Can you advise sip-providers offering t38?

Gafachi has T38 fax support. On Oct 7, 2009, at 4:26 PM, Vladimir Elizarov wrote:

Kristian Kielhofner пишет:

...

Since no one else has responded I'll chime in with some general advice.

It's troubling to see that your "provider" is using Asterisk to face you (the customer).

I've never had any luck getting T.38 to work (at all, in any mode) using Asterisk. I've heard of other people making it work but consider the chain here:

Their provider (they appear to be using DIDx, hmmm) -> Callwithus (using Asterisk) -> You (FreeSWITCH) -> Your device

I don't see anything specific in your case but there are many, many places for T.38 to go wrong in this scenario. I'd recommend attempting T.38 against a local gateway or known Tier 1 provider supporting T.38 with known equipment.

Can you advise sip-providers offering t38?

...

Otherwise, do you have any updates for us?

No. I can not add anything.

...

On Tue, Oct 6, 2009 at 3:06 PM, Vladimir Elizarov <xengelpublicx@gmail.com> wrote:

...

Hello.

I want to set up faxing via the gateway linksys spa-3102 (with support for t38) via SIP.

SIP-client -> linksys spa3102 -> fs -> provider

<gateway name="callwithus"> .... <param name="inbound-late-negotiation" value="true"/> <param name="stun-enabled" value="true"/> </gateway>

<extension name="Fax"> <condition field="destination_number" expression="^(30)(\d{11})$"> <action application="set" data="proxy_media=true"/> <action application="set" data="bypass_media=false"/> <action application="bridge" data="sofia/gateway/callwithus/*4087$2@sip.callwithus.com"/> </condition> </extension>

I make a call, press start on the fax. Fax not sent. Get this in the log (UA freeswitch not trunk-12805):

------------------------------------------------------------------------ INVITE sip:*408774957558438@sip.callwithus.com SIP/2.0 Via: SIP/2.0/UDP 77.239.230.202:5080;rport;branch=z9hG4bK8yjrHeUprX2Hr Max-Forwards: 99 From: "Mihailova Ludmila" <sip:544620048@sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN To: <sip:*408774957558438@sip.callwithus.com> Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f CSeq: 121302430 INVITE Contact: <sip:gw+callwithus@77.239.230.202:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 250 Remote-Party-ID: "Mihailova Ludmila" <sip:214@sip.callwithus.com>;party=calling;screen=yes;privacy=off

v=0 o=- 216896551 216896551 IN IP4 192.168.50.51 s=- c=IN IP4 77.239.230.202 t=0 0 m=audio 32404 RTP/AVP 0 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30

------------------------------------------------------------------------ send 1333 bytes to udp/[38.99.70.232]:5060 at 15:05:34.485226:

------------------------------------------------------------------------ INVITE sip:*408774957558438@sip.callwithus.com SIP/2.0 Via: SIP/2.0/UDP 77.239.230.202:5080;rport;branch=z9hG4bK97BHK9BtN6r4K Max-Forwards: 99 From: "Mihailova Ludmila" <sip:544620048@sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN To: <sip:*408774957558438@sip.callwithus.com> Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f CSeq: 121302431 INVITE Contact: <sip:gw+callwithus@77.239.230.202:5080;transport=udp> Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 250 Remote-Party-ID: "Mihailova Ludmila" <sip:214@sip.callwithus.com>;party=calling;screen=yes;privacy=off

v=0 o=- 216896551 216896551 IN IP4 192.168.50.51 s=- c=IN IP4 77.239.230.202 t=0 0 m=audio 32404 RTP/AVP 0 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30

------------------------------------------------------------------------ SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.50.10:5080;rport=5080;branch=z9hG4bK97BHK9BtN6r4K From: "Mihailova Ludmila" <sip:544620048@sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN To: <sip:*408774957558438@sip.callwithus.com> Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f CSeq: 121302431 INVITE Server: CWU SIP GW Content-Length: 0

------------------------------------------------------------------------ recv 836 bytes from udp/[38.99.70.232]:5060 at 15:05:49.641312:

------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.50.10:5080;rport=5080;branch=z9hG4bK97BHK9BtN6r4K Record-Route: <sip:38.99.70.232;lr=on;ftag=HXjrDKpHHX8ZN;did=e8e.089fc2b6> From: "Mihailova Ludmila" <sip:544620048@sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN To: <sip:*408774957558438@sip.callwithus.com>;tag=as22d3d3b0 Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f CSeq: 121302431 INVITE User-Agent: CWU SIP-GW Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:*408774957558438@204.74.213.1:5062> Content-Type: application/sdp Content-Length: 209

v=0 o=root 31797 31797 IN IP4 4.55.17.66 s=session c=IN IP4 4.55.17.66 t=0 0 m=audio 14104 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv

