[asterisk-users] Penalty based Cascading Queue - possible ?
Hi, folks. I have a call queue called 'support' with several members, for example : [support] member = Agent/65,5 member = Agent/78,5 member = Agent/74,1 member = Agent/62,1 With this configuration, I can configure an extension to send calls to agent 74 and 62 when they are logged in, and calls to Agent 65 and 78 when the first agents are busy, or not logged in. This works perfectly. I would like to configure a queue such that if Agents 74 and 62 do not answer, then the call is then presented to all of the four agents. This is described on voip-info as a cascading queue, and it's normally configured such that an extension to call the queue is made with a timeout, and if this fails, the call is presented to an alternative queue. This is far from ideal. I would like the call to be presented to the *same* queue, but to be able to specify a penalty that is associated with which queue members receive the call, e.g. exten = s,1,Answer exten = s,n,Queue(support|t|||10) -- penalty 1 gets the call this time .. exten = s,n,Queue(support) --- but somehow specify penalty 5 and below here Is this possible ? Many thanks Andy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie: How to remote test a call prolem in an Asterisk site?
On 1 May 2008, at 08:17, Lee, John (Sydney) wrote: I will be installing Asterisk in a few offices which I don't have any colleagues over there to help me. Let's suppose I installed Asterisk in such a site. I tested it to my satisfaction and I went back to my home office. One day, a customer called me to say that he had a problem calling out or something. Is there any way I could test out the problem myself remotely (apart from READING the message on the console when the customer tested) or I just have to believe what the customer tells me? Can anyone share their experience with me please? Send the logs to a syslog server. logger.conf [logfiles] syslog.local0 = debug, warning, error, notice, verbose ... then configure a syslog program on the asterisk box to send syslogs to your centralised syslog server that you use for clients you support. You will then be able to see the log messages generated on your own equipment, without needing access to the asterisk box. However, you will need to log into the asterisk box to make changes as per your customers' requirements ! Andy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Preflight check / lint
Hi, Am writing scripts to manage configuration management and Asterisk. I would like to be able to point the asterisk binary at config directory with an asterisk.conf in it, and for asterisk to run a pre- flight check. A bit like a pint check in php, 'apachectl configtest' and lots of other tools. asterisk will then exit with 0 on a safe config, and 1 on a bad config. I can reject bad config and stop my config management script in the event of an error. Looking at the man page, it looks like this feature is missing. Anyone got another tool which can do this instead ? Best wishes, Andy Davidson ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question on G.729 (was: H.264 *Not Patented*)
On 1 Feb 2007, at 14:14, Lacy Moore - Aspendora wrote: On 2/1/07, Andy Davidson [EMAIL PROTECTED] wrote: What I would expect to happen, is that Asterisk would transcode between the ulaw/alaw party, and me, wanting to listen via g729. Is this what *should* happen ? Worth noting that my provider does not support G.729. Is what is happening a bug ? Any patches I can try to see if they work ? Or is it my config which is broken ? How many g729 licenses do you have? Just one - my interpretation was that one license bought one inbound, and one outbound transcoder, so my scenario would work with this (phone and * talk g.729, then * turns g.729 into ulaw for my upstream..) Do I need to buy more licenses ? -a ___ --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] Question on G.729 (was: H.264 *Not Patented*)
Hi, I asked some questions here about G.729 earlier in the week, and it looks like it would fit the bill for compressing audio between my * server in colocation and sip phone at home. This is what I want my setup to look like. (Wont make sense unless you are using a fixed width font) [my phone] [asterisk] [third parties] Snom 360-- v 1.4 - ??? SIP IAX/SIP G.729 Don't care (probably something other than G.729, my preferred supplier today likes ulaw and alaw) My phone sees the * box over a relatively slow consumer connectivity link. The * box is colocated and has excellent connectivity. Therefore the tighter compression between * and my phone is important, hence why I want to use g.729 here. The config for my phone, and my preferred voice supplier looks like this : [[[from sip.conf]]] [andydesk] type=friend context=default secret=xxx host=dynamic dtmfmode=rfc2833 username=andydesk mailbox=1001 vmexten=500 disallow=all allow=g729 allow=alaw allow=ulaw allow=gsm regexten=1001 allowreinvite=no [[[from iax.conf]]] [thing] type=friend host=dynamic username=thing secret=xxx trunking=off bridging=on context=thing disallow=all allow=ulaw allow=alaw allow=gsm When I place a call, the other party's line rings as normal. When the other party answers, I get a sip 'denied' packet, and the call is aborted. Asterisk says : No path to translate from SIP/mydeskphone to IAX/myprovider and Had to drop call because I couldn't make SIP/ mydeskphone ompatible with IAX/myprovider. This looks similar to this bug : http://bugs.digium.com/view.php?id=8781nbn=4 What I would expect to happen, is that Asterisk would transcode between the ulaw/alaw party, and me, wanting to listen via g729. Is this what *should* happen ? Worth noting that my provider does not support G.729. Is what is happening a bug ? Any patches I can try to see if they work ? Or is it my config which is broken ? Inbound calls work ok, I guess this is because they are presented as alaw and asterisk is just passing them through (which of course isn't what i really want). Thanks for any suggestions, Andy ___ --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] Queue Dial Plan
On 31 Jan 2007, at 14:32, Rob Schall wrote: Perfect. Here's a quick and hopefully doable followup question. We have Polycom Soundpoint 501 phones. Is there a way to have a phone check 2 voicemail boxes? If we have a queue, and we want the MWI to show for say that users's extension 1000 and the special billing vm box of 2000. In the absence of any other replies, I'll mention that I'm pretty sure that the Snoms will check multiple voicemail-boxes for the MWI if you comma-seperate the mailbox= options in sip.conf e.g. [EMAIL PROTECTED],[EMAIL PROTECTED] Give that a try ! Cheers -a -- Regards, Andy Davidson http://www.devonshire.it/ - 0844 704 704 7 - Sheffield, UK ___ --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] H.264 *Not Patented*
On 27 Jan 2007, at 16:33, Lee Jenkins wrote: Although I wouldn't complain about a free G.729 codec, I have to be honest in saying that $10.00 isn't that great of an expense considering the better call quality you get. Does G.729 work by pushing up the compression, therefore moving from work from the Network to the CPU ? Across my LAN, I'd probably be able to handle *fewer*, rather than more calls across my * exchange if this was the case. If it's cleverer that this, I think I'll have to speculate a few dollars, assuming my Snoms can talk in G.729. cheers -a -- Regards, Andy Davidson http://www.devonshire.it/ - 0844 704 704 7 - Sheffield, UK ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hello Everybody, my problem with voicemail.conf
On 26 Jan 2007, at 10:43, Ashish Barot wrote:
Upto this moment the voicemail is generating, but it is not e-mail
to any email id. But it comes on [EMAIL PROTECTED]
[...]
[worldbiz]
exten = _111X,1,Dial(SIP/${EXTEN},4)
exten = s-BUSY,2,Goto(s,1)
exten = _111X,2,VoiceMail([EMAIL PROTECTED])
[...]
[worldbiz]
= 1234,Barot,[EMAIL PROTECTED]
1112 = 1234,Ashish Barot,[EMAIL PROTECTED],,attach=yes|format=wav
All of your extensions are configured to pass voicemail into mailbox
- this is configured to send VM notifications to [EMAIL PROTECTED]
(i.e. no voicemail is ever hitting mailbox 1112).
-a
--
Regards, Andy Davidson
http://www.devonshire.it/ - 0844 704 704 7 - Sheffield, UK
___
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