[asterisk-users] Calls disconnects after short time

[B.Masoud @ SH]

Hello,

I have grabbed again a whole call when it hangs up debug, I dono what else I
can read??

What exactly you want me to look for?

And assuming there is a firewall at my ISP, how to diagnose it?

Thanks for the advise,

Here is another log:

 

    -- Called 99999/0557202919

    -- Call accepted by xxx.xxx.xxx.xxx (format ulaw)

    -- Format for call is ulaw

elastix*CLI>

<--- SIP read from xx.xx.xx.xx:5060 --->

CANCEL sip:966557202919 at xx.xx.xx.xx SIP/2.0

Max-Forwards: 70

To: 966557202919 <sip:966557202919 at xx.xx.xx.xx>

From: <sip:9998646 at xx.xx.xx.xx>;tag=3466014864-147468

Call-ID: 19773310-3466014864-147460 at aaa.bbb.net

CSeq: 1 CANCEL

Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
SUBSCRIBE, PRACK, UPDATE

Via: SIP/2.0/UDP
xx.xx.xx.xx:5060;branch=z9hG4bKd6a812de72f094987d1363161c853aec

Contact: <sip:9998646 at xx.xx.xx.xx:5060>

Content-Length: 0

 

 

<------------->

--- (10 headers 0 lines) ---

Sending to xx.xx.xx.xx : 5060 (no NAT)

 

<--- Reliably Transmitting (no NAT) to xx.xx.xx.xx:5060 --->

SIP/2.0 487 Request Terminated

Via: SIP/2.0/UDP
xx.xx.xx.xx:5060;branch=z9hG4bKd6a812de72f094987d1363161c853aec;received=xx.
xx.xx.xx

From: <sip:9998646 at xx.xx.xx.xx>;tag=3466014864-147468

To: 966557202919 <sip:966557202919 at xx.xx.xx.xx>;tag=as717c0994

Call-ID: 19773310-3466014864-147460@ aa.bb.net

CSeq: 1 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Content-Length: 0

 

 

<------------>

 

<--- Transmitting (no NAT) to xx.xx.xx.xx:5060 --->

SIP/2.0 200 OK

Via: SIP/2.0/UDP
xx.xx.xx.xx:5060;branch=z9hG4bKd6a812de72f094987d1363161c853aec;received=xx.
xx.xx.xx

From: <sip:9998646 at xx.xx.xx.xx>;tag=3466014864-147468

To: 966557202919 <sip:966557202919 at xx.xx.xx.xx>;tag=as717c0994

Call-ID: 19773310-3466014864-147460 at aa.bb.net

CSeq: 1 CANCEL

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Content-Length: 0

 

 

<------------>

    -- Hungup 'IAX2/99999-8610'

 

Thanks.

 

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of C. Savinovich
Sent: Sunday, November 01, 2009 4:11 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Calls disconnects after short time

 

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.

 

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.

 

You should always cover your real IP numbers when showing samples of your
logs

 

CS

 

 

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of B.Masoud @ SH
Sent: Sunday, November 01, 2009 12:37 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Calls disconnects after short time

 

My server use public ip, so no nat issues, here is the out of sip debug:

 

 

 

thanks

 

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of C. Savinovich
Sent: Sunday, November 01, 2009 1:18 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Calls disconnects after short time

 

Where is the log for the actual hang up of the call?.. can you do a sip
debug?

 

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.

 

CS

 

 

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of B.Masoud @ SH
Sent: Saturday, October 31, 2009 8:12 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Calls disconnects after short time

 

Hello,

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,

Does it help to know if there is a problem that can be resolved from my
side?

 

    

 

 

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