Sure,
Start call
[Jun 10 10:01:17] VERBOSE[27776] netsock2.c: == Using SIP RTP CoS mark 5
[Jun 10 10:01:17] VERBOSE[7269] pbx.c: -- Executing [8004815122@from-sip:1]
Dial("SIP/7081-000005ed", "DAHDI/g1/18004815122") in new stack
[Jun 10 10:01:17] DEBUG[7269] sig_pri.c: sig_pri_request 5
[Jun 10 10:01:17] DEBUG[7269] sig_pri.c: CALLER NAME: Pascal Honscher NUM: 7081
[Jun 10 10:01:17] VERBOSE[7269] sig_pri.c: -- Requested transfer
capability: 0x00 - SPEECH
[Jun 10 10:01:17] VERBOSE[7269] app_dial.c: -- Called DAHDI/g1/18004815122
[Jun 10 10:01:17] VERBOSE[7269] app_dial.c: -- DAHDI/i1/18004815122-1db is
proceeding passing it to SIP/7081-000005ed
[Jun 10 10:01:17] DTMF[7267] channel.c: DTMF begin '1' received on
SIP/7134-000005eb
[Jun 10 10:01:17] DTMF[7267] channel.c: DTMF begin ignored '1' on
SIP/7134-000005eb
[Jun 10 10:01:17] DTMF[7267] channel.c: DTMF end '1' received on
SIP/7134-000005eb, duration 130 ms
[Jun 10 10:01:17] DTMF[7267] channel.c: DTMF end passthrough '1' on
SIP/7134-000005eb
[Jun 10 10:01:17] VERBOSE[7267] file.c: -- <SIP/7134-000005eb> Playing
'vm-first.ulaw' (language 'en')
[Jun 10 10:01:18] VERBOSE[7267] config.c: == Parsing
'/var/spool/asterisk/voicemail/default/7134/INBOX/msg0000.txt': [Jun 10
10:01:18] VERBOSE[7267] config.c: == Found
[Jun 10 10:01:18] VERBOSE[7267] file.c: -- <SIP/7134-000005eb> Playing
'vm-message.ulaw' (language 'en')
[Jun 10 10:01:18] VERBOSE[7267] file.c: -- <SIP/7134-000005eb> Playing
'vm-received.ulaw' (language 'en')
[Jun 10 10:01:19] VERBOSE[7268] app_dial.c: -- DAHDI/i1/18778603058-1da is
ringing
[Jun 10 10:01:19] VERBOSE[7268] app_dial.c: -- DAHDI/i1/18778603058-1da is
making progress passing it to SIP/7065-000005ec
[Jun 10 10:01:19] VERBOSE[7268] app_dial.c: -- DAHDI/i1/18778603058-1da
answered SIP/7065-000005ec
[Jun 10 10:01:19] DEBUG[7268] channel.c: setting peeraccount to "Sharon
Cordesse" for SIP/7065-000005ec from data on channel DAHDI/i1/18778603058-1da
[Jun 10 10:01:19] VERBOSE[7269] app_dial.c: -- DAHDI/i1/18004815122-1db is
making progress passing it to SIP/7081-000005ed
[Jun 10 10:01:19] VERBOSE[7267] file.c: -- <SIP/7134-000005eb> Playing
'digits/at.ulaw' (language 'en')
[Jun 10 10:01:20] VERBOSE[7269] app_dial.c: -- DAHDI/i1/18004815122-1db
answered SIP/7081-000005ed
[Jun 10 10:01:20] DEBUG[7269] channel.c: setting peeraccount to "Pascal
Honscher" for SIP/7081-000005ed from data on channel DAHDI/i1/18004815122-1db
Hangup and again dial
[Jun 10 10:01:57] DEBUG[7269] sig_pri.c: sig_pri_hangup 5
[Jun 10 10:01:57] DEBUG[7269] sig_pri.c: Not yet hungup... Calling hangup once
with icause, and clearing call
[Jun 10 10:01:57] VERBOSE[7269] chan_dahdi.c: -- Hungup
'DAHDI/i1/18004815122-1db'
[Jun 10 10:01:57] VERBOSE[7269] pbx.c: == Spawn extension (from-sip,
8004815122, 1) exited non-zero on 'SIP/7081-000005ed'
[Jun 10 10:02:01] VERBOSE[7224] file.c: -- <DAHDI/i2/9789790769-138>
Playing 'vm-intro.ulaw' (language 'en')
[Jun 10 10:02:02] VERBOSE[27776] netsock2.c: == Using SIP RTP CoS mark 5
[Jun 10 10:02:02] VERBOSE[7286] pbx.c: -- Executing [7069@from-sip:1]
Macro("SIP/7134-000005f0", "stdexten,7069,SIP/7069") in new stack
