satish patel
2011-May-04 17:12 UTC
[asterisk-users] asterisk-1.8 crash if no extension specified in Dial
Hey All
;satish testing
exten => 7778,1,Verbose(System crash when no extension specified in dial)
exten => 7778,2,Dial(SIP/)
*CLI> == Using SIP RTP CoS mark 5
-- Executing [7778 at from-sip:1] Verbose("SIP/7527-00000003",
"System crash when no extension specified in dial") in new stack
System crash when no extension specified in dial
-- Executing [7778 at from-sip:2] Dial("SIP/7527-00000003",
"SIP/") in new stack
Segmentation fault
root at shirley:~#
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satish patel
2011-May-04 17:56 UTC
[asterisk-users] asterisk-1.8 crash if no extension specified in Dial
Issue created: https://issues.asterisk.org/view.php?id=19228
is there anybody could your please try this ??
-S
From: satish_lx at hotmail.com
To: asterisk-users at lists.digium.com
Date: Wed, 4 May 2011 17:12:29 +0000
Subject: [asterisk-users] asterisk-1.8 crash if no extension specified in Dial
Hey All
;satish testing
exten => 7778,1,Verbose(System crash when no extension specified in dial)
exten => 7778,2,Dial(SIP/)
*CLI> == Using SIP RTP CoS mark 5
-- Executing [7778 at from-sip:1] Verbose("SIP/7527-00000003",
"System crash when no extension specified in dial") in new stack
System crash when no extension specified in dial
-- Executing [7778 at from-sip:2] Dial("SIP/7527-00000003",
"SIP/") in new stack
Segmentation fault
root at shirley:~#
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Christian Gansberger
2011-May-05 17:47 UTC
[asterisk-users] asterisk-1.8 crash if no extension specified in Dial
I had that problem too,
I wastesting with asterisk 1.8.3.2 and come across this:
Call from one extension to another with:
[macro-internal-call] ;ARG1=extension to call
exten => s,1,Set(TOCALL=${DB(SIP/${ARG1})})
exten => s,2,Dial(SIP/${TOCALL},60,tT)
...
As I had no entry in the asteriskdb, so the SIP uri was empty, and
asterisk core dumped with:
gdb output:
#0 0xb7c7db33 in strchr () from /lib/libc.so.6
crs
On 4 May 2011 19:56, satish patel <satish_lx at hotmail.com>
wrote:> Issue created: https://issues.asterisk.org/view.php?id=19228
>
> is there anybody could your please try this ??
>
> -S
>
> ________________________________
> From: satish_lx at hotmail.com
> To: asterisk-users at lists.digium.com
> Date: Wed, 4 May 2011 17:12:29 +0000
> Subject: [asterisk-users] asterisk-1.8 crash if no extension specified in
> Dial
>
> Hey All
>
> ;satish testing
> exten => 7778,1,Verbose(System crash when no extension specified in
dial)
> exten => 7778,2,Dial(SIP/)
>
>
>
>
> *CLI>?? == Using SIP RTP CoS mark 5
> ??? -- Executing [7778 at from-sip:1]
Verbose("SIP/7527-00000003", "System
> crash when no extension specified in dial") in new stack
> System crash when no extension specified in dial
> ??? -- Executing [7778 at from-sip:2] Dial("SIP/7527-00000003",
"SIP/") in new
> stack
> Segmentation fault
> root at shirley:~#
>
>
> -- _____________________________________________________________________ --
> Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
> Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or
> update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> ? ? ? ? ? ? ? http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> ? http://lists.digium.com/mailman/listinfo/asterisk-users
>
satish patel
2011-May-05 18:26 UTC
[asterisk-users] asterisk-1.8 crash if no extension specified in Dial
This issue has been resolved in latest branch 1.8 and will be resolved 1.8.5 version. Thanks for report. -S> Date: Thu, 5 May 2011 19:47:50 +0200 > From: support at accm.at > To: asterisk-users at lists.digium.com > Subject: Re: [asterisk-users] asterisk-1.8 crash if no extension specified in Dial > > I had that problem too, > I wastesting with asterisk 1.8.3.2 and come across this: > Call from one extension to another with: > > [macro-internal-call] ;ARG1=extension to call > exten => s,1,Set(TOCALL=${DB(SIP/${ARG1})}) > exten => s,2,Dial(SIP/${TOCALL},60,tT) > ... > As I had no entry in the asteriskdb, so the SIP uri was empty, and > asterisk core dumped with: > gdb output: > #0 0xb7c7db33 in strchr () from /lib/libc.so.6 > > crs > > On 4 May 2011 19:56, satish patel <satish_lx at hotmail.com> wrote: > > Issue created: https://issues.asterisk.org/view.php?id=19228 > > > > is there anybody could your please try this ?? > > > > -S > > > > ________________________________ > > From: satish_lx at hotmail.com > > To: asterisk-users at lists.digium.com > > Date: Wed, 4 May 2011 17:12:29 +0000 > > Subject: [asterisk-users] asterisk-1.8 crash if no extension specified in > > Dial > > > > Hey All > > > > ;satish testing > > exten => 7778,1,Verbose(System crash when no extension specified in dial) > > exten => 7778,2,Dial(SIP/) > > > > > > > > > > *CLI> == Using SIP RTP CoS mark 5 > > -- Executing [7778 at from-sip:1] Verbose("SIP/7527-00000003", "System > > crash when no extension specified in dial") in new stack > > System crash when no extension specified in dial > > -- Executing [7778 at from-sip:2] Dial("SIP/7527-00000003", "SIP/") in new > > stack > > Segmentation fault > > root at shirley:~# > > > > > > -- _____________________________________________________________________ -- > > Bandwidth and Colocation Provided by http://www.api-digital.com -- New to > > Asterisk? Join us for a live introductory webinar every Thurs: > > http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or > > update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > > _____________________________________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > New to Asterisk? Join us for a live introductory webinar every Thurs: > > http://www.asterisk.org/hello > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110505/e3fd0bda/attachment.htm>