> >> > Codec negotiation successful for media line 1
>>> >> > payload_type=0, codec_bytes=160, codec=g711ulaw,
>>> >> > dtmf_relay=rtp-nte
>>> >> > stream_type=voice+dtmf (1), dest_ip_address=64.154.41.101,
>>> >> > dest_port=45846
>>> >> > *Oct 27 12:34:10.836:
>>> >> > //846/8094E28C1800/SIP/State/sipSPIChangeStreamState:
>>> >> > Stream (callid = -1) State changed from (STREAM_DEAD) to
>>> >> > (STREAM_ADDING)
>>> >> > *Oct 27 12:34:10.836:
>>> >> > //846/8094E28C1800/SIP/Media/sipSPIUpdCallWithSdpInfo:
>>> >> > Preferred Codec : g711ulaw, bytes :160
>>> >> > Preferred DTMF relay : rtp-nte
>>> >> > Preferred NTE payload : 100
>>> >> > Early Media : No
>>> >> > Delayed Media : No
>>> >> > Bridge Done : No
>>> >> > New Media : No
>>> >> > DSP DNLD Reqd : No
>>> >> > *Oct 27 12:34:10.840:
>>> >> > //846/8094E28C1800/SIP/Info/resolve_media_ip_address_to_bind: Media
>>> >> > already
>>> >> > bound, use existing source_media_ip_addr
>>> >> > *Oct 27 12:34:10.840:
>>> >> > //846/8094E28C1800/SIP/Media/sipSPISetMediaSrcAddr:
>>> >> > Media src addr for stream 1 = 173.14.220.57
>>> >> > *Oct 27 12:34:10.840:
>>> >> > //846/8094E28C1800/SIP/Info/sipSPI_ipip_report_media_to_peer:
>>> >> > callId 846 peer 845 flags 0x200005 state STATE_RECD_PROCEEDING
>>> >> > *Oct 27 12:34:10.840:
>>> >> > //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
>>> >> > CallID 846, sdp 0x497E29C0 channels 0x4A35926C
>>> >> > *Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/copy_channels:
>>> >> > callId 846 size 240 ptr 0x4A170B28)
>>> >> > *Oct 27 12:34:10.840:
>>> >> > //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
>>> >> > Hndl ptype 0 mline 1
>>> >> > *Oct 27 12:34:10.840:
>>> >> > //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
>>> >> > Selecting
>>> >> > codec g711ulaw
>>> >> > *Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/codec_found:
>>> >> > Codec to be matched: 5
>>> >> > *Oct 27 12:34:10.840:
>>> >> > //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: ADD
>>> >> > AUDIO
>>> >> > CODEC 5
>>> >> > *Oct 27 12:34:10.840:
>>> >> > //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values
>>> :Codec:
>>> >> > g711ulaw codecbytes :160, ptime: 20
>>> >> > *Oct 27 12:34:10.840:
>>> >> > //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
>>> Media
>>> >> > negotiation done:
>>> >> > stream->negotiated_ptime=20,stream->negotiated_codec_bytes=160,
>>> coverted
>>> >> > ptime=20 stream->mline_index=1, media_ndx=1
>>> >> > *Oct 27 12:34:10.840:
>>> >> > //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
>>> >> > Adding codec 5 ptype 0 time 20, bytes 160 as channel 0 mline 1 ss 1
>>> >> > 64.154.41.101:45846
>>> >> > *Oct 27 12:34:10.840:
>>> >> > //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
>>> Copy
>>> >> > sdp to
>>> >> > channel- AFTER CODEC FILTERING:
>>> >> > ccb->pld.ipip_caps.codecInfo[channel_ndx].codec = 5
>>> >> > *Oct 27 12:34:10.840:
>>> >> > //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
>>> Copy
>>> >> > sdp to
>>> >> > channel- AFTER CODEC FILTERING:
>>> >> > ccb->pld.ipip_caps.codecInfo[channel_ndx].codec = -1
>>> >> > *Oct 27 12:34:10.840:
>>> >> > //846/8094E28C1800/SIP/Info/sipSPI_ipip_report_media_to_peer:
>>> >> > callId 846 flags 0x100 state STATE_RECD_PROCEEDING
>>> >> > *Oct 27 12:34:10.840:
>>> >> > //846/8094E28C1800/SIP/Info/sipSPI_ipip_report_media_to_peer:
>>> >> > Report initial call media
>>> >> > *Oct 27 12:34:10.840:
>>> >> > //846/8094E28C1800/SIP/Info/sipSPI_ipip_report_media_to_peer:
>>> ccb->flags
>>> >> > 0x400018, ccb->pld.flags_ipip 0x200005
>>> >> > *Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/copy_channels:
>>> >> > callId 846 size 240 ptr 0x4DEC000C)
>>> >> > *Oct 27 12:34:10.840:
>>> >> > //846/8094E28C1800/SIP/Info/ccsip_update_srtp_caps:
>>> >> > 5030: Posting Remote SRTP caps to other callleg.
