Group, I have been going through all the chit-chat about TTS and the various engines available to integrate with Asterisk incl. flite/festival, espeak, Nuance etc but I am wondering if anyones tried any or all of these to compile on a Sparc based Sola..
How can I run the sip service on asterisk on another port beside 5080?
I mean asterisk will still take sip requests on port:5080 and another custom
port, lets say port:6080
Thanks for any h..
all. I have an issue with T.38 and re-invites. Topology: provider -> A (asterisk 1.6) -> B (asterisk 1.6) -> extension -> -> (software fax, gateway whatever). When between A and B trunk is canreinvite=no everything is working smooth. When I switch canreinv..
The Asterisk Development Team has announced the release of Asterisk 22.214.171.124.This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 126.96.36.199 resolves several issues reported..
The Asterisk Development Team has announced the release of Asterisk 1.4.37. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.37 resolves several issues reported by ..
I dial on A* from a linphonec to a Playback() extension, then suddenly the sound stops after a while, without any notice. I enabled debug both in linphone and A*, and the RTP packets are sent from A* and received from linphone. It doesnt matter whet..
Limiting the call duration with the L-option of the Dial()-command is working fine, however the announcement is not played. Dialplan : exten => _367,n,Set(LIMIT_PLAYAUDIO_CALLER=yes) exten => _367,n,Dial(SIP/test6,,L(11000,5000,5000)) The call la..
All i have one issue regarding caller id, once i received a call from my SIP provider it always set caller id with append 1 into original callerID if a call from USA then there is no problem , but if i receive a call from other country like INDIA i h..