[asterisk-users] Moderated News Aggregation for Asterisk
Hi all. Just wanted to let people know about a small project I started over the weekend to help me keep up with news about Asterisk. http://asterisktimes.xdev.net/ Some of the other new sites are either not there anymore or slow to update, so I've come up with a different idea for keeping Asterisk news up-to date and in one place. For the moment, I call it Asterisk Times. OK, so maybe not the best name, but it's a work in progress. So what is it? Well, this is an attempt to create a moderated, aggregated news platform for Asterisk. We want developers, 3rd party companies, open source tools, in fact anyone who does anything noteworthy with Asterisk to tell us. And the best way to do it, is by letting us have an RSS feed into your own announcements or news. With that, we can then review and submit news to the aggregator on your behalf, which then shows up on the homepage of this website. I hope people see value in this, as I know I do. This isn't run for profit or commercial reasons, it's just because I think as a community we deserve a better, more frequently updated news site. The URL will change (or at least get its own dedicated URL) once the project is off the ground and I can see people getting value from it. Any suggestions or feedback welcome. Thanks, Ben (aka skrusty on irc) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Index of AGI Scripts
Hi, I wasn't sure quite where to inform people about this, but often I find it hard to find good AGI scripts written by the community, and voip-info is so often out of date. So I create a simple website for people to list their own, or freely available AGI scripts all in one place. http://www.theagigallery.co.uk Any feedback would be welcomed at this time, improvements, issues, general comments etc as I am sure there will be plenty. I've added some that I use and that I've found online already, but I would ask that others add any they know of too, and help build up a nice big free database of AGI scripts for the general Asterisk community. Thanks, Ben -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cdr_odbc with tds
Does anyone know why, using latest cvs head, freetds 0.62.1-0 and unixODBC, when running cdr_odbc, it says it's logged the call successfully, however, when checking the table, nothing is there! I checked through the bug tracker; and a problem very much like mine was in there, with status resolved as of last year (1339). Can anyone shed some light on this please? Cheers, Ben ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
RE: [Asterisk-Users] cdr_odbc with tds
What should I do? :) Add it to the bug tracker? Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sergey Okhapkin Sent: 20 October 2005 16:48 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] cdr_odbc with tds Looks like a bug to me, I just took a look at cdr_odbc.c. I'm not a specialist in ODBC, but what seems to me wrong is the module does INSERT into the database, but does not make COMMIT. On Thu, 2005-10-20 at 16:31 +0100, Ben merrills wrote: Does anyone know why, using latest cvs head, freetds 0.62.1-0 and unixODBC, when running cdr_odbc, it says it's logged the call successfully, however, when checking the table, nothing is there! I checked through the bug tracker; and a problem very much like mine was in there, with status resolved as of last year (1339). Can anyone shed some light on this please? Cheers, Ben ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
RE: [Asterisk-Users] Voicemail while in queue.
In your queue entry, add the line context = voicemail_context make 0 in that context goto voicemail! Should work a treat! Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shad Mortazavi Sent: 11 October 2005 14:44 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voicemail while in queue. Dear Group, I have the following requirement; I would like our users to be able to press 0 while they are in a call queue and have the option of leaving a voicemail, also when nobody is logged in, drop directly to a voicemail box. Is this possible? Thanks Shad Mortazavi -- Nexus Group Technical Manager n|m Nexus Management Inc ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] Not a local SIP domain
Just compiled from cvs head and enabled Real-time config However, I now seem to have an odd problem, which I think Ive tracked to a patch in CVS head that adds domain support to SIP (http://bugs.digium.com/view.php?id=4466). The problem is as following: - Every time a sip peer/friend attempts to register with Asterisks SIP proxy, I get the following error on the console: Not a local SIP domain Has anyone else had this, and can anyone shed some light on it? :s Cheers, Ben Merrills Development Manager tel: 0845 123 fax: 0845 123 2221 email: [EMAIL PROTECTED] Node4 Ltd Millennium Way Pride Park Derby DE24 8HZ www.node4.co.uk Broadband (ADSL/SDSL) - VPN - (Hosted) IP Telephony - Hosting Co-location - Hosted Microsoft Exchange - Hosted Microsoft Office - Web Application Development ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] Asterisk-UK Website
Just if anyone is interested, the asterisk-uk community website is now up. If you're a voip company or service provider operating or based here in the UK, please add your details to the site. If you're an asterisk user or admin, please signup and help us build content for the uk community. The url is http://asterisk-uk.org.uk Cheers, Ben ___ 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
RE: [Asterisk-Users] Cisco 7960 firmware upgrade promblems
And if it fails still, check for a buffer overrun on the configuration file SIPDefault.cnf, the lower firmware versions had less memory assigned for this file during the upgrade process. Caused me all sorts of problems :) Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Geoff Manning Sent: 12 July 2005 16:12 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Cisco 7960 firmware upgrade promblems Sergio Chersovani wrote: I know it's hard to find out infos at the cisco site. Maybe you can open a TAC case Sergio I did find this info: http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20cisco%20 79xx comments_threshold=0comments_offset=0comments_sort_mode=commentDate_d esc comments_maxComments=10comments_parentId=353#threadId358 snip The two phones I purchased had Application Load ID (AKA: firmware) of P003AM30. This is their skinny protocol load. If you're trying to do sIP, you need a load that starts out POS.. You can not upgrade from P00 to P0S, you need to downgrade to P0S30203 to get it using POS firmware, then you can upgrade to the newer releases of the SIP firmware, with one extra thing to know. You do not need to step through every version of he firmware, you can jump versions of firmware, but what you encouter is the issue with their signed binaries (ie: *.sbn files) that they have converted to. If you have both a *.bin and a *.sbn file in the TFTP server root directory, it will default to loading the *.bin (ie: unsigned binary), which you do not want to do, since you need to convert over to signed binaries, in order to continue upgrading to get to the higher versions which only come signed. If you try to load higher version binaries that are not signed, the phone will fail to load and give an error as such (which I dont have the exact verbiage of). So, bottomline, go down to SIP 2.3, then go up to the first signed binary, then go to the final signed binary, then you ought to be there. /snip ___ 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 ___ 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
[Asterisk-Users] RE: [Asterisk-uk] Meet
The feedback we are getting so far has been excellent! As more is decided the list will be updated, if you'd like to be involved in helping, please join us on the IRC channel, #asterisk-uk on irc.freenode.net. If your company would like more involvement with the event, please email me directly. I would really like to hear from people/companies who would like to: - # Exhibit a product or service. # Sponsor a specific area of the event or subject matter. # Be involved in organizing attendees (other large companies, manufacturers, telecoms providers, regulators, speakers etc). # Give a talk or presentation - send me your suggestions, I'd like to hear peoples ideas here! Join the IRC channel for information on ideas in this area we've already had! So far, we are looking at a room(s) at Pride Park Stadium in Derby (Derbyshire), the stadium is very well located, with EMA only 20 minutes away, M1 passing directly by (J23a J24 (South), J28 (North)) and good rail links (the stadium is opposite the station). For those who are thinking of traveling, there is good accommodation, however as more arrangements are made, all this information will be made available. Again, thanks to all who are offering their assistance, I look forwards to hearing from you, and all being well meeting you soon. Cheers, Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael J. Tubby B.Sc (Hons) G8TIC Sent: 22 March 2005 15:31 To: Asterisk UK Community mailing list Cc: Simon Clifton Subject: Re: [Asterisk-uk] Meet - Original Message - From: Peter Bowyer [EMAIL PROTECTED] To: Asterisk UK Community mailing list [EMAIL PROTECTED] Sent: Tuesday, March 22, 2005 1:47 PM Subject: Re: [Asterisk-uk] Meet On Tue, 22 Mar 2005 09:05:41 -, Ben Merrills [EMAIL PROTECTED] wrote: Yes, We have been talking about a meet / mini voip/asterisk-con... thing! You get the idea :) Up till now we've been talking about it on IRC, if you'd like to voice an opinion, please join us (#asterisk-uk irc.freenode.net). We've talked about getting telecom companies together with UK based voip providers (software and hardware). We would like to organize some Asterisk based activities and see some people there with Asterisk related products, so if that's you, get in touch. If you'd like to be involved, please email me directly or visit the irc channel. Hope to hear from people soon, Definitely interested Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] All, You can count myself and my Simonb Clifton (Sales Director, Thorcom Systems) in as well. Regards Mike -- Michael J. Tubby, B.Sc. (Hons) Technical Director Thorcom Systems Limited Tel: +44 1905 888 007 ___ Asterisk-uk mailing list [EMAIL PROTECTED] https://xdev.net/cgi-bin/mailman/listinfo/asterisk-uk IRC Channel: #asterisk-uk on irc.freenode.net - Join us now! ___ 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
RE: [Asterisk-Users] Attended xfer
Does anyone know if the attended transfer in CVS head works with app_queue (and more importantly, chan_agent ?) This is the only thing stopping me from deploying the attended transfer patches. Cheers, Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Benson Sent: 16 February 2005 14:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Attended xfer CVS in a production environment? Is that advisable? [EMAIL PROTECTED] wrote: I am running 1.0.5 and can happily blind xfer from extension to extension, but I can't blind xfer. I have read various snippets about #2 or #8 or other such key combos, but nothing seems to let me do attended xfer. From xlite I can blind xfer without problem but no attended xfer. For attendant transfer you should use CVS Head, in Asterisk stable is not implemented that feature! ___ 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 ___ 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 ___ 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
RE: [Asterisk-Users] # Transfers.
