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Im facing completely choppy sound. The wireshark trace shows, that there are a lot of codec changes without any trigger (means no options or reinvite or any other package).Please note: PJSIP is a free and open source multimedia communication libr..
I am using the Asterisk REST API in order to establish a call to an endpoint and to send over a remote file (HTTP). The issue is that I am experiencing an audio quality issue. I have tried encoding the file differently, but everytime Asterisk is cutt..
I have a scenario that I am failing to implement using the Queue app, but which I had thought would be commonplace…First of all, I would like to recommend this short cookbook that offers recipes for tackling dialplan fundamentals, making and controll..
I have made a success installation from source of Asterisk 14.4.0 on Debian Jessie (8.7). And I am running the Asterisk server, with several extensions and dialplans, all working well.However I am struggling to get app_jack to run.In menuselect I ..
I think many people here connect their mobile phone with Asterisk. Can someone suggest me an App that allow me to add a VoIP-number in the contact?With my old Samsung Galaxy S2 it was possible direct in system without additional Apps, but with the Gal..
Ive MP-114 that is working configured and working OK with my Asterisk but I just obtained MP-112 (2xFXS) and I can register OK with asterisk but I can only dial 3-digit extension.Anything longer than 3-digits is cut off, example I dial extension 1000:[..
Iam lookong for an Softphone for iPhor oder Android smartphone using togehter with an headset. I tried Zoiper and CSipSimple but quality was bad compared to an desktop SIP phone.Is there an better softphone?Or are there softphone solutions for PC desk..
I`ve recently upgraded a server from 1.8 to Asterisk 13. While everything is under control, I have one issue with the way CDRs are kept for queues. And I ..
In France, years ago, there was some discussions about a new regulation forcing some providers to not charge anything to callers while those are waiting for a call center agent to become available. Once caller and agent are on call with each other, nomi..
When I look at the lastest UniMRCP manual, they only mention as high as Asterisk 13.Does anybody know if I need to do anything to allow it to work on Asterisk 14 and, if so, wha..