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Hello, Could anyone please give some information on configuring asterisk as a gateway. What contents have to add in h.323 .conf and extensions.conf files ? Thanks & Regards Bincy K Philip -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071102/53be51ce/attachment.htm

Morning All, Quick question that has me stumped. Have a queue with several members (Statically defined in queues.conf at this stage for testing) who use Cisco 7960's. The queue is configured to use rrmemory and generally this works correctly. However if a member is already on a call their phone will still ring (The 7960 can show multiple incoming calls for one line). I really don't want

Hello, When I tried to install chan_h323-1.0.1-module.i386 RPM i got these errors. Failed dependencies: libh323_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 libpt_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 But i found the same files in /usr/lib/libh323_linux_x86_r.so.1 /usr/lib/libpt_linux_x86_r.so.1 What to do for asterisk to detect the same

Morning All, Just wondering if anyone can confirm that peridoic-announce and periodic-announce-frequency are still valid options within queues.conf? For testing purposes my queue includes; periodic-announce-frequency = 10 periodic-announce = demo-congrats When in the queue however I'm not hearing the message, the context we break out to works fine, its just the messages that are not being

Morning All, Has anyone here successfully implemented skills based routing within queues? The concept behind skills based routing is fairly straight forward, and I know I could do it with multiple queues, agent penalties and a bit of AGI to put the call into the right queue. However doing this is going to require the addition of several extra queues and isn't a very clean solution. The

Another quick question (Spending the day trying to get this project sorted and tucked away) If I am dynamically adding queue members, they will not abide to settings within agents.conf will they? Ie. I need the equivalent of Autologoff however want my agents to receive calls when someone joins the queue, not have to sit on hold all day. I see AgentCallbackLogin has finally been removed. Has

We use Asterisk 1.4.7 on CentOS with Bandwidth.com as our provider and Polycom 330's for endpoints. When one of our end points places a call on hold we get the following in CLI. There is no music on hold provided for the caller. The SIP.CONF entry for our connection to bandwithd.com specifies disallow=all and allow=ulaw. Should there be a similar setting on the user.conf entries? An

Does anyone know if this phone can or does work with Asterisk ? I have a potential client who wants to throw away their current mitel ip system (keeps crashing on a regular basis - not good for a phone-based company), but are reluctant to purchase new handsets. Thanks. Julian.

Afternoon All, Is anyone aware of a way to generate ringing as opposed to starting music on hold for the party originating a call with followme? I'm assuming its doable as it looks like FreePBX users get the option (Not to say that FreePBX haven't got their own followme implementation though). Cheers Nick.

Afternoon All, Today rolled a pre-production box from Trunk back to 1.4.7 (In an attempt to get a working SCCP channel). During the process Music On Hold appears to have died (Not, just when calling from a SCCP device, but coming in on SIP also). CLI is showing -- Executing [XXXXXXXX at unauthed-inbound:2] MusicOnHold("SIP/10.97.1.33-09f0cfc8", "sounds") in new stack