dosh
1
Hello,
I have no issues with asterisk and now I have one more difficult.
what observed:
- One way audio is for outgoing and incoming calls. External calls.
- missing stream is always outgoing from my internal phones
- I use trunk
- I checked IVR calling by GSM and I was able to hear it
- I checked conference bridge - one caller internal one external and then everthing was OK, both directions stream were hearable
- used codecs in tests g711ulaw and g711alaw
- til now I only tested on phone cisco 7911 sip load
Please advice
dosh
jcolp
2
What is the configuration? Are you using chan_sip or chan_pjsip? Are you behind NAT? If behind NAT have you configured things for that?
dosh
3
I use asterisk 1.8.26.1 and chan_sip.
It is behind NAT, but conference works fine. I suppose I configured for NAT. What to check?
I just realized:
- with issue going to transfer but not finalizing so going back to connection ip phone - gsm and audio is OK in both directions
- answering a call and pressing HOLD and UNHOLD - after than call is OK with both directions audio
Don’t think you’ll get much help if you are using version 1.8 which went EOL in 2015. Even chan_sip is deprecated in latest versions of Asterisk.
system
Closed
5
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