[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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller Prompts in a Queue??
DOH!! I didn't see the context option, or how to use it. Thanks Matt!! ~~Aaron - Original Message - From: Matt [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, July 05, 2006 3:41 PM Subject: Re: [asterisk-users] Caller Prompts in a Queue?? Yes, it is available out of the box. Have a look at the wiki and queues. Something like this should do the trick: [201] wrapuptime=30 timeout=30 strategy=rrmemory retry=15 queue-youarenext=queue-youarenext queue-thereare=queue-thereare queue-thankyou=custom/YOUR-MENU-FOR-CUSTOMER-HERE queue-callswaiting=queue-callswaiting music=default monitor-join=yes monitor-format=wav member=Agent/1231 member=Agent/1016 member=Agent/1001 member=Agent/1006 member=Agent/1000 member=Agent/1063 member=Agent/1045 member=Agent/1036 member=Agent/1022 member=Agent/1011 member=Agent/1012 member=Agent/1014 maxlen=0 leavewhenempty=no joinempty=Yes context=YOUR-CONTEXT-FOR-MENU-HERE announce-holdtime=yes announce-frequency=60 Just record the prompt.. it will be played once every X minutes. And, at any time while on hold, if the customer presses a button in your 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 voicemail, rather than waiting. Is this available out-of-the-box, without writing an AGI? Thanks!! ~~Aaron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 show up. Zaptel: # make clean # make linux26 # make install This is good. I've modprobe'd the cards, and everything comes up: # lsmod | grep zaptel zaptel 196740 1 wcte11xp crc_ccitt6081 2 zaptel,hisax So, I then re-compiled asterisk, so it can build the chan_zap.so: # make clean # make make install But the chan_zap.so module never gets built. What could I be missing? Thanks! ~~Aaron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 To: asterisk-users@lists.digium.com Sent: Friday, June 30, 2006 2:45 PM Subject: [Asterisk-Users] Switchtype Our PRI vendor is using a Nortel DMS500 switch. Which switch type should I use. I have been using national but we are having issues with our connectivity. national dms100 4ess 5ess euroisdn ni1 qsig Thank You James Hawks ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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: [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 show up. Zaptel: # make clean # make linux26 # make install This is good. I've modprobe'd the cards, and everything comes up: # lsmod | grep zaptel zaptel 196740 1 wcte11xp crc_ccitt6081 2 zaptel,hisax So, I then re-compiled asterisk, so it can build the chan_zap.so: # make clean # make make install But the chan_zap.so module never gets built. What could I be missing? Thanks! ~~Aaron ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 what I can do to my queues, in order to force each call down the ordering of my members list? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Queue NOT using RoundRobin ?!?
The linear function helps me too. I've built an extensive multi-queue technical support system strategy. Based on the initial queue, ALL calls goes to Tier1 first. Then, if Tier1 does not get the call (on the phone/away from desk), Tier2 should get it, so on, and so forth. In Tier1, the primary helpdesk technician (like your receptionist idea) takes ALL calls (That's what they were hired for). However, others can help out, if the pri technician is on the phone. Here's my question: If roundrobin strategy remembers the last call made, and sends the next call to the next number (and this is by design), then why on earth was the RRMemory strategy created?? Thanks for your response, Alessio. ~~Aaron - Original Message - From: Alessio Focardi To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Sent: Thursday, June 29, 2006 1:31 PM Subject: Re: [Asterisk-Users] Call Queue NOT using RoundRobin ?!? Welcome to my personal hell ! :)I'have been discussing this previously on the list and also with some digium staff: to my experience there is NO way to archieve a linear distribution of calls from a queue.I mean When a call comes in first member of the queue is ring, then second, etcSubsequent calls take the same path: first, second and so on.Someone has suggested to use "ringall" with penalties (pretty esotic!) but also this is not working for the purpose. I was also told that "nobody wants that" (you insensitive clod!) even if this call distribution seems pretty logic in some case scenarios. (hint: a receptionist is first member of a queue and another person is the second ... receptionist goes for a pee and magically 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 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 what I can do to my queues, in order to force each call down the ordering of my members list?___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Queue NOT using RoundRobin ?!?
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 used. Is app_queue.c the file that does the calling? - Original Message - From: Alessio Focardi To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Sent: Thursday, June 29, 2006 2:07 PM Subject: Re: [Asterisk-Users] Call Queue NOT using RoundRobin ?!? Will you (or anyone else) be able to code this proposed "circular" or "linear" (what sounds more appropriate?) strategy and submit it for inclusion in HEAD ?Should be pretty easy, unfortunately I have very few programming skills. Regards !P.S.here is a snippet from the wiki, whatever it means ! :)roundrobin mode remembers the last agent it _started_ with for a new call, and starts with the next agent in the list. If you have three agents, the first call will go to agent 1-2-3, the next call will go to 2-3-1, the next call will go to 3-2-1, etc. rrmemory mode remembers the last agent it tried to _call_, regardless of who it started with, so that the next call will go the agent after the last one who answered. If you have three agents and the first call rings 1-2 (and is answered), then the next call will ring 3-1 (and is answered), 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: The linear function helps me too. I've built an extensive multi-queue technical support system strategy. Based on the initial queue, ALL calls goes to Tier1 first. Then, if Tier1 does not get the call (on the phone/away from desk), Tier2 should get it, so on, and so forth. In Tier1, the primary helpdesk technician (like your receptionist idea) takes ALL calls (That's what they were hired for). However, others can help out, if the pri technician is on the phone. Here's my question: If roundrobin strategy remembers the last call made, and sends the next call to the next number (and this is by design), then why on earth was the RRMemory strategy created?? Thanks for your response, Alessio. ~~Aaron - Original Message - From: Alessio Focardi To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Sent: Thursday, June 29, 2006 1:31 PM Subject: Re: [Asterisk-Users] Call Queue NOT using RoundRobin ?!? Welcome to my personal hell ! :)I'have been discussing this previously on the list and also with some digium staff: to my experience there is NO way to archieve a linear distribution of calls from a queue.I mean When a call comes in first member of the queue is ring, then second, etcSubsequent calls take the same path: first, second and so on.Someone has suggested to use "ringall" with penalties (pretty esotic!) but also this is not working for the purpose. I was also told that "nobody wants that" (you insensitive clod!) even if this call distribution seems pretty logic in some case scenarios. (hint: a receptionist is first member of a queue and another person is the second ... receptionist goes for a pee and magically 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 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 what I can do to my queues, in order to force each call down the ordering of my members list?___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _
Re: [Asterisk-Users] Call Queue NOT using RoundRobin ?!?
