About Codec

[LeeFrederick]

1. Brekeke Product Name and Version:
Version 3.2.4.3 Standard
Version 2.4.7.3 Standard

2. Java version:6

3. OS type and the version:win2003

4. UA (phone), gateway or other hardware/software involved:yealink

5. Your problem:
I have two sip servers, Server A's version is 2.4.7.3,Server B's version is 3.2.4.3.

I add the dailplan in Server A:

Matching Patterns

$request = ^INVITE To = sip:(1[0-9]{10})@ $geturi(From) = (.+)

Deploy Patterns

From = "%1"<%2> To = sip:%1@192.168.66.30:5060

All phones register to Server A.And 192.168.66.30 is server B's ip address.

And I add the dailplan in Server B:

Matching Patterns

$request = ^INVITE To = sip:(139[0-9]{8})@ $geturi(From) = (.+)

Deploy Patterns

$auth = false &net.rtp.audio.payloadtype = 18 From = "%1"<%2> To = sip:%1@192.168.2.10:2080

192.168.2.10 is a gateway of E1's ip address.And Server B' B2B-UA mode is on.

Most of time the call can reach the gateway and dail out.But about 10% calls can not call out.At this time,I view the Active Sessions and see the session's detail,I found under these circumstances the payload's values is empty,and in the log,the failure call's error code is 500 or 617 or 603.How do I change the setting?

Thanks you!

[janP]

> &net.rtp.audio.payloadtype = 18

Do you know what it means?
The payload type 18 means G729.

Both UA support G729 audio codec?

[janP]

Which gateway product are you using??

If the caller (UAC) or callee (UAS) does't support G729, a call will not be established.

Use WireShark to check who sent an error response such as 500, 617 and 603.

[LeeFrederick]

[janP]

Which gateway product is it?

[LeeFrederick]

[LeeFrederick]

[janP]

> The codec error message is sent from gateway.

It will be the gateway's issue.
Contact the gateway's manufacturer.

[LeeFrederick]