Loss of audio (inbound) when call on hold is resumed

Hi,
The following issue is intermittent.

When a call is made and during the call, the call is placed on hold, Upon resuming the call no audio is heard on my end but the other end can hear me.

Thanks for any help.

So this is a really old thread, but I’m experiencing this on one of my extensions. When the call is resumed I get no incoming audio. Outgoing still works.

This particular phone is over a VPN (yealink T46, openvpn built in). Calling and receiving calls works, so not sure what’s going on.

So narrowed this down a bit further, its only on incoming calls for some reason. Outgoing calls can hold just fine.

When an endpoint puts a call on hold, or when it picks up a hold call, it sends a SIP reINVITE to the PBX. One (or both) of these invites has an SDP payload that is misdirecting the media. You will have to do SIP traces to find out what’s going on, buy my money is on buggy firmware.

Thanks Igaetz.

Setting Media Use Received Transport on the extension to Yes, the audio works but then the call is not returned on the callers side. They stay on hold forever.

Thanks, that info from stack overflow suggested the parties don’t agree on the codec again (for some reason). Changing from iLBC to G722 or G711 else fixed the issue.

As per my other message, setting Media Use Received Transport did work, but then the other party wasn’t resumed.

Any ideas on a workaround?

Sounds like a bug with asterisk and your codec.

This is strange:
[[email protected] ~]# cat /etc/schmooze/pbx-version
12.7.4-1804-1.sng7
[[email protected] ~]# asterisk -r
Asterisk 13.19.1, Copyright © 1999 - 2014, Digium, Inc. and others.

No it’s not.

Oops my mistake, you’re right - i glazed over the documentation. Asterisk seems to be on the latest of the 13 branch, I dare not try and update to 15 and all the headaches that will ensue.

Easy to change and change back:

https://wiki.freepbx.org/display/PPS/Changing+Major+Asterisk+Versions+on+the+Fly