Pjsip Codec Priority
Hi, following this discussion on FreePBX forums:It appears that Yealink phones are the culprit.I captured pcap files that I believe will help understand the problem. The pcap is for a call from a Yealink T46g phone to an asterisk 13.5 box, the call is to the Echo service (.43).The call should go like this:- First, Asterisk plays a message that explains how the echo service works. This message lasts for about 25 seconds;- Then the echo test begins and everything I say is echoed back to me.Looking at the pcap, it happened like this:- Asterisk on SDP offered codecs ALAW and G722;- Yealink T46G offered G722 and ALAW;- Asterisk started sendng G711 but then 1 second after, it switched to G722 to play the message. I just tested this new firmware on a T46g and can attest it fixes the codec issue described in this thread when using PJSIP on Asterisk.Kudos for Yealink for fixing it! Although the issue persisted for many months, Yealink is the first brand affected by this bug that has actually issued a fix.Now. Can this fix ship on a 'public' firmware release anytime soon?( 09:24 PM)YealinkLucia Wrote: Hi Sir,About the issue, we have solve it and modify it in the new firmware, you can download the firmware in the FTP link below, and it will be invalid in 2017.03.05:Note: The link include the firmware of T46、T48、T49.BRLucia. ( 09:24 PM)Lucia Wrote: Hi Sir,About the issue, we have solve it and modify it in the new firmware, you can download the firmware in the FTP link below, and it will be invalid in 2017.03.05:Note: The link include the firmware of T46、T48、T49.BRLuciaLucia, I have begun to see this issue again and have been using firmware 35.81.0.20 on a T48G.
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Pjsip Codec Priority Windows 10
Pjsip Asterisk
There have been some changes in my Asterisk installation that may trigger this issue, though I do not see this issue on any of my other phones/devices/softphones. Can you elaborate on the 'fix' that was provided? What did it do to improve the codec handling? I'd like to check in on the Yealink side first since it isn't affecting other phones.
Pjsip Nat
How to configure SIP Trunking for Asterisk IP PBX based systemsOur service is 100% compatible with Asterisk using either standard SIP registration or IP authentication where SIP trunks are configured as such. Asterisk is the base software behind many open-source PBX distributions, including FreePBX, Trixbox and Elastix, and is also the enabler behind many other ITSPs and commercial PABX manufacturers.This guide is aimed at Asterisk's SIP stack via the sip.conf file, it does not deal with real-time configuration via a back-end database, however, the principles are the same and the appropriate options should be transposed as such.