<p dir="ltr">So, the problem is that calls made from a direct connected user, falls to voicemail? Even if the other user is online?</p>
<p dir="ltr">All the users are on the same asterisk server? Or using a trunk outside?</p>
<p dir="ltr">As a test, tried to register to the asterisk server directly and test the call?<br></p>
<p dir="ltr">That's why I was asking to elaborate, and show a bit more about the call flow behavior... A small text diagram and desired behavior would be useful<br><br></p>
<div class="gmail_quote">El mar 31, 2014 8:13 AM, "Slava Bendersky" <<a href="mailto:volga629@networklab.ca">volga629@networklab.ca</a>> escribió:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div><div style="font-size:12pt;font-family:lucida console,sans-serif"><div>Hello Olle,<br></div><div>Overlap is disabled on asterisk. I more wonder about this message.<br></div><div><br></div><div>Mar 31 08:40:20 dsm01 /usr/sbin/kamailio[6101]: WARNING: sanity [sanity.c:833]: check_parse_uris(): sanity_check(): check_parse_uris(): failed to parse From uri</div>
<div><br></div><div>Because from direct connected network, call failing to voicemail.</div><div> <br></div><div>Slva.<br></div><hr><div style="font-size:12pt;font-style:normal;font-family:Helvetica,Arial,sans-serif;text-decoration:none;font-weight:normal">
<b>From: </b>"Olle E. Johansson" <<a href="mailto:oej@edvina.net" target="_blank">oej@edvina.net</a>><br><b>To: </b>"Kamailio (SER) - Users Mailing List" <<a href="mailto:sr-users@lists.sip-router.org" target="_blank">sr-users@lists.sip-router.org</a>><br>
<b>Sent: </b>Monday, March 31, 2014 3:33:11 AM<br><b>Subject: </b>Re: [SR-Users] message 484<br><div><br></div><div>Hi!</div>I guess this is a poorly configured Asterisk server that has "Allowoverlap" enabled.<div>
A 484 is used for overlap dialing. The server says "I need more digits to complete this call".</div><div><br></div><div>/O</div><div><br><div><div>On 31 Mar 2014, at 02:30, Pedro Niño <<a href="mailto:nino.pedro@gmail.com" target="_blank">nino.pedro@gmail.com</a>> wrote:</div>
<br><blockquote><p dir="ltr">I think this is the correct behavior, as asterisk server is complaining about the address/request not containing all the necesary data to process the message</p><p dir="ltr">Can you please elaborate with a bit more of detail? Also can use tools like sngrep, tcpdump (or wireshark) to have a better view of the complete call flow.</p>
<p dir="ltr">Maybe that way we can help.</p><div class="gmail_quote">El mar 29, 2014 1:59 AM, "Slava Bendersky" <<a href="mailto:volga629@networklab.ca" target="_blank">volga629@networklab.ca</a>> escribió:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div style="font-size:12pt;font-family:lucida console,sans-serif"><div>Hello Everyone,<br></div><div>How to correct message 484 <br>
</div><div>Is need use txt module to fill string with correct information ?<br></div><div><br></div><div><--- SIP read from UDP:<a href="http://192.168.100.145:5060/" target="_blank">192.168.100.145:5060</a> ---><br>
SIP/2.0 484 Address Incomplete<br>Via: SIP/2.0/UDP 192.168.100.145:5062;branch=z9hG4bK5ec564e6<br> From: "asterisk" <<a>sip:1300@networklab.loc</a>>;tag=as0a530a8d<br>To: <<a>sip:192.168.100.145</a>>;tag=b27e1a1d33761e85846fc98f5f3a7e58.93df ---> This line ins question.<br>
Call-ID: <a href="mailto:631e893f75da720865e8468132884367@networklab.loc" target="_blank">631e893f75da720865e8468132884367@networklab.loc</a><br> CSeq: 102 OPTIONS<br>Contact: <<a href="http://sip:1300@192.168.100.145:5062/" target="_blank">sip:1300@192.168.100.145:5062</a>>;expires=3600<br>
Server: kamailio (4.1.2 (x86_64/linux))<br>Content-Length: 0<br><div><br></div></div><div><br></div><div>Slava.<br></div></div></div><br>_______________________________________________<br> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>
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