Re: [asterisk-users] Require dialplan
ViciDial doesn't work that way, you have to use the agent web interface or the API to disposition a call. MATT--- On Mon, Apr 11, 2011 at 10:04 AM, mahesh katta maheshka...@flexydial.comwrote: Hi , In vicidial dialer I need small Dialplan require. when i call from hardphone , in that has 1to9 no.s i want define the dipositions like when i press the 1 it will goes NotIntrest, press 2 for NotAvailable. How can i configure for this. -- Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E) Mumbai 400069 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634 Web http://www.buzzworks.com -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Extension notation in default ViciDial installation
On Wed, Oct 27, 2010 at 3:43 PM, Jose P. Espinal j...@slackware-es.comwrote:
Hello List,
A few days ago I installed ViciDial on a server, and while looking to
the default 'extensions.conf' file, I saw this line:
exten = _010*010*010*015*.,1,Dial(${TRUNKTESTast}/${EXTEN:16},55,oT)
Can someone point me out to the Asterisk documentation part where
explains how to use server IP's as extension number?
I could not see it in the ATFOT2 book, and I would like to understand
better that part.
Note:
Or might it be a fully dependent setting of ViciDial?
In the installation documentation of VD says (just above the exten =
... line mentioned previously):
; local server extens:
; BE SURE TO CHANGE THIS LINE FOR YOUR IP ADDRESS!
Regards,
--
Jose P. Espinal
http://www.eSlackware.com
IRC: Khratos @ #asterisk / -doc / -bugs
This is a ViciDial feature and it depends on ViciDial being configured
properly. As of the current release version 2.2.1, the dialplan will
automatically configure for however many servers you have in your cluster,
including the dial-by-ip extensions you mention. An associated iax.conf
registration is also done from each server in your cluster to every other
server in your cluster, which allows this to work.
MATT---
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Re: [asterisk-users] Call Center: scripting for call routing, reporting, login and logout, CTI
On Tue, Sep 7, 2010 at 2:56 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; I would like to use Asterisk for a call center, but really does not know if Asterisk support the following in a good way: 1) Ability to do an inteligent routing, so to route the call to the proper skill group based on the caller information? 2) If I can create skill groups and then the agent will login to this skill group. 3) What about reporting to check the call center performance? How can I get it? 4) To have integration with the CRM, how to be done? Is it using CTI or how? 5) Is it possible that agent to login and logout and be ready and not ready? Appreciate your kindly advise and help. Regards Bilal Hello, ViciDial can also do this out of the box. It has skills-based routing as well as queue prioritization and a web-based agent screen that easily integrates with web-based CRM systems. It is also Open Source and has no licensing costs: http://www.vicidial.org MATT--- -- _ -- 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
Re: [asterisk-users] Vicibox vs VicidialNow
ViciBox actually gives you the option of using the 2.2.1 release or SVN/trunk versions ViciDial Also, ViciBox is the officially supported ISO installer of the ViciDial project. But, both ViciBox and ViciDialNow are Linux ISO installers that will give you a functional ViciDial system. Thanks, MATT--- On Sun, Jul 25, 2010 at 8:29 PM, Juan David Diaz juanch...@gmail.comwrote: The only big difference I know, is: VicidialNow - *based on CentOS* - Vicidial 2.0.5.1rc1 ViciBox - *Based on OpenSuse* - Vicidial 2.0.5 The core of the call center for both of them is Vicidial. Regards. 2010/7/25 Alejandro Cabrera Obed aco1...@gmail.com Dear all, I need a call center asterisk's based solution and I see there are two important solution for 120+ agents: VicidialNow and ViciBox Can you tell me the difference between these open source call center solution please ??? Special thanks Alejandro -- _ -- 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 -- Juan. Linux User #441131 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] How to see STDERR message?
Hello, STDERR goes to the original Asterisk process only, not any asterisk -r connections that you may use. If you launch Asterisk in a screen like we do, then you can see it and log it in context with when the output is happening. We find it very useful to do it this way. MATT--- On 1/7/10, Zhang Shukun bit...@gmail.com wrote: Thank you for you reply? is that mean STDERR couldn't show under Asterisk CLI mode? it's only saved to some file? 2010/1/7 Steve Edwards asterisk@sedwards.com: On Thu, 7 Jan 2010, Zhang Shukun wrote: i use agi to send message back to Asterisk by STDERR, but why i could't see the message in asterisk CLI? Output to STDERR does nothing for me either. I prefer to use syslog() to log the messages via syslogd. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Sucan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What happened to netxusa?
DNS doesn't seem to resolve, looks like one of those unfortunate Domain name registration decisions where the DNS servers and all contact email addresses for the domain are from the domain itself: NETXUSA.COM Administrative Contact: x...@netxusa.com Technical Contact: x...@netxusa.com Domain servers in listed order: NS1.NETXUSA.COM NS2.NETXUSA.COM They had a nice booth at Astricon and everything. Haven't heard anything about them going down, this might just be an unfortunate IT management incident. MATT--- On 11/11/09, Matt Darnell mattdarn...@gmail.com wrote: Anyone know what happened to netxusa? Seemed like they dropped off the web overnight. -Matt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RAMDisk vs Extarnal server for recording
Hello, We use RAM to record to on almost all systems we set up, although we usually use tmpfs, instead of a fixed RAM drive, because it is more flexible. The number of recordings you can handle is dependant on how long the calls are. What would your average, minimum, maximum recording lengths be? We usually do not do more than 100 concurrent recordings on a single server, but we have done up to 250 before successfully. MATT--- On 10/21/09, Robin ro...@zoap.org wrote: Thanks for your response. The hardware I have now is not sufficient to set up a ramdisk (just 4 gb)... But memory is rather cheap nowadays. If i'd buf up the server with 8 extra gigs for use as a ramdrive, do you think that might be enough to record between 30-60 simultanious streams? Or should it be way more? btw, I found this thread somewhere: http://lists.digium.com/pipermail/asterisk-users/2005-October/120930.html, but this is rather old info. Is this documentation still usefull? And if not, do you happen to have any idea/url/doc where I can find a bit less old info? thanks, robin On Wed, Oct 21, 2009 at 13:21, Zoa zoach...@securax.org wrote: There are 2 issues i think, one is the seek time on harddisks and the lack of a big buffer in Asterisk (saving 10 streams at the same time will cause a lt of random writes). The other one is the interrupts being taken up by the harddisk. So an SSD might help, saving to an network drive might help (it moves the issue to another server, where it might not cause a problem), buffering to ram (but you will lack space). The best solution depends on your exact hardware and the amount of writes you want to do. Buffering to a ramdrive before moving it over NFS seems like the best idea to me. Zoa Robin wrote: I'm having loads of problems with recordings, as in crappy audio quality and lost pieces of the recordings. I've been searching for a solution and the solutions i find on the interwebs include a ramdisk, for local recording, or another machine, handling the recording. I guess the ramdisk would be the easy solution and the external machine would be little harder to set up. I do actually prefer the external machine, but i'm not exaclty sure how to set that one up... The reason I prefer the external machine, is that the recording have to be moved to an external machine anyway. Although I've come across a post somewhere, talking about recording to ramdisk and then move the files over a crosscable directly to another disk over 1000mbit. Which sound nice as well... What do you advise for bringing serverload down and get rid of the harddisk bottleneck? Is a ramdisk a better solution then an external machine? And if so, why? Sorry about this pro-con question, but I cannot find an answer which compares these pro-cons anywhere. thanks, robin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RAMDisk vs Extarnal server for recording
Hello, Yep, I'm the ViciDial Guy :) In our most recent release we do have some instructions in the SCRATCH_INSTALL.txt doc on setting up a tmpfs partition for recording. 8GB should be fine for the 60 concurrent recordings under the times you gave, although with MySQL and Apache/PHP you may run into issues, so I would recommend moving MySQL/Apache/PHP off to a different server ASAP. Thanks for the compliments! MATT--- On 10/21/09, Robin ro...@zoap.org wrote: Hi Matt, ain't you the vicidial guy? I'm actually trying to get this stuff fixed on a vicidial system. Anyway, the minimum length is 10-20 seconds, maximum can get as long as 15-20 minutes, and on average it's about 2-5 minutes, depending on the campaign. The server is now doing everything btw, but I'm going to dedicate it to only handle calling and recording. The rest (database and http) will be moved to other servers, which might help a bit too. Off topic: the company I work for went bankrupt a few months ago, but is back in business and we are making heavy use of vicidial (awesome stuff). Going to do loads of work on it, so hope to give loads of (usefull) code to the vicidial project by the end of the year. Looking forward to it! On Wed, Oct 21, 2009 at 17:11, Matt Florell astma...@gmail.com wrote: Hello, We use RAM to record to on almost all systems we set up, although we usually use tmpfs, instead of a fixed RAM drive, because it is more flexible. The number of recordings you can handle is dependant on how long the calls are. What would your average, minimum, maximum recording lengths be? We usually do not do more than 100 concurrent recordings on a single server, but we have done up to 250 before successfully. MATT--- On 10/21/09, Robin ro...@zoap.org wrote: Thanks for your response. The hardware I have now is not sufficient to set up a ramdisk (just 4 gb)... But memory is rather cheap nowadays. If i'd buf up the server with 8 extra gigs for use as a ramdrive, do you think that might be enough to record between 30-60 simultanious streams? Or should it be way more? btw, I found this thread somewhere: http://lists.digium.com/pipermail/asterisk-users/2005-October/120930.html, but this is rather old info. Is this documentation still usefull? And if not, do you happen to have any idea/url/doc where I can find a bit less old info? thanks, robin On Wed, Oct 21, 2009 at 13:21, Zoa zoach...@securax.org wrote: There are 2 issues i think, one is the seek time on harddisks and the lack of a big buffer in Asterisk (saving 10 streams at the same time will cause a lt of random writes). The other one is the interrupts being taken up by the harddisk. So an SSD might help, saving to an network drive might help (it moves the issue to another server, where it might not cause a problem), buffering to ram (but you will lack space). The best solution depends on your exact hardware and the amount of writes you want to do. Buffering to a ramdrive before moving it over NFS seems like the best idea to me. Zoa Robin wrote: I'm having loads of problems with recordings, as in crappy audio quality and lost pieces of the recordings. I've been searching for a solution and the solutions i find on the interwebs include a ramdisk, for local recording, or another machine, handling the recording. I guess the ramdisk would be the easy solution and the external machine would be little harder to set up. I do actually prefer the external machine, but i'm not exaclty sure how to set that one up... The reason I prefer the external machine, is that the recording have to be moved to an external machine anyway. Although I've come across a post somewhere, talking about recording to ramdisk and then move the files over a crosscable directly to another disk over 1000mbit. Which sound nice as well... What do you advise for bringing serverload down and get rid of the harddisk bottleneck? Is a ramdisk a better solution then an external machine? And if so, why? Sorry about this pro-con question, but I cannot find an answer which compares these pro-cons anywhere. thanks, robin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http