What needs to be done to make this work? For me, this would be the only time we'd really use attended transfers, on the way from an agent to either another agent, or a member of staff. At the moment we have to make all transfers from agents (i.e. queue calls) via blind transfer. Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Kou Sent: 21 January 2005 02:02 To: Asterisk List; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] # Transfers. No, it's doesn't work. Asterisk List on 2005/1/21 01:48 wrote: I have no idea if atxfer works with app_queue/chan_agent. Can anyone try it? Best regards, --JJL44 On Thu, 20 Jan 2005 17:38:25 -, Ben Merrills [EMAIL PROTECTED] wrote: Does this work with app_queue/chan_agent? Cheers, Ben -- Jim Kou IT Engineer Malico Inc. Site: http://www.malico.com.tw No.5, Ming-Lung Road, Yang-Mei, Tao-Yuang, Taiwan 32643 Tel: +886-3-472-8155#2218Fax: +886-3-472-5979 __ ______ ___ _ _ _ ___ ( \/ ) /__\ ( ) (_ _)/ __)( _ ) (_ _)( \( )/ __) )( /(__)\ )(__ _)(_( (__ )(_)(_)(_ ) (( (__ (_/\/\_)(__)(__)()()\___)(_) ()(_)\_)\___)() ___ 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 ___ 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
[Asterisk-Users] Recording a meetme conference
Is it possible to record a meetme conference? What channel would you monitor, is there a main channel that all audio goes too? If so, is it possible to use the ast_monitor (iirc) to record that channel? Cheers, Ben ___ 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
RE: [Asterisk-Users] queue log analyser?
I've not released the source yet, I asked last week on the mailing list for people to send me over some example queue_logs, because so far I've only been able to test the software against my own. I have however made a lot of changes to it since last I posted about it. Template engine has been improved Allows for recursion of a directory of templates Allows for different output directories (so you can do a daily, weekly and monthly all from the same set of templates say) And quite a few other bits As soon as I get some sample data that people don't mind the results being posted for then I can show it off a bit more. Hope to get some sample data soon, Cheers, Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of João Amaro Sent: 20 January 2005 11:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] queue log analyser? -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Ben Merrills wrote: | There's a few (open source/free) ones in development. I myself am | developing one of them. | | Ben | Hi. Why not join all the project in just one ? Actually which queue log analyzers projects are beeing developed ? Check the mail from Ben Merrills sent to the list 14-10-2004 15:10. I don't know if he releases the source code, but, from the screenshots it seems to be a good one. Jo?o Amaro - -- Begin Mail | I've been doing some work on a queue log analyser for a while now, | getting the basics in place, an example of which you can find at | the URL below. However, just wondering what information people | think is most useful in a log analyser? | | At present it includes the following features: | | # Time periods - specify a period of days from the log which you | want to generate statistics for (e.g. only the last 14 days) # | Templating - allows the stats to be inserted into any html/text | template using specific tags to insert stats. This means you could | create a number of templates and execute the analyser against them | to give different information on different pages (quite flexible). | # Specify start and end dates - similar to the first feature, | except you can specify a tight period from your log, not just the | last x number of days # Channels/Agents to names - simple text file | allows you to specify a name, agent number and a channel - e.g. | Ben, Agent/1, Sip/ben. This is then used in the output # instead | of raw data # JPG graphs - includes a custom class to generate line | graphs of information (e.g. hourly call volumes etc) | | What I want to know though is, what output people would like. At | the moment there is an overview of all queues, which includes: | | Total Calls, total connected calls, total abandoned calls, calls | abandoned within x seconds, calls exited with key press, Average | hold time, max hold time, average talk time | | Agent overview includes: Calls taken, Average talk time | | Graph of call volume per hour of the day Graph of call volume per | day (over the period specified) | | Runs under windows (.NET or mono required) or any other OS that | support .NET/mono (Linux, Mac, BSD etc) | | http://muad.xdev.net/Projects/qig/sample.html | | | Not really done anything like this before, so as much input as | possible would be appreciated. | | Cheers, | | Ben -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFB75EfJUm/Bor63CERAv/UAKCwiYZ96RLqX0m7Ks9eL1f7iG4IDQCcCWvK gafg+vLAgQpjl75Hp5y8tug= =PwR8 -END PGP SIGNATURE- ___ 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 ___ 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
RE: [Asterisk-Users] # Transfers.
Does this work with app_queue/chan_agent? Cheers, Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk List Sent: 20 January 2005 17:28 To: Bruce Komito Cc: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] # Transfers. I justed edited the Wiki Asterisk config file features.conf for this attended transfer features. Please check Wiki again for details. Best regards, --JJL44 On Thu, 20 Jan 2005 09:04:40 -0800 (PST), Bruce Komito [EMAIL PROTECTED] wrote: Sorry if I missed the beginning of this thread, but I've never heard of the ** transfer key sequence, nor have I found a way to make it work. Would you mind, please explaining this further or pointing me to somewhere where it's documented? (I checked Wiki and Google but no joy.) Thanks Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ 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 ___ 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
RE: [Asterisk-Users] queue log analyser?