>>> >> > *Oct 27 12:34:10.840:
>>> >> > //846/8094E28C1800/SIP/Info/sipSPI_ipip_report_media_to_peer: do
>>> >> > cc_api_caps_ind()
>>> >> > *Oct 27 12:34:10.840:
>>> >> > //846/8094E28C1800/SIP/Media/sipSPIUpdCallWithSdpInfo:
>>> >> > Stream type : voice+dtmf
>>> >> > Media line : 1
>>> >> > State : STREAM_ADDING (2)
>>> >> > Stream address type : 1
>>> >> > Callid : 846
>>> >> > Negotiated Codec : g711ulaw, bytes :160
>>> >> > Nego. Codec payload : 0 (tx), 0 (rx)
>>> >> > Negotiated DTMF relay : rtp-nte
>>> >> > Negotiated NTE payload : 100 (tx), 100 (rx)
>>> >> > Negotiated CN payload : 0
>>> >> > Media Srce Addr/Port : [173.14.220.57]:16462
>>> >> > Media Dest Addr/Port : [64.154.41.101]:45846
>>> >> > *Oct 27 12:34:10.840:
>>> >> > //846/8094E28C1800/SIP/Info/sipSPIProcessHistoryInfoHeader: No HI
>>> >> > headers
>>> >> > recvd from app container
>>> >> > *Oct 27 12:34:10.840:
>>> //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentQSIG:
>>> >> > No
>>> >> > QSIG Body found in inbound container
>>> >> > *Oct 27 12:34:10.840:
>>> //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentQ931:
>>> >> > No
>>> >> > RawMsg Body found in inbound container
>>> >> > *Oct 27 12:34:10.840:
>>> //-1/xxxxxxxxxxxx/SIP/Info/sipSPICreateNewRawMsg:
>>> >> > No
>>> >> > Data to form The Raw Message
>>> >> > *Oct 27 12:34:10.840:
>>> >> > //846/8094E28C1800/SIP/Info/HandleSIP1xxSessionProgress:
>>> >> > ccsip_api_call_cut_progress returned: SIP_SUCCESS
>>> >> > *Oct 27 12:34:10.840:
>>> //846/8094E28C1800/SIP/State/sipSPIChangeState:
>>> >> > 0x4A357FCC : State change from (STATE_RECD_PROCEEDING,
>>> >> > SUBSTATE_PROCEEDING_PROCEEDING) to (STATE_RECD_PROCEEDING,
>>> >> > SUBSTATE_NONE)
>>> >> > *Oct 27 12:34:10.844:
>>> >> > //846/8094E28C1800/SIP/Info/HandleSIP1xxSessionProgress: Transaction
>>> >> > Complete. Lock on Facilities released.
