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<DIV><FONT face=Arial size=2>Has anyone managed to get a stutter tone working on
the X-Lite clients when that extension has voicemail ??</FONT></DIV>
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<DIV> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
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<DIV><FONT face=Arial
size=2>--------------------------------------------------------------------------------<BR>From:
<A
href="mailto:asterisk-users-admin@lists.digium.com">asterisk-users-admin@lists.digium.com</A>
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of J Poz<BR>Posted At:
05 May 2004 22:46<BR>Posted To: Asterisk-Users<BR>Conversation: [Asterisk-Users]
Re: Simple SIP X-Lite Configuration Failing<BR>Subject: Re: [Asterisk-Users] Re:
Simple SIP X-Lite Configuration Failing</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV><FONT face=Arial size=2>
<DIV><BR>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/">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>"<BR>Correct Syntax:
callerid="Jay" <400></DIV>
<DIV> </DIV>
<DIV>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>J Poz <<A
href="mailto:jpoz0000@yahoo.com">jpoz0000@yahoo.com</A>>
wrote:<BR>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</DIV>
<DIV> </DIV>
<DIV>Girish Gopinath <<A
href="mailto:gopinath_girish@hotmail.com">gopinath_girish@hotmail.com</A>>
wrote:<BR>Hello,</DIV>
<DIV> </DIV>
<DIV>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.</DIV>
<DIV> </DIV>
<DIV>Here is my Asterisk version:<BR>Asterisk CVS-02/21/04-16:21:31 built by <A
href="mailto:root@localhost.localdomain">root@localhost.localdomain</A> on a
i686 <BR>running Linux</DIV>
<DIV> </DIV>
<DIV>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?</DIV>
<DIV> </DIV>
<DIV>Regards, Girish</DIV>
<DIV> </DIV>
<DIV>BTW, J Poz, dont use reinvite=, it does not exist, use canreinvite=
instead.</DIV>
<DIV> </DIV>
<DIV>>From: J Poz <BR>>Reply-To: <A
href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</A><BR>>To:
<A
href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</A><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 do n'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
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>></FONT></DIV></BODY></HTML>