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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