There's a few (open source/free) ones in development. I myself am developing one of them. Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roy Sigurd Karlsbakk Sent: 19 January 2005 15:33 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] queue log analyser? hi I really want a good queue_log analyser, but I don't want to waste EUR 1000 for something like that, so I thought starting a small project for it. I started off in php just creating a basic parser for the log, and I'll go on extending it. See http://karlsbakk.net/asterisk/scripts/queue_log_analyser-0.0.1- pre1.php.txt Does anyone want to join this effort? roy ___ 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 ___ 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
RE: [Asterisk-Users] SS7 and Asterisk solution
Steve, I also would be very interested in getting those details. We would very much like to move forward with SS7, please feel free to contact me off list. Cheers, Ben Merrills Griffin Internet -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Felix Skwarczynski Sent: 14 January 2005 09:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SS7 and Asterisk solution Hi Steve, I also want the commercial details, so if you can send them to me or put me in touch with somebody who can it would be very helpfull. Thank you in advance, Felix Skwarczynski Steve Underwood wrote: Hi Bartosz, We have a commercial SS7 for Asterisk that is running at a few test sites, and which we are just about ready to supply to a broader range of customers. This actually links into Asterisk, so we need to use a commercially licenced copy of Asterisk. If this sounds interesting to you, I can put you in touch with someone who will give you the commercial details. Regards, Steve Bartosz Jozwiak wrote: Hello, We are looking for commercial solution SS7 with Asterisk. It does not need to be build-in with Asterisk. Could anybody suggest something? Thank you in advance. Bart ___ 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 ___ 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 ___ 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
[Asterisk-Users] Queue Log Parser
I don't know if anyone noticed my post a few months ago on the asterisk-user mailing list, but I've been writing a queue log parser. I was wondering if anyone had any queue_logs (the bigger the better) that I would use as sample data? I would of course be willing to post the stats up for the people who send me their sample logs. Post them to me off list, and please tell me if you are willing to have your data posted (in stat form) for demo purposes. Ben ___ 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
RE: [Asterisk-Users] Re: [Asterisk-biz] SS7 and Asterisk solution
When are 'we' going to have this solution Steve? :) You keep talking about it, and we keep asking when it's going to come about. I know myself, SS7 will be a make or break for our continued use of Asterisk. Even if we had some price indications would be good, and/or a timeframe? Don't want to seem pushy, but it's been on the cards for quite some time now. Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: 12 January 2005 12:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: [Asterisk-biz] SS7 and Asterisk solution Tracy R Reed wrote: On Tue, Jan 11, 2005 at 10:44:19PM -0300, Bartosz Jozwiak spake thusly: We are looking for commercial solution SS7 with Asterisk. It does not need to be build-in with Asterisk. Could anybody suggest something? I see a lot of people asking for asterisk and ss7. Just what exactly do these people intend to use SS7 for? There are other platforms you can buy which do speak SS7 for not unreasonable money. Asterisk seems like it might be the wrong place to be putting SS7. For many people SS7 is simply a requirement they have no control over. Asterisk is a perfectly good platform for many SS7 users. You must be very rich of you think many SS7 solutions are not unreasonably priced. On the other hand, we have a very reasonably priced solution for SS7 on Asterisk :-) Steve ___ 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 ___ 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
RE: [Asterisk-Users] Re: [Asterisk-biz] SS7 and Asterisk solution
We'd be more than willing to test :) Our PRI provider has been trying to push us over to SS7 for ages. So we'd be very interested in assisting you. Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: 12 January 2005 12:49 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: [Asterisk-biz] SS7 and Asterisk solution Ben Merrills wrote: When are 'we' going to have this solution Steve? :) You keep talking about it, and we keep asking when it's going to come about. I know myself, SS7 will be a make or break for our continued use of Asterisk. Even if we had some price indications would be good, and/or a timeframe? Don't want to seem pushy, but it's been on the cards for quite some time now. Ben Not on the cards, but on test. Its about time to start wider deployments now. It seems to be running OK at a few sites now. Regards, Steve ___ 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 ___ 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
RE: [Asterisk-Users] SS7 and Asterisk solution
We have the problem that our telecoms provider deals mainly in SS7 (C7, and it seems most in the UK do). For us to take EuroISDN off them, with the same features as SS7, we have to be put through a protocol converter, now this isn't an issue for us, but it is for them. Most UK phone companies (i.e. BT or the smaller regional carriers) all use SS7, everywhere! For the most part they don't accept VoIP termination (although I think BT might have some facilities for this). So they very much try and push SS7 on interconnects. And that's why SS7, for me (and I think for quite a few others taking PRI style links in the UK) is so important. Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: 12 January 2005 17:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SS7 and Asterisk solution We terminate local calls over PRI. Everything else goes out VoIP via SIP to national carriers and they terminate it. Can't you use a channel bank or an FX card to connect to PSTN? Or PRI.. -Matthew - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, January 12, 2005 8:09 AM Subject: Re: [Asterisk-Users] SS7 and Asterisk solution Matthew Boehm wrote: Isn't the goal to move away from SS7? SS7 is pretty old technology. That is what we are doing. We are dropping 2 SS7 carriers and will now send traffic to them directly as SIP. So how do I connect to a PSTN line by SIP? :-) SS7 is the basis for the entire world's telephone network signalling. New forms which run over IP are being deployed. Even if the last remaining fragments of the PSTN are shut down, and everything on land lines is IP based, SS7 is still the core protocol for the cellular networks - GSM-A, for example, is built upon SS7. Of course, its always possible WiMAX might shut those down too. :-) Regards, Steve ___ 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 ___ 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 ___ 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
[Asterisk-Users] UK * group
Is there a UK Asterisk users group? Would be interested in contacting others in the UK who use asterisk for either home or business applications. If there is, could someone provide me with some contact details, else anyone whos also interested, contact me off list. Cheers, Ben Merrills ___ 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
[Asterisk-Users] Asterisk UK Community
To those who are/were interested J Ok, Ive setup a mailman mailing list. I dont so this often, so Ive done a bad job of it, please let me know. Oh, and I know the SSL cert has expired was self signed J Now Ive apologised for my crapness, the URL to signup to the mailing list is: https://xdev.net/cgi-bin/mailman/listinfo/asterisk-uk Hope this is ok with people, once were all on there, we can talk about how we want to take this further, from speaking to you individually, sounds like weve got an excellent group of people here! Cheers, Ben Ps. This was just in case I missed anyone who emailed me off list. Please ensure all replies again are off list! Thanks :D ___ 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
RE: [Asterisk-Users] Queues without members
When a new member is rejected from the queue (because there's a limit, or there's no agents logged into the queue), is it possible to either set an announcement, or to elevate the caller to a new priority (i.e. n+100) or something? Regards Ben Merrills -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas Roedl Sent: 21 December 2004 14:52 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Queues without members Hello! Am Dienstag, 21. Dezember 2004 13:59 schrieb Senad Jordanovic: It seems that Queue() won't continue at a specific priority - like n+101 - if there are no members in the queue. Use... Joinempty=yes Perfect! Thanks. Andi -- - Andreas Roedl- Senior IT Manager - NATIVE INSTRUMENTS GmbH - [EMAIL PROTECTED] - Schlesische Strasse 28 - http://www.native-instruments.de/ - D-10997 Berlin - Tel. +49-30-61 10 35-430 - Germany - Fax +49-30-61 10 35-35 ___ 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 ___ 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
RE: [Asterisk-Users] Queue without #
ackcall=no in agents.conf -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peer Oliver Schmidt Sent: 03 December 2004 12:49 To: asterisk-users ML Subject: [Asterisk-Users] Queue without # Hi, I want to run a queue with CallBacklogin which works fine. However, I want the system to directly connect without the user having to press # Ideas anyone?! TIA rgds pos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
