Re: [Asterisk-Users] Two SIP servers communicating without IAX

2004-06-23 Thread Philipp von Klitzing
Hi!

 If I call from an MX250 phone to an Asterisk phone, the conversation is
 ok, but there is a noticeable delay in the voice stream.

You will probably want to start by analysing your ethernet network (using 
ping, traceroute, ethereal etc). You might also want to eliminate a 
switch in between and directly link the two systems to see if things 
improve.

 If I call from an Asterisk phone to an MX250 phone, I can talk FROM the
 MX250 phone TO the Asterisk phone, but not the other way around. In
 other words, the Asterisk phone will hear everything that is said from
 the MX250 phone, but if I say anything on the Asterisk phone the MX250
 phone never gets it.

Look at your codec configuration (disallow=all followed byallow= 
statements). Do a SIP DEBUG on the Asterisk CLI to learn more about the 
codec negotiation before/during a call. Finally check your phones VAD/ 
silence suppression settings, make sure those are turned off.

Cheers, Philipp


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Re: [Asterisk-Users] Two SIP servers communicating without IAX

2004-06-23 Thread Alex Malinovich
On Wed, 2004-06-23 at 05:01, Philipp von Klitzing wrote:
--snip--
  If I call from an Asterisk phone to an MX250 phone, I can talk FROM the
  MX250 phone TO the Asterisk phone, but not the other way around. In
  other words, the Asterisk phone will hear everything that is said from
  the MX250 phone, but if I say anything on the Asterisk phone the MX250
  phone never gets it.
 
 Look at your codec configuration (disallow=all followed byallow= 
 statements). Do a SIP DEBUG on the Asterisk CLI to learn more about the 
 codec negotiation before/during a call. Finally check your phones VAD/ 
 silence suppression settings, make sure those are turned off.

That took care of it. I added a disallow=all followed by an allow=ulaw
and it worked fine. What's strange is that I had manually set the codec
on the phone earlier to use ulaw, but it didn't appear to want to
listen. Forcing it via sip.conf took care of the problem. Thanks a lot
for the help.

-- 
Alex Malinovich
Golden Technologies, Inc.
(219) 462-7200 x 216
http://www.golden-tech.com


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