[cisco-voip] dtmf from cucm to 2821 cube to sip trunk
Dane Newman
dane.newman at gmail.com
Tue Oct 27 12:56:40 EDT 2009
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
> >
> >
>
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