Sorry to ask about this again, but Im trying to get my head around a few things. If a call is redirected from a PRI (quad e1) back out into the PRI (same quad e1) what processes does asterisk undertake on the call? Were getting terrible quality issues (and only with redirected calls). Is the bridging done at the PRI interface level? Or does this require that the call be constantly running through Asterisk? I.e. would asterisk count for any of the quality issues? I noticed that CAPI had some low level redirection instructions, does EuroISDN/E1 posses the same qualities? Sorry if this is a bit confusing, just trying to get my head round it myself! Cheers, Ben ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI Logging
Is there a way to log all PRI events to a logfile? Cheers, Ben Merrills Griffin Internet T: 0870 8040862 F: 0870 8040805 W: www.griffin.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PRI Logging
But there's no way of just specifying PRI events? I'd prefer to have a logfile that simple had all the PRI output (e.g. output of pri debug span n) Cheers for the suggestions though, Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Svensson Sent: 23 November 2004 13:52 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] PRI Logging On Tue, 23 Nov 2004, Ben Merrills wrote: Is there a way to log all PRI events to a logfile? Maybe pri intense debug span ??? is what you are after? If you set up a logging file in /etc/asterisk/logger.conf that logs everyting you should get all the pri events. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR Question
How can I tell the dialled number from CDR records? We need to be able to bill our provider based on the dialled number. Is this possible? Ben Merrills Griffin Internet T: 0870 8040862 F: 0870 8040805 W: www.griffin.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dropping incompatible voice frame
I keep getting console messages like the following: Nov 17 11:22:33 NOTICE[16767]: channel.c:1320 ast_read: Dropping incompatible voice frame on SIP/sipuser-5240 of format ALAW since our native format has changed to ULAW I think this is causing some incoming queue calls to ring on an agents extension, however when they pick up the phone, no one is there and the caller gets taken off onhold music, however they are still shown in the queue (show queues). Sometimes, after maybe 30 seconds the caller is bridged to the agent, and sometimes the channel is dropped. Anyone know why this might be happening? All SIP phones are registered to use ULAW. Cheers, Ben Merrills Griffin Internet T: 0870 8040862 F: 0870 8040805 W: www.griffin.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PSTN - Asterisk - PSTN Call quality
Hi there, Having some issues with call quality when taking calls from E1, using Asterisk to reroute the call back out onto E1. Sometimes theres quite a big echo and others the line is just very scratchy. Call quality for incoming calls to VoIP is fine. To redirect the incoming call I use an AGI that fires off the Dial command to redial the extension back out over PSTN. Is this the right way to redirect a call out over PSTN? Would doing this cause any kind of call quality loss? Cheers for any suggestions or help, Ben Griffin Internet T: 0870 8040862 F: 0870 8040805 W: www.griffin.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PSTN - Asterisk - PSTN Call quality
Hmm, When we first got our TE410p (we have 2 at the minute) it was in a normal IDE ATA100 box with a P4, and the static on the line was really, REALLY bad. We don't have this issue now we use a Compaq with SCSI, unless we reroute the call back onto PSTN. It doesn't always happen I might add though! It seems to be some landlines that have more problems than others. Mobiles tend to be fine. Is this a Digital - Analogue issue? Strange that it should be fine the rest of the time :( Yes, a hardware guide for Asterisk would be a god send! I'm getting two new DELLs next week to play with (SCSI again). I'll let you know how well they work with Asterisk! Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Goryachev Sent: 15 November 2004 12:01 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] PSTN - Asterisk - PSTN Call quality On Mon, 2004-11-15 at 21:18, Ben Merrills wrote: Hi there, Having some issues with call quality when taking calls from E1, using Asterisk to reroute the call back out onto E1. Sometimes there's quite a big echo and others the line is just very scratchy. Call quality for incoming calls to VoIP is fine. To redirect the incoming call I use an AGI that fires off the Dial command to redial the extension back out over PSTN. Is this the right way to redirect a call out over PSTN? Would doing this cause any kind of call quality loss? I had (still have) the same problem. So far, I've found two possible solutions: a) Get a new motherboard b) Get a new motherboard and a new TE410p (instead of my current TE405p) I am still suffering this problem since I can't really afford to buy a new motherboard, cpu, memory besides, how do I know that after buying all that, it will even solve the problem? Looking forward to a motherboard whitelist ? someone? Regards, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] non blind call transfers
Does anyone know if that patch now works with chan_agent.c ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Arvanitis Kostas Sent: 10 November 2004 15:01 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] non blind call transfers On Friday 29 October 2004 23:17, lenz wrote: Hello list, I was looking for a way to implement non-blind call transfers with *, i.e. the usual behaviour of most PBXs when pressing the flash button: - A and B are talking - A pushes flash - A is free to compose a new number - B hears music on hold - A's call is answered by C - A hangs up - B and C are in conversation As much as I can understand, * only supports blind transfers, where if C does not answer the phone there is no way for A to get back to B. Is there a way to have a standard flash behaviour? Yes, * only supports blind (unattended) transfers by default. However there is a patch by anthm (real name) in the Digium bug database (bug number 0002460, http://bugs.digium.com/bug_view_page.php?bug_id=0002460 ) that provides this functionality for the CVS version, and some of the older patches there are usable with the 1.0.1 version of *. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Queue / Agent Problem
Is anyone planning on patching chan_agent.c to reflect the new transfer method (using the patch linked below)? I had a stab at it, but my c skills are next to none :) Cheers, Ben Merrills -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Jackson Sent: 22 October 2004 16:38 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Queue / Agent Problem -Original Message- From: Joseph [mailto:[EMAIL PROTECTED] Sent: Friday, October 22, 2004 11:22 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Queue / Agent Problem But if the agent does a consultive transfer, the queue system thinks the agent still has the call and does not send anymore calls to agent. Is there a work around to this? One is to use the blind (#) transfer, but that is less than professional. Check out bug 2460. It adds attended transfers via '#'. http://bugs.digium.com/bug_view_page.php?bug_id=0002460 Also, your agents could place the call on hold, call the person they are going to transfer the call to, announce the call, hangup, pickup the original call, then transfer via '#'. It may sound complicated, but it works pretty well for us, and it allows the person recieving the transfer to let it go to voicemail or choose to answer it. (Also, posted to the bug tracker.) Hope this helps, Robert Jackson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Manager API / Agents
Is there a way to find out which agents are logged in, without waiting for the Agentcallbacklogin/off event? Id like to be able to get the login status of all Agents when I connect to the Manager API. Is this possible? Cheers, Ben Merrills Griffin Internet T: 0870 8040862 F: 0870 8040805 W: www.griffin.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] (Another) Queue log analyser
Hi there, Cheers for your suggestions, would be great to see the output of some other reports. Logins and logouts are available within the engine, just need to represent them in some way now. What do you suggest would be a good format? Typical duration of login? Only problem might be where someone hasn't logged out before their next login statement (no one here ever logs out, because they're all to slack :) Anything you can send me over would be much appreciated, I have no problems in giving you a pre-release copy so you can give some feedback too. Regards, Ben Merrills Griffin Internet T: 0870 8040862 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wayne Sheppard Sent: 14 October 2004 19:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] (Another) Queue log analyser Very nice work Ben, thanks. Here are some additional thoughts - One segmentation that might be useful would be to add outbound calling activities as a either a separate column or even view. On agent stats, it would be useful to see login/logout stamps, login time, ready/not ready time (if this can be tracked, not sure). If you would like, I can send you some example reports that are used in a typical call center, contact me directly if you would find that helpful. Cheers, Wayne Ben Merrills wrote: I've been doing some work on a queue log analyser for a while now, getting the basics in place, an example of which you can find at the URL below. However, just wondering what information people think is most useful in a log analyser? At present it includes the following features: # Time periods - specify a period of days from the log which you want to generate statistics for (e.g. only the last 14 days) # Templating - allows the stats to be inserted into any html/text template using specific tags to insert stats. This means you could create a number of templates and execute the analyser against them to give different information on different pages (quite flexible). # Specify start and end dates - similar to the first feature, except you can specify a tight period from your log, not just the last x number of days # Channels/Agents to names - simple text file allows you to specify a name, agent number and a channel - e.g. Ben, Agent/1, Sip/ben. This is then used in the output # instead of raw data # JPG graphs - includes a custom class to generate line graphs of information (e.g. hourly call volumes etc) What I want to know though is, what output people would like. At the moment there is an overview of all queues, which includes: Total Calls, total connected calls, total abandoned calls, calls abandoned within x seconds, calls exited with key press, Average hold time, max hold time, average talk time Agent overview includes: Calls taken, Average talk time Graph of call volume per hour of the day Graph of call volume per day (over the period specified) Runs under windows (.NET or mono required) or any other OS that support .NET/mono (Linux, Mac, BSD etc) http://muad.xdev.net/Projects/qig/sample.html Not really done anything like this before, so as much input as possible would be appreciated. Cheers, Ben ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_queue manager API
