Hi,
I have a couple of questions about MeetMe and call queues. I'm still
pretty new to Asterisk, but already having to write a Service Center
call manager for it (which I might add, our director has agreed to make
open source!).
MeetMe:
How can I get MeetMe (does it even do this) to ask the user to speak
their name first, and play that as the new member announcement. It seems
like a common feature in most hardware PBX systems we've used that
support Call Conferences.
Has anyone found a way of doing this? Is there an alternative to MeetMe
that would support this feature (that's as good if not better?).
Queues:
I'm running the 1.0 stable from the cvs server, and I've added the queue
status announcement directives to the queues.conf - yet asterisk gives
me the following errors:
Apr 21 11:22:58 WARNING[950286]: Unknown keyword in queue 'Sales':
monitor-format at line 9 of queue.conf
Apr 21 11:22:58 WARNING[950286]: Unknown keyword in queue 'Sales':
announce-frequency at line 10 of queue.conf
Apr 21 11:22:58 WARNING[950286]: Unknown keyword in queue 'Sales':
announce-holdtime at line 11 of queue.conf
Apr 21 11:22:58 WARNING[950286]: Unknown keyword in queue 'Sales':
queue-youarenext at line 12 of queue.conf
Apr 21 11:22:58 WARNING[950286]: Unknown keyword in queue 'Sales':
queue-thereare at line 13 of queue.conf
Apr 21 11:22:58 WARNING[950286]: Unknown keyword in queue 'Sales':
queue-callswaiting at line 14 of queue.conf
Apr 21 11:22:58 WARNING[950286]: Unknown keyword in queue 'Sales':
queue-holdtime at line 15 of queue.conf
Apr 21 11:22:58 WARNING[950286]: Unknown keyword in queue 'Sales':
queue-minutes at line 16 of queue.conf
Apr 21 11:22:58 WARNING[950286]: Unknown keyword in queue 'Sales':
queue-thankyou at line 17 of queue.conf
These directives I found in the asterisk wiki!
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On Wed, 21 Apr 2004, Ben Merrills waxed:> Hi, > > I have a couple of questions about MeetMe and call queues. I'm still > pretty new to Asterisk, but already having to write a Service Center > call manager for it (which I might add, our director has agreed to make > open source!).That's great news.> MeetMe: > > How can I get MeetMe (does it even do this) to ask the user to speak > their name first, and play that as the new member announcement. It seems > like a common feature in most hardware PBX systems we've used that > support Call Conferences. > > Has anyone found a way of doing this? Is there an alternative to MeetMe > that would support this feature (that's as good if not better?).I don't think this is currently supported, could be wrong, tho. Would take some modification to app_meetme.c, or else just have people say there name when it beeps them in. :) Sort of the flip side, but maybe it would be more helpful to have the person entering the conference hear the name of everyone already in it. That could be done via Record and Playback apps, before executing MeetMe. Every time someone enters, have Record take their name. Then, run Playback for each of the Recorded files.> Queues: > > I'm running the 1.0 stable from the cvs server, and I've added the queue > status announcement directives to the queues.conf - yet asterisk gives > me the following errors: > > Apr 21 11:22:58 WARNING[950286]: Unknown keyword in queue 'Sales': > monitor-format at line 9 of queue.confI think this only works in development, not stable, CVS. :( --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May
Hi All,
As a company, we are looking to rationalize our phone system
infrastructure and have come across using a digium quad port E1 PRI
cards in conjunction with the Asterisk PBX software. I'm hoping you'll
be able to answer the following questions and maybe give me a few
configuration hints.
Presently I have an Asterisk installation using a Fritz card and a BRI
line for testing, and unfortunately, I don't have any DDI's configured
on the BRI line.
We have several C/T servers with PRI lines that are under utilised, in
the following configuration
eISDN -> PRI -> C/T Server 1
eISDN -> PRI -> C/T Server 2
eISDN -> PRI -> C/T Server 3
We wish to use Asterisk as a switch to direct calls (based on the
dialled number) to the correct C/T server
-> PRI -> C/T Server 1
/
eISDN -> PRI -> Asterisk --> PRI -> C/T Server 2
\
-> PRI -> C/T Server 3
For our C/T applications we need the Dialed Number passing from the PRI
to the C/T server - is this possible ?
If we install 2 or more of the Quad port ISDN cards, and a call came in
on the first card, but was re-directed out of a second card, is there a
dedicated bus between the cards (as with Dialogic cards) or would it use
the Server's PCI bus ?
Do you have any idea of the extra load this would put on the CPU ?
We also have a Samsung DCS phone switch that connects to 4 BRI lines, do
you have or know of any product that will work with asterisk and allow
us to connect this to the Asterisk server ?