>>> >> > *Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Info/ccsip_bridge:
>>> confID =
>>> >> > 6,
>>> >> > srcCallID = 846, dstCallID = 845
>>> >> > *Oct 27 12:34:10.844:
>>> >> > //846/8094E28C1800/SIP/Info/sipSPIUupdateCcCallIds:
>>> >> > Old src/dest ccCallids: -1/-1, new src/dest ccCallids: 846/845
>>> >> > *Oct 27 12:34:10.844:
>>> >> > //846/8094E28C1800/SIP/Info/sipSPIUupdateCcCallIds:
>>> >> > Old streamcallid=846, new streamcallid=846
>>> >> > *Oct 27 12:34:10.844:
>>> >> > //846/8094E28C1800/SIP/Info/ccsip_gw_set_sipspi_mode:
>>> >> > Setting SPI mode to SIP-H323
>>> >> > *Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Info/ccsip_bridge:
>>> >> > xcoder_attached = 0, xmitFunc = 1131891908, ccb xmitFunc =
>>> 1131891908
>>> >> > *Oct 27 12:34:10.844:
>>> >> > //846/8094E28C1800/SIP/Media/sipSPIProcessRtpSessions:
>>> >> > sipSPIProcessRtpSessions
>>> >> > *Oct 27 12:34:10.844: //846/8094E28C1800/SIP/Media/sipSPIAddStream:
>>> >> > Adding
>>> >> > stream 1 of type voice+dtmf (callid 846) to the VOIP RTP library
>>> >> > *Oct 27 12:34:10.844:
>>> >> > //846/8094E28C1800/SIP/Info/resolve_media_ip_address_to_bind: Media
>>> >> > already
>>> >> > bound, use existing source_media_ip_addr
>>> >> > *Oct 27 12:34:10.844:
>>> >> > //846/8094E28C1800/SIP/Media/sipSPISetMediaSrcAddr:
>>> >> > Media src addr for stream 1 = 173.14.220.57
>>> >> > *Oct 27 12:34:10.844:
>>> >> > //846/8094E28C1800/SIP/Media/sipSPIUpdateRtcpSession:
>>> >> > sipSPIUpdateRtcpSession for m-line 1
>>> >> > *Oct 27 12:34:10.848:
>>> >> > //846/8094E28C1800/SIP/Media/sipSPIUpdateRtcpSession:
>>> >> > rtcp_session info
>>> >> > laddr = 173.14.220.57, lport = 16462, raddr = 64.154.41.101,
>>> >> > rport=45846, do_rtcp=TRUE
>>> >> > src_callid = 846, dest_callid = 845, stream type =
>>> voice+dtmf,
>>> >> > stream direction = SENDRECV
>>> >> > media_ip_addr = 64.154.41.101, vrf tableid = 0
>>> media_addr_type =
>>> >> > 1
>>> >> > *Oct 27 12:34:10.848:
>>> >> > //846/8094E28C1800/SIP/Media/sipSPIUpdateRtcpSession:
>>> >> > RTP session already created - update
>>> >> > *Oct 27 12:34:10.848:
>>> >> > //846/8094E28C1800/SIP/Media/sipSPIUpdateRtpSession:
>>> >> > stun is disabled for stream:4A1709F8
>>> >> > *Oct 27 12:34:10.848:
>>> >> > //846/8094E28C1800/SIP/Media/sipSPIGetNewLocalMediaDirection:
>>> >> > New Remote Media Direction = SENDRECV
>>> >> > Present Local Media Direction = SENDRECV
>>> >> > New Local Media Direction = SENDRECV
>>> >> > retVal = 0
>>> >> > *Oct 27 12:34:10.848:
>>> >> > //846/8094E28C1800/SIP/State/sipSPIChangeStreamState:
>>> >> > Stream (callid = 846) State changed from (STREAM_ADDING) to
>>> >> > (STREAM_ACTIVE)
>>> >> > *Oct 27 12:34:10.848: //846/8094E28C1800/SIP/Info/ccsip_bridge:
>>> really
>>> >> > can't
>>> >> > find peer_stream for
>>> >> > dtmf-relay
>>> interworking
>>> >> > *Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/ccsip_caps_ind:
>>> Entry
>>> >> > *Oct 27 12:34:11.140:
>>> >> > //846/8094E28C1800/SIP/Info/ccsip_get_rtcp_session_parameters:
>>> CURRENT
>>> >> > VALUES: stream_callid=846, current_seq_num=0x23ED
>>> >> > *Oct 27 12:34:11.140:
>>> >> > //846/8094E28C1800/SIP/Info/ccsip_get_rtcp_session_parameters: NEW
>>> >> > VALUES:
>>> >> > stream_callid=846, current_seq_num=0x11D9
>>> >> > *Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/ccsip_caps_ind:
>>> Load
>>> >> > DSP
>>> >> > with negotiated codec: g711ulaw, Bytes=160
>>> >> > *Oct 27 12:34:11.140: //846/8094E28C1800/SIP/Info/ccsip_caps_ind:
>>> Set
>>> >> > forking flag to 0x0
>>> >> > *Oct 27 12:34:11.140:
>>> >> > //846/8094E28C1800/SIP/Info/sipSPISetDTMFRelayMode:
>>> >> > Set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_NTE_AND_OOB with rx
>>> payload =
>>> >> > 100, tx payload = 100
>>> >> > *Oct 27 12:34:11.140:
>>> //846/8094E28C1800/SIP/Info/sip_set_modem_caps:
>>> >> > Preferred (or the one that came from DSM) modem relay=0, from CLI
>>> >> > config=0
>>> >> > *Oct 27 12:34:11.140:
>>> //846/8094E28C1800/SIP/Info/sip_set_modem_caps:
>>> >> > Disabling Modem Relay...
