I think you are correct, thank you for pointing it out. I just switch entries in sip.Cong put [pstn-9998] "first' and [pstn-4444] "second" and the second entry was selected :-( (so you are right on). Audiocodes gateway, has two FXO ports, I was convinced that entry is selected based on registration context in [square-bracket] of sip.conf but it doesn't appear to be the case; the last registered entry is selected as default.
Is it a limitation how SIP works or asterisk limitation? Is it possible to split registration into two different sip.conf files (sip1.conf and sip2.conf)? -- Joseph On 02/19/10 09:05, Ioan Indreias wrote: >I hope I'm not wrong but I think the problem is related to the fact >that on incoming calls Asterisk find the peers based on their IP and >not on their IP+PORT. Thus, if you have several extensions on the same >devices (=> one single IP with different SIP ports), the last entry >into your sip.conf file is taken into consideration => all calls are >sent to the context of that last extension. > >You could check this if you configure a higher verbose/debug level >(like more than 10) and check into the Asterisk logs the information >displayed by chan_sip.c > >HTH, >Ioan Indreias >www.modulo.ro > >### extract from: >http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20Channels >### >Incoming SIP Connections >=================== >When Asterisk receives an incoming SIP call, the SIP Channel Module > + first tries to find a [user] section matching the caller name >(From: username), > + then tries to find a [peer] section matching the caller's IP address. > + If no matching user or peer is found, the call is sent to the >context defined in the [general] section of sip.conf. >See: Asterisk SIP user vs peer >### > >On Fri, Feb 19, 2010 at 7:48 AM, Joseph <syscon...@gmail.com> wrote: >> Yes, it should but it doesn't. >> And the gurus at Audiocodes support can not explain why? >> >> -- >> Joseph >> >> On 02/18/10 19:27, C F wrote: >>>It should use the context of the device >>> >>>On Wed, Feb 17, 2010 at 8:40 PM, Joseph <syscon...@gmail.com> wrote: >>>> Is there any asterisk guru who can explain me how how asterisk knows which >>>> context forward the call to? >>>> >>>> -- >>>> 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