[Asterisk-Users] No path to translatefrom SIP/615-25c8(256)toSIP/601-27b6(4)

[Joshua Colp]

That made no sense to me. Please try again. If you mean why did it not go to
the next line when it tried to bridge it's because you can't switch codecs
in the middle of a call.

- Joshua Colp. 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ronald
Wiplinger
Sent: Saturday, June 11, 2005 2:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] No path to translatefrom
SIP/615-25c8(256)toSIP/601-27b6(4)

Joshua Colp wrote:

> Codecs are negotiated between asterisk and the device, not device to 
>device... So since you specify G729, one side negotiated to G729 first...
>Then when you dialed the other device, that one negotiated at ULAW... 
>And then when they attempted to be bridged together - voila, failure.
>  
>

Exactly here I see the problem!
Why did one phone could negotiate to use g729, although it is not on
Asterisk???? It should therefore go to the next line, shouldn't it?


bye

Ronald

>- Joshua Colp.
>
>-----Original Message-----
>From: asterisk-users-bounces at lists.digium.com
>[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ronald
>Wiplinger
>Sent: Saturday, June 11, 2005 1:20 PM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [Asterisk-Users] No path to translate from
>SIP/615-25c8(256)toSIP/601-27b6(4)
>
>Joshua Colp wrote:
>
>  
>
>>One side is using G729, the other is using ULAW. Asterisk is having to 
>>convert between the two and can not, probably because you do not have 
>>the
>>G729 codec with the proper license ($10/channel from Digium).
>>
>> 
>>
>>    
>>
>
>You are right! I removed g729, but I still wonder, why it did not go to the
>next line!!!!
>
>[snom-190]
>...
>disallow=all
>allow=g729
>allow=ulaw                    
>allow=alaw
>
>
>bye
>
>Ronald
>
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>  
>


-- 
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