Hello, thanks or the quick replies. I tried with both 1.6.2.9 and 1.6.2.17 My config is sipaxclient-asterisk-(siptrunk)-telcooperator-analogfax
All i know is telco uses cisco on their side..Not sure which version they are using. I got t38pt_udptl = yes parameter on sip.conf general. Didnt make any other special settings or trunk config itself. On monday, i better run a debug on sip protocol and paste what errors do i have on that.. PLus if i can manage i will ask for version info about telco side. Thank you. > > On Apr 16, 2011, at 9:27 AM, Steve Underwood wrote: > >> On 04/16/2011 01:24 PM, Oguzhan Kayhan wrote: >>> Hello, >>> We have a sip trunk end point with cisco media gateway. >>> VoIP works fine. >>> But when we try to send faxes thru this trunk, we simply can not. >>> >>> Is there anybody experienced such problem and solved? >>> How should i set sip.conf and udptl.conf. >>> >>> I already have t38pt_udptl=yes in sip.conf >>> >>> Thank you. >> How old is the Cisco software? It appears they completely changed their >> T.38 software platform a couple of years ago. Before that is was awful. >> I wasted a lot of time, while developing my T.38 platform, hunting down >> problems that turned out to be broken Ciscos. Since the new software has >> spread into the field, the complaints have largely gone away. > > > I have the following in my dial-peers, but *KEEP IN MIND*, for calls > placed to a POTS dial-peer on a Cisco, it won't do 'fax rate disable' > etc.. on that side of the session if the origin doesn't match a dial peer > as well, so it may be worthwhile to have a high priority (catchall) peer > that has something like .T as the pattern with your catch-all parameters. > > PBX TIE: > > dial-peer voice 7700 pots > answer-address 77.. > destination-pattern 77.. > fax rate disable > port 0/0/0:23 > prefix 77 > ! > > Asterisk PEER: > > dial-peer voice 1000 voip > preference 1 > answer-address 1... > destination-pattern 1... > session protocol sipv2 > session target ipv4:10.0.0.1 > session transport udp > dtmf-relay rtp-nte > codec g711ulaw > fax-relay ecm disable > fax rate disable > fax protocol pass-through g711ulaw > no vad > ! > > DID Setup: > > dial-peer voice 214915135 voip > destination-pattern 214915135. > session protocol sipv2 > session target ipv4:10.0.0.1 > session transport udp > dtmf-relay rtp-nte > codec g711ulaw > fax-relay ecm disable > fax rate disable > fax protocol pass-through g711ulaw > no vad > ! > dial-peer voice 1350 pots > incoming called-number 214915135. > fax rate disable > direct-inward-dial > ! > > > > > -- > _____________________________________________________________________ > -- 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
