<html><body><div style="font-family: lucida console,sans-serif; font-size: 12pt; color: #000000"><div>Hello Pedro,<br></div><div>I meant when removing option line kamailio starts getting message 484.<br></div><div><br></div><div><br></div><div>Slava.<br></div><div><br></div><hr id="zwchr"><div style="color:#000;font-weight:normal;font-style:normal;text-decoration:none;font-family:Helvetica,Arial,sans-serif;font-size:12pt;"><b>From: </b>"Pedro Niño" <nino.pedro@gmail.com><br><b>To: </b>"Kamailio (SER) - Users Mailing List" <sr-users@lists.sip-router.org><br><b>Sent: </b>Wednesday, April 9, 2014 2:35:01 PM<br><b>Subject: </b>Re: [SR-Users] message 484<br><div><br></div><p dir="ltr">Getting....?<br>
</p>
<div class="gmail_quote">El abr 7, 2014 1:21 PM, "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 Pedro,<br></div><div>I just come back on line. <br></div><div>If i remove this line I start getting<br></div><div><br></div><div><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>"Pedro Niño" <<a href="mailto:nino.pedro@gmail.com" target="_blank">nino.pedro@gmail.com</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>Tuesday, April 1, 2014 8:40:58 PM<br><b>Subject: </b>Re: [SR-Users] message 484<br>
<div><br></div><p dir="ltr">I think you should remove this section: or comment it, its behavior is not the one we want at this moment.</p>
<p dir="ltr">-------</p>
<p dir="ltr">if (is_method("OPTIONS") || allow_trusted("$si","$(proto)")) { #if (is_method("OPTIONS")) { # send reply for each options request sl_send_reply("200", "OK"); }</p>
<p dir="ltr">-----<br>
</p>
<div class="gmail_quote">El abr 1, 2014 7:58 PM, "Pedro Niño" <<a href="mailto:nino.pedro@gmail.com" target="_blank">nino.pedro@gmail.com</a>> escribió:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<p dir="ltr">Sorry, I was out for a while. Still have this issue?</p>
<p dir="ltr">From what I am seeing, asterisk is expecting for the password. Is the voicemail configured ? Check username and password.</p>
<p dir="ltr">Somewhere there it says that couldn't read username and password from the voicemail. Have the extensions.conf at asterisk dialplan configured properly?</p>
<div class="gmail_quote">El mar 31, 2014 2:25 PM, "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 Pedro,<br></div><div><br></div><div>Here SDP from asterisk. Asterisk it just don't know where to send traffic.<br></div><div>Sip peer on asterisk connects no issue. </div>
<div> <br></div><div>[voice]<br>type=peer<br>host=kamailio ip<br>defaultuser=1300<br>fromuser=1300<br>user=1300<br>secret=test<br>permit=local subnet<br>disallow=all<br>allow=ulaw<br>dtmfmode=rfc2833<br>context=voicemailbox<br>
canreinvite=no<br>insecure=port,invite<br>qualify=yes<br>directrtpsetup=no<br><div><br></div></div><div><br></div><div><br></div><div><br></div><div> -- Incorrect password '' for user '1200' (context = default)<br>
-- <SIP/1200-00000004> Playing 'vm-incorrect-mailbox.gsm' (language 'en')<br>Retransmitting #9 (no NAT) to <a href="http://10.237.236.207:5060" target="_blank">10.237.236.207:5060</a>:<br>SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP 10.237.236.207;branch=z9hG4bKf682.cc9d98383fa97727d9968596f24c2c0a.0;received=10.237.236.207<br>Via: SIP/2.0/UDP 10.237.236.212:64609;branch=z9hG4bK-d8754z-f319541e694ad32f-1---d8754z-<br>Record-Route: <sip:10.237.236.207;lr=on><br>
From: "Slava Bendersky"<sip:1200@networklab.loc;transport=UDP>;tag=6358d712<br>To: <sip:120@networklab.loc;transport=UDP>;tag=as3b53c4ae<br>Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI.<br>CSeq: 2 INVITE<br>
Server: Asterisk PBX 12.0.0<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>Session-Expires: 1800;refresher=uas<br>Contact: <<a href="http://sip:120@10.237.236.207:5062" target="_blank">sip:120@10.237.236.207:5062</a>><br>
Content-Type: application/sdp<br>Require: timer<br>Content-Length: 183<br><div><br></div>v=0<br>o=root 1990993471 1990993471 IN IP4 10.237.236.207<br>s=Asterisk PBX 12.0.0<br>c=IN IP4 10.237.236.207<br>t=0 0<br>m=audio 15070 RTP/AVP 0<br>
a=rtpmap:0 PCMU/8000<br>a=ptime:20<br>a=sendrecv<br><div><br></div>---<br>Retransmitting #10 (no NAT) to <a href="http://10.237.236.207:5060" target="_blank">10.237.236.207:5060</a>:<br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 10.237.236.207;branch=z9hG4bKf682.cc9d98383fa97727d9968596f24c2c0a.0;received=10.237.236.207<br>
Via: SIP/2.0/UDP 10.237.236.212:64609;branch=z9hG4bK-d8754z-f319541e694ad32f-1---d8754z-<br>Record-Route: <sip:10.237.236.207;lr=on><br>From: "Slava Bendersky"<sip:1200@networklab.loc;transport=UDP>;tag=6358d712<br>
