Hi,

Rich I stand corrected you are absolutely right - see http://www.voip-info.org/wiki-Asterisk+config+sip.conf

The following appears on the page:


   Please note

   * Asterisk does not yet support SIP over TCP. It only supports SIP
     <http://www.voip-info.org/wiki/view/SIP> over UDP.
   * For Grandstream <http://www.voip-info.org/wiki/view/Grandstream>
     phones: set *dtmfmode=info*
   * Asterisk uses the incoming RTP
     <http://www.voip-info.org/wiki/view/RTP> Stream as a timing source
     for sending its outgoing Stream. If the incoming stream is
     interrupted due to silence suppression then musiconhold will be
     choppy. So in conclusion, you cannot use silence suppression.
     *Make sure ALL SIP phones have disabled silence suppression.*
     There is a solution for the silence suppression problem, see bug
     5374 <http://bugs.digium.com/view.php?id=5374> for details.

Thanks


Rich Adamson wrote:

I don't believe asterisk has any sip "tcp" support. Its all "udp".

------------------------
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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