03-29-2010 07:29 AM - edited 03-15-2019 10:01 PM
Hello everybody,
I enabled Built In Bridge because we use Nice Recording, recording work fine but we can not make conference, thanks to help me to fix this problem,
regards,Mohamed
03-29-2010 07:51 AM
You don't use the built-in bridge for ad-hoc or meet-me, that config is not related.
Check you have a CFB available in the phones MRGL and, if the CUCM SW CFB, that you're using G711
HTH
java
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03-29-2010 07:54 AM
Java is correct. The built-in bridge functionality is used for the Barge feature.
Hailey
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03-29-2010 08:04 AM
in advance thank you for you answer,
but when just i disable the built-in bridge the conference work fine ?!!!
regards,
Mohamed
03-29-2010 08:26 AM
Exactly what happens when you try a conference???
HTH
java
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03-30-2010 02:34 PM
I had this problem the other day. The issue is a recorded line in combination with the G.722 codec. In my situation, when a station to station call was placed, the phones negotiated to use the G.722 codec. You can verify this by calling another voip phone and press the ? button twice. Normally, if a phone is using G.722 and is added to a conference, it will renegotiate to G.711 or some other supported codec. However, on a recorded line this renegotiation doesn't happen. Therefore, the call gets dropped because of the unsupported codec.
To resolve this problem you need to go into System -> Service Parameters and select the Cisco CallManager Service. Under "Clusterwide Parameters (System - Location and Region)" set the value for "G.722 Codec Enabled" to "Enabled for All Devices Except Recording Enabled Devices"
Hope this helps
Robert
03-31-2010 01:21 AM
Good tip (+5)
01-28-2011 01:11 AM
I have exactly same issue.
I think that would be a good suggestion.
I will give it a try, and I will update you.
Regards,
Aziz
01-28-2011 09:04 AM
This is why you ran into that problem
The BiB (built in bridge) supports other codecs too, but be aware that it will do codec locking. This means if phone A is on a call with phone B using g711, and the region between phone B (who has recording enabled with the BiB) and the SIP trunk to the recording server is g729,recording will need to invoke a transcoder since the phone will only send out g711 which matches the original call codec. If the recorded phone was using g729 for the RTP stream between phones, the BiB can only send g729 out.
HTH
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