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<p class=MsoNormal><span style='color:#1F497D'>Hello,<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>I have grabbed again a whole
call when it hangs up debug, I dono what else I can read??<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>What exactly you want me to look
for?<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>And assuming there is a firewall
at my ISP, how to diagnose it?<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>Thanks for the advise,<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>Here is another log:<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'> -- Called
99999/0557202919<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'> -- Call
accepted by xxx.xxx.xxx.xxx (format ulaw)<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'> -- Format for
call is ulaw<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>elastix*CLI><o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><--- SIP read from
xx.xx.xx.xx:5060 ---><o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>CANCEL
sip:966557202919@xx.xx.xx.xx SIP/2.0<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>Max-Forwards: 70<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>To: 966557202919
<sip:966557202919@xx.xx.xx.xx><o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>From:
<sip:9998646@xx.xx.xx.xx>;tag=3466014864-147468<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>Call-ID:
19773310-3466014864-147460@aaa.bbb.net<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>CSeq: 1 CANCEL<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>Allow: INVITE, BYE, OPTIONS,
CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>Via: SIP/2.0/UDP
xx.xx.xx.xx:5060;branch=z9hG4bKd6a812de72f094987d1363161c853aec<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>Contact:
<sip:9998646@xx.xx.xx.xx:5060><o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>Content-Length: 0<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><-------------><o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>--- (10 headers 0 lines) ---<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>Sending to xx.xx.xx.xx : 5060
(no NAT)<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><--- Reliably Transmitting
(no NAT) to xx.xx.xx.xx:5060 ---><o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>SIP/2.0 487 Request Terminated<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>Via: SIP/2.0/UDP
xx.xx.xx.xx:5060;branch=z9hG4bKd6a812de72f094987d1363161c853aec;received=xx.xx.xx.xx<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>From:
<sip:9998646@xx.xx.xx.xx>;tag=3466014864-147468<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>To: 966557202919
<sip:966557202919@xx.xx.xx.xx>;tag=as717c0994<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>Call-ID:
19773310-3466014864-147460@ aa.bb.net<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>CSeq: 1 INVITE<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>User-Agent: Asterisk PBX<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>Allow: INVITE, ACK, CANCEL,
OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>Supported: replaces<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>Content-Length: 0<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><------------><o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><--- Transmitting (no NAT) to
xx.xx.xx.xx:5060 ---><o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>SIP/2.0 200 OK<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>Via: SIP/2.0/UDP
xx.xx.xx.xx:5060;branch=z9hG4bKd6a812de72f094987d1363161c853aec;received=xx.xx.xx.xx<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>From:
<sip:9998646@xx.xx.xx.xx>;tag=3466014864-147468<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>To: 966557202919
<sip:966557202919@xx.xx.xx.xx>;tag=as717c0994<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>Call-ID:
19773310-3466014864-147460@aa.bb.net<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>CSeq: 1 CANCEL<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>User-Agent: Asterisk PBX<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>Allow: INVITE, ACK, CANCEL,
OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>Supported: replaces<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>Content-Length: 0<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><------------><o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'> -- Hungup
'IAX2/99999-8610'<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>Thanks.<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<div>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>C.
Savinovich<br>
<b>Sent:</b> Sunday, November 01, 2009 4:11 PM<br>
<b>To:</b> 'Asterisk Users Mailing List - Non-Commercial Discussion'<br>
<b>Subject:</b> Re: [asterisk-users] Calls disconnects after short time<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><span style='color:#1F497D'>The only informative part are
the 2 paragraphs of the sip debug, but can’t tell much since you only
show a very small portion of the sip log. There is a “487
Request terminated” there screaming at you but can’t tell if
meaning that provider is not handling the ACKs. That section of the
[macro-hangupcall] context is useless as it is caused by the hangup, and not an
effect.<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>The usage of a public IP is not
indicative of the existence of a firewall which can be blocking any necessary
ports for tcp and/or udp.<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>You should always cover your
real IP numbers when showing samples of your logs<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>CS<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<div>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>B.Masoud @
SH<br>
<b>Sent:</b> Sunday, November 01, 2009 12:37 AM<br>
<b>To:</b> 'Asterisk Users Mailing List - Non-Commercial Discussion'<br>
<b>Subject:</b> Re: [asterisk-users] Calls disconnects after short time<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><span style='color:#1F497D'>My server use public ip, so no
nat issues, here is the out of sip debug:<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>thanks<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<div>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>C.
Savinovich<br>
<b>Sent:</b> Sunday, November 01, 2009 1:18 PM<br>
<b>To:</b> 'Asterisk Users Mailing List - Non-Commercial Discussion'<br>
<b>Subject:</b> Re: [asterisk-users] Calls disconnects after short time<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><span style='color:#1F497D'>Where is the log for the actual
hang up of the call?.. can you do a sip debug?<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>Although there can be many
reasons, my first suspect is always a nat issue, which manifest as the
inability of asterisk to receive the incoming packets. In that case, you
should be getting a message saying “hanging up call XXXX, no reply to our
critical package”… see if you receive a message like that in your
debugging.<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>CS<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<div>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>B.Masoud @
SH<br>
<b>Sent:</b> Saturday, October 31, 2009 8:12 PM<br>
<b>To:</b> 'Asterisk Users Mailing List - Non-Commercial Discussion'<br>
<b>Subject:</b> [asterisk-users] Calls disconnects after short time<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal style='text-indent:9.0pt'>Hello,<o:p></o:p></p>
<p class=MsoNormal style='text-indent:9.0pt'>My client customers complaining
that their calls suddenly get hung-up, I am just investigating if the problem
from my side, I had a log of a hang-up case,<o:p></o:p></p>
<p class=MsoNormal style='text-indent:9.0pt'>Does it help to know if there is a
problem that can be resolved from my side?<o:p></o:p></p>
<p class=MsoNormal style='text-indent:9.0pt'><o:p> </o:p></p>
<p class=MsoNormal> <o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
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