Is there a way from the manager interface to obtain a listing of all the channels (callers) in a queue? I know as they join/part events are fired, but I'd like to obtain a listing of them when I connect to the manager interface. Any ideas how this can be done? Cheers, Griffin Internet T: 0870 8040862 F: 0870 8040805 W: www.griffin.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] app_queue manager API
Ah cheers, It seems to have changed to add the event ' QueueEntry' from when I last looked at the src. Cheers for your help Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Crick Sent: 15 October 2004 18:04 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] app_queue manager API Ben Check out Action: QueueStatus - it'll list the stats for each queue as well as listing each queue member verbosely. Cheers Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (Another) Queue log analyser
I've been doing some work on a queue log analyser for a while now, getting the basics in place, an example of which you can find at the URL below. However, just wondering what information people think is most useful in a log analyser? At present it includes the following features: # Time periods - specify a period of days from the log which you want to generate statistics for (e.g. only the last 14 days) # Templating - allows the stats to be inserted into any html/text template using specific tags to insert stats. This means you could create a number of templates and execute the analyser against them to give different information on different pages (quite flexible). # Specify start and end dates - similar to the first feature, except you can specify a tight period from your log, not just the last x number of days # Channels/Agents to names - simple text file allows you to specify a name, agent number and a channel - e.g. Ben, Agent/1, Sip/ben. This is then used in the output # instead of raw data # JPG graphs - includes a custom class to generate line graphs of information (e.g. hourly call volumes etc) What I want to know though is, what output people would like. At the moment there is an overview of all queues, which includes: Total Calls, total connected calls, total abandoned calls, calls abandoned within x seconds, calls exited with key press, Average hold time, max hold time, average talk time Agent overview includes: Calls taken, Average talk time Graph of call volume per hour of the day Graph of call volume per day (over the period specified) Runs under windows (.NET or mono required) or any other OS that support .NET/mono (Linux, Mac, BSD etc) http://muad.xdev.net/Projects/qig/sample.html Not really done anything like this before, so as much input as possible would be appreciated. Cheers, Ben ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FYI - Zoom X5v built-in VoIP DSL router
Just thought I would let the list know, as we got our pre release versions today of the new Zoom X5 that supports VoIP. The device comes with an RJ11 phone socket on the back and lets you configure your ADSL router to become a SIP phone (using your existing PSTN phone). Better still, it also allows you to switch the phone between landline and SIP, and does it automatically for incoming calls. No idea what the price of these devices will be when they hit the shops, but setting one up today, if anyone thinks it would be helpful I dont mind doing a little review of the hardware once its tested. Model Number is 5565 Cheers, Ben Merrills Griffin Internet T: 0870 8040862 F: 0870 8040805 W: www.griffin.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_queue
Has anyone else experienced a problem with app_queue where after a time, calls can still come into asterisk, but once they enter a queue, they just get silence, any calls in the queue get frozen in it, and never get sent to an agent, yet calls can be made in or out of the phone system. The other thing is, the CLI deadlocks. Anyone had anything like this before? Cheers, Ben ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Redirecting incoming PRI to PSTN
HI, Id like to redirect an incoming E1 call to a local landline, at the moment I just do Exten = thenumber,1,Dial(Zap/g1/localnumber) However this seems to cause all sorts of problems with the fax machine on the end of that landline. Is there a better way to redirect a call? Cheers, Ben ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 1.0 Mirrors
Is there a release of the zaptel drivers too for 1.0 release? Or should I just get the latest from cvs? Cheers, Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Boehnlein Sent: 23 September 2004 15:21 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] 1.0 Mirrors Hello, Please be conscious of Digium's bandwidth and use a Mirror when downloading 1.0. I have mirrored the tarballs at: ftp://ftp.nacs.net/asterisk/ Direct links: ftp://ftp.nacs.net/asterisk/asterisk-1.0.0.tar.gz ftp://ftp.nacs.net/asterisk/asterisk-sounds-1.0.0.tar.gz ftp://ftp.nacs.net/asterisk/libpri-1.0.0.tar.gz -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_queue cisco transfer
There seems to be a bug in app_queue that prevents calls from reaching agents. If a call is directed to an agent, and that agent transfers the call using the transfer facility on Cisco phones (SIP firmware 7.2) then show agents shows the call still at that same agent. This prevents calls from being sent to that agent for the duration of the transferred call! Is this a bug in app_queue or something more fundamental? Cheers, Ben Merrills Griffin Internet ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queues Transfers
If someone takes a call from a queue on a CISCO 7960, then does a Transfer to another agent (using the transfer button), the queue system seems to think they still have the call, and wont assign them another call till the other agent finishes the transferred call. Is this a known bug? Is it something that can be overcome? Cheers, Ben ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] English vs American voice files
Hash! When we had the American voices on our system a lot of people complained, not only that it was American (no offence to Americans!) but also because of the terminology used, e.g. `pound`. We re-recorded all our voice files and use `hash`. Ben Merrills Griffin Internet -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: 17 September 2004 16:51 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] English vs American voice files Ah, this brings up an interesting point. I've noted that BT are calling # square rather than hash. What do the other providers call it back in Blighty? Before someone goes recording the files we'd better get the language straight. Mark rant Especially when asked to press pound! Pound! This is a pound £ not this # rant-end -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Broken sound in voicemail
Hi, Been using asterisk for a little while now, but were starting to notice that a number of voicemails left have very broken sound. Were using a Digium Quad E1 card, we had some problems with broken audio at first that turned out to be a problem with interrupts, however those issues seem to have been solved. Yet we are getting broken sound in voicemail. Could this be something to do with silence detection? Has anyone else experienced similar problems? Any help much appreciated, Regards, Ben ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Virtual queue member
I was wondering if anyone knew how to do the following Call comes in, gets put into a Queue, say `Sales`. Then the queue member is presented with the option to exit the queue, yet have the phone system sit in their place for them. When the virtual member reaches the front, call back the caller and connect them to the agent. Any ideas? Did i explain that ok? :) Cheers, Ben Merrills skrusty. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI issue
Hi, I recompiled asterisk today from CVS and Ive been having a number of problems, Ive read the deadlock page on the wiki and some of it sounds like that, however, the latest issue were having it that sometimes Asterisk doesnt seem to know the PRI channel has dropped, and assumes its still busy. However, that same channel can be used to make an outgoing call?! Has anyone experienced anything similar? Regards, Ben ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PRI issue