>>> >> > *Oct 27 12:34:11.140:
>>> //846/8094E28C1800/SIP/Info/sip_set_modem_caps:
>>> >> > Negotiation already Done. Set negotiated Modem caps and generate SDP
>>> >> > Xcap
>>> >> > list
>>> >> > *Oct 27 12:34:11.140:
>>> //846/8094E28C1800/SIP/Info/sip_set_modem_caps:
>>> >> > Modem
>>> >> > Relay & Passthru both disabled
>>> >> > *Oct 27 12:34:11.144:
>>> //846/8094E28C1800/SIP/Info/sip_set_modem_caps:
>>> >> > nse
>>> >> > payload = 0, ptru mode = 0, ptru-codec=0, redundancy=0, xid=0,
>>> relay=0,
>>> >> > sprt-retry=12, latecncy=200, compres-dir=3, dict=1024, strnlen=32
>>> >> > *Oct 27 12:34:11.144:
>>> //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:
>>> >> > 1
>>> >> > Active Streams
>>> >> > *Oct 27 12:34:11.144:
>>> //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:
>>> >> > Adding stream type (voice+dtmf) from media
>>> >> > line 1 codec g711ulaw
>>> >> > *Oct 27 12:34:11.144:
>>> //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:
>>> >> > caps.stream_count=1,caps.stream[0].stream_type=0x3,
>>> >> > caps.stream_list.xmitFunc=
>>> >> > *Oct 27 12:34:11.144:
>>> //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:
>>> >> > voip_rtp_xmit, caps.stream_list.context=
>>> >> > *Oct 27 12:34:11.144:
>>> //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:
>>> >> > 0x497E0B60 (gccb)
>>> >> > *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/ccsip_caps_ind:
>>> Load
>>> >> > DSP
>>> >> > with codec : g711ulaw, Bytes=160, payload = 0
>>> >> > *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/ccsip_caps_ind:
>>> >> > ccsip_caps_ind: ccb->pld.flags_ipip = 0x200405
>>> >> > *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: No
>>> >> > video
>>> >> > caps detected in the caps posted by peer leg
>>> >> > *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/ccsip_caps_ind:
>>> >> > Setting
>>> >> > CAPS_RECEIVED flag
>>> >> > *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/ccsip_caps_ind:
>>> >> > Calling
>>> >> > cc_api_caps_ack()
>>> >> > *Oct 27 12:34:11.144: //846/8094E28C1800/SIP/Info/ccsip_caps_ack:
>>> Set
>>> >> > forking flag to 0x0
>>> >> > *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/ccsip_caps_ind:
>>> Entry
>>> >> > *Oct 27 12:34:11.168:
>>> >> > //846/8094E28C1800/SIP/Info/ccsip_get_rtcp_session_parameters:
>>> CURRENT
>>> >> > VALUES: stream_callid=846, current_seq_num=0x11D9
>>> >> > *Oct 27 12:34:11.168:
>>> >> > //846/8094E28C1800/SIP/Info/ccsip_get_rtcp_session_parameters: NEW
>>> >> > VALUES:
>>> >> > stream_callid=846, current_seq_num=0x11D9
>>> >> > *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/ccsip_caps_ind:
>>> Load
>>> >> > DSP
>>> >> > with negotiated codec: g711ulaw, Bytes=160
>>> >> > *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/ccsip_caps_ind:
>>> Set
>>> >> > forking flag to 0x0
>>> >> > *Oct 27 12:34:11.168:
>>> >> > //846/8094E28C1800/SIP/Info/sipSPISetDTMFRelayMode:
>>> >> > Set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_NTE_AND_OOB with rx
>>> payload =
>>> >> > 100, tx payload = 100
>>> >> > *Oct 27 12:34:11.168:
>>> //846/8094E28C1800/SIP/Info/sip_set_modem_caps:
>>> >> > Preferred (or the one that came from DSM) modem relay=0, from CLI
>>> >> > config=0
>>> >> > *Oct 27 12:34:11.168:
>>> //846/8094E28C1800/SIP/Info/sip_set_modem_caps:
>>> >> > Disabling Modem Relay...