Re: [asterisk-users] RAMDisk vs Extarnal server for recording
On 10/21/09, David Backeberg dbackeb...@gmail.com wrote: On Wed, Oct 21, 2009 at 7:36 AM, Robin ro...@zoap.org wrote: Thanks for your response. The hardware I have now is not sufficient to set up a ramdisk (just 4 gb)... But memory is rather cheap nowadays. If i'd buf up the server with 8 extra gigs for use as a ramdrive, do you think that might be enough to record between 30-60 simultanious streams? Or should it be way more? I'm doing ramdisk recordings of about the same number of streams you're talking, in 4GB. I move out completed recordings once every 15 minutes or so via NFS, and as such, I never use very much of the ramdisk. There's no rule that says you have to use the whole 4GB of ram for recordings. I'm probably staying below 100MB or so. Strictly speaking, I'm using both ramdisk and external server, but the external server is just a centralized system with larger disks. However, I know that this arrangement isn't working for my load which is about to double again, so I'm upgrading to better hardware (and maintaining the status quo with my asterisk arrangement) If you read every single title of asterisk-users in the last few months, you'll find a similar discussion on this topic which went through the pros and cons of ramdisk versus centralized server. Somebody at that time mentioned particular names of programs that can do the centralized recordings by doing network hardware level replication and picking off the SIP packets. I've never done this, but if you find that mailing list thread you'll be able to find names of people who say they've done that. We have a few clients that use Oreka(from OrecX) that does network-based SIP packet-capture recording. It works very well on their multi-server setups and the core of Oreka is Open Source. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IVR seleCtion
Hello, ViciDial has IVR logging(pre-Queue) of IVRs set up through our web interface(we call them Call Menus), but ViciDial does not use Asterisk queues at all and it's logging is done entirely in a MySQL database. As a side note, the logging done by ViciDial (non-IVR of course) is also fully compatible with QueueMetrics. MATT--- On 9/16/09, Maria Cristina Bayno falls_m...@yahoo.com wrote: Hello Team, IVR selection of QUEUEMETRICS As we know queuemetrics had an IVR selection functionality where it can get the IVR keypress of a caller. We saw this link http://forum.queuemetrics.com/index.php?action=printpage;topic=503.0 and upon checking, its only determined the Queue, I want to get is the per IVR of a caller. Can you help me guys regarding this? I want to implement this with the trixbox asterisk. Any idea? Thank you Cristina Bayno Technical Support Bitstop Network Services, inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The o dial option
Hello, This changed years ago, and originally it was the 'p' dial option(for preserve CallerID). The reason we are told for the change was for calls being transferred within a company that originated on outside lines, so that you would know who the transfer was coming from. I didn't understand it either, but there it is. MATT--- On 9/14/09, John A. Sullivan III jsulli...@opensourcedevel.com wrote: Hello, all. I see there is an o option for the Dial() command which reverts to the previous behavior of using the original callerid throughout the call - I suppose more specifically, using the callerid from leg 1 for leg 2 in B2BUA if I understand it correctly. That seems to be highly desirable behavior; I know we are seeing some problems with call history and call forwarding because of the default use of callerid. However I'm assuming it was changed to the current behavior for a good reason. Before we revert to the old behavior, I'd like to ask, why was it changed? What problems arose from the old behavior that provoked the change? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vicidial now extension setup
Hello, You know we do have a very active ViciDial forum: http://www.vicidial.org/VICIDIALforum/index.php MATT--- On 8/15/09, Tareq Kibria mtk_...@yahoo.com wrote: Plz mention what type of information do u need. So that i can collect .. ---tareq --- On Sat, 8/15/09, Alex Balashov abalas...@evaristesys.com wrote: From: Alex Balashov abalas...@evaristesys.com Subject: Re: [asterisk-users] Vicidial now extension setup To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Saturday, August 15, 2009, 10:59 AM Insufficient information. Tareq Kibria wrote: Dear Alex , Definitely there is reason behind it which i dont understand properly...I am novice to the system. For Inbound calling i create ingroups and assign it to the campaigns...USER logged in to that campaign ..But no call forwarded to that user.. Could u plz give me an idea about it ---tareq --- On *Fri, 8/14/09, Alex Balashov /abalas...@evaristesys.com/* wrote: From: Alex Balashov abalas...@evaristesys.com Subject: Re: [asterisk-users] Vicidial now extension setup To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, August 14, 2009, 9:59 AM This would have to do with the reason they're being rejected, wouldn't it? What is the reason? Tareq Kibria wrote: Dear All, I am trying to using E1 PRI Connection with vicidialnow setup..Calls are landed in asterisk .From Asterisk CLI i can see the caller id from where the call came ..but the calls are rejects. So what should i do now for forwarding this call to a agent who is using softphone.Is there any Dialplanning required? Plz Help me to sort it out. ---tareq ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone actively using RLT for mobile phone forwarding?
On 8/4/09, Brian Thompson br...@eng.wayne.edu wrote: Hello, We currently have a scenario where a large percentage of inbound calls on a telco PRI are intended for professors who aren't currently in their offices/at their desks. My question is, is anyone actively using the Asterisk RLT (Release Link Trunking) feature to bounce these sorts of calls back to the telco? The idea being to forward the call to their mobile phone without tying up two of the PRI channels (one inbound and one outbound) for the duration of the call. If so, any caveats pertaining to the combination of RLT and Asterisk that I should be aware of before attempting to build such a system? Thanks, Brian Hello, We have set up several systems using TBCT(Two B-Channel Transfer), which is the NI2 protocol version of RLT(which is for DMS100 only). As long as the carrier supports it(which is always the biggest problem in our experience), the actual functionality works great with Asterisk. Some things to keep in mind: - Your carrier will bill you for the time on the off-circuit legs of the call, and will send you a PRI message when the call hangs up. Asterisk throws up a warning for this event and ignores it(not logged in any way). - With RLT(on DMS100) you can only do RLT with calls that come in to your circuit, whereas TBCT on NI2 can work for inbound or outbound initiated calls. Setup is fairly easy for these, just set zapata.conf facilityenable=yes and transfer=yes and upon a native bridge on the same trunk group the calls will be released to the carrier and free up those two lines. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Scalability and stability matters
On 7/21/09, Jose Arias cyr2...@gmail.com wrote: Hi all, I'm planning to develop a custom autodialer application which will be dealing with its own model for agents and queues, therefore it won't use neither asterisk agents nor asterisk queues, nor asterisk cdr. The application will supply the whole reporting and agent managing features by itself. The application will command asterisk through an AMI telnet connection using only the originate, redirect and hangup AMI commands plus the stream file AGI command (AsyncAGI patch will be required). The application will make outbound calls, then they will be redirected on the fly to dynamically defined meetme rooms, then the application will call extensions (registered endpoints) where it will know there are available agents in order to redirect them to the previous meetme rooms. If the application launched more calls than available agents it would play prompts while waiting for agents to become available. Since the planned features set from asterisk to be used by the application will be very short, but the figures can be very large (in terms of concurrent calls, registered endpoints, traffic on the AMI port, etc..) I would appreciate if anybody can help me to find out what's the more suitable asterisk version to use in terms of scalability and stability: - concurrent registered endpoints (SIP and IAX) - concurrent two and tree party meetme rooms (whatever codec can be used) - concurrent mixmonitor recordings - concurrent playings for prompts - commands and events rate on the AMI port It's important to notice the advanced features from asterisk aren't a priority. I already looked over some links like http://www.voip-info.org/wiki/view/Asterisk+dimensioning and others but I found more questions than answers there. Thanks in advance Jose This sounds a lot like ViciDial, which does use meetme instead of Asterisk Queues/Agents, is already engineered to be multi-server, is capable of placing 200,000+ outbound calls per server per day, has a web-based GUI for configuring the system and a web-based agent interface. - concurrent registered endpoints (SIP and IAX) Doesn't really matter, we've done 500+ on a single server before and it didn't really affect load much. As for number of agents, we are usually conservative on that front, usually we keep it under 50 agents per outbound server, but we have done 100 before. - concurrent two and tree party meetme rooms (whatever codec can be used) Everything is transcoded in a meetme room to slin. ViciDial does everything in Meetme, and while it does use slightly more resources than Asterisk Queues, it is more stable and offers more flexibility - concurrent mixmonitor recordings We do not recommend using mxmonitor. It is better to have a custom recording handling script. And if you are using Meetme for everything you don't have to bother mixing recordings anyway. - concurrent playings for prompts This depends on a lot of different things, if load or playback quality becomes an issue then you should put prompts on a RAM drive or tmpfs - commands and events rate on the AMI port Use a single point(or a few limited points) of entry to the AMI to keep it working well. You should not have an AMI connection for each agent. We currently use a version of 1.4.21.2 that has about 8 patches applied to it, and we have found it to be very stable in production. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Scalability and stability matters
