Archives : November-2015
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,yesterday I have run into a show stopper problem with Asterisk 13.6 / PJSIP 2.4.5.My users couldnt use their phones during the whole day, nor incoming nor outgoing calls were working. This has been mysterious because Asterisk used to work stably..
I just purchased an Amfeltec USB-FXO adapter and am trying to compile DAHDI 2.10 on a Raspberry PI running Pidora 2014 R3.I have all the dependencies but I get an error and cannot finish.Is it even possible to compile DAHDI for the ARM plataform?H..
Issue 375978 :Support AGSM audio codec https://code.google.com/p/chromium/issues/detail?id=375978#c12Please click on the link and at least Star / Favorite the issue to vote it up.Chrome can play many audio files directly within the browser.We want..
,I have a very strange problem :* external calls work perfectly,* internal calls between some phones too,* but internal call between two similar phones dont work !!! (Snom 710)When we have sound, there are no errors in asterisk. When we do not have sou..
Asterisk unable to receive DTMF tone from sip client. Im using the (d) flag in dial application to perfume one digit exit during ringing state. But unfortunately doesnt work. Here is my sip configuration :-[100]type=friend username0host=dynamic nat=..
How can I update asterisk to send back move temporarily with updated IPaddress to incoming INVITE.i.e, Incoming call from ITSP to server 1 with x DID and there is a need to update the ITSP that the specified x DID number is allocated in server 2. Thanks,..
How to accept DMTF tone during ringing mode? Its possible.Regards-..
For those wandering web-searching souls:https://issues.asterisk.org/jira/browse/ASTE..
im evaluating performance of CentOS7i did tests on CentOS6 x86_64/distro kernel 2.6.32, asterisk 11.16.0 with 500calls (7sec alaw, simple dialplan, pass trough – sipp generators/asterisk receiver with answer/playback)scenario – sipp generators – aster..
all,My Asterisk is between my ITSP and a SIP phone. I cannot do direct media between the provider and the SIP phone, but I would like Asterisk to locally RTP bridge the two channels using native_rtp.Example:My SIP phone supports G722 and PCMA, as d..