HI I am Using asterisk-1.6.0.5 i cannot originate call from AMI interface here are my Originate action Packet Action: Originate Channel: SIP/111 Context: default Exten: 8135551212 Priority: 1 Callerid: 3125551212 Timeout: 30000 Variable: var1=23|var2=24|var3=25 ActionID: ABC45678901234567890 where 111 Is my SIP phone number which registered with my asterisk server and here are my manager.conf [mark] secret = mysecret read = system,call,log,verbose,agent,user,config,dtmf,reporting,cdr,dialplan write = system,call,agent,user,config,command,reporting,originate I can login with this manager User and while trying with above action i got Response: Error Message: Channel Not Specified can anybody help me? regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090529/2c93cecc/attachment.htm
On Fri, May 29, 2009 at 4:22 AM, DHAVAL INDRODIYA <dhaval.it01034 at gmail.com> wrote:> i cannot originate call from AMI interface here are my Originate action > Packet > Channel: SIP/111 > where 111 Is my SIP phone number which registered with my asterisk server > I can login with this manager User and while trying with above action i got > Response: Error > Message: Channel Not SpecifiedYou need a destination. SIP/111 needs an @destination to be a complete channel name.
On Friday 29 May 2009 11:20:31 am David Backeberg wrote:> On Fri, May 29, 2009 at 4:22 AM, DHAVAL INDRODIYA > > <dhaval.it01034 at gmail.com> wrote: > > i cannot originate call from AMI interface here are my Originate action > > Packet > > Channel: SIP/111 > > where 111 Is my SIP phone number which registered with my asterisk server > > I can login with this manager User and while trying with above action i > > got Response: Error > > Message: Channel Not Specified > > You need a destination. SIP/111 needs an @destination to be a complete > channel name.i apologize for not being able to get to the right bug # right now, but there was a manager bug that was fixed in following versions of asterisk. the patch that does the fix is simple: http://cvs.fedoraproject.org:80/viewvc/rpms/asterisk/F-10/0016-Fix-a-reversed- logic-ast_strlen_zero.patch?revision=1.1&view=markup -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: This is a digitally signed message part. Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090529/52a08a35/attachment.pgp
hi i update patch and its working now On Fri, May 29, 2009 at 11:59 PM, Anthony Messina <amessina at messinet.com>wrote:> On Friday 29 May 2009 11:20:31 am David Backeberg wrote: > > On Fri, May 29, 2009 at 4:22 AM, DHAVAL INDRODIYA > > > > <dhaval.it01034 at gmail.com> wrote: > > > i cannot originate call from AMI interface here are my Originate action > > > Packet > > > Channel: SIP/111 > > > where 111 Is my SIP phone number which registered with my asterisk > server > > > I can login with this manager User and while trying with above action i > > > got Response: Error > > > Message: Channel Not Specified > > > > You need a destination. SIP/111 needs an @destination to be a complete > > channel name. > > i apologize for not being able to get to the right bug # right now, but > there > was a manager bug that was fixed in following versions of asterisk. > > the patch that does the fix is simple: > > > http://cvs.fedoraproject.org:80/viewvc/rpms/asterisk/F-10/0016-Fix-a-reversed- > logic-ast_strlen_zero.patch?revision=1.1&view=markup<http://cvs.fedoraproject.org:80/viewvc/rpms/asterisk/F-10/0016-Fix-a-reversed-%0Alogic-ast_strlen_zero.patch?revision=1.1&view=markup> > > -- > Anthony - http://messinet.com - http://messinet.com/~amessina/gallery > 8F89 <http://messinet.com/%7Eamessina/gallery%0A8F89> 5E72 8DF0 BCF0 10BE > 9967 92DC 35DC B001 4A4E > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > 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/20090530/cf1d9ff5/attachment.htm