[asterisk-users] Calls disconnects after short time

Sun Nov 1 07:10:43 CST 2009

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:

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

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

Sending to 213.165.32.100 : 5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 213.165.32.100:5060 --->

SIP/2.0 487 Request Terminated

Via: SIP/2.0/UDP
213.165.32.100:5060;branch=z9hG4bKa95f47a28fdc61714dc862cefe1a326a;received=
213.165.32.100

From: <sip:9991558 at 213.165.32.100>;tag=3466008105-77358

To: 966599740196 <sip:966599740196 at 213.165.32.100>;tag=as54d7ac3d

Call-ID: 19751463-3466008105-77352 at dalmsx01.vincomm.net

CSeq: 1 INVITE

User-Agent: Asterisk PBX

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

Supported: replaces

Content-Length: 0

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

elastix*CLI>

<--- Transmitting (no NAT) to 213.165.32.100:5060 --->

SIP/2.0 200 OK

Via: SIP/2.0/UDP
213.165.32.100:5060;branch=z9hG4bKa95f47a28fdc61714dc862cefe1a326a;received=
213.165.32.100

From: <sip:9991558 at 213.165.32.100>;tag=3466008105-77358

To: 966599740196 <sip:966599740196 at 213.165.32.100>;tag=as54d7ac3d

Call-ID: 19751463-3466008105-77352 at dalmsx01.vincomm.net

CSeq: 1 CANCEL

User-Agent: Asterisk PBX

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

Supported: replaces

Content-Length: 0

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

    -- Hungup 'IAX2/99999-4490'

  == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on
'SIP/213.165.32.100-b7c10ad8' in macro 'dialout-trunk'

  == Spawn extension (outbound-allroutes, 966599740196, 4) exited non-zero
on 'SIP/213.165.32.100-b7c10ad8'

    -- Executing [h at macro-dialout-trunk:1]
Macro("SIP/213.165.32.100-b7c10ad8", "hangupcall|") in new stack

    -- Executing [s at macro-hangupcall:1]
ResetCDR("SIP/213.165.32.100-b7c10ad8", "w") in new stack

    -- Executing [s at macro-hangupcall:2] NoCDR("SIP/213.165.32.100-b7c10ad8",
"") in new stack

    -- Executing [s at macro-hangupcall:3]
GotoIf("SIP/213.165.32.100-b7c10ad8", "1?skiprg") in new stack

    -- Goto (macro-hangupcall,s,6)

    -- Executing [s at macro-hangupcall:6]
GotoIf("SIP/213.165.32.100-b7c10ad8", "1?skipblkvm") in new stack

    -- Goto (macro-hangupcall,s,9)

    -- Executing [s at macro-hangupcall:9]
GotoIf("SIP/213.165.32.100-b7c10ad8", "1?theend") in new stack

    -- Goto (macro-hangupcall,s,11)

    -- Executing [s at macro-hangupcall:11]
Hangup("SIP/213.165.32.100-b7c10ad8", "") in new stack

  == Spawn extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/213.165.32.100-b7c10ad8' in macro 'hangupcall'

  == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on
'SIP/213.165.32.100-b7c10ad8'

elastix*CLI>

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?

elastix*CLI>

    -- Hungup 'IAX2/99999-6813'

  == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on
'SIP/213.165.32.100-b7d21018' in macro 'dialout-trunk'

  == Spawn extension (outbound-allroutes, 966507944491, 4) exited non-zero
on 'SIP/213.165.32.100-b7d21018'

    -- Executing [h at macro-dialout-trunk:1]
Macro("SIP/213.165.32.100-b7d21018", "hangupcall|") in new stack

    -- Executing [s at macro-hangupcall:1]
ResetCDR("SIP/213.165.32.100-b7d21018", "w") in new stack

    -- Executing [s at macro-hangupcall:2] NoCDR("SIP/213.165.32.100-b7d21018",
"") in new stack

    -- Executing [s at macro-hangupcall:3]
GotoIf("SIP/213.165.32.100-b7d21018", "1?skiprg") in new stack

    -- Goto (macro-hangupcall,s,6)

    -- Executing [s at macro-hangupcall:6]
GotoIf("SIP/213.165.32.100-b7d21018", "1?skipblkvm") in new stack

    -- Goto (macro-hangupcall,s,9)

    -- Executing [s at macro-hangupcall:9]
GotoIf("SIP/213.165.32.100-b7d21018", "1?theend") in new stack

    -- Goto (macro-hangupcall,s,11)

    -- Executing [s at macro-hangupcall:11]
Hangup("SIP/213.165.32.100-b7d21018", "") in new stack

  == Spawn extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/213.165.32.100-b7d21018' in macro 'hangupcall'

  == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on
'SIP/213.165.32.100-b7d21018'

elastix*CLI>

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