To: <sip:120@networklab.loc;transport=UDP>;tag=as3b53c4ae<br>Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI.<br>CSeq: 2 INVITE<br>Server: Asterisk PBX 12.0.0<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>
Supported: replaces, timer<br>Session-Expires: 1800;refresher=uas<br>Contact: <<a href="http://sip:120@10.237.236.207:5062" target="_blank">sip:120@10.237.236.207:5062</a>><br>Content-Type: application/sdp<br>Require: timer<br>
Content-Length: 183<br><div><br></div>v=0<br>o=root 1990993471 1990993471 IN IP4 10.237.236.207<br>s=Asterisk PBX 12.0.0<br>c=IN IP4 10.237.236.207<br>t=0 0<br>m=audio 15070 RTP/AVP 0<br>a=rtpmap:0 PCMU/8000<br>a=ptime:20<br>
a=sendrecv<br>
<br>---<br>[Mar 31 14:44:25] WARNING[1834]: chan_sip.c:4259 retrans_pkt: Retransmission timeout reached on transmission YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. for seqno 2 (Critical Response) -- See <a href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions" target="_blank">https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions</a><br>
Packet timed out after 32000ms with no response<br>[Mar 31 14:44:25] WARNING[1834]: chan_sip.c:4288 retrans_pkt: Hanging up call YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI. - no reply to our critical packet (see <a href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions" target="_blank">https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions</a>).<br>
[Mar 31 14:44:25] WARNING[2801][C-0000000e]: app_voicemail.c:10590 vm_authenticate: Couldn't read username<br>Scheduling destruction of SIP dialog 'YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI.' in 32000 ms (Method: INVITE)<br>
set_destination: Parsing <sip:10.237.236.207;lr=on> for address/port to send to<br>set_destination: set destination to <a href="http://10.237.236.207:5060" target="_blank">10.237.236.207:5060</a><br>Reliably Transmitting (no NAT) to <a href="http://10.237.236.207:5060" target="_blank">10.237.236.207:5060</a>:<br>
BYE sip:1200@10.237.236.212:64609;transport=UDP SIP/2.0<br>Via: SIP/2.0/UDP 10.237.236.207:5062;branch=z9hG4bK7b8eba54<br>Route: <sip:10.237.236.207;lr=on><br>Max-Forwards: 70<br>From: <sip:120@networklab.loc;transport=UDP>;tag=as3b53c4ae<br>
To: "Slava Bendersky"<sip:1200@networklab.loc;transport=UDP>;tag=6358d712<br>Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI.<br>CSeq: 102 BYE<br>User-Agent: Asterisk PBX 12.0.0<br>X-Asterisk-HangupCause: No user responding<br>
X-Asterisk-HangupCauseCode: 18<br>Content-Length: 0<br><div><br></div><br>---<br><div><br></div><--- SIP read from UDP:<a href="http://10.237.236.207:5060" target="_blank">10.237.236.207:5060</a> ---><br>SIP/2.0 481 Call/Transaction Does Not Exist<br>
Via: SIP/2.0/UDP 10.237.236.207:5062;branch=z9hG4bK7b8eba54<br>To: "Slava Bendersky"<sip:1200@networklab.loc;transport=UDP>;tag=6358d712<br>From: <sip:120@networklab.loc;transport=UDP>;tag=as3b53c4ae<br>
Call-ID: YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI.<br>CSeq: 102 BYE<br>Accept-Language: en<br>Content-Length: 0<br><div><br></div><-------------><br>--- (8 headers 0 lines) ---<br>Really destroying SIP dialog 'YTk2YTBlYjQzYzJiZDQ4OTRkMmI3Nzk1NGU0ZDg1NTI.' Method: INVITE<br>
Reliably Transmitting (no NAT) to <a href="http://10.237.236.207:5060" target="_blank">10.237.236.207:5060</a>:<br>OPTIONS sip:10.237.236.207 SIP/2.0<br>Via: SIP/2.0/UDP 10.237.236.207:5062;branch=z9hG4bK51a7f1ef<br>Max-Forwards: 70<br>
From: "asterisk" <sip:1300@networklab.loc>;tag=as7232ca20<br>To: <sip:10.237.236.207><br>Contact: <<a href="http://sip:1300@10.237.236.207:5062" target="_blank">sip:1300@10.237.236.207:5062</a>><br>
Call-ID: 46ea55704ee7005705c98d9106904470@networklab.loc<br>CSeq: 102 OPTIONS<br>User-Agent: Asterisk PBX 12.0.0<br>Date: Mon, 31 Mar 2014 18:44:35 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>
Supported: replaces, timer<br>Content-Length: 0<br><div><br></div></div><div>Slava.<br></div><div><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>"Pedro Niño" <<a href="mailto:nino.pedro@gmail.com" target="_blank">nino.pedro@gmail.com</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 9:51:11 AM<br><b>Subject: </b>Re: [SR-Users] message 484<br><div><br></div><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>
</p><div><br></div><div class="gmail_quote">El mar 31, 2014 8:13 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 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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