Thanks, But SIP uses the caller box to send the call to the second box where is running the AGI script, the second box uses DAHDI to routes the call to E1. I've tested the codec routing a call between a E1 extension and a local one with the originate extension command and works. So that is because I'm loose with this
Regards, Gustavo On 06/03/2013, at 12:12, Gertjan Baarda <gertjan.baa...@gmail.com> wrote: > Might be a codec issue, try allow=all in your sip.conf > > Sent from my iPhone > > On 6 mrt. 2013, at 17:49, Gustavo Salvador > <gustavo.salvador...@gmail.com> wrote: > >>> >>> Hi every body, >>> >>> Please if some one could help me with this: >>> I'm writing an AGU Perl Script which basically makes a call using an >>> extension provided by other asterisk box to an E1. The asterisk version is >>> 1.6.0.28, so it hasn't the Wellington know AGI class. The code is as >>> follows: >>> >>> ============================= >>> #!/usr/bin/perl >>> use strict; >>> >>> my %AGI; >>> : >>> print "EXEC Dial(DAHDI/g2/$AGI{dnid},,W)"; >>> ============================= >>> >>> When dialplan executes the AGI, asterisk throw the following error: >>> "Dropping incompatible voice fraile on SIP/INCONCERT-00004796 of formar >>> ulaw since our native format has changed to 0x8 (alaw)." >>> >>> And connection is never make. >>> >>> It's the code OK?, I'm newbie on asterisk, and really don't know what is >>> going wrong. >>> >>> Regards, >>> >>> Gustavo >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users