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: [email protected] > To: [email protected] > 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 <[email protected]> wrote: > > Issue created: https://issues.asterisk.org/view.php?id=19228 > > > > is there anybody could your please try this ?? > > > > -S > > > > ________________________________ > > From: [email protected] > > To: [email protected] > > 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@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@from-sip:2] Dial("SIP/7527-00000003", "SIP/") in new > > stack > > Segmentation fault > > root@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
-- _____________________________________________________________________ -- 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
