AMI bridging two channels but No audio coming

[urmi.l]

Hello,

I am facing one issue with asterisk 20. I am using AMI to originate calls. I am successfully able to do calls between two SIP channels but I am not able to hear voice. The originate command first calls one SIP extension and on answer of that, dials another extension and then it creates the Bridge. But the voice is not coming.

If I dial the sip to sip calls directly from softphones then the voice is perfectly coming but with the AMI originate, there is no audio.

[david551]

Is there a translation path to the SLIN codec? If not, you will need to explicitly specify the codec.

[urmi.l]

Thanks david for quick help.

I tried with allow=slin in the sip.conf but by doing that the originate is failed and calls are not getting generated.

my sip.conf

[8002]
username=8002
secret=123
accountcode=8002
callerid=“8002” <8002>
mailbox=8002
;context=default
context=ren-manualcall
;context=ren-sip-to-sip
type=friend
host=dynamic
disallow=all
;allow=slin
allow=all

[8001]
username=8001
secret=123
accountcode=8001
callerid=“8001” <8001>
mailbox=8001
;context=default
;context=ren-sip-to-sip
context=ren-manualcall
type=friend
host=dynamic
port=5060
disallow=all
;allow=slin
allow=all

My extensions.conf
[ren-sip-to-sip]
exten => _X.,1,NoOp(=================== Calling SIP to SIP ======================)
exten => _X.,n,NoOp(=================== Variable1: ${CLIENT_NUMBER} ========================)
exten => _X.,n,Dial(SIP/${CLIENT_NUMBER},tTorR)
exten => _X.,n,Hangup()

I cant find the RTP flowing in the RTP debug on.

Below is my CLI Logs :

== Manager ‘renAMI’ logged on from x.x.x.x
== Using SIP RTP CoS mark 5
– Called 8002
– SIP/8002-0000000d is ringing
> 0x7f7c98008250 – Strict RTP learning after remote address set to: 192.168.0.193:61030
– SIP/8002-0000000d answered
– Executing [_X.@ren-sip-to-sip:1] NoOp(“SIP/8002-0000000d”, “=================== Calling SIP to SIP ======================”) in new stack
– Executing [_X.@ren-sip-to-sip:2] NoOp(“SIP/8002-0000000d”, “=================== Variable1: 8001 ========================”) in new stack
– Executing [_X.@ren-sip-to-sip:3] Dial(“SIP/8002-0000000d”, “SIP/8001,tTorR”) in new stack
== Manager ‘renAMI’ logged off from x.x.x.x
== Using SIP RTP CoS mark 5
– Called SIP/8001
– SIP/8001-0000000e is ringing
> 0x7f7c94011ba0 – Strict RTP learning after remote address set to: 192.168.0.192:58930
– SIP/8001-0000000e answered SIP/8002-0000000d
– Channel SIP/8001-0000000e joined ‘simple_bridge’ basic-bridge
– Channel SIP/8002-0000000d joined ‘simple_bridge’ basic-bridge
localhost*CLI>

[david551]

slin is generally only understood internally, not over the wire. Where you would set the codec is in the Codecs option of originate. You need to set it to what the phone use, or what Asterisk can transcode to that. In particular, if you are using G.729 pass through, you need to set it G.729 if you don’t have a, properly licensed, G.729 codec installed.