Dear All On my CentOS 5 server , I have upgraded my Asterisk from 1.4 to 1.6.2 but its CLI help does not show sip and when dialing outward sip it complains as 'sip not implemented' . Can you please let me know what is wrong my case here ? Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100202/878ececb/attachment.htm
On Tue, Feb 2, 2010 at 12:40 PM, hadi motamedi <motamedi24 at gmail.com> wrote:> Dear All > On my CentOS 5 server , I have upgraded my Asterisk from 1.4 to 1.6.2 but > its CLI help does not show sip and when dialing outward sip it complains as > 'sip not implemented' . Can you please let me know what is wrong my case > here ? > Thank you > >Sorry . Forgot to mention that I have made use of the following packages for the upgrade procedure : asterisk-1.6.2.1.tar.gz dahdi-linux-complete-2.2.1+2.2.1.tar.gz libpri-1.4.10.2.tar.gz Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100202/5a993ce5/attachment.htm
This is usually due to an error with the SIP stack not being loaded due to an error - make sure that full logging is on and check your log file and search for ERROR and see if there is any mention to SIP (chan_sip.o etc), alternatively, start asterisk from the command like with asterisk -vvvvvvvvvdddddddddc and watch the output to screen for any errors at startup. Fix the error and SIP will start up.
On Wed, Feb 3, 2010 at 12:17 AM, Ben Dinnerville <ben at voicelogic.com.au>wrote:> > This is usually due to an error with the SIP stack not being loaded due > to an error - make sure that full logging is on and check your log file > and search for ERROR and see if there is any mention to SIP (chan_sip.o > etc), alternatively, start asterisk from the command like with asterisk > -vvvvvvvvvdddddddddc and watch the output to screen for any errors at > startup. Fix the error and SIP will start up. > > > -- > _____________________________________________________________________ > -- 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 >Thank you very much for your reply . I found my mistake . It was coming from my attempt to copy the old sip.conf & extensions.conf onto the new build ones . It seems that it is not possible this way . -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100203/bdbcd135/attachment.htm