[cisco-voip] dtmf from cucm to 2821 cube to sip trunk
Dane Newman
dane.newman at gmail.com
Tue Oct 27 14:48:21 EDT 2009
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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