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Hi list, I'm having some strange problems since I updated to CVS HEAD three hours ago... First: I was using Iax Comm in some PCs, it suddenly stopped working, what I get is som pieces of audio once in a while, I mean instead of listening to the ring tone and then the voice on the other side I just hear a bit of the ring tone, maybe another bit, a bit of someone answering... like

Hi! I'm trying to transfer a incomming call from a Queue to another extension. I'm receiving a call from a queue with the AgentCallbackLogin. The queu is as following: Queue(sales|t) Which should allow transfers. So as soon as the call is answered I would like to be able to transfer it When the agent presses the # I get the dialtone but as soon as I press any digit Asterisk tells me

We're investigating audio quality issues in our system; maybe someone can help. We're using Asterisk as a basic PBX, with a single PRI on one side and SIP phones on the other: Sipura SPA-841's. We're experiencing several audio effects which seem to commonly correspond to network failures (packet loss, high jitter, etc manifested as "robot voice", dropouts, periodic

...;>To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> >> > > > >------------------------------ > >Message: 4 >Date: Sun, 27 Mar 2005 23:50:40 -0300 >From: "Matias G." <listas_ast@reliable.com.ar> >Subject: Re: [Asterisk-Users] Comedian Voicemail Issues >To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> >Message-ID: <002b01c53340$ef10db30$c900a8c0@krikkit> >Content-Type: text/plain; chars...

...clues? :) Turn off call waiting in zapata.conf callwaiting=no -Seth Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ------------------------------ Message: 24 Date: Fri, 18 Mar 2005 13:26:41 -0300 From: "Matias G." <listas_ast@reliable.com.ar> Subject: [Asterisk-Users] call a url and get a result in the dialplan To: <asterisk-users@lists.digium.com> Message-ID: <00a001c52bd7$4578b9f0$c900a8c0@krikkit> Content-Type: text/plain; charset="iso-8859-1" Hi, can a call a php script wich is located in a...

Hi, I have 3 PAP2 connected to *, they work fine but there are some things which I would like to improve, some of them are: - double ring tone when placing a call (I hear two tones it seems like the PAP2 is generating it's own tone) - some kind of noise (like glitches or something) when I pick up the phone (seems like some polarity thing) - I'd like to keep the tone after

Hi, can a call a php script wich is located in a remote server, someting like calling www.theserveraddress.com/scripts/validate?code=234234swq and get the result which this script generates (a 0 or a 1) back in the dial plan in a direct way or should I create a script which in turn does this? I'm using * CVS HEAD. Also I searched for this for I while but didn't manage to find anything

Hi i'm trying to connect to the PSTN in the following way sip ATA -> * -> gnugk -> Cisco AS5300 -> PSTN I'm using asterisk CVS-HEAD-06/01/05-14:33:15 running over RH EL3 Asterisk-Oh323 0.7.2 pre1 Open H323 v1.13.5 pwlib v1.6.6 and I'm having a lot of trouble, gnugk and * both have public ips and are not behind any type of firewall, the sip ATA is behind a firewall and