[cisco-voip] dtmf from cucm to 2821 cube to sip trunk

Dane Newman dane.newman at gmail.com
Tue Oct 27 09:14:43 EDT 2009


Thanks for this response.

I set it as below and restarted the ccm service on both pub and sub.  Still
didnt work.  I also tried to put

voice-class Sip rel1xx supported 100rel

under the dial peer and still didnt work but thanks for the response

Dane

On Tue, Oct 27, 2009 at 8:48 AM, Chris Osborne (US) <
chris.osborne at us.didata.com> wrote:

>  You could also try enabling this under Clusterwide Params (Device-SIP)  ,
> by default its turned off.
>
>
>
>
>
>
>
>
>
> I had a similar issue where when we called an at&t own 1800 number it would
> just ring forever because of how they handle calls from other carriers via
> sip. This allowed those calls to start working.
>
>
>
> Chris Osborne CCNP
>
> Sr Engineer  -  US Southeast
>
> Dimension Data
>
>
>
>
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Matthews
> Sent: Tuesday, October 27, 2009 8:44 AM
> To: Dane Newman
> Cc: cisco-voip
> Subject: Re: [cisco-voip] dtmf from cucm to 2821 cube to sip trunk
>
>
>
> 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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