<html><div style='background-color:'><P>Hey asteriskers!</P>
<P><BR>I know that may look weird, but it's happening: </P>
<P>We have an * server running in a wireless(cellular) operator for IVR services, we bill them per minute, but there is a remarkable difference between our CDR records and their billing system.</P>
<P> * server have a Sangoma, and 3 PRI_ISDN E1 lines are connected to it, * is working perfectly with no problems at all. We requested a re-analysis and they confirmed their records are correct and error free!!</P>
<P> At last we agreed on setting a DB that collects call information at their postgres DB that I post in, BUT I'm having a problem detecting hangups, as I tried DeadAgi on h extension, but the problem is the channels are destroyed and I can't do anything.</P>
<P> Anyone have any suggestions?<BR><BR></P>
<P>Truely/</P>
<P>Joe</P>
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From: <I>Jean-Yves Avenard <jyavenard@gmail.com></I><BR>Reply-To: <I>Asterisk Users Mailing List - Non-Commercial Discussion<asterisk-users@lists.digium.com></I><BR>To: <I>asterisk-users@lists.digium.com</I><BR>Subject: <I>[Asterisk-Users] Uniden UIP200 and Asterisk v1.2.4: problem notregistering</I><BR>Date: <I>Tue, 7 Feb 2006 02:45:15 +1100</I><BR><BR>Hello<BR><BR>We recently moved to Asterisk 1.2.4 (from 1.0.x) and our 10 Uniden UIP200 have stopped working ever since.<BR><BR>We can make a call with the UIP200 to any other extensions, but it can not receive a call. In fact the UIP200 always appears offline: <BR><BR>It does show up in asterisk a few seconds after the UIP200 reboot:<BR>-- Saved useragent "Uniden SIP Phone p2 Ver BS4.70" for peer uip200<BR><BR>but after about 5s I will get something like:<BR>UIP200 is now unreachable. <BR><BR>htpc*CLI> sip show
peers<BR>Name/username Host Dyn Nat ACL Port Status <BR>uip200/uip200 <A href="http://192.168.10.104/">192.168.10.104</A> D 5061 UNREACHABLE <BR><BR><BR>I have tried the latest firmware (v4.70) and the previous one we've been running for over 18 months (v4.59) without any luck<BR><BR>Here is the sip.conf I've created on a test server where Asterisk is using the port 5061 , same for the UIP200 using port 5061. There is no NAT, the UIP200 is on the same subnet as the asterisk server: <BR><BR>(I'm trying to isolate the issue without affecting
our main asterisk server)<BR>[uip200]<BR>type=friend<BR>port=5061<BR>secret=uip200 ; password for registration<BR>nat=never ; phone may be behing nat <BR>host=dynamic<BR>reinvite=no<BR>canreinvite=no<BR>qualify=3000 ; send udp every 2 seconds, to keep nat open<BR>callerid="Jean-Yves <200>"<BR>dtmfmode=rfc2833 ; DTMF mode <BR>context =
jya-in ; Default context for incoming calls<BR>disallow=all<BR>allow=ulaw<BR>allow=alaw<BR>allow=g729<BR><BR>If I unable: sip debug ip <A href="http://192.168.10.104/">192.168.10.104</A> (the UIP200 IP address), I get every 3-4 seconds on the console: <BR>---<BR>Retransmitting #2 (no NAT) to <A href="http://192.168.10.104:5061/">192.168.10.104:5061</A>:<BR>OPTIONS sip:uip200@192.168.10.104:5061 SIP/2.0<BR>Via: SIP/2.0/UDP <A href="http://192.168.10.11:5061/">192.168.10.11:5061 </A>;branch=z9hG4bK08dcd4a8;rport<BR>From: "asterisk" <sip:asterisk@192.168.10.11:5061>;tag=as67a81892<BR>To: <sip:uip200@192.168.10.104:5061><BR>Contact: <sip:asterisk@192.168.10.11:5061><BR>Call-ID: <A
href="mailto:008a5e9e721f01da0115c67b4c7cbdb1@192.168.10.11">008a5e9e721f01da0115c67b4c7cbdb1@192.168.10.11</A><BR>CSeq: 102 OPTIONS<BR>User-Agent: Asterisk PBX<BR>Max-Forwards: 70<BR>Date: Mon, 06 Feb 2006 15:40:37 GMT <BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<BR>Content-Length: 0<BR><BR>Any help would be greatly appreciated.<BR><BR>Thank you in advance.<BR>Regards<BR>JY<BR><BR><BR><BR>
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