Hello
I have recently added g729 on my astpp system.
I have added PCMA,PCMU,G729 in my sip profile.
But then if I don’t add the same value in my trunk then call not going.
So it doesn’t matter if I add the PCMA,PCMU,G729 in sip profile or not.
It’s alaways take from the trunk.
Is it a bug? Or it’s normal.
I want the call go by sip profile codec list.
And keep my trunk side codec empty
What can I do?
Thanks
@sokalsondha,
I’ve checked in my ASTPP and it’s working fine & sending the codec from the sip profile itself, sharing a screenshot below for your reference, thanks.
no my one not taking from sip profile. for sure.
do you mind if you share me your sip profile. so i can match it.
i just notice that my calling card access number provider also using G729 and my outbound trunk also use G729.
so when my customer call it should use the passthrough g729 without transcoding the media?
but its not working this way. is it normal?
Here is the configuration from sip profile “PCMA,PCMU,G729”, by add G729 over there you can add G729 for passthrough mode. and it’s working fine.
Thanks.
okay. i have done this.
but i had to add one thing in the sip profile
inbound-bypass-media it was set false but i set it true now its working through my calling card access number.
but when i make call i can see in the live call report not showing the call. but if i setup the profile false its showing in the live call report.
any idea?
@sokalsondha,
I’ve default configurations and without any change, live calls from switch -> Live Calls section, working fine for me.
Thanks
All you need to do is remove any codec set at trunk level and and preferred codec will be default to the source device.