Re: [asterisk-users] Crash asterisk res_odbc

Hi, Thank you for the reply. My OS is : Debian 7. But i have more than 20 servers with the same features/resources (OS, material, ... ) without the issue. Regards. Le 28/02/2016 21:49, Leandro Dardini a écrit : Which operating system are you using? I have experienced the same problem on

Re: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

2016-02-19 12:01 GMT+01:00 Marek Červenka : > on my own server > Today, I'm back from holidays trip. First of all, thanks for replying ! I'll try to use jssip as you suggested. Anyway, I'm still failing to understand if wiki's page [1] is still valid with Asterisk 13, and

[asterisk-users] Zoiper on Windows Phone

Hello everyone, I have some problems to enable push the Zoiper Windows Phone in my Asterisk 11. Below is the result of CLI == Using SIP RTP CoS mark 5 -- Executing [1033@ramais:1] Answer("SIP/1030-0201", "") in new stack > 0x7efc90024190 -- Probation passed - setting RTP source

Re: [asterisk-users] PJSIP signaling question

On Mon, Feb 29, 2016 at 2:04 PM, Kevin Long wrote: > > > Greetings. > > > I am using the PJSIP driver with TLS transport, and my endpoints are SIP > mobile apps operating in environments that I do not control. > > I would like Asterisk to default to sending INVITES

[asterisk-users] PJSIP signaling question

Greetings. I am using the PJSIP driver with TLS transport, and my endpoints are SIP mobile apps operating in environments that I do not control. I would like Asterisk to default to sending INVITES and all other SIP signals to endpoints via the existing SIP TLS connection which is already