Hi Chuck,2005/12/17, Chuck Bunn [EMAIL PROTECTED]:
If you do not have QOS assigned to the SIP protocol it is quite possiblethat there are packets time outs and the packets are discarded. Is itpossible to test the network during the evening or at a time whentraffic is at it lowest?
It took me some
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
Hi Chuck,2005/12/17, Chuck Bunn [EMAIL PROTECTED]:
What are you codec and dmtfmode settings in sip.conf and in the sipphone settings.I use gsm.
If you dmtfmode is set to 'inband' and you are usinganything other than ulaw or alaw codec it wont work.I changed the settings and tried:---cut---exten =
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
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
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
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.
