I can call but we cannot hear each other

Asterisk Asterisk Support
1

hello, i have a problem with asterisk concerning the external calls.
i can call to any landline and stablish contact and talk without problem , but when try to call a mobile phone we stablish contact but we cannot hear eachother.
any guesses?

This is what we have found out so far:
When we call to a landline we can find these lines in asterisk:

-- Executing [0xxxxxxxxx@INTERNO:1] Dial("SIP/2202-00000000", "SIP/xxxxxxxxx@-------,30,tr") in new stack

== Using SIP RTP CoS mark 5
– Called SIP/xxxxxxxxx@-------
> 0x74204c98 – Strict RTP learning after remote address set to: (IP and port)
– SIP/sarenet-00000001 is making progress passing it to SIP/2202-00000000
– SIP/sarenet-00000001 answered SIP/2202-00000000
– Channel SIP/sarenet-00000001 joined ‘simple_bridge’ basic-bridge <58fd0e72-31bd-4cc6-a616-8172b4ed6f88>
– Channel SIP/2202-00000000 joined ‘simple_bridge’ basic-bridge <58fd0e72-31bd-4cc6-a616-8172b4ed6f88>
> 0x73f04808 – Strict RTP switching to RTP target address (IP-2 and port) as source
> 0x74204c98 – Strict RTP switching to RTP target address (IP and port) as source
> 0x73f04808 – Strict RTP learning complete - Locking on source address (IP-2 and port)
> 0x74204c98 – Strict RTP learning complete - Locking on source address (IP and port)
– Channel SIP/sarenet-00000001 left ‘simple_bridge’ basic-bridge <58fd0e72-31bd-4cc6-a616-8172b4ed6f88>
– Channel SIP/2202-00000000 left ‘simple_bridge’ basic-bridge <58fd0e72-31bd-4cc6-a616-8172b4ed6f88>
== Spawn extension (INTERNO, 0xxxxxxxxx, 1) exited non-zero on ‘SIP/2202-00000000’

And these are the ones we get when we use a mobile phone:

-- Executing [0xxxxxxxxx@INTERNO:1] Dial("SIP/2202-00000002", "SIP/xxxxxxxxx@--------,30,tr") in new stack

== Using SIP RTP CoS mark 5
– Called SIP/xxxxxxxxx@---------
> 0x74405dd8 – Strict RTP learning after remote address set to: (IP-1 and Port)
– SIP/sarenet-00000003 is ringing
– SIP/sarenet-00000003 is making progress passing it to SIP/2202-00000002
> 0x74405dd8 – Strict RTP learning after remote address set to: (IP-1 and Port)
– SIP/sarenet-00000003 answered SIP/2202-00000002
– Channel SIP/sarenet-00000003 joined ‘simple_bridge’ basic-bridge
– Channel SIP/2202-00000002 joined ‘simple_bridge’ basic-bridge
> 0x73f02ba0 – Strict RTP switching to RTP target address (IP-2 and Port) as source
> 0x73f02ba0 – Strict RTP learning complete - Locking on source address (IP-1 and Port)
– Channel SIP/2202-00000002 left ‘simple_bridge’ basic-bridge
– Channel SIP/sarenet-00000003 left ‘simple_bridge’ basic-bridge
== Spawn extension (INTERNO, 0xxxxxxxxx, 1) exited non-zero on ‘SIP/2202-00000002’

the text in bold is where we think the problem is but we where not able to find out anything

2

If Asterisk doesn’t receive media then it can’t forward it, and it can’t lock on to the source of media. Based on the log messages that doesn’t appear to be happening so media isn’t reaching Asterisk. You’d need to do a packet capture to see if it is even getting to the machine. You should also check the SDP to make sure the correct IP address is in it and that the firewall is open.

3

How do IP-1 and IP-2 relate to particular components of your system?

Is “------” the same as “--------”?

4

IP-1 is the IP of the telephone we are using and IP-2 is the IP where we get the service to be able to have outbound and inbound calls.

And “------” and “--------” are the same.