2009/7/21 Matt Florell astma...@gmail.com On 7/21/09, Jose Arias cyr2...@gmail.com wrote: Hi all, I'm planning to develop a custom autodialer application which will be dealing with its own model for agents and queues, therefore it won't use neither asterisk agents nor asterisk queues, nor asterisk cdr. The application will supply the whole reporting and agent managing features by itself. The application will command asterisk through an AMI telnet connection using only the originate, redirect and hangup AMI commands plus the stream file AGI command (AsyncAGI patch will be required). The application will make outbound calls, then they will be redirected on the fly to dynamically defined meetme rooms, then the application will call extensions (registered endpoints) where it will know there are available agents in order to redirect them to the previous meetme rooms. If the application launched more calls than available agents it would play prompts while waiting for agents to become available. Since the planned features set from asterisk to be used by the application will be very short, but the figures can be very large (in terms of concurrent calls, registered endpoints, traffic on the AMI port, etc..) I would appreciate if anybody can help me to find out what's the more suitable asterisk version to use in terms of scalability and stability: - concurrent registered endpoints (SIP and IAX) - concurrent two and tree party meetme rooms (whatever codec can be used) - concurrent mixmonitor recordings - concurrent playings for prompts - commands and events rate on the AMI port It's important to notice the advanced features from asterisk aren't a priority. I already looked over some links like http://www.voip-info.org/wiki/view/Asterisk+dimensioning and others but I found more questions than answers there. Thanks in advance Jose This sounds a lot like ViciDial, which does use meetme instead of Asterisk Queues/Agents, is already engineered to be multi-server, is capable of placing 200,000+ outbound calls per server per day, has a web-based GUI for configuring the system and a web-based agent interface. - concurrent registered endpoints (SIP and IAX) Doesn't really matter, we've done 500+ on a single server before and it didn't really affect load much. As for number of agents, we are usually conservative on that front, usually we keep it under 50 agents per outbound server, but we have done 100 before. - concurrent two and tree party meetme rooms (whatever codec can be used) Everything is transcoded in a meetme room to slin. ViciDial does everything in Meetme, and while it does use slightly more resources than Asterisk Queues, it is more stable and offers more flexibility - concurrent mixmonitor recordings We do not recommend using mxmonitor. It is better to have a custom recording handling script. And if you are using Meetme for everything you don't have to bother mixing recordings anyway. - concurrent playings for prompts This depends on a lot of different things, if load or playback quality becomes an issue then you should put prompts on a RAM drive or tmpfs - commands and events rate on the AMI port Use a single point(or a few limited points) of entry to the AMI to keep it working well. You should not have an AMI connection for each agent. We currently use a version of 1.4.21.2 that has about 8 patches applied to it, and we have found it to be very stable in production. MATT--- On 7/21/09, Jose Arias cyr2...@gmail.com wrote: Many thanks Matt, I heard asterisk had some problems with registering over 100 SIP endpoints and I was worried about how much the transcoding load could be for over 100 concurrents calls too. I expect to be over these figures. Regarding the AMI connection, yes, there will be only one, like any third-party cti-link but my concern was about how many commands an events asterisk is able to handle without becoming in a bottleneck. You said you're using about 8 patches. Are all of them to make sure the stability and scalability of the system? Well, one of them is the AsyncAGI patch, isn't? Is there anyone to mach originate commands with new_channel events? I'm planning to use asterisk 1.4.18 Regards Jose Hello, We don't use AsyncAGI at all, the patches to Asterisk are mostly for issues with waitforsilence, chan_sip, AGI defunct channels and gsm codec. We also add patches for changed meetme enter/leave sounds and Sangoma CPD SIP message processing. We match Originates to new channel using code in our AMI listener app. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo
Re: [asterisk-users] Skill based routing
On 7/17/09, Alex Balashov abalas...@evaristesys.com wrote: Rupert Utteridge - Digital Techniques (Austalia) Limited wrote: We are trying to implement skill based routing for agents in a support centre based on the agent login. Has anyone had any experience with this and what was the outcome? It can't really be done using Asterisk queues, unless you want to create a large number of queues for every relevant skill factor and have agents join various combinations of these simultaneously--which would take quite a bit of dial plan and/or AGI logic to pull off. Plus, that doesn't scale any given queue beyond one host. I suggest you look into using FastAGI[1] to simulate the queue experience by generating hold music and announcements without actually using Asterisk queues per se. This is quite possible to do, and, this allows you to distribute queues across multiple hosts, as well as distribute calls within those queues by whatever logic you choose. No shoehorning--just write it yourself. -- Alex [1] Yes, FastAGI. Not local AGI. And most especially not in PHP; contrary to a lot of the info out there, PHP could not possibly be a less suitable language in which to write AGI scripts. I don't know who comes up with these lavish heights of mediocrity. If you are not looking to write it yourself you could always try ViciDial which has skills-based routing built in, and it's free and Open Source. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller ID (name) - where does it come from?
On 7/8/09, Steve Totaro stot...@first-notification.com wrote: On Wed, Jul 8, 2009 at 2:14 AM, Olivieroza-4...@myamail.com wrote: Hi, Reading this thread, is this correct to say CallerName is widely used in the US ? Here in France, this service is optional but I don't think many companies are subscribing to it and I'm not aware of any non-Telco CNAM providers. I would curious to know how the situation is elsewhere. Regards Whether true or not, I was told that nearly 80% of people in the US have caller ID. I would say that number is much higher for business, especially on PRI circuits. I think the two big motivators there were packaging of services, for X amount extra, you get caller ID, call waiting, voicemail on at the telco, etc The other factor was the proliferation of telemarketing. Before the DNC, a white pages listed home phone could ring a dozen times a day by people selling stuff. -- Thanks, Steve Totaro In Canada, their telephone network is set up to allow for dynamic CallerIDname on PRIs just like how CallerIDnumber works here in the USA. We didn't believe it at first until we tried it, but they seem to be the only country we've worked in, out of a few dozen countries, that allows dynamic CallerIDname defined on a per-call basis. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Scaling
On 6/17/09, Gordon Henderson gordon+aster...@drogon.net wrote: On Wed, 17 Jun 2009, Steve Totaro wrote: Hi, Quick question to the real world. Approx what specs would I need on server to handle 95 ZAP or Dahdi - SIP gateway using G729 on the SIP to carrier side (nothing else, just media conversion)? Does the latest Asterisk/DAHDI significantly improve these numbers over say, Asterisk 1.2.X? Sure, there is plenty to read but nothing I could find quickly on my exact needs that was clear and I want to be fairly sure before ordering a server. Obviously load avg has something to do with it but CPU and mem seems to be the biggest factors. Transcoding - It's CPU grunt and multi processors what you need (more so than memory) - handling that many calls ought to be a breeze on any modern hardware without the transcoding. Personally, I think I'd be looking at a TC400B card which can handle 96 concurrent g729 transcodes... But as a rough benchmark, I can do 12 concurrent g729 transcodes on a 1GHz VIA processor before it's totally maxed out, (stupid) extrapolation to 96 would suggest 8 x 1GHz processors, 4 x 2GHz, or 3 x 3GHz processors... However you gain more with the faster processors in terms of bigger cache (but that can also be a loss too) I'd start with a quad core system and see how it goes under benchmark... Gordon The TC400B is up to 120 channels of G729a now: http://www.digium.com/en/products/voice/tc400b.php MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] callcenter / dialer / predictive dialer / vicidial program is now open
OK, enough with the ViciDial bashing. Have you taken a look at the most recent release? Have you seen all of the new features including the new APIs? Why exactly is it so bad? Does it not do what it is supposed to do? Is there something better out there that does the same thing and is open source? MATT--- On 5/18/09, Alex Balashov abalas...@evaristesys.com wrote: Nah, the real problem with this post is that Vicidial is just so *bad*. No offense to Matt Florell and all; it just is. Martin wrote: On Mon, May 18, 2009 at 4:16 PM, Jeff LaCoursiere j...@jeff.net wrote: Not the business - the list (is non commercial). Meaning if you want to advertise your cool new service, do it on asterisk-biz. He knew that for sure. j LOL All the time I thought the message went only to asterisk-biz My bad ... Well his logic was to get the biggest possible exposure (looking for users ??? :) Martin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] callcenter / dialer / predictive dialer / vicidial program is now open
LOL, I blame Getty images for taking over the stock images industry, this is clearly all their fault for not having enough images of attractive females with headsets on. No, there is no affiliation that I know of between our two companies. MATT--- On 5/18/09, Martin asteriskl...@callthem.info wrote: BTW Is vicidial related to http://www.contacttel.com/ ? http://www.contacttel.com/ http://www.vicidial.com/ the same female face is looking from these websites :) Martin On Mon, May 18, 2009 at 5:07 PM, Matt Florell astma...@gmail.com wrote: OK, enough with the ViciDial bashing. Have you taken a look at the most recent release? Have you seen all of the new features including the new APIs? Why exactly is it so bad? Does it not do what it is supposed to do? Is there something better out there that does the same thing and is open source? MATT--- On 5/18/09, Alex Balashov abalas...@evaristesys.com wrote: Nah, the real problem with this post is that Vicidial is just so *bad*. No offense to Matt Florell and all; it just is. Martin wrote: On Mon, May 18, 2009 at 4:16 PM, Jeff LaCoursiere j...@jeff.net wrote: Not the business - the list (is non commercial). Meaning if you want to advertise your cool new service, do it on asterisk-biz. He knew that for sure. j LOL All the time I thought the message went only to asterisk-biz My bad ... Well his logic was to get the biggest possible exposure (looking for users ??? :) Martin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD Not Working
Hello,
Well, depending on the version of app_amd that you used when you added
it to Asterisk 1.2, you might need to use HUMAN and MACHINE as the
possible AMDSTATUS instead of AMD_PERSON and AMD_MACHINE. The
AMDSTATUS was changed at some point in the app_amd code, not sure why
they changed it, but that might be your issue.
Also, since you are calling your own number you might want to do an
Answer on the call before running AMD, not sure if that would cause
the hangups you are seeing or not, but it's something to try.
MATT---
On 4/24/09, Sam Hawkin gvrt...@gmail.com wrote:
Hi,
Thanks for your reply
We are using the Asterisk 1.2.4.
and below the dialplan path. we are orginating the call to
my number and connection it to context cdtest and extension 1.
[cdtest]
exten = 1,1,NoOp( cb amd issue testing )
exten =
1,2,Macro(Cb-old|/root/business_hours|/root/business_hours)
[macro-Cb]
exten = s,1,NoOp( values in CB arg1 ${ARG1} arg2 ${ARG1} )
exten = s,2,AMD
exten = s,3,GotoIf($[${AMDSTATUS}=AMD_PERSON]?4:7)
exten =
s,4,NoOp(Humanplaying--${ARG1})
exten = s,5,Playback(${ARG1})
exten = s,6,Hangup
exten = s,7,GotoIf($[${AMDSTATUS}=AMD_MACHINE]?8:11)
exten =
s,8,NoOp(Machine---playing--${ARG2})
exten = s,9,Playback(${ARG2})
exten = s,10,Goto(s|12)
exten = s,11,Playback(${ARG1})
please suggest our what might be the problem.
Any help is highly appreciated.
Thanks.
On Thu, Apr 23, 2009 at 8:36 PM, Matt Florell astma...@gmail.com wrote:
On 4/23/09, Sam Hawkin gvrt...@gmail.com wrote:
Hi All,
I am trying to use the AMD (Answering Machine Detect).
But it is not sending the AMD_Status as either
the Human or Machine, it hangs up in middle.
can any one suggest us, what might be the problem
and possible solution to it.
below is the log
-- Executing AMD(SIP/sip-ffe0, ) in new stack
-- AMD: SIP/sip-ffe0 14082284927 (null) (Fmt: 4)
Apr 23 08:00:26 NOTICE[28319]: app_amd.c:134 isAnsweringMachine: AMD
using
the default parameters.