I think this issues stems from the (in my case) wct4xxp driver. When updating libpri, I also updated zaptel, however, I'm unsure if I installed it correctly (i.e. updated to the newly compiled version). After stopping asterisk, doing rmmod wct4xxp, make install on zaptel and then restarting asterisk, so far, it seems to be working. I'm not 100% sure this is the problem, but it would seem this resolved the issue... time will tell :) Cheers, Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian D'Arcy Sent: 08 September 2004 16:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] PRI issue Ben, I ran into a similar issue on the 8/31 cvs, except it was backwards. Outbound calls would report a busy on the channel selected, yet a few minutes later the channel would be used for an inbound call. I had to revert back to my previous checkout from 8/16 to resolve the issue. The problems didn't break the channels completely, it happened probably every 5-10 minutes. Brian D'Arcy From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Merrills Sent: Wednesday, September 08, 2004 7:51 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] PRI issue Hi, I recompiled asterisk today from CVS and I've been having a number of problems, I've read the deadlock page on the wiki and some of it sounds like that, however, the latest issue we're having it that sometimes Asterisk doesn't seem to know the PRI channel has dropped, and assumes it's still busy. However, that same channel can be used to make an outgoing call?! Has anyone experienced anything similar? Regards, Ben ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Broken sound in VoiceMail
It seems voicemail recordings have broken sound. It cuts out randomly throughout the recording. Has anyone had any similar experiences? Ive included some snips of my voicemail.conf Cheers, Ben --SNIP--- [general] ; Default formats for writing Voicemail ;format=g723sf|wav49|wav format=wav ; Who the e-mail notification should appear to come from [EMAIL PROTECTED] ;[EMAIL PROTECTED] ; Should the email contain the voicemail as an attachment attach=yes ; Maximum length of a voicemail message in seconds ;maxmessage=180 ; Minimum length of a voicemail message in seconds ;minmessage=3 ; Maximum length of greetings in seconds ;maxgreet=60 ; How many miliseconds to skip forward/back when rew/ff in message playback skipms=3000 ; How many seconds of silence before we end the recording maxsilence=10 ; Silence threshold (what we consider silence, the lower, the more sensitive) silencethreshold=100 ; Max number of failed login attempts maxlogins=3 ; If you need to have an external program, i.e. /usr/bin/myapp ; called when a voicemail is left, delivered, or your voicemailbox ; changes, uncomment this: ;externnotify=/usr/bin/myapp ; For the directory, you can override the intro file if you want ;directoryintro=dir-intro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transfer to queue
Hi, Using a cisco 7960, if I try and transfer someone using the transfer button, when I transfer them to a queue, it seems to disconnect them. Does anyone know why? I simply have an extension that points to a queue (e.i. exten = 281,1,Queue(Sales) ). Cheers, Ben Merrills ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Empty Queues
Hi, Is there a way to detect if the caller will be entering an agentless queue? Id like to be able to redirect any caller who tried to join a queue with no logged in agents, to be redirected to the groups voicemail. Is this possible? I know I could create a menu and an announcement for voicemail (should the user wish to drop from the queue); but they wouldnt know no one was taking calls :/ Any help much appreciated. Regards, Ben ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Creating 79xx Configs
Sounds good, sounds like a handy thing to have around! :) Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph Sent: 20 August 2004 14:04 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Creating 79xx Configs I made a little php script that creates a 79xx config if you give it the mac address, ext, etc. Is this something that would be of interest to anyone? Likely it could be improved on. And there may be some variations that I have not thot of. -- respectfully, Joseph === -= ** = ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Stream File (AGI) question
Can this be used with alaw audio files? I have an AGI that generates alaw audio files, then tries to use STREAM FILE to get asterisk to play them. The file is created in /var/lib/asterisk/sounds and, if I put Background(filename) in the next priority, it plays fine. I dont quite know why its not playing when I issue the Stream FILE?! Any suggestions? Cheers, Ben Merrills
RE: [Asterisk-Users] Creating 79xx Configs
If you don't have somewhere to host it, drop me an email. Else yeah, just stick it in the wiki, somewhere under the Cisco 79XX section? Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph Sent: 20 August 2004 17:45 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Creating 79xx Configs On Fri, 2004-08-20 at 12:35, Ben Merrills wrote: Sounds good, sounds like a handy thing to have around! :) Ben I don't know where to post it? I could not see a way to put it on the wiki... -- respectfully, Joseph === -= ** = ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Static on outgoing calls using either X100P or TDM400P
Have you had a play with irqmasking on your drivers? Are they IDE drivers? If so, try: hdparm -u1 /dev/drive Do this for each of your IDE drives, this seems to have fixed a number of peoples issues in the past. Also, when you say nothing is sharing an IRQ, is this according to proc/interrupts? Don't forget that unless the kernel has a driver for the device, it wont show up in that listing. Double check using lspci -v Apart from that, get a line test done. Your telcom provider should do one at your request. In the end I changed the box they were running in, and all was resolved. Extreme I know, but it worked :) Cheers, Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lenny Self Sent: 13 August 2004 21:33 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Static on outgoing calls using either X100P or TDM400P Hello. I've seen several posts talking about line quality using Digium cards that are sharing IRQs or on machines where X is running but after trying all of those fixes I am still having a problem with line static on outoing calls. BTW, calls that are from one extension to another extension have no static, however, they have occasional clicks and pops. At any rate, I was wondering if someone might be able to help figure out how to fix this problem. Here is some information about my setup: Celeron 2.4Ghz w/ 256MB of Ram running Mandrake 9.2 with the 2.4.22-36mdk kernel installed with X not installed. This happens to be running on an Intel MB Model (D865GBF) 1 - X100P 1 - TDM400P configured with 4 FXO ports each of these cards have their own IRQ. I am running RC2 *, zaptel drivers, etc. I've fiddled wit hteh tx/rx gain and have succeeded only in decreasing the volume of the static. Any suggestions? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call stealing
Hi, How can I (through the manager interface) steal a call from one phone, and transfer it to another? Does asterisk provide for actions like this? Its a common action in Lucent systems it seems. Cheers, Ben
RE: [Asterisk-Users] Static on outgoing calls using either X100P or TDM400P
I'd just ask for an end to end test, and line survey if needs be. You want them to be testing for static along the call route; your line might in fact be a number of lines tagged together (for example if you have a EuroISDN line). Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ryan Courtnage Sent: 16 August 2004 17:24 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Static on outgoing calls using either X100P or TDM400P Ben Merrills wrote: Hello. I've seen several posts talking about line quality using Digium cards that are sharing IRQs or on machines where X is running but after trying all of those fixes I am still having a problem with line static on outoing calls. Apart from that, get a line test done. Your telcom provider should do one at your request. Any idea what you would ask the telco to test for? I too have seen tdm400ps stop working and output only static noise (requiring wcfxs module reload). The solution to the problem is never the same. Sometimes changing PCI slots, relocating the computer, using a noise reducing powerbar, even changing computers completely. In all cases, I'm not 100% confident that the problem has been resolved for good. If there is a phone-line condition that will trigger the tdm400p driver to go into this non-functional state, I'd love to know what it is, and how to test for it. Thanks Ryan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Static on outgoing calls (Quad E1)
Just FYI, It turned out that the problem was a simple one, but not something you'd expect to be a big issue these says (with ACPI etc): IRQ sharing. There was an onboard gfx card using IRQ 11, the same IRQ as the digium card. lspci -v showed the true story, even though cat /proc/interrupts didn't. Hope that helps anyone else out there with similar issues. Cheers, Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Merrills Sent: 12 August 2004 09:08 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Static on outgoing calls (Quad E1) I placed a call as follows: SIP (Cisco Phone) - Asterisk PRI (outgoing) - Asterisk PRI (incoming) - Sip (Cisco) The call exhibited the same problems as before, static crackle on the line. (Dialled party) I still think this is an issue with the Digium card, but I'm unsure as to what. I've been playing with the rxgain and txgain, although I think this just has the effect of making the call so quiet that there's little noise to hear. Hope this provides some help... confused :( Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andres Sent: 11 August 2004 18:53 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Static on outgoing calls (Quad E1) Ben Merrills wrote: If I call a number from my mobile say, it sounds fine! Nothing is wrong with the call quality at all. If I call asterisk (via the digium card) then route that call out to another mobile, that sounds just as bad as making the call from asterisk... So, to cap off, OK: Mobile - EuroISDN - Asterisk (TE410P E1) - Anything BAD: SIP (Cisco) - Asterisk (TE410P E1) - Anything (Mobile, landline etc) Bad: Mobile - Asterisk (TE410P E1) ReDialed - Anything (Mobile, landline etc) Can you try a SIP - Asterisk -PSTN - SIP call? What I want to know is if you dial out of your Cisco SIP Phone via the PRI and force that call to loop back immediately (by dialing one of your DIDs) to your Asterisk an then get answered by another SIP Phone, does the call sound bad? I think the idea here is to isolate wether the bad sound is due to your hardware and immediate PSTN switch or some farther away PSTN switch. Calls that go out of your Asterisk get routed one way in the PSTN jungle and calls coming to you probably get routed another way. Thats is one way to explain why there is bad audio on outgoing calls and good on incoming. It's rather confusing... Any help with this would of course be greatly appreciated. I called Digium last week, and they didn't seem to know what to do. Except replace the card, which I'm sure isn't what needs doing. It really does seem like it's some obscure problem with the configuration... gah! :/ Cheers, Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andres Sent: 11 August 2004 17:19 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Static on outgoing calls (Quad E1) Ben Merrills wrote: We're using a TE410P connected directly to Kingston Telecom (C7 - EuroISDN conversion is done along the route to us). Unsure what to change within my setup really. I've played around with the rxgain and txgain, although it's made some difference, nothing special! Does anyone know why I would get static on outgoing calls via the TE410P and not incoming? What happens when you call one of your numbers..ie have the call go out the PRI and then come back into your Asterisk. Does it also sound bad? -- Andres Network Admin http://www.telesip.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Static on outgoing calls (Quad E1)