Ie FROM:-
eISDN 4xBRI -> DCS
To :-
Asterisk -> 4xBRI -> DCS
Thanks in advance for any information
Mark Wilkinson
2PM Technologies Ltd.
clive18@webmail.co.za
2004-Aug-04 06:17 UTC
[Asterisk-Users] A few questions - isdn call routing
On Wed, 4 Aug 2004 13:34:02 +0200 (CEST) Peter Svensson <psvasterisk@psv.nu> wrote:> On Tue, 3 Aug 2004 clive18@webmail.co.za wrote: > > > There is a device called a "parlay" made by a crowd > called > > voxtream which will route the ISDN calls based on the > DID > > and/or the callerid, before the call is answered. > > > > It would be nice if this feature could be done in > Asterisk > > as well, but at this point in time, it first answers > the > > call. > > Are you sure about this? When I looked at the traces on > our setup it seems > that CONNECT was only sent on the incoming leg after it > was received from > the outgoing leg. As a graph: > > > pstn <-pri-> asterisk <-pri-> other_device > > > pstn asterisk other device > -SETUP-> > dial(...) > <-PROCEEDING- > -SETUP-> > <-ALERTING- > <-PROCEEDING- > <-ALERTING- > <-CONNECT > -CONNECT ACK-> > <-CONNECT- > -CONNECT ACK-> > > > Peter >Peter, if thats correct, then thats great! Clive> > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users _____________________________________________________________________ For super low premiums ,click here http://www.dialdirect.co.za/quote
I've been watching the mailing list for a few days, have done some archive searching and still have a handfull of questions (I've looked for most of these on voip-info.org and a slew of other "asterisk related" sites). I'm going to throw the ones that are foremost in this email and will add more as the need arises. Before I do so, let me explain my setup. I have Asterisk running on a FreeBSD machine that is also my router/firewall and MySQL server. It is running fine and I've gotten it working with FWD and will be testing a direct IAX server in the next few days. I'm migrating from a Packet8 Virtual Office setup and have managed to get their "DTA-310" working on my installation. Here are my questions.- 1. Does anyone have suggestions for license-friendly MOH sources? Same for reworks of the voicemail and autoattendant prompting? 2. Help! I got the MWI light on the phone (an Astra powered by the DTA-310) but now it won't go off. 3. Is there any way to have asterisk take a phone back to a plain dialtone instead of a fast busy when a call ends? 4. Even though I've got the basics working I keep wondering what else is available. For example I see on the * website that things like transferring and web access to voicemail are available but I don't even know where to begin looking for that stuff. Where are the guides that I'm missing for all of the different configuration issues? 5. All my other boxes are Windows machines - can someone recommend a config tool that I can run on Windows to help me get everything straightened out? 6. What hardware is really needed to bring in a copper pair? I have a single CO line that we're using for faxes and I'd like to be able to include it in our outgoing call system for 911 capabilities. At the same time I don't want to throw down a bill for a card. Thanks! Hatton
You should checkout Asterisk@Home, that's what we have installed here. It gives you web access to voicemail as well as a web configuration tool (Asterisk Management Portal). For the POTS line, you will need an FXO card. We are a Digium reseller so I can get you what you need as far as hardware. The Digium TDM01B is a 1 FXO port card that runs around $130. It's about $70 for an additional port FXS or FXO. You can also try the X100P card, look on ebay. I just saw one for about $20. As I am not in anyway an advanced user, I'm not sure on your other questions, I just started using Asterisk about 2 weeks ago. I know there are others who will be able to answer your other questions. If you need anything else let me know. "C. Hatton Humphrey" <chumphrey@gmail.com> wrote:I've been watching the mailing list for a few days, have done some archive searching and still have a handfull of questions (I've looked for most of these on voip-info.org and a slew of other "asterisk related" sites). I'm going to throw the ones that are foremost in this email and will add more as the need arises. Before I do so, let me explain my setup. I have Asterisk running on a FreeBSD machine that is also my router/firewall and MySQL server. It is running fine and I've gotten it working with FWD and will be testing a direct IAX server in the next few days. I'm migrating from a Packet8 Virtual Office setup and have managed to get their "DTA-310" working on my installation. Here are my questions.- 1. Does anyone have suggestions for license-friendly MOH sources? Same for reworks of the voicemail and autoattendant prompting? 2. Help! I got the MWI light on the phone (an Astra powered by the DTA-310) but now it won't go off. 3. Is there any way to have asterisk take a phone back to a plain dialtone instead of a fast busy when a call ends? 4. Even though I've got the basics working I keep wondering what else is available. For example I see on the * website that things like transferring and web access to voicemail are available but I don't even know where to begin looking for that stuff. Where are the guides that I'm missing for all of the different configuration issues? 5. All my other boxes are Windows machines - can someone recommend a config tool that I can run on Windows to help me get everything straightened out? 6. What hardware is really needed to bring in a copper pair? I have a single CO line that we're using for faxes and I'd like to be able to include it in our outgoing call system for 911 capabilities. At the same time I don't want to throw down a bill for a card. Thanks! Hatton _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050605/36122b75/attachment.htm
