Hello,
nobody use an ip phone on these mailing lists !
your call will put on queue .
I just need some people to dial sip:[EMAIL PROTECTED] to
check and debug my config .
Regards
Harry
Dear Harry,
What would you like to be debugged ?
Is nxs.yi.org your server ?
Rehan
Hello,
nobody use an ip phone on these mailing lists !
your call will put on queue .
I just need some people to dial sip:[EMAIL PROTECTED] to
check and debug my config .
Regards
Harry
hello,
Is this so difficult to call an ip phone towards
another via sip ?
Does ser and asterisk projects are dedicated to the
telephony or mail servers ?
Harry
___
Nouveau : téléphonez
A previous poster mentioned the same thing, with no response:
http://lists.digium.com/pipermail/asterisk-users/2004-
December/080161.html
Fresh asterisk 1.0.5 install on FC3, started with make samples,
nothing fancy. It's so bland, I'm surprised the list isn't full of
people having the same
Thanks to Pau (the original person to pose the question on this list), it's fixed. The firewall was getting in the way. I needed to open up UDP ports 1 to 2 for RTP traffic.
See the following for more info:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20rtp.conf
if it needs
to be on a public IP vs a private ip however.
Lyle
- Original Message -
From:
Gary Carr
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, August 24, 2004 1:29
PM
Subject: [Asterisk-Users] sip to sip
calls thru asterisk
That was coming from the register statements in the sip.conf file. Once I
removed those and restarted the sip clients everything started to work.
Thanks!
Gary
It's not clear how you are making the call.
You should be able to call directly from either phone to the other by
dialing 5011
I have a test box setup and I can make outbound
calls on the PSTN thru the diguim card, however I can not make a sip user to sip
user call by dialing the extensions. I am getting the following
error.
-- Called cisco7960 -- Got
SIP response 482 "Loop Detected" back from 208.218.14.123 == No
It's not clear how you are making the call.
You should be able to call directly from either phone to the other by
dialing 5011 or 5012, respectively, if
your context local indeed contains the those extensions, which is not
clear from your configuration excerpts.
But it seems you are calling
Hi All,
I have a weird problem. I have asterisk setup using the G729 Codec to
receive Incoming calls both from a SIP Gateway (SER and Quintum) and via
ISDN using i4l and have rules setup in extensions.conf for sending calls out
either back via the SIP Gateway or ISDN. What I want to do is
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