FYI: I have T.38 disabled and have found the delay/jitter acceptable in our environment over the network with g711ulaw passthrough.
- Jared On Apr 16, 2011, at 11:28 AM, Oguzhan Kayhan wrote: > 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 -- _____________________________________________________________________ -- 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