<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 TRANSITIONAL//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV="Content-Type" CONTENT="text/html; CHARSET=UTF-8">
<META NAME="GENERATOR" CONTENT="GtkHTML/3.0.10">
</HEAD>
<BODY>
Hi,<BR>
<BR>
I tried to make a call to extension 2001 with the setting 2000@provider.com (Detailed:<BR>
exten => _7.,2,Dial(SIP/2000@provider.com/${EXTEN:1}) <BR>
which does not work at all - i always get the failure message: No such host provider.com/2001 (the number i dialed) - why ??<BR>
when i try the same with a peer agent (exten => _7.,2,Dial(SIP/provider_out/${EXTEN:1}) - i always get the failure message <BR>
WARNING[-178521168]: chan_sip.c:680 retrans_pkt: Maximum retries exceeded on call 0ac244f146d85c325ca3357b4db1d05a@192.168.0.7 for seqno 102 (Critical Request)<BR>
<BR>
What am i missing ??<BR>
I am running out if ideas !.<BR>
<BR>
Johannes<BR>
<BR>
Am Fr, den 10.09.2004 schrieb Begumisa Gerald M um 12:04:
<BLOCKQUOTE TYPE=CITE>
<PRE><FONT COLOR="#737373"><I> On Thu, 9 Sep 2004, Karl Brose wrote:
> In order to dial out to a sip provider, you need to configure that
> provider in your sip.conf file as a peer with your proper username
> and secret, etc.
Cool! Just found that in the handbook too a second or two ago :-)
Thanks for taking time to answer this.
Three Cheers!
Gerald
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com</FONT>
<A HREF="http://lists.digium.com/mailman/listinfo/asterisk-users"><U>http://lists.digium.com/mailman/listinfo/asterisk-users</U></A>
<FONT COLOR="#737373">To UNSUBSCRIBE or update options visit:
</FONT><A HREF="http://lists.digium.com/mailman/listinfo/asterisk-users"><U>http://lists.digium.com/mailman/listinfo/asterisk-users</U></I></A></PRE>
</BLOCKQUOTE>
</BODY>
</HTML>