Re: [asterisk-users] t.38 interop with metaswitch
On 10/11/2011 01:50 AM, Jeremy Kister wrote: I'm trying to receive a t.38 fax from a Metaswitch 7.3. I have full control over the metaswitch, but it is in production. I have the Metaswitch hooked to Asterisk 1.8.7.0 (middleman named s3). Then I have the target Asterisk 1.8.7.0 with res_fax_digum 1.8.4_1.3.0 (named pbx1) registered to s3. attempts to receive fax over t.38 always error in res_fax with fax session timed-out i have debug output at: http://jeremy.kister.net/tmp/t38/pbx1.txt http://jeremy.kister.net/tmp/t38/s3.txt It appears that the FAX transaction attempted training a few times and failed, then fell back to audio mode (or the calling endpoint hung up). is the UDPTL debug on pbx1.txt (near line 474) interesting in that LOG_TAG(s) is evaluated to 'SIP/' ? This bug with UDPTL debug messages was just fixed last week. I don't think my (sip|udptl|extensions).conf are interesting, but i'd be happy to post them. the only interesting tidbit is that when i changed 't38pt_udptl=yes' to 'yes,none' or 'yes,redundancy' the fax would fail with 't38 negotiation failed. Well, as a starting point, I'd suggest disabling directmedia (canreinvite) on s3. It should be possible for directmedia to be enabled for RTP and not interfere with UDPTL, but there could still be lingering problems there. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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
Re: [asterisk-users] t.38 interop with metaswitch
On 10/11/2011 11:48 AM, Kevin P. Fleming wrote: Well, as a starting point, I'd suggest disabling directmedia (canreinvite) on s3. It should be possible for directmedia to be enabled for RTP and not interfere with UDPTL, but there could still be lingering problems there. yep, you hit the nail on the head. setting directmedia=no on s3 allows me to receive t38 faxes on pbx1. debug for successful faxes in this case are at: http://jeremy.kister.net/tmp/t38/no-reinvite-on-s3/pbx1.txt http://jeremy.kister.net/tmp/t38/no-reinvite-on-s3/s3.txt is there further changes that can be done to allow reinvite on s3? or is this something that should go to the tracker ? thanks, -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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
Re: [asterisk-users] t.38 interop with metaswitch
On 10/11/2011 02:04 PM, Jeremy Kister wrote: On 10/11/2011 11:48 AM, Kevin P. Fleming wrote: Well, as a starting point, I'd suggest disabling directmedia (canreinvite) on s3. It should be possible for directmedia to be enabled for RTP and not interfere with UDPTL, but there could still be lingering problems there. yep, you hit the nail on the head. setting directmedia=no on s3 allows me to receive t38 faxes on pbx1. debug for successful faxes in this case are at: http://jeremy.kister.net/tmp/t38/no-reinvite-on-s3/pbx1.txt http://jeremy.kister.net/tmp/t38/no-reinvite-on-s3/s3.txt is there further changes that can be done to allow reinvite on s3? or is this something that should go to the tracker ? It should be reported as a bug; even if the UDPTL stack doesn't support directmedia, Asterisk should be able to properly setup a UDPTL media stream in spite of a previous directmedia RTP path. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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
