I wasn't sure how the lines were counted. Here is the debug output from Asterisk where it is building the invite packet. I looked at the a=T38 lines and nothing is standing out to me.
Ryan [May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 0 [ 47]: INVITE sip:[email protected]:5060 SIP/2.0 [May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK2837f4cf;rport [May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 2 [ 54]: Route: <sip:x.x.x.x;lr>,<sip:x.x.x.x;lr> [May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 3 [ 16]: Max-Forwards: 70 [May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 4 [ 59]: From: <sip:[email protected]:5060>;tag=as7d21d6f3 [May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 5 [ 53]: To: <sip:[email protected]:5060>;tag=gK0d4c48f7 [May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 6 [ 39]: Contact: <sip:[email protected]> [May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 7 [ 39]: Call-ID: [email protected] [May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 8 [ 16]: CSeq: 102 INVITE [May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 9 [ 36]: User-Agent: Asterisk PBX 1.6.2.7-rc3 [May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 10 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 12 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 14 [ 19]: Content-Length: 293 [May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 15 [ 0]: [May 6 13:29:05] DEBUG[32389] chan_sip.c: Body 0 [ 3]: v=0 [May 6 13:29:05] DEBUG[32389] chan_sip.c: Body 1 [ 48]: o=root 2048302926 2048302927 IN IP4 x.x.x.x [May 6 13:29:05] DEBUG[32389] chan_sip.c: Body 2 [ 26]: s=Asterisk PBX 1.6.2.7-rc3 [May 6 13:29:05] DEBUG[32389] chan_sip.c: Body 3 [ 21]: c=IN IP4 x.x.x.x [May 6 13:29:05] DEBUG[32389] chan_sip.c: Body 4 [ 5]: t=0 0 [May 6 13:29:05] DEBUG[32389] chan_sip.c: Body 5 [ 22]: m=image 4575 udptl t38 [May 6 13:29:05] DEBUG[32389] chan_sip.c: Body 6 [ 17]: a=T38FaxVersion:0 [May 6 13:29:05] DEBUG[32389] chan_sip.c: Body 7 [ 21]: a=T38MaxBitRate:14400 [May 6 13:29:05] DEBUG[32389] chan_sip.c: Body 8 [ 22]: a=T38FaxFillBitRemoval [May 6 13:29:05] DEBUG[32389] chan_sip.c: Body 9 [ 37]: a=T38FaxRateManagement:transferredTCF [May 6 13:29:05] DEBUG[32389] chan_sip.c: Body 10 [ 24]: a=T38FaxMaxDatagram:1400 [May 6 13:29:05] DEBUG[32389] chan_sip.c: Body 11 [ 23]: a=T38FaxUdpEC:t38UDPFEC On Thu, May 6, 2010 at 6:54 PM, Kevin P. Fleming <[email protected]> wrote: > On 05/06/2010 05:46 PM, Ryan Wagoner wrote: >> Does anybody have T.38 faxing working with Flowroute? I am running >> Asterisk 1.6.2.7-rc3 with spandsp 0.0.6-pre17. I can successfully >> receive faxes over ulaw. I enabled T.38 with t38pt_udptl=yes in >> sip.conf. When I receive a fax it tries to negotiate T.38 and >> Flowroute sends back a Bad Request response saying I have a SIP syntax >> error. >> >> Flowroute support is recommending that I try again after removing >> externip and localnet from sip.conf. They state that their service >> will recognize the private IP and rewrite the SIP packets. However >> this is going to cause issues for my remote SIP phones. >> >> Thanks, >> Ryan >> >> DEBUG[32389] app_fax.c: Negotiating T.38 for receive on >> SIP/flowroute-00000000 >> >> INVITE sip:[email protected]:5060 SIP/2.0 >> ... >> CSeq: 102 INVITE >> User-Agent: Asterisk PBX 1.6.2.7-rc3 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO >> Supported: replaces, timer >> X-asterisk-Info: SIP re-invite (External RTP bridge) >> Content-Type: application/sdp >> Content-Length: 293 >> >> v=0 >> o=root 2048302926 2048302927 IN IP4 xx.xx.xx.xx >> s=Asterisk PBX 1.6.2.7-rc3 >> c=IN IP4 xx.xx.xx.xx >> t=0 0 >> m=image 4575 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:14400 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxDatagram:1400 >> a=T38FaxUdpEC:t38UDPFEC >> >> SIP/2.0 400 Bad Request >> ... >> CSeq: 102 INVITE >> Error-Info: <sip:[email protected]>;cause="[line 023] SIP syntax error" >> Content-Length: 0 > > Which line is 'line 23' of the T.38 re-INVITE? > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > skype: kpfleming | jabber: [email protected] > Check us out at www.digium.com & www.asterisk.org > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