I placed a call as follows: SIP (Cisco Phone) - Asterisk PRI (outgoing) - Asterisk PRI (incoming) - Sip (Cisco) The call exhibited the same problems as before, static crackle on the line. (Dialled party) I still think this is an issue with the Digium card, but I'm unsure as to what. I've been playing with the rxgain and txgain, although I think this just has the effect of making the call so quiet that there's little noise to hear. Hope this provides some help... confused :( Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andres Sent: 11 August 2004 18:53 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Static on outgoing calls (Quad E1) Ben Merrills wrote: If I call a number from my mobile say, it sounds fine! Nothing is wrong with the call quality at all. If I call asterisk (via the digium card) then route that call out to another mobile, that sounds just as bad as making the call from asterisk... So, to cap off, OK: Mobile - EuroISDN - Asterisk (TE410P E1) - Anything BAD: SIP (Cisco) - Asterisk (TE410P E1) - Anything (Mobile, landline etc) Bad: Mobile - Asterisk (TE410P E1) ReDialed - Anything (Mobile, landline etc) Can you try a SIP - Asterisk -PSTN - SIP call? What I want to know is if you dial out of your Cisco SIP Phone via the PRI and force that call to loop back immediately (by dialing one of your DIDs) to your Asterisk an then get answered by another SIP Phone, does the call sound bad? I think the idea here is to isolate wether the bad sound is due to your hardware and immediate PSTN switch or some farther away PSTN switch. Calls that go out of your Asterisk get routed one way in the PSTN jungle and calls coming to you probably get routed another way. Thats is one way to explain why there is bad audio on outgoing calls and good on incoming. It's rather confusing... Any help with this would of course be greatly appreciated. I called Digium last week, and they didn't seem to know what to do. Except replace the card, which I'm sure isn't what needs doing. It really does seem like it's some obscure problem with the configuration... gah! :/ Cheers, Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andres Sent: 11 August 2004 17:19 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Static on outgoing calls (Quad E1) Ben Merrills wrote: We're using a TE410P connected directly to Kingston Telecom (C7 - EuroISDN conversion is done along the route to us). Unsure what to change within my setup really. I've played around with the rxgain and txgain, although it's made some difference, nothing special! Does anyone know why I would get static on outgoing calls via the TE410P and not incoming? What happens when you call one of your numbers..ie have the call go out the PRI and then come back into your Asterisk. Does it also sound bad? -- Andres Network Admin http://www.telesip.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Static on outgoing calls (Quad E1)
I placed a call as follows: SIP (Cisco Phone) - Asterisk PRI (outgoing) - Asterisk PRI (incoming) - Sip (Cisco) The call exhibited the same problems as before, static crackle on the line. (Dialled party) I still think this is an issue with the Digium card, but I'm unsure as to what. I've been playing with the rxgain and txgain, although I think this just has the effect of making the call so quiet that there's little noise to hear. Hope this provides some help... confused :( Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andres Sent: 11 August 2004 18:53 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Static on outgoing calls (Quad E1) Ben Merrills wrote: If I call a number from my mobile say, it sounds fine! Nothing is wrong with the call quality at all. If I call asterisk (via the digium card) then route that call out to another mobile, that sounds just as bad as making the call from asterisk... So, to cap off, OK: Mobile - EuroISDN - Asterisk (TE410P E1) - Anything BAD: SIP (Cisco) - Asterisk (TE410P E1) - Anything (Mobile, landline etc) Bad: Mobile - Asterisk (TE410P E1) ReDialed - Anything (Mobile, landline etc) Can you try a SIP - Asterisk -PSTN - SIP call? What I want to know is if you dial out of your Cisco SIP Phone via the PRI and force that call to loop back immediately (by dialing one of your DIDs) to your Asterisk an then get answered by another SIP Phone, does the call sound bad? I think the idea here is to isolate wether the bad sound is due to your hardware and immediate PSTN switch or some farther away PSTN switch. Calls that go out of your Asterisk get routed one way in the PSTN jungle and calls coming to you probably get routed another way. Thats is one way to explain why there is bad audio on outgoing calls and good on incoming. It's rather confusing... Any help with this would of course be greatly appreciated. I called Digium last week, and they didn't seem to know what to do. Except replace the card, which I'm sure isn't what needs doing. It really does seem like it's some obscure problem with the configuration... gah! :/ Cheers, Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andres Sent: 11 August 2004 17:19 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Static on outgoing calls (Quad E1) Ben Merrills wrote: We're using a TE410P connected directly to Kingston Telecom (C7 - EuroISDN conversion is done along the route to us). Unsure what to change within my setup really. I've played around with the rxgain and txgain, although it's made some difference, nothing special! Does anyone know why I would get static on outgoing calls via the TE410P and not incoming? What happens when you call one of your numbers..ie have the call go out the PRI and then come back into your Asterisk. Does it also sound bad? -- Andres Network Admin http://www.telesip.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Static on outgoing calls (Quad E1)
Hi, I have a problem with a Digium quad E1 card. It seems when I make outgoing calls to any party, when that person talks on the line, they hear scratching and static (theres also background static, but less of it). The person making the call from asterisk (via the E1) doesnt hear any of this. Ive done things like turned off irqmasking on the IDE drives in the machine, which had no effect it would seem. Has anyone come across an issue like this before? This doesnt happen on incoming calls to the Asterisk box! On incoming calls, both parties hear excellent quality audio. Here is a copy of my zaptel.conf and also a copy of Zapata.conf Ive had a line test done by our PRI provider; they cant see any faults on the line. And only span1 is active at this time. Hope someone has had a similar problem, Cheers, Ben zapata.conf Description: zapata.conf zaptel.conf Description: zaptel.conf
RE: [Asterisk-Users] Static on outgoing calls (Quad E1)
We're using a TE410P connected directly to Kingston Telecom (C7 - EuroISDN conversion is done along the route to us). Unsure what to change within my setup really. I've played around with the rxgain and txgain, although it's made some difference, nothing special! Does anyone know why I would get static on outgoing calls via the TE410P and not incoming? Cheers, Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Franks Sent: 11 August 2004 13:20 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Static on outgoing calls (Quad E1) Hi, I have a problem with a Digium quad E1 card. It seems when I make outgoing calls to any party, when that person talks on the line, they hear scratching and static (there's also background static, but less of it). The person making the call from asterisk (via the E1) doesn't hear any of this. I've done things like turned off irqmasking on the IDE drives in the machine, which had no effect it would seem. I had a similar issue before, but it was with a T100P connected to an Adtran TA750. The phones we were using were Polycom SIP IP 500 phones. What we did (and anyone of these could have resolved it) was got Polycom to send us new Power of Ethernet injectors. We also cleaned the wiring up from the POTS lines to the Adtran. Then from the Adtran to the T100P. We also, moved the server off of a Batery backup (cheapo) that seemed to be semi related. When we moved it onto the cheapo battery backup, we started noticing the static. So anyone of those seemed to resolve it. We also moved echo cancellation off of * onto a Tellabs box, and are much happier with the performance and cancellation. Best of both worlds I guess. Thanks, Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Static on outgoing calls (Quad E1)
