[Asterisk-Users] Strange problem with sjphone and 1.2.1

[Chuck Bunn]

Hi,

What are you codec and dmtfmode settings in sip.conf and in the sip 
phone settings. If you dmtfmode is set to 'inband' and you are using 
anything other than ulaw or alaw codec it wont work. Also since your 
hear the phone sometimes you may be experiencing QOS issues on your 
network. Doe you have QOS set up on your switches in the points between 
the server running asterisk and the sip client?

Hope this helps

Evil Skymarshal wrote:

> Hi,
>
> I am using a sjphone to connect via SIP to an Asterisk 1.2.1. For 
> testing reasons I but the following in extensions.conf
>
> ---cut---
> [from-sip]
> exten => 2000,1,Answer()
> exten => 2000,2,Wait(1)
> exten => 2000,3,SayDigits(123)
> exten => 2000,4,Hangup()
> ---cut---
>
> When ever I call the 2000 asterisk -vvvvvc says:
>
> ---cut---
>     -- Executing Answer("SIP/2303-1ae1", "") in new stack
>     -- Executing Wait("SIP/2303-1ae1", "1") in new stack
>     -- Executing SayDigits("SIP/2303-1ae1", "123") in new stack
>     -- Playing 'digits/1' (language 'en')
>     -- Playing 'digits/2' (language 'en')
>     -- Playing 'digits/3' (language 'en')
>     -- Executing Hangup("SIP/2303-1ae1", "") in new stack
> ---cut---
>
> BUT I don't hear it everytime! Why? Sometimes I can hear it and most 
> time I can not. I redial 20 times and I can hear it only 3-4 times. 
> Does anybody have an idea what kind of strange problem that could be?
>
> Thanx
>   ES
>
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