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:+number at xx.xx.xx.xx: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:+number at xx.xx.xx.xx>;cause="[line 023] SIP syntax error" Content-Length: 0 WARNING[32389] app_fax.c: Transmission error
Kevin P. Fleming
2010-May-06 22:54 UTC
[asterisk-users] T.38 Fax With Flowroute SIP Provider
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:+number at xx.xx.xx.xx: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:+number at xx.xx.xx.xx>;cause="[line 023] SIP syntax error" > Content-Length: 0Which 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: kfleming at digium.com Check us out at www.digium.com & www.asterisk.org