If I call a number from my mobile say, it sounds fine! Nothing is wrong with the call quality at all. If I call asterisk (via the digium card) then route that call out to another mobile, that sounds just as bad as making the call from asterisk... So, to cap off, OK: Mobile - EuroISDN - Asterisk (TE410P E1) - Anything BAD: SIP (Cisco) - Asterisk (TE410P E1) - Anything (Mobile, landline etc) Bad: Mobile - Asterisk (TE410P E1) ReDialed - Anything (Mobile, landline etc) It's rather confusing... Any help with this would of course be greatly appreciated. I called Digium last week, and they didn't seem to know what to do. Except replace the card, which I'm sure isn't what needs doing. It really does seem like it's some obscure problem with the configuration... gah! :/ Cheers, Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andres Sent: 11 August 2004 17:19 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Static on outgoing calls (Quad E1) Ben Merrills wrote: We're using a TE410P connected directly to Kingston Telecom (C7 - EuroISDN conversion is done along the route to us). Unsure what to change within my setup really. I've played around with the rxgain and txgain, although it's made some difference, nothing special! Does anyone know why I would get static on outgoing calls via the TE410P and not incoming? What happens when you call one of your numbers..ie have the call go out the PRI and then come back into your Asterisk. Does it also sound bad? -- Andres Network Admin http://www.telesip.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960 NAT question
I've got 4 Cisco 7960's and they're behind a firewall (sonic wall). The asterisk box is on a WAN connection on the other end of a DS3, the phones connect fine to the Asterisk server as you can see from the output of show sip peers below. tp3/tp3 firewall-ip D N 255.255.255.255 60665 Unmonitored tp2/tp2 firewall-ip D N 255.255.255.255 60646 Unmonitored tp1/tp1 firewall-ip D N 255.255.255.255 60649 Unmonitored Now, the Cisco phones are set to use nat (nat = 1) and in the SIP configuration, the phones are also configured for SIP. [tp1] type=friend secret=tp1 host=dynamic nat=yes callerid=Test Phone 1 I can make calls out over the phones, but can't get anything back in. If I call voicemail say, then that's fine. But if I try and call another phone behind the firewall, it just sits there :/ IS there a specific port range I need to open? Should I be using a different sip config? Cheers for any help, Ben www.griffin.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI Diad number
Is there a way of getting the dialled number from an AGI? Is it passed in the initial variables, or can it be pulled out or passed across from the dial plan? Cheers, Ben Merrills Griffin Internet
RE: [Asterisk-Users] Wiki down
I work for an ISP and we have quite a lot of hosting, I also run a number of Linux boxes with mysql and php. If a mirror was required, and those in charge want one... I wouldn't mind talking and trying to arrange a free mirror. If you want to get in touch, drop me an email. Cheers, Ben Merrills Griffin Internet -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael George Sent: 27 May 2004 15:47 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Wiki down Apparently there is no mirror or anything for it? I've been in the groove for a couple days making great progress, but I need the application documentation... On May 27, 2004, at 8:44 AM, Gregory Junker wrote: http://www.voip-info.org gives: Warning: mysql error: No Database Selected in query: select `name` ,`value` from `tiki_preferences` in /var/www/html/tikiwiki-1.8.2/lib/tikidblib.php on line 133 Values: Array ( ) $result is false $result is empty Was going to grab a link to give to Florent regarding his CTI thread and question about how to program against the Asterisk API... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Meetme Options (new one)
Seems like it would be a simple modification? Where would I post a feature request like this? J Cheers, Ben From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Sullivan Sent: 24 May 2004 17:16 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Meetme Options (new one) On May 24, 2004, at 8:21 AM, Ben Merrills wrote: Is it possible to select the audio stream thats played as a user enters a meetme conference? I was just now doing an RTFS trying to figure that out. At the moment, the sound played on entering is hard-coded. Time for a feature request?
RE: [Asterisk-Users] Troubles with Kphone
That looks more like a problem with artsd! Under KDE artsd is used to daemonize the sound system, try `killall artsd` then try making a call again. Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of enano Sent: 25 May 2004 10:53 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Troubles with Kphone Hi , I'm triying to use kphone 4.02, but when i'm make a call the programs doesn't respond any command, so i can't hear any sound .. in sip.conf that's my codec config: disallow=all allow=gsm allow=ulaw allow=ilbc and the kphone give the follow : SipClient: Sending: 06:46:28.116 ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2;rport CSeq: 6121 ACK To: sip:[EMAIL PROTECTED];tag=as12aab0bf From: ivan2 sip:[EMAIL PROTECTED];tag=7F6911ED Call-ID: [EMAIL PROTECTED] Content-Length: 0 User-Agent: kphone/4.0.2 Contact: ivan2 sip:[EMAIL PROTECTED];transport=udp res_search: NO result ! res_search: NO result ! SipClient: Sending to '192.168.0.3:5060' SipCallMember: localStatusUpdated: 200 CallAudio: Using GSM for output CallAudio: Sending to remote site 192.168.0.3:19696 UDPMessageSocket::SetTOS: Operation not permitted CallAudio: OSS device already open (readwrite) anyone can help me ?? thanks Ivan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Meetme Options (new one)
Is it possible to select the audio stream thats played as a user enters a meetme conference? If you could, it would be very simple to record a users name, and then play that as the greeting to other attendees as they join the conference. If not, could someone tell me how hard it would be to modify the source? I presume at the moment the file to be played it stored in a var somewhere, is it simply a case of allowing MeetMe() to accept another param, which could be the audio stream? Cheers, Ben Merrills
RE: [Asterisk-Users] Error compiling Zaptel
Do you have a symlink in /usr/src as follows? lrwxrwxrwx 1 root src 20 May 7 11:01 linux - kernel-source-2.4.18 (note that it may differ depending on the kernel source you have?) If youve installed via an apt style package manager, and havnt recompiled your kernel, then visit www.kernel.org, download a stable kernel (I recommended 2.4.26). Extract it to /usr/src/linux-kernelnumber Then create a symlink ln s /usr/src/kernel source /usr/src/linux This should resolve the issues youre having there, else Ive missed the point and just waffled for 5 minutes ;) Hope that helps, Ben Merrills Griffin Internet From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of San Singhania Sent: 11 May 2004 16:17 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Error compiling Zaptel Hi, I just finished downloadingasterisk and when trying to compile the zaptel drivers, get the following errors. I dont have a clue whats going on... can someone help. In file included from /usr/include/linux/module.h:20, from zaptel.c:44: /usr/include/linux/modversions.h:1:2:#error Modules should never use kernel-headers system headers, /usr/include/linux/modversions.h:2:2: error but rather headers from an appropriate kernel-source package. /usr/include/linux/modversions.h:3:2: #error Change -I/usr/src/linux/include (or similar) to /usr/include/linux/modversions.h:4:2: #error -I/lib/modules/$(uname -r)/build/include /usr/include/linux/modversions.h:5:2: #error to build against the currently-running kernel. make: *** [zaptel.o] Error 1 Thanks San
RE: [Asterisk-Users] Asterisk Rhetorical Systems
I took a look at their site and played some of the demo's - can I have any comments from Asterisk users? How did they get on with it and what is the general opinion of the quality etc? This will be used for a major service line that reports faults and outages across a network. Kind Regards, Ben Merrills -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andy Powell Sent: 10 May 2004 16:33 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk Rhetorical Systems hehehehhe Yes I know I use cepstral.. I wrote app_cepstral... (bkw messed with it too) Andy *** REPLY SEPARATOR *** On 10/05/2004 at 08:06 Eric Wieling wrote: On Mon, 2004-05-10 at 05:37, Andy Powell wrote: I'd love to hear how you get on Ben, but I get the feeling that Rhetorical's software prices are out of the reach of most people here. I think integration of this would be a very good move tho. Quite frankly Rhetoricals tts is the best I've heard so far. Try www.cepstral.com They have a wide range of voices, runs on both Linux and Windows, and is US$30 for the non-development version. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Rhetorical Systems
Has anyone tried integrating Asterisk and Rhetoricals rVoice software? Were evaluating different approaches to system announcements via T2S. Has anyone gone down this route that could give some advice? Ive installed festival and wasnt too impressed, the demo one the website seems far better quality and clarity then the defaults in the source package. However I must admit Ive not yet figured our how to change the voice, and Im sure the quality could be improved on (in which case, does anyone know how to switch to the Male British voice?) Any advice or guidance here would be greatly appreciated. Kind Regards, Ben Merrills Internet Applications Developer Griffin Internet www.griffin.com
[Asterisk-Users] A few questions
Hi, I have a couple of questions about MeetMe and call queues. Im 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 weve used that support Call Conferences. Has anyone found a way of doing this? Is there an alternative to MeetMe that would support this feature (thats as good if not better?). Queues: Im running the 1.0 stable from the cvs server, and Ive 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!