-- AMD: initialSilence [3500] greeting [1500] afterGreetingSilence
[300] totalAnalysisTime [5000] minimumWordLength [120]
betweenWordsSilence
[50] maximumNumberOfWords [5] silenceThreshold [256]
-- AMD: HANGUP
What version of Asterisk are you running this on?
What is the dialplan path that this is running through?
MATT---
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Re: [asterisk-users] timing source problem
Hello, I would suggest that you first methodically try every possible combination of zaptel.conf timing settings(each change follwed by a hard reboot of the Asterisk server) to see if there is a magic combination of settings that will work. I don't know if you have the time for that, or if it takes a while for the timing issues to appear, but that is what I would try. If that still doesn't work, we have solved similar issues with older(2 years ago) Digium quad cards by switching to Sangoma hardware that offers more options for forcing timing in it's wanpipe driver software. Although when I posted about this before in another thread the folks from Digium swear that newer Digium cards(with newer firmware) do not have this problem using the newer Dahdi drivers. What version of Zaptel are you using and how old is your Digium card? MATT--- On 4/24/09, Wolfgang Pichler wpich...@yosd.at wrote: hi all, we do have some troubles with zaptel timing source - we have a setup with 3 telco PRI lines, connected to asterisk digium card 0 - asterisk does some handling - calls are leaving on digium card 1 - going to a siemens hipath - there is some call handling - some of the calls then are going from the hipath over a qsig line to a bosch integral PBX - handling the rest of the calls. To be able to get away from the bosch system - we like to put asterisk (1 port free on each card) in the middle of the path siemens - bosch - so that it will be siemens - card 0 asterisk card 1 - bosch. Currently the Siemens hipath is playing the network side - the bosch is cpe. So the siemens hipath does provide the timing source. With asterisk in the middle i can not take the timing source from the siemens link - because i have already the telco line as timing source. But when starting it in this setup - i will get lots of timing source auto card 0! messages. So i think the siemens timing is not in sync. with the telco timing - so mixed up on asterisk with telco line as primary timing will not work when the siemens does try to deliver timing. I have not tried as /etc/zaptel.conf parameter 0 als timing parameter (0 = be master) - but i think it wont work because the siemens wont accept the timing from the asterisk box. Changing configuration of the siemens is not possible. So - here the questions... - is it possible to do what i want to do ? - do you think timing=0 in zaptel.conf will work ? - would it be possible to connect a xorcom 2 PRI channel bank to asterisk to handle the qsig line between the two ? Or will the xorcom then also take the timing from the digum cards - telco lines ? any hints would be nice... many thanks Wolfgang ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD Not Working
On 4/23/09, Sam Hawkin gvrt...@gmail.com wrote: Hi All, I am trying to use the AMD (Answering Machine Detect). But it is not sending the AMD_Status as either the Human or Machine, it hangs up in middle. can any one suggest us, what might be the problem and possible solution to it. below is the log -- Executing AMD(SIP/sip-ffe0, ) in new stack -- AMD: SIP/sip-ffe0 14082284927 (null) (Fmt: 4) Apr 23 08:00:26 NOTICE[28319]: app_amd.c:134 isAnsweringMachine: AMD using the default parameters. -- AMD: initialSilence [3500] greeting [1500] afterGreetingSilence [300] totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence [50] maximumNumberOfWords [5] silenceThreshold [256] -- AMD: HANGUP What version of Asterisk are you running this on? What is the dialplan path that this is running through? MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2B Channel Transfer on XO-based T1
On 4/14/09, Max Metral m...@povo.com wrote: I’m trying to get “blind transfer” from an incoming DAHDI line to an external number to work on an * 1.6 install using a T1 from XO. The documentation is very “distributed” and incomplete, so while it’s not working, it’s definitely more likely my error somehow. Couple questions if anybody is out there who even knows what TBCT is… 1) Is this even supported? 2) Does it require some settings in dahdi_channels, or features, or whatever? 3) Would I “trigger” it via a Dial command or commands, or via Transfer? 4) Do either or both of the legs need to be answered? Are you positive that your carrier PRI circuit has this feature enabled? If so, how much are they charging you for this service?(if they are not charging you for it monthly it is most likely not enabled) What kind of PRI do you have? (5ESS, NI2, DMS100,...) How many PRIs and trunk groups set up across them do you have? I have set up 2BCT for two different call center clients before, and neither implementation went smoothly as far as the carrier's part of it was concerned. Asterisk and zaptel(Dahdi) can handle it and will always attempt to do 2BCT on ALL native bridging of channels on the same trunk group if you have Transfer=yes in zapata.conf(or the Dahdi equivelent file). I have never configured 2BCT on an Asterisk 1.6 system, only 1.2 and 1.4 (using zaptel 1.4 for both), although I can't see any reason why it wouldn't work. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New ViciDial Call Center Suite Release: 2.0.5
On 4/10/09, ContactTel Business li...@contacttel.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rony Ron Sent: April-09-09 11:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] New ViciDial Call Center Suite Release: 2.0.5 Great ! thank you very much for your job! BR, Matt Florell a écrit : Hello, We've released another update to our VICIDIAL/astGUIclient call center suite: 2.0.5 http://astguiclient.sf.net/ The call center suite client applications run on most modern web browsers on almost any GUI-capable operating system, and it includes the VICIDIAL call center suite. This package is free and AGPLv2. This package is geared towards Asterisk installations with SIP,IAX or Zap phones and Zaptel, IAX or SIP trunks. For this release, we have added hundreds of new features including Asterisk phone, trunk and DID configuration through the VICIDIAL web interface. We have also tested the suite on Asterisk versions through 1.2.30.2 and 1.4.21.2. All client web-apps and administration pages are available in English, Spanish, Greek, German, Italian and French, with rough translations of Polish, Portuguese, Brazillian Portuguese, Slovak, Russian and Dutch for the client web-apps only. Check out the project blog for more information: http://astguiclient.blogspot.com Let me know what you think. Thanks, MATT--- ___ Just ran into a problem with a cluster fu..k.. freepbx +vicidial+bad stuff.. Isn't that still broken as hell ? with like putting registrations in between stanza's etc ? Client added new trunk + routing into FPBX and it still used old routes also.. Nice attempt, but from what i hear and see its same old pile of agi's and scripts, making a single phone call ,,running over 2000 lines of stuff , where even the see logs function stops at 2000 lines.. ;) I really think the idea is nice, however, people trying to replicate the functionality are probably running away as fast as they can when seeing the extensive, undocumented code, is that intentional ? Hello, Not sure what you're getting at here. ViciDial does not use FreePBX in any way, and we do not recommend using FreePBX in conjunction with ViciDial on the same machine at all. Are you experiencing a specific problem with ViciDial itself? MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New ViciDial Call Center Suite Release: 2.0.5
On 4/6/09, Wolfgang Pichler wpich...@yosd.at wrote: Hi, we are using version 2.0.4 (vicidialnow distribution) now for some time in productino - working quit nice. Is there any upgrade instruction out there - or will a simple yum update do the job in the feature. PS: On the astguiclient site you have April 3, 2008 - Released version 2.0.5 - i think thats not correct ;-) Hello, There is actually an UPGRADE file right in the main directory of the release that you should read over. Since there are many database and dialplan changes since 2.0.4 a software-only upgrade would only get you part of the way. Thanks for the catch on the date, it has been fixed now. MATT--- Am Freitag, den 03.04.2009, 10:30 -0400 schrieb Matt Florell: Hello, We've released another update to our VICIDIAL/astGUIclient call center suite: 2.0.5 http://astguiclient.sf.net/ The call center suite client applications run on most modern web browsers on almost any GUI-capable operating system, and it includes the VICIDIAL call center suite. This package is free and AGPLv2. This package is geared towards Asterisk installations with SIP,IAX or Zap phones and Zaptel, IAX or SIP trunks. For this release, we have added hundreds of new features including Asterisk phone, trunk and DID configuration through the VICIDIAL web interface. We have also tested the suite on Asterisk versions through 1.2.30.2 and 1.4.21.2. All client web-apps and administration pages are available in English, Spanish, Greek, German, Italian and French, with rough translations of Polish, Portuguese, Brazillian Portuguese, Slovak, Russian and Dutch for the client web-apps only. Check out the project blog for more information: http://astguiclient.blogspot.com Let me know what you think. Thanks, MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] New ViciDial Call Center Suite Release: 2.0.5
Hello, We've released another update to our VICIDIAL/astGUIclient call center suite: 2.0.5 http://astguiclient.sf.net/ The call center suite client applications run on most modern web browsers on almost any GUI-capable operating system, and it includes the VICIDIAL call center suite. This package is free and AGPLv2. This package is geared towards Asterisk installations with SIP,IAX or Zap phones and Zaptel, IAX or SIP trunks. For this release, we have added hundreds of new features including Asterisk phone, trunk and DID configuration through the VICIDIAL web interface. We have also tested the suite on Asterisk versions through 1.2.30.2 and 1.4.21.2. All client web-apps and administration pages are available in English, Spanish, Greek, German, Italian and French, with rough translations of Polish, Portuguese, Brazillian Portuguese, Slovak, Russian and Dutch for the client web-apps only. Check out the project blog for more information: http://astguiclient.blogspot.com Let me know what you think. Thanks, MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues Announce help request.
