search for: pjsua

Hi, I am new to the list, so I hope my questions aren't too stupid. I am using Asterisk 1.4.21.2 and already set it up to use realtime, so a CDR for an incoming SIP call is written in my mysql database. This works fine. The problem is that I don't want to have my phone ringing all the time. I just need a CDR of everyone how is calling me and to read out the CDR from my PHP script. I

...n't work: I get a black window and if I remember correctly, I was getting white noise in one direction (not sure if the noise thing is asterisk's fault or the phone's). I do see the video packets going through when I enable debugging on iax2, so this is strange. I tested with linphone, pjsua, and a Grandstream gxv3240. Video calls that don't go through the iax2 trunk work fine (eg: both phones in the same server). If I change the trunk to use pjsip, video works fine, but then I have other problems that am still working on. I'd rather keep using iax2 for my trunks for a while...

...est -- Met: True --> Dependency: SIPp -- Met: False --> tests/channels/SIP/sip_attended_transfer --- PASSED --> tests/channels/SIP/sip_attended_transfer_tcp --- SKIPPED --> Dependency: twisted -- Met: True --> Dependency: starpy -- Met: True --> Dependency: pjsua -- Met: True --> tests/channels/SIP/sip_attended_transfer_v6 --- SKIPPED --> Dependency: twisted -- Met: True --> Dependency: starpy -- Met: True --> Dependency: pjsua -- Met: True --> Dependency: ipv6 -- Met: True --> Dependency: pjsuav6 -- Met: True...

...ch plugged an extension into an audio amplifier, and that was the PA system. Now, the Asterisk server is quite far from the audio amplifier and it has no audio card. So my idea is to plug the audio card of another linux server, which is over the amplifier, into the amplifier. I've configured a pjsua with auto answer but the audio is very poor, very low volume compared to a normal audio playing (like 'aplay ttt.wav'). Is there any way to increase the volume of sip calls? Is a client side configuration, a server side or both :) Any ideas? Please, I'm going mad. Thanks in advance.

Hi, Am 16.02.2017 um 14:19 schrieb Annus Fictus: > And Microsip using PJSIP SIP stack :) Sorry (also, for off-topic), based on my latest experience with PJSIP, I'm not sure if this really is a sign of good quality. Maybe it's a problem with the implementation in Asterisk (I haven't tried PJSIP in other software), but after just five minutes of testing I found several bugs

Has anyone created any docker images I might be able to use on EC2 for load testing an asterisk platform? I started an instance this morning and was about to load sipp and other tools, and then thought surely someone must have done this already. I'd like to hammer a platform we have created with multiple EC2 images until it breaks, to test capacity. Cheers, j