No sound after integration webphone with vicidial

[Jitin]

VERSION: 2.14-650a
BUILD: 180111-1544
asterisk version : 11.22.0
Operating System : centos 6.9
Its a scratch install


Hello Sir
I was successfully able to integrate webphone with vicidial. I followed the following link for integration part :

https://github.com/chornyitaras/PBXWebPhone

Initially, I faced an error of srtp module missing. For that I had to re-install asterisk with srtp.

After that, when I login the agent, initial ring comes on webphone and the call is answered.
After that, there is no sound. In logs, I see that "only person in conference" is played. Even if I make a manual dial on my number, the call goes through and is connected to my number but there is no sound. Otherwise, Integration is working.

I installed g729 codec as well but still no sound.
In system settings, in webphone , I have defined codecs

Default Codecs : gsm,ulaw,g729

In phones as well, I tried by defining codec in its settings but still no sound.

Please help me out in the same.

[mflorell]

I would suggest trying VICIphone(https://viciphone.com/) and you should upgrade to Asterisk 13 if you want current WebRTC support.

[ngtechnologies]

Currently have servers running the chornyitaras webphone with no issues, however installed a new server and have the same issue........ring and answered but no sound.

Mine also are scratch installed centos6, question is:
is the upgraded Asterisk13 the base asterisk? or is it a Vici patched asterisk? if is patched, where to download for use in Centos?

VERSION: 2.14-670a
BUILD: 180424-1521
asterisk version : 11.22.0
Operating System : centos 6.9
Its a scratch install

[mflorell]

You should be able to install stock Asterisk 13.20 or higher and have it work with VICIdial.

[Jitin]

I would suggest trying VICIphone(https://viciphone.com/) and you should upgrade to Asterisk 13 if you want current WebRTC support.

How can we install this ? Can you please help me with the steps for the same.

[Jitin]

You should be able to install stock Asterisk 13.20 or higher and have it work with VICIdial.

So do I have to remove the current asterisk and re-install asterisk 13 ?

[rrb555]

You may see the instruction on the site on how to configure viciphone. I recently upgraded to Viciphone!

[Jitin]

Will viciphone work with asterisk 11.22.0 as well or upgrade is necessary.

and any suggestions on my current situation. How can I fix no sound issue in the current webphone.

[chornyi_taras]

It might be nat related issue. Check if ice is enabled. Also pls enable rtp debug.

What browser you are using?

[Jitin]

I tried using both Google chrome and Firefox

nat=yes in sip.conf

"Check if ice is enabled" : How can we check this? Didn't understand this message

[chornyi_taras]

1) please do all your test in Google Chrome
2) Make sure that you have followed all steps described on Wiki https://github.com/chornyitaras/PBXWebP ... figuration
3) add stun severer configuratin(file /etc/asterisk/rtp.conf)

Code: Select all

[general]
icesupport=yes
stunaddr=stun.l.google.com:19302

[ngtechnologies]

OK, so here is weird update

The webphone was working perfectly on OVH hosted servers, and when we installed on 1and1 server the webphone rings but no sound. So, we once again made a fresh install on OVH servers and works just fine. Perhaps the problem is in the hardware? We have seen some motherboards in the past that did not support particular OS or software.

[jfsfx]

I'm having the same exact issue. Did you find a solution for it?

[ngtechnologies]

I'm having the same exact issue. Did you find a solution for it?

1and1 support was of no help, and after 5 reinstalls we went back to OVH hosting and dropped the 1and1 hosting. Never had an issue with OVH.

[munty31]

add stun severer configuratin(file /etc/asterisk/rtp.conf)
CODE: SELECT ALL
[general]
icesupport=yes
stunaddr=stun.l.google.com:19302

This Works to solved voice issue in due to rtp.

[carpenox]

great postback, thank you im going to try this as well