On 3/20/09, Cary Fitch ca...@usawide.net wrote: I am trying to get a queue to do more than just play music and hold calls. Specifically, making some comforting voice announcements would be nice. Below is the queues.conf file relevant portions. Member phone number is munged to protect the guilty. We shouldn't need the announcement source info, but I have been trying everything. The problem is with the member busy, we get no voice announcements. (For test purposes is being on hold busy? We have also just laid the phone on the desk.) We will settle for expected hold time, Thank you announcements, Position in queue, or Dow Jones 30 Industrials news. :-) Anyone have a tip? Cary Fitch I just thought I'd mention that ViciDial has the ability to play a periodic announcement on inbound queue calls, as well as music on hold, place in line, estimated hold time and lots of other inbound-only features. ViciDial does not use Asterisk Queues so the way we got it working probably wouldn't help you much, but I just wanted to mention it's functionality and that it is open source if you wanted to give it a try. Thanks, MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 - now known as RFC 5456
On 2/20/09, John Todd jt...@digium.com wrote: Mark and Ed received word today that the long-awaited RFC for IAX2 has been approved by the IETF, and is now published: http://www.rfc-editor.org/authors/rfc5456.txt Thanks to Ed Guy, Mark Spencer, Brian Capouch, Frank Miller, and Kenny Shumard! Lots of revisions and discussions have paid off. JT Many congratulations to Mark and Ed! I have been extolling the virtues of using IAX and IAX trunking for years(much to the grief of Olle). Maybe now we'll see some more mainstream hardware with IAX as a protocol option. There are only so many iterations of PA1688 chipset devices coming out of China, it would be nice to once again run a Snom phone with IAX firmware on it. Thanks again, MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Test Lab
On 2/13/09, Alex Balashov abalas...@evaristesys.com wrote: On Fri, 13 Feb 2009 09:48:11 + (GMT), Lee Wilson leef...@yahoo.co.uk wrote: Alex, thanks for the quick response. So I can assume from your response this should work. That was easy :-) I just want to clarify before I got and buy anything the cards are not so cheap. Yep, it should work. I am not exactly sure how one goes about setting one of the cards to provide the T1 master clock as I have only configured low-level T1 settings on Sangoma cards, but it should be possible. Although, honestly, I am not sure that you're going to get a lot of timing slips on something like a 6 ft crossover cable anyway even if the clocks are both set to line or internal; timing synchronisation is much more of a concern on lengthier spans and circuit designs that go through numerous network elements. If you want to be able to set Master timing you will have to use Sangoma cards, they allow you force the timing in the wanrouter configuration. I have done extensive testing with crossover T1/E1/PRIs and I even have a testing lab set up with servers that have quad T1 cards with crossover cables going between all of the ports. There are no time slip issues, and you can run the T1 line for hundreds of feet without issues on good cable. For the cable I would suggest buying one of these: http://www.smartronixstore.com/index.cfm?fuseaction=product.displayProduct_ID=9 (SuperLooper ISDN (PRI) Crossover Adapter) They are fairly cheap and will save you the headache of making your own cables. If you are using Sangoma cards you will have to configure Wanrouter properly, but the Asterisk/zaptel side is fairly easy to configure, just make sure one side is pri_net and the other is pri_cpe and you should be good to go. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Test Lab
On 2/13/09, Tom Moore tommym2...@gmail.com wrote: The cable needed for this is a different cable than an ethernet cross over. I have actually done this same thing today with a Samsung 100 system and Asterisk 1.4.20.1 and Zaptel 1.4.11 and things work great. I would again just recommend getting one of these, they are worth it: http://www.smartronixstore.com/index.cfm?fuseaction=product.displayProduct_ID=9 (SuperLooper ISDN (PRI) Crossover Adapter) A question of my own: I know I can emulate the network side of a pri connection, but can I do this same trick with other t1 standards like ani and others? If I can be a client on the different t1 types, does this also mean I can be the server side and feed back the different standards to legacy equipment as well or are there some limitations to this? Yes you can do other T1/E1 standards like RBS D4/AMI, EM wink start and most of your other old favorites MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Test Lab
On 2/13/09, Kevin P. Fleming kpflem...@digium.com wrote: Matt Florell wrote: If you want to be able to set Master timing you will have to use Sangoma cards, they allow you force the timing in the wanrouter configuration. I have done extensive testing with crossover T1/E1/PRIs I don't believe this is true; we use Digium cards connected back-to-back to each other, with one providing the span timing (clock) all the time. This is a very common configuration and works fine. To set a Zaptel/DAHDI card to provide span timing, just set all the spans on it to 'zero' as the priority for timing source; when none of the span clocks are the timing source, the onboard clock on the card will be the timing source. Can you tell me where the setting is to force Master timing on Digium cards per port? I really didn't think Digium cards had the ability to force Master in this way. I've tried to do it with channelbanks before and couldn't force it to master, whereas I can get it to work with Sangoma. Thanks, MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Test Lab
On 2/13/09, Kevin P. Fleming kpflem...@digium.com wrote: Matt Florell wrote: Can you tell me where the setting is to force Master timing on Digium cards per port? I really didn't think Digium cards had the ability to force Master in this way. I've tried to do it with channelbanks before and couldn't force it to master, whereas I can get it to work with Sangoma. I'm not sure that the terms you are using match what I'm used to hearing and telling people, so let me try to describe how our cards work and you can tell me how that matches up with what you want to do. Digium multi-port T1/E1 cards use a single clock source for transmitting on all connected spans. That clock source can be on the onboard oscillator, or the recovered clock from one of the spans, and the source selection can change dynamically based on the configuration (in other words, you can set the first span as 'highest priority', the second span as 'second priority', etc). If a span is selected to be the clock source for the entire card, and then that span goes into red alarm, a different clock source will be chosen, until that span recovers. So, what this means is that each span port that is configured to be used is *always* transmitting a bit stream, and due to the nature of T1/E1 signaling, that bit stream includes a clock. A device connected to that port will *always* be able to recover a clock from that bitstream if it chooses to do so. For channel banks, other servers, downstream PBXes, etc. this is a common configuration, and the device will derive its clock from the bitstream it receives from the Digium card. In cases where the system admin is using spans that are generated from 'upstream' devices, where the card should slave its transmitted clock to the recovered clock from that span, then the card can be configured in this mode. Once a span has been selected to be the clock source for the card, *all* the spans on that card will use that clock source for their transmitted bit streams. I guess I'm not explaining myself very well, so I'll describe exactly what happened and how my problem was solved. We had a digium quad port T1 card with 3 carrier T1s plugged into it and one channelbank. After a few months of everything running just fine on the system the channelbank would go red alarm after a few hours of the server being on, if we reset the server, channelbank or even just unplugged the crossover T1 and plugged it back into the channelbank it would work again for a few more hours. The carrier told us it was a timing issue, so I began to mess with the timing settings and after a week of making changes and waiting to see if they would work, none of the changes to any of the timing settings in zapata.conf would do anything. At this point I swapped out the Digium card with a Sangoma card(because the quad T1 cards were cheaper than a new channelbank) and the same thing happened. I emailed Sangoma support and they suggested I try the forced Master clock setting for the channelbank port to see if that would help, and after figuring out exactly how to set it up I put it live and no more red alarms on the channelbank. My understanding is that this setting lets you ignore the timing signal coming from the other end of one of the ports, and the card will take another timing source from a port that you specify and force it to be used as a timer on that first port. If my explanation makes no sense I apologize, but this is how it happened and the problem was solved. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Test Lab
On 2/13/09, Kevin P. Fleming kpflem...@digium.com wrote: Matt Florell wrote: My understanding is that this setting lets you ignore the timing signal coming from the other end of one of the ports, and the card will take another timing source from a port that you specify and force it to be used as a timer on that first port. That is very close to what our cards do, except that it's not controllable on a port-by-port basis. In that situation, which port did you pick as the clock source for the channelbank port, and what was the clock source for it prior to that? When I got it working I had set the clock source to the 4th port on the card for the channelbank which was on the first port. We never tried any other ports because I was afraid to touch it after I got it working. Before it was working, we had tried every one of the dozens of combinations of timing settings in zapata.conf for the 4 spans, and none of them worked. As a note, I should mention that this was over 2 years ago. Thanks, MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.31.1, 1.4.22.2, 1.4.23.1, and 1.6.0.5 released
On 1/30/09, Mark Michelson mmichel...@digium.com wrote: Matt Florell wrote: Yep, my bad I found them once I searched with the dash '-' after the 1.4.23. They were lost in the flood of users list mail in my inbox. I wonder if these could also be posted on the asterisk-announce list more consistently? I see a few releases on the announce list, but last 1.4 one was December 2nd and nothing after that on that list except for a few vulnerability postings. The policy that we have been following is that only final releases will be announced to the asterisk-announce list. Betas and release candidates are not. The rationale is that asterisk-announce is supposed to be a low-volume list and that most subscribers to it would not appreciate all the noise of announcing release candidates or betas there. Got this December 1st on the asterisk-announce list: from: Asterisk Team asteriskt...@digium.com to asterisk-annou...@lists.digium.com dateDec 1, 2008 11:58 PM subject [asterisk-announce] Asterisk 1.2.30.3, 1.4.23-rc2, 1.6.0.2, 1.6.1-beta3, and Asterisk-Addons 1.6.0.1, 1.6.1-rc2 released mailed-by lists.digium.com The Asterisk.org development team has released Asterisk versions 1.2.30.3, 1.4.23-rc2, 1.6.0.2, 1.6.1-beta3, as well as Asterisk-Addons versions 1.6.0.1 and 1.6.1-rc2. These releases are available for immediate download from http://downloads.digium.com/. I know that there were official releases mentioned in this email, but an RC and a beta were also both announced. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.31.1, 1.4.22.2, 1.4.23.1, and 1.6.0.5 released
For a while we were seeing RC(release cantidates) release announcements and I can see that there were RC release for this 1.4.23 release. Any reason they aren't being publicized, or am I just looking in the wrong place? We always do compatibility testing before putting a new release in production and at this point 1.4.21.2 is the most recent stable release as far as we are concerned. Of course 1.4 wasn't really stable until 1.4.18, which is when the RC releases started too. MATT--- On 1/29/09, Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote: On Thursday 29 January 2009 13:50:19 Remco Barendse wrote: On Thu, 29 Jan 2009, Thomas Stein wrote: On Thursday 29 January 2009 09:23:41 Remco Barendse wrote: 1.4.23.1 doesn't seem to work for me. I did an in-place upgrade of Asterisk 1.4.21 and upgraded to the latest zaptel as well. Incoming calls stopped working. Whenever an extension was trying to pickup the phone by doing a group pickup with *8 the extension just got dead audio and the next phone in the group stared ringing. Yeah. Thats http://bugs.digium.com:80/view.php?id=14206 I'm also concerned about that one: http://bugs.digium.com:80/view.php?id=13488 cheers t. Thanks for your reply, indeed that is the problem. Strange that this stable release is still prominently on the asterisk.org website as the latest and greatest. Where do you see us denoting any release as stable (or defining what that term actually means)? We release when we think that we've eliminated the bugs we can find, and then people find more bugs. If you can fix bugs before they're reported, we'd love to have you contribute to the development effort. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.31.1, 1.4.22.2, 1.4.23.1, and 1.6.0.5 released