------------------------------------------------------------------------ v=0 o=root 31797 31797 IN IP4 4.55.17.66 s=session c=IN IP4 4.55.17.66 t=0 0 m=audio 14104 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20

2009-10-06 19:03:31.112988 [NOTICE] sofia.c:3449 Pre-Answer sofia/external/*408774957558438@sip.callwithus.com! 2009-10-06 19:03:31.112988 [DEBUG] switch_channel.c:1822 Send signal sofia/internal/214@pbx0.tssec.lan [BREAK] 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2095 Set Codec sofia/external/*408774957558438@sip.callwithus.com PROXY/8000 20 ms 160 samples 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2372 PROXY AUDIO RTP [sofia/external/*408774957558438@sip.callwithus.com] 192.168.50.10:32404->4.55.17.66:14104 codec: 0 ms: 20 2009-10-06 19:03:31.112988 [DEBUG] switch_rtp.c:1163 Not using a timer 2009-10-06 19:03:31.112988 [DEBUG] switch_ivr_originate.c:2154 sofia/internal/214@pbx0.tssec.lan receive message [PROGRESS] 2009-10-06 19:03:31.112988 [INFO] switch_ivr_originate.c:2154 Sending early media 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:1322 sofia/internal/214@pbx0.tssec.lan Patched SDP --- v=0 o=root 31797 31797 IN IP4 4.55.17.66 s=session c=IN IP4 4.55.17.66 t=0 0 m=audio 14104 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20

+++ v=0 o=root 31797 31797 IN IP4 4.55.17.66 s=session c=IN IP4 192.168.50.10 t=0 0 m=audio 32698 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20

2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2372 PROXY AUDIO RTP [sofia/internal/214@pbx0.tssec.lan] 192.168.50.10:32698->192.168.50.51:16474 codec: 0 ms: 30 2009-10-06 19:03:31.112988 [DEBUG] switch_rtp.c:1163 Not using a timer 2009-10-06 19:03:31.112988 [DEBUG] sofia_glue.c:2508 Set comfort noise payload to 13 2009-10-06 19:03:31.112988 [DEBUG] sofia.c:3462 sofia/internal/214@pbx0.tssec.lan receive message [PROGRESS] 2009-10-06 19:03:31.116989 [NOTICE] mod_sofia.c:1521 Pre-Answer sofia/internal/214@pbx0.tssec.lan! 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/214@pbx0.tssec.lan [BREAK] 2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_originate.c:2196 Originate Resulted in Success: [sofia/external/*408774957558438@sip.callwithus.com ] 2009-10-06 19:03:31.116989 [DEBUG] switch_channel.c:182 sofia/external/*408774957558438@sip.callwithus.com receive message [AUDIO_SYNC] 2009-10-06 19:03:31.116989 [DEBUG] switch_channel.c:182 sofia/internal/214@pbx0.tssec.lan receive message [AUDIO_SYNC] 2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_bridge.c:911 sofia/external/*408774957558438@sip.callwithus.com receive message [BRIDGE] 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send signal sofia/external/*408774957558438@sip.callwithus.com [BREAK] 2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_bridge.c:918 sofia/internal/214@pbx0.tssec.lan receive message [BRIDGE] 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/214@pbx0.tssec.lan [BREAK] 2009-10-06 19:03:31.116989 [DEBUG] switch_ivr_bridge.c:962 (sofia/external/*408774957558438@sip.callwithus.com) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2009-10-06 19:03:31.116989 [DEBUG] sofia.c:3359 Channel sofia/internal/214@pbx0.tssec.lan entering state [early][183] 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/214@pbx0.tssec.lan [BREAK] 2009-10-06 19:03:31.116989 [DEBUG] switch_core_session.c:932 Send signal sofia/external/*408774957558438@sip.callwithus.com [BREAK] 2009-10-06 19:03:31.116989 [DEBUG] switch_core_state_machine.c:306 (sofia/external/*408774957558438@sip.callwithus.com) Running State Change CS_EXCHANGE_MEDIA 2009-10-06 19:03:31.116989 [DEBUG] switch_core_state_machine.c:343 (sofia/external/*408774957558438@sip.callwithus.com) State EXCHANGE_MEDIA 2009-10-06 19:03:31.116989 [DEBUG] mod_sofia.c:430 SOFIA LOOPBACK recv 822 bytes from udp/[38.99.70.232]:5060 at 15:05:58.503701:

------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.50.10:5080;rport=5080;branch=z9hG4bK97BHK9BtN6r4K Record-Route: <sip:38.99.70.232;lr=on;ftag=HXjrDKpHHX8ZN;did=e8e.089fc2b6> From: "Mihailova Ludmila" <sip:544620048@sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN To: <sip:*408774957558438@sip.callwithus.com>;tag=as22d3d3b0 Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f CSeq: 121302431 INVITE User-Agent: CWU SIP-GW Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:*408774957558438@204.74.213.1:5062> Content-Type: application/sdp Content-Length: 209

v=0 o=root 31797 31798 IN IP4 4.55.17.66 s=session c=IN IP4 4.55.17.66 t=0 0 m=audio 14104 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv

------------------------------------------------------------------------ send 762 bytes to udp/[38.99.70.232]:5060 at 15:05:58.507702:

------------------------------------------------------------------------ ACK sip:*408774957558438@204.74.213.1:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.10:5080;rport;branch=z9hG4bKaH59m4vXjFFQF Route: <sip:38.99.70.232;lr=on;ftag=HXjrDKpHHX8ZN;did=e8e.089fc2b6> Max-Forwards: 70 From: "Mihailova Ludmila" <sip:544620048@sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN To: <sip:*408774957558438@sip.callwithus.com>;tag=as22d3d3b0 Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f CSeq: 121302431 ACK Contact: <sip:gw+callwithus@77.239.230.202:5080;transport=udp> Content-Length: 0

------------------------------------------------------------------------ 2009-10-06 19:03:39.975377 [DEBUG] sofia.c:3359 Channel sofia/external/*408774957558438@sip.callwithus.com entering state [ready][200] 2009-10-06 19:03:39.975377 [DEBUG] switch_channel.c:1935 Send signal sofia/internal/214@pbx0.tssec.lan [BREAK] 2009-10-06 19:03:39.975377 [NOTICE] sofia.c:3814 Channel [sofia/external/*408774957558438@sip.callwithus.com] has been answered 2009-10-06 19:03:39.975377 [DEBUG] switch_channel.c:182 sofia/external/*408774957558438@sip.callwithus.com receive message [AUDIO_SYNC] 2009-10-06 19:03:39.975377 [DEBUG] sofia.c:3819 sofia/internal/214@pbx0.tssec.lan receive message [ANSWER] 2009-10-06 19:03:39.975377 [DEBUG] sofia_glue.c:1322 sofia/internal/214@pbx0.tssec.lan Patched SDP --- v=0 o=root 31797 31797 IN IP4 4.55.17.66 s=session c=IN IP4 4.55.17.66 t=0 0 m=audio 14104 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20

+++ v=0 o=root 31797 31797 IN IP4 4.55.17.66 s=session c=IN IP4 192.168.50.10 t=0 0 m=audio 32698 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20

2009-10-06 19:03:39.975377 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/214@pbx0.tssec.lan [BREAK] 2009-10-06 19:03:39.975377 [NOTICE] sofia.c:3819 Channel [sofia/internal/214@pbx0.tssec.lan] has been answered 2009-10-06 19:03:39.975377 [DEBUG] switch_channel.c:182 sofia/internal/214@pbx0.tssec.lan receive message [AUDIO_SYNC] 2009-10-06 19:03:39.975377 [DEBUG] sofia.c:3359 Channel sofia/internal/214@pbx0.tssec.lan entering state [completed][200] 2009-10-06 19:03:39.991381 [DEBUG] sofia.c:3359 Channel sofia/internal/214@pbx0.tssec.lan entering state [ready][200] recv 1054 bytes from udp/[38.99.70.232]:5060 at 15:06:05.577607:

------------------------------------------------------------------------ INVITE sip:gw+callwithus@192.168.50.10:5080;transport=udp SIP/2.0 Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.0 Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK41d48157;rport=5062 From: <sip:*408774957558438@sip.callwithus.com>;tag=as22d3d3b0 To: "Mihailova Ludmila" <sip:544620048@sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN Contact: <sip:*408774957558438@204.74.213.1:5062> Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f CSeq: 102 INVITE User-Agent: CWU SIP-GW Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-info: SIP re-invite (T38 switchover) Content-Type: application/sdp Content-Length: 350

v=0 o=root 31797 31799 IN IP4 32.50.63.192 s=session c=IN IP4 32.50.63.192 t=0 0 m=image 14104 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:176 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy

------------------------------------------------------------------------ send 443 bytes to udp/[38.99.70.232]:5060 at 15:06:05.577607:

------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.0 Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK41d48157;rport=5062 From: <sip:*408774957558438@sip.callwithus.com>;tag=as22d3d3b0 To: "Mihailova Ludmila" <sip:544620048@sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805 Content-Length: 0

------------------------------------------------------------------------ 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3359 Channel sofia/external/*408774957558438@sip.callwithus.com entering state [received][100] 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3366 Remote SDP: v=0 o=root 31797 31799 IN IP4 32.50.63.192 s=session c=IN IP4 32.50.63.192 t=0 0 m=image 14104 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:176 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy

2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3683 Passing SDP to other leg. v=0 o=root 31797 31799 IN IP4 32.50.63.192 s=session c=IN IP4 32.50.63.192 t=0 0 m=image 14104 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:176 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy

2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3696 sofia/internal/214@pbx0.tssec.lan receive message [MEDIA_REDIRECT] 2009-10-06 19:03:47.049284 [DEBUG] mod_sofia.c:1195 sofia/internal/214@pbx0.tssec.lan Sending media re-direct: v=0 o=root 31797 31799 IN IP4 32.50.63.192 s=session c=IN IP4 32.50.63.192 t=0 0 m=image 14104 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:176 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy

2009-10-06 19:03:47.049284 [DEBUG] sofia_glue.c:1101 Remote address:port [192.168.50.51:16474] has not changed. 2009-10-06 19:03:47.049284 [DEBUG] sofia_glue.c:1322 sofia/internal/214@pbx0.tssec.lan Patched SDP --- v=0 o=root 31797 31799 IN IP4 32.50.63.192 s=session c=IN IP4 32.50.63.192 t=0 0 m=image 14104 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:176 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy

+++ v=0 o=root 31797 31799 IN IP4 32.50.63.192 s=session c=IN IP4 192.168.50.10 t=0 0 m=image 32698 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:176 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy

2009-10-06 19:03:47.049284 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/214@pbx0.tssec.lan [BREAK] 2009-10-06 19:03:47.049284 [DEBUG] sofia.c:3359 Channel sofia/internal/214@pbx0.tssec.lan entering state [calling][0] 2009-10-06 19:03:47.077291 [DEBUG] sofia_glue.c:1101 Remote address:port [192.168.50.51:16474] has not changed. 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3158 Passing 200 OK to other leg 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3170 sofia/external/*408774957558438@sip.callwithus.com receive message [RESPOND] 2009-10-06 19:03:47.077291 [DEBUG] mod_sofia.c:1427 Responding with 200 [OK] 2009-10-06 19:03:47.077291 [DEBUG] sofia_glue.c:1322 sofia/external/*408774957558438@sip.callwithus.com Patched SDP --- v=0 o=- 216899808 216899808 IN IP4 192.168.50.51 s=- c=IN IP4 192.168.50.51 t=0 0 m=image 16474 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:200 a=T38FaxUdpEC:t38UDPRedundancy

+++ v=0 o=- 216899808 216899808 IN IP4 192.168.50.51 s=- c=IN IP4 77.239.230.202 t=0 0 m=image 32404 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:200 a=T38FaxUdpEC:t38UDPRedundancy

2009-10-06 19:03:47.077291 [DEBUG] sofia_glue.c:1110 AUDIO RTP CHANGING DEST TO: [32.50.63.192:14104] 2009-10-06 19:03:47.077291 [DEBUG] switch_core_session.c:630 Send signal sofia/external/*408774957558438@sip.callwithus.com [BREAK] 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3359 Channel sofia/internal/214@pbx0.tssec.lan entering state [ready][200] 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3366 Remote SDP: v=0 o=- 216899808 216899808 IN IP4 192.168.50.51 s=- c=IN IP4 192.168.50.51 t=0 0 m=image 16474 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:200 a=T38FaxUdpEC:t38UDPRedundancy

send 1003 bytes to udp/[38.99.70.232]:5060 at 15:06:05.605615:

------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.0 Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK41d48157;rport=5062 From: <sip:*408774957558438@sip.callwithus.com>;tag=as22d3d3b0 To: "Mihailova Ludmila" <sip:544620048@sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f CSeq: 102 INVITE Contact: <sip:gw+callwithus@77.239.230.202:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 274

v=0 o=- 216896551 216896552 IN IP4 192.168.50.51 s=- c=IN IP4 77.239.230.202 t=0 0 m=image 32404 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:200 a=T38FaxUdpEC:t38UDPRedundancy

