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