RE: [Asterisk-Users] overriding DTMF and codec from dialplan?
OH But it is just that simple. You also have: -= Info about application 'ImportVar' =- [Synopsis]: Set variable to value [Description]: ImportVar(#n=channel|variable): Sets variable n to variable as evaluated on the specified channel (instead of current). If prefixed with _, single inheritance assumed. If prefixed with __, infinite inheritance is assumed. bkw It's not so simple. Check http://bugs.digium.com/bug_view_advanced_page.php?bug_id=928 for the details. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] overriding DTMF and codec from dialplan?
Brian West wrote: OH But it is just that simple. You also have: -= Info about application 'ImportVar' =- [Synopsis]: Set variable to value [Description]: ImportVar(#n=channel|variable): Sets variable n to variable as evaluated on the specified channel (instead of current). If prefixed with _, single inheritance assumed. If prefixed with __, infinite inheritance is assumed. I give up, my mistake. bkw It's not so simple. Check http://bugs.digium.com/bug_view_advanced_page.php?bug_id=928 for the details. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] overriding DTMF and codec from dialplan?
is it possible, from an agi script or directly in extensions.conf, to override the DTMF and codec settings? to answer my own question SetVar(SIP_CODEC=g726) allowed me to force g726, but only on outgoing calls. when dialling in from the iax server, I do the same, setting the codec etc, but this does not work. sip show channels only shows the channel using alaw Change this into SetVar(_SIP_CODEC=g726) and it will work. you sure? sipgw1:/usr/src/asterisk # grep -r _SIP_CODEC . sipgw1:/usr/src/asterisk # roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] overriding DTMF and codec from dialplan?
On Sat, 27 Nov 2004, Roy Sigurd Karlsbakk wrote: Change this into SetVar(_SIP_CODEC=g726) and it will work. you sure? sipgw1:/usr/src/asterisk # grep -r _SIP_CODEC . sipgw1:/usr/src/asterisk # The leading underscore means the variable will be inherited by the outgoing channel. Did you try it? Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] overriding DTMF and codec from dialplan?
Roy Sigurd Karlsbakk wrote: Change this into SetVar(_SIP_CODEC=g726) and it will work. you sure? sipgw1:/usr/src/asterisk # grep -r _SIP_CODEC . sipgw1:/usr/src/asterisk # I don't think the leading underscore is part of the source. . and it's defined in the channels dir: /usr/src/asterisk/channels# grep SIP_CODEC * chan_sip.c: codec=pbx_builtin_getvar_helper(p-owner,SIP_CODEC); chan_sip.c: ast_log(LOG_NOTICE, Changing codec to '%s' for this call because of ${SIP_CODEC) variable\n,codec); chan_sip.c: ast_log(LOG_NOTICE, Ignoring ${SIP_CODEC} variable because it is not shared by both ends.\n); chan_sip.c: } else ast_log(LOG_NOTICE, Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): %s\n,codec); B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] overriding DTMF and codec from dialplan?
is it possible, from an agi script or directly in extensions.conf, to override the DTMF and codec settings? to answer my own question SetVar(SIP_CODEC=g726) allowed me to force g726, but only on outgoing calls. when dialling in from the iax server, I do the same, setting the codec etc, but this does not work. sip show channels only shows the channel using alaw roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] overriding DTMF and codec from dialplan?
Roy Sigurd Karlsbakk wrote: is it possible, from an agi script or directly in extensions.conf, to override the DTMF and codec settings? to answer my own question SetVar(SIP_CODEC=g726) allowed me to force g726, but only on outgoing calls. when dialling in from the iax server, I do the same, setting the codec etc, but this does not work. sip show channels only shows the channel using alaw Change this into SetVar(_SIP_CODEC=g726) and it will work. roy Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users