[Jun 10 10:02:02] VERBOSE[7286] pbx.c: -- Executing [s@macro-stdexten:1]
Set("SIP/7134-000005f0", "_CALLED_EXT=SIP/7069") in new stack
[Jun 10 10:02:02] VERBOSE[7286] pbx.c: -- Executing [s@macro-stdexten:2]
Dial("SIP/7134-000005f0", "SIP/7069&iax2/7069,20,t") in new stack
[Jun 10 10:02:02] VERBOSE[7286] netsock2.c: == Using SIP RTP CoS mark 5
[Jun 10 10:02:02] VERBOSE[7286] app_dial.c: -- Called SIP/7069
[Jun 10 10:02:02] WARNING[7286] app_dial.c: Unable to create channel of type
'iax2' (cause 20 - Unknown)
[Jun 10 10:02:02] VERBOSE[7286] app_dial.c: -- SIP/7069-000005f1 is ringing
[Jun 10 10:02:04] VERBOSE[27776] netsock2.c: == Using SIP RTP CoS mark 5
[Jun 10 10:02:04] VERBOSE[7287] pbx.c: -- Executing [8004815122@from-sip:1]
Dial("SIP/7081-000005f2", "DAHDI/g1/18004815122") in new stack
[Jun 10 10:02:04] DEBUG[7287] sig_pri.c: sig_pri_request 5
[Jun 10 10:02:04] DEBUG[7287] sig_pri.c: CALLER NAME: Pascal Honscher NUM: 7081
[Jun 10 10:02:04] VERBOSE[7287] sig_pri.c: -- Requested transfer
capability: 0x00 - SPEECH
[Jun 10 10:02:04] VERBOSE[7287] app_dial.c: -- Called DAHDI/g1/18004815122
[Jun 10 10:02:04] VERBOSE[7287] app_dial.c: -- DAHDI/i1/18004815122-1dc is
proceeding passing it to SIP/7081-000005f2
[Jun 10 10:02:06] VERBOSE[7224] file.c: -- <DAHDI/i2/9789790769-138>
Playing 'beep.ulaw' (language 'en')
[Jun 10 10:02:06] VERBOSE[7287] app_dial.c: -- DAHDI/i1/18004815122-1dc is
making progress passing it to SIP/7081-000005f2
[Jun 10 10:02:07] VERBOSE[7224] app_voicemail.c: -- Recording the message
[Jun 10 10:02:07] VERBOSE[7224] app.c: -- x=0, open writing:
/var/spool/asterisk/voicemail/default/7000/tmp/L7WLl1 format: wav49, 0x2b98578
[Jun 10 10:02:07] VERBOSE[7224] app.c: -- x=1, open writing:
/var/spool/asterisk/voicemail/default/7000/tmp/L7WLl1 format: gsm, 0x2c56eb8
[Jun 10 10:02:07] VERBOSE[7224] app.c: -- x=2, open writing:
/var/spool/asterisk/voicemail/default/7000/tmp/L7WLl1 format: wav, 0x29e9ae8
[Jun 10 10:02:07] VERBOSE[7287] app_dial.c: -- DAHDI/i1/18004815122-1dc
answered SIP/7081-000005f2
[Jun 10 10:02:07] DEBUG[7287] channel.c: setting peeraccount to "Pascal
Honscher" for SIP/7081-000005f2 from data on channel DAHDI/i1/18004815122-1dc
Date: Fri, 10 Jun 2011 13:51:21 -0500
From: [email protected]
To: [email protected]
Subject: Re: [asterisk-users] asterisk 1.8 PRI random call drop issue
On Fri, Jun 10, 2011 at 1:27 PM, satish patel <[email protected]> wrote:
Hi,
We having some PRI call drop issue on asterisk 1.8.x but we had no issue never
ever on asterisk 1.2. Anybody else having this issue ?
-S
What troubleshooting have you already done? Do you have logs? Debug
information? What error messages are displaying when the call is dropped? The
more information you can provide us, the better we can help you.
I would suggest as a minimum that you provide us with the CLI output of a
complete, dropped call. We can move forward from there.
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com
--
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_____________________________________________________________________
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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