>>> >> > *Oct 27 12:34:11.168:
>>> //846/8094E28C1800/SIP/Info/sip_set_modem_caps:
>>> >> > Negotiation already Done. Set negotiated Modem caps and generate SDP
>>> >> > Xcap
>>> >> > list
>>> >> > *Oct 27 12:34:11.168:
>>> //846/8094E28C1800/SIP/Info/sip_set_modem_caps:
>>> >> > Modem
>>> >> > Relay & Passthru both disabled
>>> >> > *Oct 27 12:34:11.168:
>>> //846/8094E28C1800/SIP/Info/sip_set_modem_caps:
>>> >> > nse
>>> >> > payload = 0, ptru mode = 0, ptru-codec=0, redundancy=0, xid=0,
>>> relay=0,
>>> >> > sprt-retry=12, latecncy=200, compres-dir=3, dict=1024, strnlen=32
>>> >> > *Oct 27 12:34:11.168:
>>> //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:
>>> >> > 1
>>> >> > Active Streams
>>> >> > *Oct 27 12:34:11.168:
>>> //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:
>>> >> > Adding stream type (voice+dtmf) from media
>>> >> > line 1 codec g711ulaw
>>> >> > *Oct 27 12:34:11.168:
>>> //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:
>>> >> > caps.stream_count=1,caps.stream[0].stream_type=0x3,
>>> >> > caps.stream_list.xmitFunc=
>>> >> > *Oct 27 12:34:11.168:
>>> //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:
>>> >> > voip_rtp_xmit, caps.stream_list.context=
>>> >> > *Oct 27 12:34:11.168:
>>> //846/8094E28C1800/SIP/Media/sipSPISetStreamInfo:
>>> >> > 0x497E0B60 (gccb)
>>> >> > *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/ccsip_caps_ind:
>>> Load
>>> >> > DSP
>>> >> > with codec : g711ulaw, Bytes=160, payload = 0
>>> >> > *Oct 27 12:34:11.168: //846/8094E28C1800/SIP/Info/ccsip_caps_ind:
>>> >> > ccsip_caps_ind: ccb->pld.flags_ipip = 0x200425
>>> >> > *Oct 27 12:34:11.172: //846/8094E28C1800/SIP/Info/ccsip_caps_ind: No
>>> >> > video
>>> >> > caps detected in the caps posted by peer leg
>>> >> > *Oct 27 12:34:11.172: //846/8094E28C1800/SIP/Info/ccsip_caps_ind:
>>> Second
>>> >> > TCS
>>> >> > received for transfers across trunk - set CAPS2_RECEIVED
>>> >> > *Oct 27 12:34:15.876:
>>> >> > //846/8094E28C1800/SIP/Media/sipSPIUpdateRtpSession:
>>> >> > stun is disabled for stream:4A1709F8
>>> >> > *Oct 27 12:34:15.876:
>>> //846/8094E28C1800/SIP/Info/ccsip_call_statistics:
>>> >> > Stats are not supported for IPIP call.
>>> >> > *Oct 27 12:34:15.876: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo:
>>> >> > Queued
>>> >> > event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
>>> >> > *Oct 27 12:34:15.880:
>>> >> > //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event:
>>> >> > ccsip_spi_get_msg_type returned: 3 for event 7
>>> >> > *Oct 27 12:34:15.880: //846/8094E28C1800/SIP/Info/sipSPISendCancel:
>>> >> > Associated container=0x4E310C1C to Cancel
>>> >> > *Oct 27 12:34:15.880:
>>> //846/8094E28C1800/SIP/Transport/sipSPISendCancel:
>>> >> > Sending CANCEL to the transport layer
>>> >> > *Oct 27 12:34:15.880:
>>> >> > //846/8094E28C1800/SIP/Transport/sipSPITransportSendMessage:
>>> >> > msg=0x4DF0D994,
>>> >> > addr=64.154.41.200, port=5060, sentBy_port=0, is_req=1, transport=1,
>>> >> > switch=0, callBack=0x419703BC
>>> >> > *Oct 27 12:34:15.880:
>>> >> > //846/8094E28C1800/SIP/Transport/sipSPITransportSendMessage:
>>> Proceedable
>>> >> > for
>>> >> > sending msg immediately
>>> >> > *Oct 27 12:34:15.880:
>>> >> > //846/8094E28C1800/SIP/Transport/sipTransportLogicSendMsg: switch
>>> >> > transport
>>> >> > is 0
>>> >> > *Oct 27 12:34:15.880:
>>> >> > //846/8094E28C1800/SIP/Transport/sipTransportLogicSendMsg: Set to
>>> send
>>> >> > the
>>> >> > msg=0x4DF0D994
>>> >> > *Oct 27 12:34:15.880:
>>> >> > //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting
>>> >> > send
>>> >> > for msg=0x4DF0D994, addr=64.154.41.200, port=5060, connId=2 for UDP