-Outstanding. I missed that one. I'll check out HEAD tomorrow, and apply the patch. We'll see if it works. Thanks Michael! ~~Aaron On Jun 29, 2006, at 3:44 PM, Michael Konietzny wrote: hey, a patch for linear mode is posted to bugs.digium.com already: http://bugs.digium.com/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 used. Is app_queue.c the file that does the calling? - Original Message - *From:* Alessio Focardi mailto:[EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com *Cc:* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] *Sent:* Thursday, June 29, 2006 2:07 PM *Subject:* Re: [Asterisk-Users] Call Queue NOT using RoundRobin ?!? Will you (or anyone else) be able to code this proposed circular or linear (what sounds more appropriate?) strategy and submit it for inclusion in HEAD ? Should be pretty easy, unfortunately I have very few programming skills. Regards ! P.S. here is a snippet from the wiki, whatever it means ! :) roundrobin mode remembers the last agent it _started_ with for a new call, and starts with the next agent in the list. If you have three agents, the first call will go to agent 1-2-3, the next call will go to 2-3-1, the next call will go to 3-2-1, etc. rrmemory mode remembers the last agent it tried to _call_, regardless of who it started with, so that the next call will go the agent after the last one who answered. If you have three agents and the first call rings 1-2 (and is answered), then the next call will ring 3-1 (and is answered), 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] mailto:[EMAIL PROTECTED] wrote: The linear function helps me too. I've built an extensive multi-queue technical support system strategy. Based on the initial queue, ALL calls goes to Tier1 first. Then, if Tier1 does not get the call (on the phone/away from desk), Tier2 should get it, so on, and so forth. In Tier1, the primary helpdesk technician (like your receptionist idea) takes ALL calls (That's what they were hired for). However, others can help out, if the pri technician is on the phone. Here's my question: If roundrobin strategy remembers the last call made, and sends the next call to the next number (and this is by design), then why on earth was the RRMemory strategy created?? Thanks for your response, Alessio. ~~Aaron - Original Message - *From:* Alessio Focardi mailto:[EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com *Cc:* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] *Sent:* Thursday, June 29, 2006 1:31 PM *Subject:* Re: [Asterisk-Users] Call Queue NOT using RoundRobin ?!? Welcome to my personal hell ! :) I'have been discussing this previously on the list and also with some digium staff: to my experience there is NO way to archieve a linear distribution of calls from a queue. I mean When a call comes in first member of the queue is ring, then second, etc Subsequent calls take the same path: first, second and so on. Someone has suggested to use ringall with penalties (pretty esotic!) but also this is not working for the purpose. I was also told that nobody wants that (you insensitive clod!) even if this call distribution seems pretty logic in some case scenarios. (hint: a receptionist is first member of a queue and another person is the second ... receptionist goes for a pee and magically 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] mailto:[EMAIL PROTECTED] wrote: 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
[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 Asterisk? i.e. If it can't find the extension, send it out Zap/g1? My dialplan works with patterns, but patterns isn't what I need here. Is anyone doing anything like this? Thanks! ~~Aaron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Default dialplan??
Thanks Cosmin!! I didn't realize that the dialplans ran in sequential order. I'll try that. thanks! --Aaron - Original Message - From: Cosmin Prund [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, May 18, 2006 9:44 AM Subject: Re: [Asterisk-Users] Default dialplan?? I'll give this one a try, but don't trust me, test it yourself :-) Of course Asterisk can do it! All you need to do is set up a rule for matching ALL extensions in the PBX in it's own separate context and include that context into your normal context. In the following example, asterisk will try matching all extensions in context Normal (all extensions defined on *) and, if no match was found, start searching the context secondary_pbx. In my sample this secondary context will 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 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 Asterisk? i.e. If it can't find the extension, send it out Zap/g1? My dialplan works with patterns, but patterns isn't what I need here. Is anyone doing anything like this? Thanks! ~~Aaron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 not sure why this is such a bad idea.
It's two lines, in a centralized dialplan. Rather than searching endless
XML labels across all your Polycoms.
- Original Message -
From:
Chad
Osmond
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, May 18, 2006 3:01
PM
Subject: RE: [Asterisk-Users] Polycom -
missed calls dial back
Prepend outgoing calls by using Dial(9{exten}) instead of dialing 9 to
get out on your system...
Or, add a 9 to caller id.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill
GibbsSent: May 18, 2006 11:37 AMTo: Asterisk Users
Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users]
Polycom - missed calls dial back
This is not necessarily Asterisk
specific but if I have Polycom 301/501 and 601s and want to dial a missed call
back, how do I prepend a 9 can I do this via the polycom config? I
cant find anything in the docs.
Bill
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