[cisco-voip] dtmf from cucm to 2821 cube to sip trunk
Dane Newman
dane.newman at gmail.com
Tue Oct 27 08:51:38 EDT 2009
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
> >> >
> >> >
> >
> >
>
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