[Asterisk-Users] Strange problem with sjphone and 1.2.1

[Chuck Bunn]

Hi,

If you do not have QOS assigned to the SIP protocol it is quite possible 
that there are packets time outs and the packets are discarded. Is it 
possible to test the network during the evening or at a time when 
traffic is at it lowest? Also try several traceroutes and see if there 
is a wide variation in return times (widely varying treceroutes could 
indicate network saturation). You are using gsm are you using 
dmtfmode=rfc2833 or something else (this must be set in the sip.conf and 
on the sip soft phone and they must match!)

Thanks

Evil Skymarshal wrote:

> Hi Chuck,
>
> 2005/12/17, Chuck Bunn <chuck.bunn at networkdoc.com 
> <mailto:chuck.bunn at networkdoc.com>>:
>
>     What are you codec and dmtfmode settings in sip.conf and in the sip
>     phone settings.
>
>
> I use gsm.
>
>     If you dmtfmode is set to 'inband' and you are using
>     anything other than ulaw or alaw codec it wont work.
>
>
> I changed the settings and tried:
> ---cut---
> exten => 2000,1,Answer()
> exten => 2000,2,Wait(1)
> exten => 2000,3,Playback(hello-world)
> exten => 2000,4,Hangup()
> ---cut---
>
> Same problem. Sometimes it works but most of the times it doesn't.
>  
>
>     Also since your
>     hear the phone sometimes you may be experiencing QOS issues on your
>     network.
>
>
> Of course it could be a QOS problem. But should I hear at least something?
>
> cu
>   ES
>
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