>>> >> > *Oct 27 12:34:15.880:
>>> >> > //846/8094E28C1800/SIP/Info/sentCancelDisconnecting:
>>> >> > Sent Cancel Request, starting CancelWaitResponseTimer
>>> >> > *Oct 27 12:34:15.880:
>>> //846/8094E28C1800/SIP/State/sipSPIChangeState:
>>> >> > 0x4A357FCC : State change from (STATE_RECD_PROCEEDING,
>>> SUBSTATE_NONE)
>>> >> > to
>>> >> > (STATE_DISCONNECTING, SUBSTATE_NONE)
>>> >> > *Oct 27 12:34:15.888: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>> >> > Sent:
>>> >> > CANCEL sip:18774675464 at 64.154.41.200:5060 SIP/2.0
>>> >> > Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK4A18DE
>>> >> > From: <sip:6782282221 at sip.talkinip.net<sip%3A6782282221 at sip.talkinip.net>
>>> >;tag=2EDA9C8-25D6
>>> >> > To: <sip:18774675464 at 64.154.41.200<sip%3A18774675464 at 64.154.41.200>
>>> >
>>> >> > Date: Tue, 27 Oct 2009 12:34:09 GMT
>>> >> > Call-ID: DB9895B8-C22B11DE-801EC992-790F56B7 at 173.14.220.57
>>> >> > CSeq: 102 CANCEL
>>> >> > Max-Forwards: 70
>>> >> > Timestamp: 1256646855
>>> >> > Reason: Q.850;cause=16
>>> >> > Content-Length: 0
>>> >> > *Oct 27 12:34:15.900:
>>> >> > //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads:
>>> >> > Msg enqueued for SPI with IP addr: [64.154.41.200]:5060
>>> >> > *Oct 27 12:34:15.900:
>>> >> > //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event:
>>> >> > ccsip_spi_get_msg_type returned: 2 for event 1
>>> >> > *Oct 27 12:34:15.900:
>>> >> > //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg:
>>> >> > context=0x00000000
>>> >> > *Oct 27 12:34:15.900:
>>> >> > //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor:
>>> >> > Checking Invite Dialog
>>> >> > *Oct 27 12:34:15.900: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>> >> > Received:
>>> >> > SIP/2.0 200 OK
>>> >> > Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK4A18DE
>>> >> > From: <sip:6782282221 at sip.talkinip.net<sip%3A6782282221 at sip.talkinip.net>
>>> >;tag=2EDA9C8-25D6
>>> >> > To: <sip:18774675464 at 64.154.41.200<sip%3A18774675464 at 64.154.41.200>
>>> >
>>> >> > Call-ID: DB9895B8-C22B11DE-801EC992-790F56B7 at 173.14.220.57
>>> >> > CSeq: 102 CANCEL
>>> >> > Content-Length: 0
>>> >> > *Oct 27 12:34:15.900:
>>> //846/8094E28C1800/SIP/Info/sipSPICheckResponse:
>>> >> > non-INVITE response with no RSEQ - do not disable IS_REL1XX
>>> >> > *Oct 27 12:34:15.900: //846/8094E28C1800/SIP/Info/sipSPIIcpifUpdate:
>>> >> > CallState: 3 Playout: 0 DiscTime:4913670 ConnTime 0
>>> >> > *Oct 27 12:34:15.912:
>>> >> > //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads:
>>> >> > Msg enqueued for SPI with IP addr: [64.154.41.200]:5060
>>> >> > *Oct 27 12:34:15.912:
>>> >> > //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event:
>>> >> > ccsip_spi_get_msg_type returned: 2 for event 1
>>> >> > *Oct 27 12:34:15.912:
>>> >> > //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg:
>>> >> > context=0x00000000
>>> >> > *Oct 27 12:34:15.912:
>>> >> > //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor:
>>> >> > Checking Invite Dialog
>>> >> > *Oct 27 12:34:15.912: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>> >> >
>>> >> > On Mon, Oct 26, 2009 at 7:36 PM, Nick Matthews <matthnick at gmail.com
>>> >
>>> >> > wrote:
>>> >> >>
>>> >> >> You would want to check the SDP of 200 OK the provider sends for
>>> your
>>> >> >> outgoing call. It will list the payload type for the dtmf in the
>>> >> >> format a=fmtp 101 1-16, or something similar. You want to find out
>>> >> >> what payload type they are advertising (or if they are at all). It
>>> >> >> would be worth checking the incoming INVITE from them to see what
>>> >> >> they're using when they send the first SDP.