Yep, my bad I found them once I searched with the dash '-' after the 1.4.23. They were lost in the flood of users list mail in my inbox. I wonder if these could also be posted on the asterisk-announce list more consistently? I see a few releases on the announce list, but last 1.4 one was December 2nd and nothing after that on that list except for a few vulnerability postings. I know it would help me to get those release notices on that list, then I could flag them better so my mail viewer will smack me on the head to read them when they come in. Thanks, MATT--- On 1/29/09, Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote: On Thursday 29 January 2009 16:00:14 Matt Florell wrote: For a while we were seeing RC(release cantidates) release announcements and I can see that there were RC release for this 1.4.23 release. Any reason they aren't being publicized, or am I just looking in the wrong place? http://lists.digium.com/pipermail/asterisk-users/2008-December/222727.html http://lists.digium.com/pipermail/asterisk-users/2008-December/223668.html http://lists.digium.com/pipermail/asterisk-users/2009-January/224940.html I'd say you just missed them, as they were published to this list, as evidenced by the archives. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vicidialnow
The astguiclient/VICIDIAL project is very much alive. There were several updates to the 2.0.4 release put out over the last year and we have actually been so busy and so much has changed in the development trunk that we have not had time to put out a full proper release. The latest stable code is of course available through our SVN server as well and most companies that we do installs for are working off of the SVN codebase. Thanks, MATT--- On 1/26/09, David fire ddf...@gmail.com wrote: the vicidial proyect is alive? the last stable releace is from 2007 thanks David 2009/1/26 David fire ddf...@gmail.com where i can buy the vicidial manual? thanks David 2009/1/26 ram talk2...@gmail.com On 1/23/09, David @ULC ucoms2...@gmail.com wrote: But after installing it with CD , I guess we have to change SIP file and do few more changes .. I am looking for those steps.. On Fri, Jan 23, 2009 at 2:55 PM, David @ULC ucoms2...@gmail.com wrote: Anyone have properly formatted document ? On Fri, Jan 23, 2009 at 1:42 AM, David @ULC ucoms2...@gmail.com wrote: But I believe even after doing that , there are few setting and changes required before we can start using it for production I guess... On Fri, Jan 23, 2009 at 12:27 AM, David @ULC ucoms2...@gmail.com wrote: Anyone using VicidialNow ? I have documents for Vicidial scratch install but how to install step by step Vicidialnow ? Buy manuals and understand the Dialplan logic Ram ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vicidialnow
You can download the free manuals and buy the full paid-for manuals at: http://www.eflo.net MATT--- On 1/26/09, David fire ddf...@gmail.com wrote: where i can buy the vicidial manual? thanks David 2009/1/26 ram talk2...@gmail.com On 1/23/09, David @ULC ucoms2...@gmail.com wrote: But after installing it with CD , I guess we have to change SIP file and do few more changes .. I am looking for those steps.. On Fri, Jan 23, 2009 at 2:55 PM, David @ULC ucoms2...@gmail.com wrote: Anyone have properly formatted document ? On Fri, Jan 23, 2009 at 1:42 AM, David @ULC ucoms2...@gmail.com wrote: But I believe even after doing that , there are few setting and changes required before we can start using it for production I guess... On Fri, Jan 23, 2009 at 12:27 AM, David @ULC ucoms2...@gmail.com wrote: Anyone using VicidialNow ? I have documents for Vicidial scratch install but how to install step by step Vicidialnow ? Buy manuals and understand the Dialplan logic Ram ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Asterisk-users] DTMF pass-through question
On 12/28/08, jonathan augenstine jaugenst...@gmail.com wrote: I am trying to resolve an issue and I believe it is my configuration. The scenario is that I have a SIP detected on the server. The dial plan then makes a local connection to another part of the dial plan. The new dial plan extension then places another SIP call out to a SIP phone. When the call is accepted there is streamed from the calling SIP phone. When the audio is complete a DTMF is transmitted to Asterisk. The DTMF is detected by Asterisk but it does not get passed through to the other SIP phone. I would like the DTMF to pass-through to the other SIP phone. Is this a configuration issue? Or do I need to handle this on the dial plan level? Jonathan Asterisk version? What are both dtmfmodes set to for each SIP endpoint? Are the calls natively bridged or bridged through Asterisk? MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Asterisk-users] DTMF pass-through question
On 12/28/08, jonathan augenstine jaugenst...@gmail.com wrote: Matt, Asterisk version == 1.4.22 dtmfmode == info calls are bridged through Asterisk (canreinvite=no) Jonathan Have you tried setting dtmfmode to 'inband' for both SIP endpoints? MATT--- On Sun, Dec 28, 2008 at 3:23 PM, Matt Florell astma...@gmail.com wrote: On 12/28/08, jonathan augenstine jaugenst...@gmail.com wrote: I am trying to resolve an issue and I believe it is my configuration. The scenario is that I have a SIP detected on the server. The dial plan then makes a local connection to another part of the dial plan. The new dial plan extension then places another SIP call out to a SIP phone. When the call is accepted there is streamed from the calling SIP phone. When the audio is complete a DTMF is transmitted to Asterisk. The DTMF is detected by Asterisk but it does not get passed through to the other SIP phone. I would like the DTMF to pass-through to the other SIP phone. Is this a configuration issue? Or do I need to handle this on the dial plan level? Jonathan Asterisk version? What are both dtmfmodes set to for each SIP endpoint? Are the calls natively bridged or bridged through Asterisk? MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How long will Asterisk 1.4.x supported/maintained
On 11/20/08, Steve Totaro [EMAIL PROTECTED] wrote: On Thu, Nov 20, 2008 at 3:38 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Nov 20, 2008 at 08:25:54AM +0100, Olivier wrote: 2008/11/17 Philipp Kempgen [EMAIL PROTECTED] Tilghman Lesher schrieb: On Thursday 13 November 2008 08:16:42 Klaus Darilion wrote: Is there somewhere a statement from Digium how long they will support Asterisk 1.4? 0 There is no statement, because we haven't even discussed when the EOL for 1.4 will be reached. Certainly that means it won't happen for at least the next 60 days, but beyond that, I really don't know. For the average non-techie user who does not want to compile themselves that may sound funny (if not scary). When Debian Lenny (featuring Asterisk 1.4) is finally going to be released that version might not even be supported any more. I think to a large extend, Asterisk is not to be considered as binary distributed at all, as many hardware it supports is not directly managed by kernel team. Interesting consideration. Debian Etch and RHEL5 are based on kernel 2.6.18, but support quite a few hardware devices not included in that kernel. If this issue bothers you, please help test the alternative timing mechanism support now included in trunk. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir I still compile and install 1.2 for the most part, for call centers and large systems. The few 1.4 installs that I have done have been for medium sized PBXs, say 50-70 phones/users and they have been trouble free for the most part. Safe_asterisk may make some troubles transparent. I am not really sure what 1.4 has over 1.2 for the average PBX installation. Then you have the OpenPBX guys who forked 1.2, I know they have added functionality to 1.2, but the following puts me off. Perhaps vaporware, perhaps not, it all relies on the devs. You also have people like Matt Florell who have continued to add functionality to 1.2 but since Digium won't take them, or the dev doesn't want to sign over their first born, they are hard to come by but certainly out there. 1.4 may follow the same path, being forked. 1.6 is not on my radar. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) Hello, We really just maintain a set of patches for 1.2 (just updated waitforsilence a couple weeks ago in fact) and we regularly install 1.2.30.2 in call center setups. It is rock solid and extremely proven in high-call-volume situations. We have started installing 1.4.21.2 on some systems that are not high load as well (1.4.22 has some strange issues with it we have noticed) because we do have clients requesting to use 1.4 for some of the nicer PBX functionality that it has as well as better SIP support. We test 1.6 periodically and we are very much looking forward to some of the great new features of it, but it crashes very quickly when trying to use it in call center situations. just keep in mind that in my opinion the 1.4 tree did not become usable until 1.4.18 when most of the major bugs were finally fixed. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How long will Asterisk 1.4.x supported/maintained
On 11/21/08, Tilghman Lesher [EMAIL PROTECTED] wrote: On Friday 21 November 2008 09:42:12 Matt Florell wrote: On 11/20/08, Steve Totaro [EMAIL PROTECTED] wrote: You also have people like Matt Florell who have continued to add functionality to 1.2 but since Digium won't take them, or the dev doesn't want to sign over their first born, they are hard to come by but certainly out there. We really just maintain a set of patches for 1.2 (just updated waitforsilence a couple weeks ago in fact) and we regularly install 1.2.30.2 in call center setups. It is rock solid and extremely proven in high-call-volume situations. We have started installing 1.4.21.2 on some systems that are not high load as well (1.4.22 has some strange issues with it we have noticed) because we do have clients requesting to use 1.4 for some of the nicer PBX functionality that it has as well as better SIP support. We test 1.6 periodically and we are very much looking forward to some of the great new features of it, but it crashes very quickly when trying to use it in call center situations. just keep in mind that in my opinion the 1.4 tree did not become usable until 1.4.18 when most of the major bugs were finally fixed. Are you reporting these crashes in 1.6? I'd like to know where they are, so we can track them down and fix them. By the time we get around to testing for any length of time there is always another version released(including RCs and betas), we haven't tested on the most recent 1.6 release and we don't really have the resources to do intense debugging and bug reporting on 1.6 anytime in the near future. We have tested two of the original beta releases as well as the first 1.6.0.1 RC and they all had crash problems. We have also had issues adjusting to using Dahdi on 1.6 since it is manditory and you cannot use zaptel as you can with 1.4. I am hoping to set up a system over the Holidays that I will only put 1.6 on that I will be able to do bug testing on, but from our experience it is not easy to move from 1.2/1.4 over to 1.6 and back again in a timely manner because of all of the major changes made to Asterisk between 1.4 and 1.6. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How long will Asterisk 1.4.x supported/maintained
On 11/21/08, Alex Balashov [EMAIL PROTECTED] wrote: On Fri, Nov 21, 2008 at 10:42 AM, Matt Florell [EMAIL PROTECTED] wrote: just keep in mind that in my opinion the 1.4 tree did not become usable until 1.4.18 when most of the major bugs were finally fixed. The longer you drag out the adoption curve, the longer it will take for 1.6 to catch up to that state. Alex Balashov We tried using 1.4 many times, and posted many bugs to the tracker. Some of those bugs were ignored because I was told that I posted too much information. We tried using most of the 1.4 releases and we did post our results, even going as far as posting on the dev list and in IRC, and I was always ignored or not gotten back to. I even offered several times to donate my time to set up a system at Digium to reproduce these bugs on demand and still had noone would take up my offer. I talked to several VPs at Digium in-person and even Mark and was always referred to someone else and nothing was ever done about it. Then after 1.4.17 was released is when bug fixing became a higher priority and they started implementing the release-cantidate process, and myself and many others participated in that process and 1.4.18 went through several RCs with many many bug fixes and a lot of testing, and 1.4.18 was the first fully tested release of the 1.4 tree. As for 1.6, we haven't had anywhere near the time we did with 1.4 to try to get it working for us, and there is a much steeper upgrade path from 1.4 to 1.6 than there was between 1.4 and 1.6 which causes a lot of other small issues in testing and implementation. Hopefully in the next month or so we will have the time to spend on this. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Intergrating vicidial with trixbox
