Re: [asterisk-users] Caller Prompts in a Queue??

menu options it will 'go' since they are holding in that context. On 7/5/06, Aaron Paxson <[EMAIL PROTECTED]> wrote: Can I have caller prompts in a queue? If so, anyone know of an example or documentation? Inside my queue, I want to give the callers a choice to leave a voicemai

[asterisk-users] Caller Prompts in a Queue??

Can I have caller prompts in a queue?  If so, anyone know of an example or documentation?   Inside my queue, I want to give the callers a choice to leave a voicemail, rather than waiting.   Is this available out-of-the-box, without writing an AGI?   Thanks!!   ~~Aaron

Re: [Asterisk-Users] Cannot get back chan_zap.so module!??

I get the chan_zap.so if I recompile under asterisk-1.2.7.1, but not under subversion TRUNK   Anyone able to do this? - Original Message - From: Aaron Paxson To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, June 30, 2006 1:44 PM Subject

Re: [Asterisk-Users] Switchtype

I would work that out with your vendor, as the settings must be the same on both sides.   If national won't work for you, ask them if they can change to something else.    What kinds of connectivity issues?  Could be line problems too. - Original Message - From: James Hawks

[Asterisk-Users] Cannot get back chan_zap.so module!??

Hey list!   I keep getting the error:   "Unable to create channel of type 'Zap' (cause 66 - Channel not implemented)" error.    In looking on my filesystem, I seemed to have "lost" the chan_zap.so module from /usr/lib/asterisk/modules.  I've re-compiled Zaptel and Asterisk, but it doesn't sh

Re: [Asterisk-Users] Call Queue NOT using RoundRobin ?!?

m/view.php?id=7279 greetings, Michael Aaron Paxson schrieb: If someone can point me in the right direction, I'll look into it. I'm not a C programmer, but I *should* be able to find my way. I'm looking at app_queue.c I see the strategies defined, but nothing about how they are

Re: [Asterisk-Users] Call Queue NOT using RoundRobin ?!?

red), then the next call will ring 2->3->1, etc. For the first call, if agent 2 answered it in roundrobin mode, they would still be the first agent for the next call, but rrmemory mode will move past them. On 6/29/06, Aaron Paxson <[EMAIL PROTECTED]> wrote:

Re: [Asterisk-Users] Call Queue NOT using RoundRobin ?!?

ally calls are rerouted to the backup operator after ringing to the first). Hope you can find out something to share, maybe we can also launch a "count us" initiative :)Alessio Focardi On 6/29/06, Aaron Paxson <[EMAIL PROTECTED]> wrote: I have setup

[Asterisk-Users] Call Queue NOT using RoundRobin ?!?

I have setup several Calling Queues, each setup with RoundRobin strategy.   When I call the queue, the first member/agent phone rings.  Great!  I call it again, the second member/agent rings??   I thought that was the RRMemory strategy, but it seems RoundRobin is also doing it.   Anyone know

Re: [Asterisk-Users] Polycom - missed calls dial back

Well, if you need to add a 9, because it's an "outside line", then just create a pattern for it.  Anything that is 7 or 10 digits "MUST" be an outside line (assuming your internal extensions are < 7 or 10).   exten => _XXX,1,Dial(9${EXTEN}) exten => _XX,1,Dial(9${EXTEN})   I'm no

Re: [Asterisk-Users] Default dialplan??

ill match any 3-digit number and send it to the other PBX. Should work... [Normal] include => secondary_pbx exten => 101,1,Dial(sip/101) [secondary_pbx] exten => _XXX,Dial(Zap/g1) Aaron Paxson wrote: Hey all! I've got my Asterisk box tied into my PBX. Currently, if a call comes

[Asterisk-Users] Default dialplan??

Hey all!   I've got my Asterisk box tied into my PBX.  Currently, if a call comes into my PBX, and can't find the extension, it forwards it through my Asterisk trunk to Asterisk.   This works great!   Is there a special dialplan function (or common usage pattern) that can do the same thing in