Re: [asterisk-users] T38 REINVITe issue
Anyone for this ? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ujjval Karihaloo Sent: Monday, October 05, 2009 11:02 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] T38 REINVITe issue Hi My call flow is T38 static IP gateway -- Asterisk -- Sip Provider-- PSTN Call is placed in reverse direction - from PSTN to T38 Gateway. T38 GW reinvites with T38, and asterisk passes it along to the SIP provider. The SIP provider challenges it and asterisk reponds to the Challenge with INVITE with Auth credentials...however, the Asterisk changes the SDP and replaces the T38 info in SDP with G711uLawand fax fails. How do I configure the host entry in users.conf such that it maintains the T38 reinvite as it responds to the SIP INVITE challenge from the Sip Provider. Her eis my users.conf entry for Asterisk registration to the Sip Provider. (I know I don't have T38 as allowed codecs, not sure what to add for T38) [trunk_66] ;register allow = ulaw dialformat = ${EXTEN:1} canreinvite = no hasexten = no hasiax = no hassip = yes host = provider.com insecure = very port = 5060 registeriax = no registersip = yes trunkname = abc username = abc disallow = gsm,g726,alaw contact = abc secret = abc Any ideas appreciated. Thx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38 REINVITe issue
Ujjval Karihaloo wrote: Her eis my users.conf entry for Asterisk registration to the Sip Provider. (I know I don’t have T38 as allowed codecs, not sure what to add for T38) [trunk_66] ;register allow = ulaw dialformat = ${EXTEN:1} canreinvite = no hasexten = no hasiax = no hassip = yes host = provider.com insecure = very port = 5060 registeriax = no registersip = yes trunkname = abc username = abc disallow = gsm,g726,alaw contact = abc secret = abc You'll need to put t38pt_udptl = yes somewhere in your sip.conf, probably in the general section for T.38 to work properly. -- Trevor Peirce Digital Conceptions Canada http://www.digitalcon.ca 1-888-606-3030 / 250-391-7822 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38 REINVITe issue
Already have it... If provider does not challenge re- invite Fax works fine! Ujjval On Oct 6, 2009, at 11:33 PM, Trevor Peirce tpei...@digitalcon.ca wrote: Ujjval Karihaloo wrote: Her eis my users.conf entry for Asterisk registration to the Sip Provider. (I know I don’t have T38 as allowed codecs, not sure wha t to add for T38) [trunk_66] ;register allow = ulaw dialformat = ${EXTEN:1} canreinvite = no hasexten = no hasiax = no hassip = yes host = provider.com insecure = very port = 5060 registeriax = no registersip = yes trunkname = abc username = abc disallow = gsm,g726,alaw contact = abc secret = abc You'll need to put t38pt_udptl = yes somewhere in your sip.conf, probably in the general section for T.38 to work properly. -- Trevor Peirce Digital Conceptions Canada http://www.digitalcon.ca 1-888-606-3030 / 250-391-7822 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] T38 REINVITe issue
Hi My call flow is T38 static IP gateway -- Asterisk -- Sip Provider-- PSTN Call is placed in reverse direction - from PSTN to T38 Gateway. T38 GW reinvites with T38, and asterisk passes it along to the SIP provider. The SIP provider challenges it and asterisk reponds to the Challenge with INVITE with Auth credentials...however, the Asterisk changes the SDP and replaces the T38 info in SDP with G711uLawand fax fails. How do I configure the host entry in users.conf such that it maintains the T38 reinvite as it responds to the SIP INVITE challenge from the Sip Provider. Her eis my users.conf entry for Asterisk registration to the Sip Provider. (I know I don't have T38 as allowed codecs, not sure what to add for T38) [trunk_66] ;register allow = ulaw dialformat = ${EXTEN:1} canreinvite = no hasexten = no hasiax = no hassip = yes host = provider.com insecure = very port = 5060 registeriax = no registersip = yes trunkname = abc username = abc disallow = gsm,g726,alaw contact = abc secret = abc Any ideas appreciated. Thx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users