<DIV>Girish and All,</DIV>
<DIV> </DIV>
<DIV>I got my SIMPLE SIMPLE X-Lite to X-Lite configuration to work.</DIV>
<DIV> </DIV>
<DIV>First, I want to thank "William Ray" who helped me offline. I found him by one of his postings in the <A href="http://asterisk.xvoip.com/viewtopic.php?t=155&highlight=xlite">http://asterisk.xvoip.com/</A> forum where he helped someone else with a similar issue. He gave me a working configuration for what I was trying to do and it worked.</DIV>
<DIV> </DIV>
<DIV>However, I still needed to debug and find out what was wrong with my configuration since I know it can help someone else in the future (if they search the mailing list for the same or similar problem).</DIV>
<DIV> </DIV>
<DIV>The main problem that PREVENTED the X-Lite softphones from communicating with eachother was I had invalid syntax in the sip.conf file for the caller-id field as follows:</DIV>
<DIV> </DIV>
<DIV>Incorrect Syntax: callerid="Jay <400>"</DIV>
<DIV>Correct Syntax: callerid="Jay" <400></DIV>
<DIV><BR>It blows my mind why asterisk would spit out "Auto-congesting" and "is circuit-busy" errors since I don't know how they're related. </DIV>
<DIV> </DIV>
<DIV>That was the main problem that needed to get fixed for the configuration to work. Also, I then changed the DTMF mode to "rfc2833" from "inband" to get rid of "Unable to process inband DTMF on 2 frames" warnings. Everything is PEACHY now..</DIV>
<DIV> </DIV>
<DIV>I hope this proves useful to others in the future. I invested over 30 hours on this problem so hopefully it can be avoided by someone else.</DIV>
<DIV> </DIV>
<DIV>J..........</DIV>
<DIV> </DIV>
<DIV><BR><B><I>J Poz <jpoz0000@yahoo.com></I></B> wrote:</DIV>
<BLOCKQUOTE class=replbq style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid">
<DIV>
<DIV>Girish,</DIV>
<DIV> </DIV>
<DIV>Thanks for replying and trying to work my "simple configuration". Nobody on the list has replied with any help and I still have the problem.</DIV>
<DIV> </DIV>
<DIV>I've invested well over 20 hours on this problem and still don't have a solution (I have everything else within Asterisk working including IVR menus, X100 interfaces, etc). However, I am not able to get a simple Softphone to Softphone configuration to work.</DIV>
<DIV> </DIV>
<DIV>Can anyone on the LIST help us<BR><BR><B><I>Girish Gopinath <gopinath_girish@hotmail.com></I></B> wrote:</DIV>
<BLOCKQUOTE class=replbq style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid">Hello,<BR><BR>Replying to the mail which was posted 3 days back. I tested the <BR>configuration here with SJphones, and got the same error: "circuit-busy". I <BR>tried with sip debug turned on, and found that asterisk receives a CANCEL <BR>request from the user agent immediately after it receives INVITE. When i <BR>first saw this mail, i thought it was a simple config issue, but even after <BR>trying for more than 2 hours, i am not able to figure out why it is <BR>happening. I tried changing the sip.conf entries with the minimum required <BR>values, but no success. I started evaluating Asterisk a few months ago, i <BR>also tried with such simple configurations and did not have issues like <BR>this.<BR><BR>Here is my Asterisk version:<BR>Asterisk CVS-02/21/04-16:21:31 built by root@localhost.localdomain on a i686 <BR>running Linux<BR><BR>I am really curious if you were able to solve
the problem. If so, what was <BR>the reason behind that weird behaviour and how did you solve it? If not, can <BR>anyone please tell what is going wrong?<BR><BR>Regards, Girish<BR><BR>BTW, J Poz, dont use reinvite=, it does not exist, use canreinvite= instead.<BR><BR>>From: J Poz <JPOZ0000@YAHOO.COM><BR>>Reply-To: asterisk-users@lists.digium.com<BR>>To: asterisk-users@lists.digium.com<BR>>Subject: Re: [Asterisk-Users] Re: Simple SIP X-Lite Configuration Failing<BR>>Date: Sun, 2 May 2004 16:41:52 -0700 (PDT)<BR>><BR>>Sorry for any confusion.........But in my latest error, instead of calling <BR>>my clients "jay" and "jtest", I'm calling them "400" and "410".. Everything <BR>>else is still the same and it's same problem.<BR>><BR>>My guess is that I've set a parameter incorrectly and therefore Asterisk <BR>>thinks there's only one client so any calls I try to make between the two <BR>>fail since it thinks the other client is busy. But I don't
understand <BR>>enough to interpret the error message. I thought the SIP part would be the <BR>>easy part - I already have the FXO and FXS interfaces working.<BR>><BR>>Again, thanks for anyone who can help me since I am at a loss!<BR>><BR>>J Poz <JPOZ0000@YAHOO.COM>wrote:<BR>>Can anyone help. I've changed the extensions.conf file as follows:<BR>><BR>>extensions.conf<BR>>[sip] ; context for X-Lite Clients<BR>>exten =>11,1,Dial(SIP/jay,20,tr)<BR>>exten =>22,1,Dial(SIP/jtest,20,tr)<BR>><BR>>I'm still getting the Auto-congesting error (and circuit-busy). Does anyone <BR>>know what is causing this in such a simple configuration?<BR>><BR>><BR>>localhost*CLI><BR>> -- Executing Dial("SIP/400-c3de", "SIP/410|20|tr") in new stack<BR>> -- Called 410<BR>>May 2 19:15:56 NOTICE[1133742896]: chan_sip.c:1021 auto_congest: <BR>>Auto-congesting SIP/410-a4a1<BR>> -- SIP/410-a4a1 is circuit-busy<BR>> == Everyone is busy at
this time<BR>><BR><BR>_________________________________________________________________<BR>Send flowers in 24 hours! <BR>http://www.fabmall.com/affiliatehtml/redir/nl7.asp At MSN Shopping.<BR></BLOCKQUOTE></DIV>
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