-users] one way rtp stream (Sent alwax to 127.0.0.1)
On 12/29/06 06:04 Hans-Jürgen Brand said the following:
Found problem
xlite client Version 3 Build 34025 send wrong rtp port to asterisk. But
I don't know how to change this at xlite
have you tried nat=yes in sip.conf
Asterisk version 1.2.14
I use snom190 and xliteV3 as sip phones.
asterisk send the rtp stream never to the xlite softphone.
Any hits for me?
*CLI rtp debug
RTP Debugging Enabled
-- Executing Dial(SIP/xlite-007918f0, SIP/snom) in new stack
-- Called snom
-- SIP/snom-00797110 is
;rinstance=45385da6efafa3ea;tag=7b512144
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
Original-Nachricht
Datum: Thu, 28 Dec 2006 22:30:24 +0100
Von: Hans-Jürgen Brand [EMAIL
I'm looking for some doc's what is new in 1.4
in deep for the comand stun debug
neptun:/usr/src/asterik_pgk/asterisk-1.4.0/main # asterisk -r
Asterisk 1.4.0, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer [EMAIL PROTECTED]
Asterisk comes with ABSOLUTELY NO
First I nor sure if you can use the 7970 with SIP. LoadID looks for me the
bootimage does not match with the applicationimage. Mabe you have to erase the
flash (I'm not sure)
here is my config for Skinny Channel
venus:/srv/tftpboot # cat XMLDefault.cnf.xml
Default
callManagerGroup
members
just have a look at http://sourceforge.net/projects/oscsta
Hi,
Can anybody tell me that is their CSTA support for asterisk
sanchal
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