etc/php5/cli/conf.d/mcrypt.ini on line 1 in Unknown on line 0 PHPĥ.3.2-1ubuntu4.10 with Suhosin-Patch (cli) (built: 00:09:58) Copyright (c) 1997-2009 The PHP Group Zend Engine v2.3.0,Ĭopyright (c) 1998-2010 Zend TechnologiesĪsterisk -v Asterisk 1.8.8.0~rc4-1digium0+1~lucid, Copyright (C) 1999 ~# php -v PHP Deprecated: Comments starting with '#' are deprecated in My current config is like the ones in the turorial extensions,sips,iax'as I made links to both var and usr agi-bin folder back to placed the a2billing.php with the lib link but never writes anything to the log file.Īpache2 runs runs as www-data:www-data asterisk runs as asterisk:asterisk freepbx (i disabled it for now but runs as asterisk) I set all the version to 1_6, permissions are good- i even wrote test into the a2billing_agi.log and it shows the test string on the admin end. step by step idiot tutorial and i still cant get it working. and all the steps and verifications are just like he shows (execpt i use /usr/share/asterisk instead of /var/lib/asterisk. and just error q931(zoiper on iphone) if i use the freepbx extension i can talk with anybody iusing freepbx routes.īasically i skipped freepbx out of the picture, so its just asterisk direct to a2billing. I connect to a2billing user, place call- silence. I made a ln -s to /usr/src/a2billing/AGI/a2billing.php /usr/share/asterisk/agi-bin (the asterisk conf defines this as the agi location-and freepbx time,freebpx echo works)Īnother problem is that a2billing has not made 1 single log file and i cannot determine what the heck is going on(the permissions are set write global and belong to asterisk). I am on Ubuntu 10.4LTS - Asterisk was installed using apt-get -> /usr/share/asterisk but i also have a /var/lib/asterisk IN the simulator i use the same number that works in there on my sip. Which means the host(a2billing and asterisk) knows about the call but activbely denied routing the call. The diagnostic field mayĬontain additional information about the supplementary service and This cause mayĪlso be generated by the network, indicating that the call was clearedĭue to a supplementary service constraint. Sending this cause is neither busy nor incompatible. although it could have accepted the call because the equipment That the equipment sending this cause does not wish to accept thisĬall. every single time i dial I get error q931_disconnect_code_21 I can make a user log in using a sip device. all that works - i can log in add rates,trunks, the simulator get the correct routes,rates and trunks. My Asterisk works fine, freepbx i can make extension and place calls on trunks OK, internals perfect. I have been trying to get a2billing to work for the past 4 days now, i followed 4 tutorials and they all gave me some insight into how things work.
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