Is it possible to split sip.conf into two files (sip1.conf sip2.conf)? I have an Audiocodes gateway with two FXO ports, and (according to info I received, and it appears to be correct) Asterisk find the peers based on their IP and not on their IP+PORT. Thus, Audiocodes with two FXO ports registered on the same devices (=> one single IP with different SIP ports), the last entry into my sip.conf file is taken into consideration => all calls are sent to the context of that last extension. So I can only use one "context" for incoming calls. If I split the sip.conf into two files will it make any difference. Is it a limitation/bug in Asterisk or sip.conf? -- Joseph
I think sip.conf will allow the inclusion of a second (or greater) sip2.conf file. This might only apply to extensions.conf, but I'm betting all .conf files are processed with the same parser. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joseph Sent: Friday, February 19, 2010 10:22 AM To: asterisk-users at lists.digium.com Subject: [asterisk-users] splitting sip.conf to two files Is it possible to split sip.conf into two files (sip1.conf sip2.conf)? I have an Audiocodes gateway with two FXO ports, and (according to info I received, and it appears to be correct) Asterisk find the peers based on their IP and not on their IP+PORT. Thus, Audiocodes with two FXO ports registered on the same devices (=> one single IP with different SIP ports), the last entry into my sip.conf file is taken into consideration => all calls are sent to the context of that last extension. So I can only use one "context" for incoming calls. If I split the sip.conf into two files will it make any difference. Is it a limitation/bug in Asterisk or sip.conf? -- Joseph -- _____________________________________________________________________ -- 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
On Fri, Feb 19, 2010 at 09:21:46AM -0700, Joseph wrote:> Is it possible to split sip.conf into two files (sip1.conf sip2.conf)? > > I have an Audiocodes gateway with two FXO ports, and (according to info I received, and it appears to be correct) Asterisk find the peers based on their IP > and not on their IP+PORT. Thus, Audiocodes with two FXO ports registered on the same devices (=> one single IP with different SIP ports), the last entry > into my sip.conf file is taken into consideration => all calls are sent to the context of that last extension. > > So I can only use one "context" for incoming calls. If I split the sip.conf into two files will it make any difference. > > Is it a limitation/bug in Asterisk or sip.conf?I assume you use '#include to separate sip.conf to two files. #include is a verbatim inclusion, and thus for all prictical purposes it is the same as if everything were in a single file. Configuration [sections] cannot be repeated in the Asterisk configuration files. If you want to add later on anything to [foo], you can't just add a second [foo] . Rather, you should add: [foo](+) This will add the content of that section after the content of the existing section [foo]. See http://svn.digium.com/svn/asterisk/trunk/doc/tex/configuration.tex (Any better direct link?) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.cohen at xorcom.com +972-50-7952406 mailto:tzafrir.cohen at xorcom.com http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir
Joseph wrote:> Is it possible to split sip.conf into two files (sip1.conf sip2.conf)? > > I have an Audiocodes gateway with two FXO ports, and (according to info I received, and it appears to be correct) Asterisk find the peers based on their IP > and not on their IP+PORT. Thus, Audiocodes with two FXO ports registered on the same devices (=> one single IP with different SIP ports), the last entry > into my sip.conf file is taken into consideration => all calls are sent to the context of that last extension. > > So I can only use one "context" for incoming calls. If I split the sip.conf into two files will it make any difference.there might be an "include" directive in sip.conf (i can't confirm) however Asterisk will see it as one big sip.conf so it will do absolutely nothing for you in this situation. what you can do is setup automatic dial to different extensions on the 2 ports on audiocodes. -- Edwin Lam <edwin.lam at officegeneral.com> Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20