>>> >> >>
>>> >> >> On that note, I would also remove the asymmetric payload command -
>>> to
>>> >> >> my knowledge it doesn't do what you're expecting it to. You may
>>> want
>>> >> >> to try this command:
>>> >> >> voice-class sip dtmf-relay force rtp-nte
>>> >> >>
>>> >> >>
>>> >> >> -nick
>>> >> >>
>>> >> >> On Mon, Oct 26, 2009 at 5:16 PM, Dane Newman <
>>> dane.newman at gmail.com>
>>> >> >> wrote:
>>> >> >> > Hello,
>>> >> >> >
>>> >> >> > I am having an issue with dtmf working outbound. Inbound dtmf
>>> works
>>> >> >> > fine.
>>> >> >> > It took some playing around with it. At first it didnt work till
>>> the
>>> >> >> > payload was ajusted. I am now trying to get outbound dtmf
>>> working
>>> >> >> > properly.
>>> >> >> >
>>> >> >> > On my 2821 I debugged voip rtp session named-events and then made
>>> a
>>> >> >> > call
>>> >> >> > to
>>> >> >> > a 1800 number and hit digits. I didn't see any dtmf output on
>>> the
>>> >> >> > router
>>> >> >> > nothing showed up in the debug. Does this mean I can safely
>>> asume
>>> >> >> > that
>>> >> >> > the
>>> >> >> > problem for right now is not on the ITSP side but on my side
>>> since
>>> >> >> > dtmf
>>> >> >> > is
>>> >> >> > not being sent down the sip trunk?
>>> >> >> >
>>> >> >> > I have my cuc 7.x configured to talk to my 2821 via h323. The
>>> >> >> > configuration
>>> >> >> > of the cisco 2821 is shown below. Does anyone have any ideas
>>> what I
>>> >> >> > can
>>> >> >> > do
>>> >> >> > so dtmf digits process properly outbound?
>>> >> >> >
>>> >> >> > The settings in my cuc 7.x to add the gateway h323 are
>>> >> >> >
>>> >> >> > h323 cucm gateway configuratration
>>> >> >> > Signaling Port 1720
>>> >> >> > media termination point required yes
>>> >> >> > retry video call as auto yes
>>> >> >> > wait for far end h.245 terminal capability set yes
>>> >> >> > transmit utf-8 calling party name no
>>> >> >> > h.235 pass through allowed no
>>> >> >> > significant digits all
>>> >> >> > redirect number IT deliver - inbound no
>>> >> >> > enable inbound faststart yes
>>> >> >> > display IE deliver no
>>> >> >> > redirect nunmber IT deliver - outbound no
>>> >> >> > enable outbound faststart yes
>>> >> >> >
>>> >> >> >
>>> >> >> > voice service voip
>>> >> >> > allow-connections h323 to h323
>>> >> >> > allow-connections h323 to sip
>>> >> >> > allow-connections sip to h323
>>> >> >> > allow-connections sip to sip
>>> >> >> > fax protocol pass-through g711ulaw
>>> >> >> > h323
>>> >> >> > emptycapability
>>> >> >> > h225 id-passthru
>>> >> >> > h245 passthru tcsnonstd-passthru
>>> >> >> > sip
>>> >> >> >
>>> >> >> >
>>> >> >> > voice class h323 50
>>> >> >> > h225 timeout tcp establish 3
>>> >> >> > !
>>> >> >> > !
>>> >> >> > !
>>> >> >> > !
>>> >> >> > !
>>> >> >> > !
>>> >> >> > !
>>> >> >> > !
>>> >> >> > !
>>> >> >> > !
>>> >> >> > !
>>> >> >> > voice translation-rule 1
>>> >> >> > rule 1 /.*/ /190/
>>> >> >> > !
>>> >> >> > voice translation-rule 2
>>> >> >> > rule 1 /.*/ /1&/
>>> >> >> > !
>>> >> >> > !
>>> >> >> > voice translation-profile aa
>>> >> >> > translate called 1
>>> >> >> > !
>>> >> >> > voice translation-profile addone
>>> >> >> > translate called 2
>>> >> >> > !
