Update I have found that setting notransfer=yes enables me to call Telappliant numbers (09XX) and not get disconnected but if I call a BT number the call goes out via Telappliant to the BT phone, it rings, the client answers, they can hear you, but firefly does not know the other end has been answered and continues to ring, obviously then you cant hear the client. Is this a symptom of notransfer=yes or is there another problem? Firewall related maybe?
Anyone with a working Telappliant account using IAX? Cheers! Roy.... > -----Original Message----- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Roy Eddleston > Sent: 28 July 2004 09:13 > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Iax unable to transfer > > Dimitri > > Did you get a resolution to this problem? I am seeing the same, my * box > talks to Telappliant using AIX, anybody else seen this? > > Roy.. > > > -----Original Message----- > > From: [EMAIL PROTECTED] [mailto:asterisk-users- > > [EMAIL PROTECTED] On Behalf Of reseaux > > Sent: 23 June 2004 10:47 > > To: [EMAIL PROTECTED] > > Subject: [Asterisk-Users] Iax unable to transfer > > > > Dear List > > I have notice this kind of problem between my two * box. > > My scenario is : > > Iax GSM > > IaxClient----->PBX1------------>PBX2-->TDM > > today CVS Stable V1 > > > > I use as Client FireFly with IAX2/GSM and try to call my PBX1 this > server call > > PBX2 to terminate the call trought a TDM line (TE410P) but after PBX2 > join > > the two call i can see the log below from my PBX1, i can speak for few > second > > and after the FireFly hangup. > > I have try to change * version from Stable to today CVS but no success > same > > problem. > > I have enabled the IAX Debug and seems the RX side (PBX1) dont accept > > something from PBX2 and show the "unable to transfer" (im not expert) > :-) > > > > The strange thing is if i call from Sip Phone/client i dont have a > problem the > > Call is bridged! > > > > The events from the CLI: > > --------- > > Executing Dial("[EMAIL PROTECTED]/5", > > "IAX2/out:[EMAIL PROTECTED]/[EMAIL PROTECTED]|60|g") in new > stack > > -- Called out:[EMAIL PROTECTED]/[EMAIL PROTECTED] > > -- Call accepted by 213.215.xx.xx (format GSM) > > -- Format for call is GSM > > -- IAX2[out]/6 stopped sounds > > -- IAX2[out]/6 is ringing > > -- IAX2[out]/6 stopped sounds > > -- IAX2[out]/6 answered [EMAIL PROTECTED]/5 > > -- Attempting native bridge of [EMAIL PROTECTED]/5 and > IAX2[out]/6 > > -- Channel 'IAX2[out]/6' unable to transfer > > -- Hungup 'IAX2[out]/6' > > -- Executing Hangup("[EMAIL PROTECTED]/5", "") in new stack > > == Spawn extension (incoming,0012234456666, 4) exited non-zero on > 'IAX2 > > [EMAIL PROTECTED]/5' > > -- Executing Hangup("[EMAIL PROTECTED]/5", "") in new stack > > ----------- > > > > Thanks in advance for possible help! > > Dimitri > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
