Hi,
Something else I should mention. Sip uses UDP and TCP packets. TCP
packets are used if there is congestion on the network. I am unclear
about what mechanism causes sip to switch between UDP and TCP but I
believe it is controllable - I believe It would be easier to use QOS
though. If UDP is used that packets could time out and you would never
know it since UDP is dumb and has no packet loss recovery mechanism.
What is the topology of your network. Is the Asterisk box and the client
on the same backbone and switch?
Thanks
Evil Skymarshal wrote:
Hi Chuck,
2005/12/17, Chuck Bunn <[EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]>>:
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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