adam.collard@sbcglobal.net wrote: [SNIP]> line, you will need an FXO card. We are a _Digium reseller_ so I can get[SNIP]> other questions, I just started using Asterisk about _2 weeks ago_. I know[SNIP] How do you end up being a Digium reseller after using Asterisk for two weeks? Do you plan to provide your customers with support? -- Cheers, Matt Riddell _______________________________________________ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
I've got my own Asterisk tech. He's been with the company for about 2 years, finally convinced me to switch to Asterisk. Matt Riddell <matt.riddell@sineapps.com> wrote:adam.collard@sbcglobal.net wrote: [SNIP]> line, you will need an FXO card. We are a _Digium reseller_ so I can get[SNIP]> other questions, I just started using Asterisk about _2 weeks ago_. I know[SNIP] How do you end up being a Digium reseller after using Asterisk for two weeks? Do you plan to provide your customers with support? -- Cheers, Matt Riddell _______________________________________________ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050606/877867f5/attachment.htm
Dear Sir, Lol, with respect....that is the dumbest idea I have heard today. Your decision to not to invest time is a fallacy. A@H will have you up and running in 30 minutes or your money (it's free) back. Kind Regards, Dean> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of C. Hatton Humphrey > Sent: Monday, 6 June 2005 10:09 AM > Cc: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] A Few Questions > > On 6/5/05, adam.collard@sbcglobal.net <adam.collard@sbcglobal.net>wrote:> > You should checkout Asterisk@Home, > > Thanks for the link - unfortunately A@H won't quite do the trick for > me here as it is a complete OS replacement from what I can tell... I > can't do that. I have too much time and money invested in the box > that I'm running Asterisk on to wipe it and reload. > > Besides that, I already have Asterisk installed and running; maybe my > next step should be to get AMP working on it (which would entail > getting a webserver and whatever other requirements AMP has). > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
On Tue, 7 Jun 2005, marek cervenka wrote:> can you someone post tftp template for gxp-2000? > like > http://www.grandstream.com/DOWNLOAD/Configuration_Tool/Windows/Grandstream_Configuration_File_Template_1.0.6.x.txtI think it will be released with the 1.0.1.9 firmware. You may be able to get it by asking their support for it. YMMW. Peter
we are beginning to test asterisk for our office one of the features of the current phone system that is very heavily used is overhead paging now i came accross this post http://forums.digium.com/viewtopic.php?t=2844&highlight=features that basically says it is not possible with asterisk. let's hope that i am not understanding it right, since i am new to the telephony. can any one help me out and explain it to the unfortunate ? :)) second question is as follows: when you access voice mail the default msg is 'welcome to comedian mail' is there any way to get rid of this par of the greeting ? thanks
buy a grandstream bt101, cut the handpiece cables and connect this to your speakers, program auto answer, you can have as many zoes as you want. ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Kerry Garrison Sent: Fri 12/9/2005 1:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] a few questions On 12/9/05, Kerry Garrison <support@techdatapros.com> wrote:> Overhead paging is totally possible, there are several articles > available on how to do it. But you cannot have multiple zones today > unless you use a sip device that has autoanswer. >>Why can mutilple zones not be done?????????????????????, why do I need asip device at all for the >paging? any of the follwing (and I'm sure>more) will do, even for multiple zones: >* PC Sound Card >* Digum hardware >* any type of ata type gateway (SIP/h323 or whatever else that willinterface with an analog >>>port), even one without auto answerI have yet to see an example of overhead paging with multiple zones using a soundcard, digium hardware, or an ata. -Kerry _______________________________________________ --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 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 4473 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20051209/d3f4a8a5/attachment.bin
Hey - best trick is not to take any of it personally. We all fell off our bikes while learning to ride! (old quote, but still valid) PaulH> Stas Khromoy <stas@edpausa.com> wrote: > > my apologies if anything > but as i said i am not that knowledgeable > and most probably misunderstood the post. > > as it looks from your reply i have > > if you don't mind letting me know what i got wrong > i would greatly appreciate it. > > > > > On 12/9/05, Stas Khromoy <stas@edpausa.com> wrote: > >> we are beginning to test asterisk for our office > >> one of the features of the current phone system that is very heavily > >> used is overhead paging > > > > Overhead paging can be done with asteirsk in anyway you want, you can > > even do mutilple zones, all zones, or whatever you want. > > > >> now i came accross this post > >> http://forums.digium.com/viewtopic.php?t=2844&highlight=features > >> > > > > I cound't find *anything* on that page that has to do with paging. > > > >> that basically says it is not possible with asterisk. > >> > > > > Exactly where on that page??????????????? > > > >> let's hope that i am not understanding it right, since i am new to > the > >> telephony. > >> > >> can any one help me out and explain it to the unfortunate ? :)) > >> > >> second question is as follows: > >> > >> when you access voice mail > >> the default msg is 'welcome to comedian mail' > >> is there any way to get rid of this par of the greeting ? > >> > > > > Yeah, just rerecord that massage. > > > >> thanks > >> > >> > >> _______________________________________________ > >> --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