>>> >> >> > !
>>> >> >> > voice-card 0
>>> >> >> > dspfarm
>>> >> >> > dsp services dspfarm
>>> >> >> > !
>>> >> >> > !
>>> >> >> > sccp local GigabitEthernet0/1
>>> >> >> > sccp ccm 10.1.80.11 identifier 2 version 7.0
>>> >> >> > sccp ccm 10.1.80.10 identifier 1 version 7.0
>>> >> >> > sccp
>>> >> >> > !
>>> >> >> > sccp ccm group 1
>>> >> >> > associate ccm 1 priority 1
>>> >> >> > associate ccm 2 priority 2
>>> >> >> > associate profile 1 register 2821transcode
>>> >> >> > !
>>> >> >> > dspfarm profile 1 transcode
>>> >> >> > codec g711ulaw
>>> >> >> > codec g711alaw
>>> >> >> > codec g729ar8
>>> >> >> > codec g729abr8
>>> >> >> > codec g729r8
>>> >> >> > maximum sessions 4
>>> >> >> > associate application SCCP
>>> >> >> > !
>>> >> >> > !
>>> >> >> > dial-peer voice 100 voip
>>> >> >> > description AA Publisher
>>> >> >> > preference 1
>>> >> >> > destination-pattern 1..
>>> >> >> > voice-class h323 50
>>> >> >> > session target ipv4:10.1.80.10
>>> >> >> > dtmf-relay h245-alphanumeric
>>> >> >> > codec g711ulaw
>>> >> >> > no vad
>>> >> >> > !
>>> >> >> > dial-peer voice 1000 voip
>>> >> >> > description incoming Call
>>> >> >> > translation-profile incoming aa
>>> >> >> > preference 1
>>> >> >> > rtp payload-type nse 101
>>> >> >> > rtp payload-type nte 100
>>> >> >> > incoming called-number 6782282221
>>> >> >> > dtmf-relay rtp-nte
>>> >> >> > codec g711ulaw
>>> >> >> > ip qos dscp cs5 media
>>> >> >> > ip qos dscp cs5 signaling
>>> >> >> > no vad
>>> >> >> > !
>>> >> >> > dial-peer voice 101 voip
>>> >> >> > description AA Subscriber
>>> >> >> > preference 2
>>> >> >> > destination-pattern 1..
>>> >> >> > voice-class h323 50
>>> >> >> > session target ipv4:10.1.80.11
>>> >> >> > dtmf-relay h245-alphanumeric
>>> >> >> > codec g711ulaw
>>> >> >> > no vad
>>> >> >> > !
>>> >> >> > dial-peer voice 2000 voip
>>> >> >> > description outbound
>>> >> >> > translation-profile outgoing addone
>>> >> >> > preference 1
>>> >> >> > destination-pattern .T
>>> >> >> > rtp payload-type nse 101
>>> >> >> > rtp payload-type nte 100
>>> >> >> > voice-class sip asymmetric payload dtmf
>>> >> >> > session protocol sipv2
>>> >> >> > session target ipv4:64.154.41.200
>>> >> >> > dtmf-relay rtp-nte
>>> >> >> > codec g711ulaw
>>> >> >> > no vad
>>> >> >> > !
>>> >> >> > !
>>> >> >> > sip-ua
>>> >> >> > credentials username ***** password 7 ***** realm
>>> sip.talkinip.net
>>> >> >> > authentication username ***** password 7 *****
>>> >> >> > authentication username ***** password 7 ***** realm
>>> >> >> > sip.talkinip.net
>>> >> >> > set pstn-cause 3 sip-status 486
>>> >> >> > set pstn-cause 34 sip-status 486
>>> >> >> > set pstn-cause 47 sip-status 486
>>> >> >> > registrar dns:sip.talkinip.net expires 60
>>> >> >> > sip-server dns:sip.talkinip.net:5060
>>> >> >> > _______________________________________________
>>> >> >> > cisco-voip mailing list
>>> >> >> > cisco-voip at puck.nether.net
>>> >> >> > https://puck.nether.net/mailman/listinfo/cisco-voip
>>> >> >> >
>>> >> >> >
>>> >> >
>>> >> >
>>> >
>>> > _______________________________________________
>>> > cisco-voip mailing list
>>> > cisco-voip at puck.nether.net
>>> > https://puck.nether.net/mailman/listinfo/cisco-voip
>>> >
>>> >
>>>
>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>
>
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