------------------------------------------------------------------------ 2009-10-06 19:03:47.077291 [DEBUG] sofia.c:3359 Channel sofia/external/*408774957558438@sip.callwithus.com entering state [completed][200] 2009-10-06 19:03:47.285347 [INFO] switch_rtp.c:1905 Auto Changing port from 32.50.63.192:14104 to 4.55.17.66:14104 recv 527 bytes from udp/[38.99.70.232]:5060 at 15:06:05.817672:

------------------------------------------------------------------------ ACK sip:gw+callwithus@192.168.50.10:5080;transport=udp SIP/2.0 Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKc406.1d5670f7.2 Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK5ffd72cd;rport=5062 From: <sip:*408774957558438@sip.callwithus.com>;tag=as22d3d3b0 To: "Mihailova Ludmila" <sip:544620048@sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN Contact: <sip:*408774957558438@204.74.213.1:5062> Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f CSeq: 102 ACK User-Agent: CWU SIP-GW Max-Forwards: 69 Content-Length: 0

------------------------------------------------------------------------ 2009-10-06 19:03:47.289349 [DEBUG] sofia.c:3359 Channel sofia/external/*408774957558438@sip.callwithus.com entering state [ready][200] recv 476 bytes from udp/[38.99.70.232]:5060 at 15:06:47.588933:

------------------------------------------------------------------------ BYE sip:gw+callwithus@192.168.50.10:5080;transport=udp SIP/2.0 Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKd406.f8553e14.0 Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK0f2cff01;rport=5062 From: <sip:*408774957558438@sip.callwithus.com>;tag=as22d3d3b0 To: "Mihailova Ludmila" <sip:544620048@sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f CSeq: 103 BYE User-Agent: CWU SIP-GW Max-Forwards: 69 Content-Length: 0

------------------------------------------------------------------------ 2009-10-06 19:04:29.060609 [NOTICE] sofia.c:328 Hangup sofia/external/*408774957558438@sip.callwithus.com [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-10-06 19:04:29.060609 [DEBUG] switch_channel.c:1726 Send signal sofia/external/*408774957558438@sip.callwithus.com [KILL] 2009-10-06 19:04:29.060609 [DEBUG] switch_core_session.c:932 Send signal sofia/external/*408774957558438@sip.callwithus.com [BREAK] 2009-10-06 19:04:29.060609 [DEBUG] switch_core_state_machine.c:437 thread mismatch skipping state handler. send 574 bytes to udp/[38.99.70.232]:5060 at 15:06:47.588933:

------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 38.99.70.232;branch=z9hG4bKd406.f8553e14.0 Via: SIP/2.0/UDP 204.74.213.1:5062;branch=z9hG4bK0f2cff01;rport=5062 From: <sip:*408774957558438@sip.callwithus.com>;tag=as22d3d3b0 To: "Mihailova Ludmila" <sip:544620048@sip.callwithus.com;transport=udp>;tag=HXjrDKpHHX8ZN Call-ID: 8978ece2-2d2c-122d-9c9d-00163e90370f CSeq: 103 BYE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12805 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Length: 0

------------------------------------------------------------------------

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On Wed, Oct 7, 2009 at 6:46 PM, Klaus Hochlehnert <Mailings@kh-dev.de>wrote:

Hi,

currently I'm playing around with call pickup and Snom phones. I'm using the intercept function for that.

My "problem" is now that after the call pickup (which works fine) I don't see the caller id of the original call. Instead I see the pickup code, e.g. *820

I've tried to change nearly every channel variable (before doing the call pickup), but the phone won't display the caller id.

Does anybody know how to change that?

Can you paste a debug log of the call flow? Also, are you using just the default dialplan or have you made any changes? Lastly, what mods, if any, did you make to your Snom configs? I have a bricked Snom that I'm waiting for TFTP to fix but when it does I will see about testing this. -MC

Sorry for hijacking. Didn't know that the mailing list system finds out about that... ;-) Will do the update in the next days and try again. Today I tried with the command send_display. This workaround currently works for me. Thanks, Klaus From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, October 09, 2009 12:54 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Changing callerid before/after call pickup Also, I failed to mention two things: #1 - Please don't hijack threads. :) #2 - Update to the latest SVN and try again. You might be pleasantly surprised. -MC On Wed, Oct 7, 2009 at 6:46 PM, Klaus Hochlehnert <Mailings@kh-dev.de<mailto:Mailings@kh-dev.de>> wrote: Hi, currently I'm playing around with call pickup and Snom phones. I'm using the intercept function for that. My "problem" is now that after the call pickup (which works fine) I don't see the caller id of the original call. Instead I see the pickup code, e.g. *820 I've tried to change nearly every channel variable (before doing the call pickup), but the phone won't display the caller id. Does anybody know how to change that? Thanks, Klaus _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org