Hello, The paid VICIDIAL user manuals do not cover installing on Trixbox. Mostly because it can be very difficult to install VICIDIAL on Trixbox due to the many different versions of Trixbox and the dialplan complexity of Trixbox.(also I want to mention that there are FREE versions of the VICIDIAL manuals, and all admin-based documentation is in the open-source codebase) We do not recommend putting VICIDIAL on the same machine as Trixbox, mostly due to the performance hit of just running trixbox which effectively cuts the functionaly capacity of the machine in half. We recommend using IAX trunks to connect a separate VICIDIAL machine to your Trixbox machine, that way you can still use your trixbox phones and inbound DIDs if needed with VICIDIAL while still allowing VICIDIAL to efficiently dial out through it's own trunks if you like, all without messing with the internals of the Trixbox-generated dialplan and utilities. MATT--- On 10/29/08, Ron Byer Jr. [EMAIL PROTECTED] wrote: I noticed that the vicidial site has documentation available which probably covers the topics required. However, I also see that they want $50-$100 to download the docs. Seems harsh. Ron Byer Jr. NetWeave Integrated Solutions, Inc. +1.732.786.8830 x120 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Wednesday, October 29, 2008 12:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Intergrating vicidial with trixbox I would contact the vendor. James Mutuku wrote: Hello, I am searched the net for tutorials on how I can Integrate vicidial with trixbox. I can't find any. Anyone who knows where I can get one? James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG. Version: 7.5.549 / Virus Database: 270.8.4/1753 - Release Date: 10/28/2008 9:20 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New VICIDIAL astGUIclient Release: 2.0.4
Hello, I have never tried using Aastra phones as user agent. If they support Javascript and AJAX then it should work. VICIDIAL is tested with IE, Firefox, Opera and Safari. At the very least they may be able to use the remote agent interface that does not use Javascript, but there is reduced functionality as compared to the full agent interface. Thanks, MATT--- On 10/1/08, broadband Voice [EMAIL PROTECTED] wrote: Can I used Aastra phones as agents instead of web-base on astGUIclient-VICIDIAL suite: 2.0.4? Thanks. Our Asterisk is remote and call center will be using Aastra phones or Linksys ATA. On Mon, Dec 3, 2007 at 3:03 AM, Matt Florell [EMAIL PROTECTED] wrote: Hello, We've released another update to our astGUIclient-VICIDIAL suite: 2.0.4 http://astguiclient.sf.net/ The client suite runs on most modern web browsers on almost any GUI-capable operating system, and it includes the VICIDIAL call center suite and the astGUIclient client-side web app which extends your phone's functionality. This package is free and GPL. (the suite is not an asterisk configuration tool) This package is geared towards Asterisk installations with SIP,IAX or Zap phones and Zaptel, IAX or SIP trunks. For this release, we have focused on adding new features to inbound call handling such as custom music-on-hold, agent alert messages per inbound group and agent-rank call routing per skill as well as several other new administrative features. We have also tested the suite on Asterisk versions through 1.2.24. All client web-apps and administration pages are available in English, Spanish, Greek and German, with rough translations of French, Polish, Italian, Portuguese and Brazillian Portuguese for the client web-apps only. Check out the project blog for more information: http://astguiclient.blogspot.com Let me know what you think. Thanks, MATT--- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Aheeva With Asterisk
Hello, If you are looking for a list of Call Center software packages that work with Asterisk then take a look here: http://www.voip-info.org/wiki/view/Predictive+dialer There are over 20 now I believe. MATT--- On 10/1/08, broadband Voice [EMAIL PROTECTED] wrote: I stumbled upon this call center software that works with Asterisk calles Aheeva. Does anyone else use it? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Callcenter monitoring tool
On 9/16/08, Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], Alex Balashov [EMAIL PROTECTED] wrote: Al Baker wrote: Steve Totaro wrote: Although it is commercial, Queuemetrics is a good place to look if you want to pay for a feature rich turn-key solution. does QUEMETRICS gather an ADDITIONAL stats or info than is in the CDRs or is it principally and easy way to view/process CDRS ? It gathers all the info that is present in the Asterisk queue log, which is somewhat more detailed and focused on agent performance. It also uses AGI scripts to place additional entries in the queue log for outgoing calls. Cheers Tony We have done quite a bit of queue_log QueueMetrics integration in the VICIDIAL project, and there are several things that QM can do now that are beyond what basic queue_log entries out of base Asterisk can provide, such as adding call status codes and agent pause codes, and several other features. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] vicidial mysql problem
I just wanted to note that we do have a very active community forum for VICIDIAL available at: http://www.eflo.net/VICIDIALforum/index.php MATT--- On 8/20/08, Alex Balashov [EMAIL PROTECTED] wrote: You need to install the MySQL client libraries and MySQL driver for Perl-DBI. mahboob zaman wrote: I installed asterisk, astguiclient, php and mysql. but when i dialled one number to another number my asterisk server give the following error: /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi install_driver(mysql) failed: Can't load '/usr/lib/perl5/site_perl/5.8.8/i486-linux-thread-multi/auto/DBD/mysql/mysql.so' for module DBD::mysql: libmysqlclient.so http://libmysqlclient.so/.15: cannot open shared object file: No such file or directory at /usr/lib/perl5/5.8.8/i486-linux-thread-multi/DynaLoader.pm line 230. -- Mahboob Zaman System Engr Systems Services Limited Cell: +8801712280308 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI TBCT - Practical Experience, Anybody?
On 8/15/08, Don Kelly [EMAIL PROTECTED] wrote: 1. The carrier you are connected to must be licensed for it and have the necessary software, if the carrier requires, your circuit(s) must be provisioned for it. The originating/destination carriers shouldn't matter. Most carrier sales people don't know what TBCT is unfortunately, and even if a carrier is capable of doing it, it is a possiblity that not all of their equipment is capable of doing it. One client of mine tried to get TBCT working across all 16 of their PRIs(all on the same carrier) and it only worked on 4 of them, supposedly because not all of the telco equipment was capable of the feature. 2. Both incoming and outgoing calls can be transferred to a second outgoing call; I think it's theoretically possible to connect two incoming calls, but I haven't done that. This actually depends on the kind of PRI service you have. For instance with DMS100 circuits you can only do TBCT with calls that come in to your circuit, not with outgoing calls. As for connecting two incoming calls, since that is not possible in Asterisk(to natively bridge two incoming calls together) I can't see how you would get that to work even if it is possible in TBCT. I believe that only DMS100, NI2 and 5ESS PRI signalling protocals are capable of TBCT with the current zaptel code-base. Also, the two B channels involved in the TBCT have to use the same D channel. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HP server and Meetme applications
On 8/11/08, aymen warfalli [EMAIL PROTECTED] wrote: I got one HP ProLiant DL380 G5 - Quad-Core Xeon E5440 2.83 with 4 gig RAM I install Centos 5.2 64 bit and it is rumming pretty well and I need to use it as voice conferencing application (Meetme) server for high number of users fit to 8 E1 links (240 users ) with echo cancellation using same coding use g711 my qustion is this server is this server suitable for 240 users on meetme application on the same asterisk at the same time ? and what is the dimensions of one conference room should I biuld ? and finally if i can go for more users at same server ? I have set up a system with 180 users in meetme rooms on a single server (4 x dual core Xeons) using a Sangoma a108D(8 port T1/E1 card with hardware EC with 8 x T1s connected) and the machine was running at high load but it was usable with good audio. Not sure what adding another 60 channels to that would do in terms of load or audio quality. What is the exact application you are trying to build? What capacity does the meetme room need to have in total? I have actually built distributed meetme applications where you have multiple servers that you can connect meetme rooms on one server to another and have essentially unlimited capacity in a single functional conference room as long as you have the hardware for it. Shameless plug I am going to be talking about this very subject at Astricon next month, along with 2 other presentations I'm giving there, if you happen to be going. /Shameless plug MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HP server and Meetme applications
On 8/11/08, aymen warfalli [EMAIL PROTECTED] wrote: I got one HP ProLiant DL380 G5 - Quad-Core Xeon E5440 2.83 with 4 gig RAM I install Centos 5.2 64 bit and it is rumming pretty well and I need to use it as voice conferencing application (Meetme) server for high number of users fit to 8 E1 links (240 users ) with echo cancellation using same coding use g711 my qustion is this server is this server suitable for 240 users on meetme application on the same asterisk at the same time ? and what is the dimensions of one conference room should I biuld ? and finally if i can go for more users at same server ? I have set up a system with 180 users in meetme rooms on a single server (4 x dual core Xeons) using a Sangoma a108D(8 port T1/E1 card with hardware EC with 8 x T1s connected) and the machine was running at high load but it was usable with good audio. Not sure what adding another 60 channels to that would do in terms of load or audio quality. What is the exact application you are trying to build? What capacity does the meetme room need to have in total? I have actually built distributed meetme applications where you have multiple servers that you can connect meetme rooms on one server to another and have essentially unlimited capacity in a single functional conference room as long as you have the hardware for it. Shameless plug I am going to be talking about this very subject at Astricon next month, along with 2 other presentations I'm giving there, if you happen to be going. /Shameless plug MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VICIDial error
If that's the only set of errors it might be a PHP/Apache error. I would recommend posting on the VICIDIAL Forums if you can't get it to work: http://www.eflo.net/VICIDIALforum MATT--- On 8/8/08, Brad [EMAIL PROTECTED] wrote: Warning: Cannot modify header information - headers already sent by (output started at /home/telecom/public_html/vicidial/admin.php:1175) in /home/telecom/public_html/vicidial/admin.php on line 1187 Warning: Cannot modify header information - headers already sent by (output started at /home/telecom/public_html/vicidial/admin.php:1175) in /home/telecom/public_html/vicidial/admin.php on line 1188 Has anyone ever seen this? I am getting a double header sent with all aspects of the Astisk GUI including VICIDial ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI and Call Center to do CRM integration
We have done this several times for customers with VICIDIAL. I have also seen companies use AGI scripts to enable this kind of application as well. So, yes it is possible. MATT--- On 8/7/08, bilal ghayyad [EMAIL PROTECTED] wrote: CRM: Customer Record Module which is any kind of application. For example, a bank has an application and the agent sit on his PC, when call come, the application fetched with the customer information based on the card number which is entered with the IVR, How the application of the bank was able to fetch the infomation? It was passing to it from the call center. Also another example: when call come to call center, and before call routing to the proper skill group, then we need to check the data related to caller, based on these data we determine which skill group need to be routed, how this to be done? AGI can do? Regards Bilal --- On Thu, 8/7/08, Steve Totaro [EMAIL PROTECTED] wrote: From: Steve Totaro [EMAIL PROTECTED] Subject: Re: [asterisk-users] AGI and Call Center to do CRM integration To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, August 7, 2008, 6:12 PM On Thu, Aug 7, 2008 at 5:55 PM, bilal ghayyad [EMAIL PROTECTED] wrote: Hi All; Did anyone used AGI to do te CRM integration in the Asterisk call center? If yes, I would like to know the overview to know from where to start? Regards Bilal What CRM? FastAGI to hit a box that has logic to update the backend DB of the CRM. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Experience with Vicidial
