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Hi All, We are experiencing a a problem when running calls over IAX with g.729. The call flow is as follows: Sip handset -(SIP)> Asterisk1 -(IAX)> Asterisk2 -(SIP)> Carrier The first Asterisk system is running 1.2 and the second is running 1.0. When using g726 from the handset all the way thru to Asterisk2(then 729 for the carrier leg) calls go thru fine, but when using g729, there

Dear All On my CentOS 5 server , I have upgraded my Asterisk from 1.4 to 1.6.2 but its CLI help does not show sip and when dialing outward sip it complains as 'sip not implemented' . Can you please let me know what is wrong my case here ? Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I seem to have stumbled on a bit of a problem. When trying to send a fax with Fax For Asterisk on 1.6.2.x (have tried 1.6.2.5, 1.6.2.7 and the current svn version, with FFA 1.2 I get a core dump each time. Here is an extract form the console: [May 12 22:47:09] DEBUG[22584]: app_queue.c:1084 handle_statechange: Device 'SIP/vltb-sbc01' changed to state '1' (Not in use)

I have the following dialplan: ; calls prefix by '8' are recorded exten = _8[01]./_251,1,Set(something=shortened) exten = _8[01]./_251,n,Set(WAV=filename) exten = _8[01]./_251,n,Monitor(wav,${WAV},mb) exten = _8[01]./_251,n,Dial(mISDN/2/${EXTEN:1},,g) exten = _8[01]./_251,n,System(send-recorded-conversation ${WAV}.wav ${EXTEN:1} emailaddr) exten = _8[01]./_251,n,Hangup() The idea is that

Hi All, I hate to post yet another "bloody hangup detection issue" on the list, but I have been pulling my hair out no end of late with a hangup detection issue on 1 system (have a few others out there with TDM400's and no issue but this one is causing a real headache) The scenario is - system with TDM04B, a call comes in on a analogue line, rings internally and then diverts to a

Hi All, I am in the process of migrating from 1.4.20 to 1.6.2.x and have stumbled across a number of "differences" between the 2 versions that are forcing me to use local channels in my dialplan (mainly centered around the different behavior of CDR fields in the 2 versions) . Previously, I would place a call via an AMI Originate action similar to: action:.Originate..

Hi All, I have recently installed a TC400B card into a system and am trying to get it to work. As far as I ca tell from the docco on Digiums website, there is no config as such unless you want to enable / disable only 1 codec, otherwise by default it runs as 92 channels of either. I have tried asterisk 1.4.9, 1.4.10 and 1.4.10.1 along with zaptel 1.4.4 and addons 1.4.2. The zaptel modules

Hi All, I have been running a environment with asterisk 1.4.20.1 for some time now with no issue but have recently added some extra functionality (enabled call recording via MixMonitor) and ran into some deadlock issues which seem to be well documented with earlier 1.4.x releases so have decided to take the plunge and upgrade. I decided to start testing with 1.6.2 but have run into a couple