On 7/17/08, Alex Balashov [EMAIL PROTECTED] wrote: Ein Bielaczyc wrote: I have a small customer looking to update their aged telemarketing system. I ran across astGUIclient and Vicidial (http://astguiclient.sourceforge.net/) during a Google search and was wondering if anyone had any experiences to share, positive or negative. Well, you do have to realise that you're putting almost anyone who may have had a negative experience with ViciDIAL in the difficult position of effectively slandering it, or at least earning the ire and distaste of many other list members, most certainly including the authors. But, that's no reason for self-censorship. So, with apologies to Matt Florell and others: Personally, I've found that ViciDIAL generally works - in a practical, functional, utilitarian sort of sense in which programs work deterministically in accordance with their underlying instructions. I think if you want it to do what most people want it to do more or less out of the box, it's probably a good choice. The story is a bit different, however, in the unlikely event that you actually do care about what's under the hood. It proved to be impractical for my intended use because customisation was required, and the code is an absolute nightmare from a developer's perspective. It is a hodge-podge of naive, inefficient PHP and Perl written with absolutely zero regard for maintainability. It is impossible to read, has no discernable formatting characteristics, is often obfuscated, poorly spaced, arbitrarily indented, and follows no consistent or useful nomenclature or conventions. It's a lot of spaghetti code, and it does not leave one united with the impression that functional decomposition, abstraction or modularity is valued at all, let alone as a guiding value. I suspect many -- although certainly not all -- of limitations to its scalability stem from resource consumption caused by extremely inefficient algorithms, flow control constructs, and serial database interactions that involve repeatedly swapping data in and out of the programmatic layer in high-volume transactions, transforming it, and sending it back to the database. There also appears to be considerable reliance upon the database as a real-time IPC mechanism--another very deadly anti-pattern. Additionally, it has far too many processes, many of whose essence is not clearly or easily understood by the naked eye. If the code were readable, this wouldn't be so bad. But as it stands, in addition to the ugly hack that results from retrofitting astguiclient in this fashion, there are plethoric, innumerable Perl / AGI scripts whose coherence can only be depicted with evocations of a Rube Goldberg device. So, as long as you are interested only in the superficie, it seems to work pretty well, although I can't comment on the overall stability, bugs too much. However, if you are interested in development or customisation, you need to run for the hills, because nothing short of a complete, categorical, wholesale from-scratch rewrite -- one with some evidence of method -- is going to untangle the catastrophe that boils under the deck. -- Alex Hello, No apologies necessary, I think a lot of what you said is mostly true. To address the points one by one: The PHP and Perl code is not the prettiest around, and a lot of it is not commented or formatted as well as it should be. However, I would disagree that there is absolutely zero regard for maintanability or readability. As with many other GPL projects out there VICIDIAL is free to use and modify, and there are many people outside of our company who have worked with the code to provide patches and added functionality. The 50,000+ lines of code were mostly written by me over the course of 5 years, and as we are now having more people working on the code and we are going through the scripts we are working to make the code easier to understand. It is important to mention that VICIDIAL is quite complex and offers a lot of features that add to the complexity of the code. Many of these features were not even conceived when the project was started so they were added in in the most efficient manner that was available. At this point there are over 1000 database fields across 60+ tables that control how VICIDIAL works. As for scalability, VICIDIAL scales to hundreds of seats across multiple Asterisk servers. It can do this because of it's reliance on the MySQL database that acts as the core of a VICIDIAL system. We chose to use MySQL instead of a dedicated communications protocol so that the data could be accessed and used by almost any programming or user interface, and still remain extremely fault-tolerant and resistant to issues on any individual system. As for efficiency, yes there are a lot of small inefficiencies in the code, but most of the major bottlenecks were removed after changing many AGI
Re: [asterisk-users] Experience with Vicidial
On 7/17/08, Alex Balashov [EMAIL PROTECTED] wrote: Matt Florell wrote: No apologies necessary, I think a lot of what you said is mostly true. Well, thank you. I really appreciate that you're willing to entertain what I am saying without construing it as some sort of attack; it is not in the least bit intended that way. The PHP and Perl code is not the prettiest around, and a lot of it is not commented or formatted as well as it should be. However, I would disagree that there is absolutely zero regard for maintanability or readability. As with many other GPL projects out there VICIDIAL is free to use and modify, and there are many people outside of our company who have worked with the code to provide patches and added functionality. Indeed, and I acknowledge that this is a challenge with an open-source project - a problem that is probably best solved by means that are not necessarily received well politically, and is probably seen as orthogonal to the spirit and philosophy of open-source. It requires some degree of centralisation of the patch management process and high standards for acceptance, testing, and coding conventions. This leads to a process that gives the perception of being more closed, a la the Linux kernel, and, as some would have it, perhaps the Asterisk source tree. (I can't really say, as I've never attempted to contribute any of my modifications of the Asterisk source to the project. *hangs head in shame*) Bad programmer, no beer for you :) But seriously, Asterisk is a better example of doing things right more recently, a couple of years ago all sorts of stuff went into the stable releases of Asterisk without enough testing resulting in some pretty big bugs (like in asterisk 1.2.10-12) And even further back the code conventions of Asterisk 6 years ago were in many places about as good as VICIDIAL is today. I think it's the evolution of a project from being in its early years as opposed to its more mature years like Asterisk has started to enter. If I were to start writing VICIDIAL today I would have done many things a lot differently from how they are, but then again if I started from scratch today it wouldn't be a functioning product for many months. Right now, we do basic testing of everything before it is committed to SVN. We are planning on starting an official QA process when we are able to that would test a full range of functions before every commit. But, maintainability and extensibility are probably the biggest challenges to the adoption of an open-source project by a commercial organisation, although those challenges are even more formidable for proprietary, closed-source products. The stark, naked economic realities of adopting something are still there. There are a few large companies that have adopted VICIDIAL and internalized the maintenance of the code for their systems, so we know that the state of the code has not been a road-block for all companies. The integration paths, APIs, transparency, modularity and extensibility are the most important central concerns. For instance, many likely use cases of ViciDIAL entail at least some degree of integration with existing business systems, rules, and logic; after all, the data that goes into the hopper must come from somewhere. :-) Most companies will do some degree of basic lead loading and lead exporting integration on the back-end along with integration with web-based CRM systems on the agent front-end. And this has proven to usually be not too difficult for many companies to accomplish. ... Which leads me to my point: contrary to what is often zealously claimed by purveyors and advocates of open-source solutions, the simple fact that the code is open **does not, ipso facto, offer the necessary level of integration and extensibility.** Version changes, feature changes, bug fixes, and other revisions cannot be readily or easily applied if an organisation chooses to essentially fork an internal revision of stock code, so invariably the preoccupation of good engineering project management becomes whether the custom code can be kept outboard in modules entirely separate from the main code tree, so that the latter can remain more or less pristine across upgrades, updates, add-ons, etc. This is a big reason that we use SVN and we started maintaining a stable branch that we apply only minor changes and bug fixes to and a development branch that all new development takes place on. This is where the importance of data import/export, APIs, and other integration paths comes in. It's the same reason why Asterisk worked out so well for you in creating ViciDIAL; you can do pretty much everything in AGI, instead of having to hack the source or even do a whole lot in the configs. And, of course, in closed-source situations, these integration paths become the lifeline - the only possible path
Re: [asterisk-users] Experience with Vicidial
On 7/17/08, Alex Balashov [EMAIL PROTECTED] wrote: Matt Florell wrote: But seriously, Asterisk is a better example of doing things right more recently, a couple of years ago all sorts of stuff went into the stable releases of Asterisk without enough testing resulting in some pretty big bugs (like in asterisk 1.2.10-12) And even further back the code conventions of Asterisk 6 years ago were in many places about as good as VICIDIAL is today. I think it's the evolution of a project from being in its early years as opposed to its more mature years like Asterisk has started to enter. If I were to start writing VICIDIAL today I would have done many things a lot differently from how they are, but then again if I started from scratch today it wouldn't be a functioning product for many months. I know this dilemma well, and sympathise. In my experience, with some projects a gradual evolution to more sophisticated approaches is possible, but with others the work involved in making that happen would be greater - and the result less elegant and cohesive - than simply rewriting the thing. It depends on a lot of things, some of them important - but intangible - aesthetic judgments. There are a few large companies that have adopted VICIDIAL and internalized the maintenance of the code for their systems, so we know that the state of the code has not been a road-block for all companies. Oh, sure, and I doubt that anything is a roadblock with sufficient spiritual commitment. The ViciDIAL code *is* modifiable and maintainable, logically; it's code, after all. At the end of the day, it is still Perl or PHP. The issues I've run into have been with the willingness of developers and managers to dive into code that looks like that. The claim is never that it's conceptually too complicated, just that any semblance of organisation, structure, readability, modularisation or abstraction is conspicuously absent. For that reason, it is seen as too much work to deal with it. Most companies will do some degree of basic lead loading and lead exporting integration on the back-end along with integration with web-based CRM systems on the agent front-end. And this has proven to usually be not too difficult for many companies to accomplish. A lot depends on the size and scope of the operation and the intended application of the dialer. It uses an open database, and you can put your stuff into the database is a surprisingly ineffective integration path in settings with either high volume or dense business rules requirements. This is actually one of the biggest obstacles I ran into in the previously mentioned endeavours. For one, by requiring that leads be entered into ViciDIAL's lead table, you force the source to shoehorn its data exactly to your schema for leads, which usually results in some degree of information loss and redundancy. And any derivative consequences elsewhere in the database arising from those importations must also be dealt with, e.g., managing IDs used for keying the rows, etc. For another, unless the data used by the company conveniently resides in MySQL - yea, the same instance of it - as ViciDIAL's, some sort of synchronisation job or process will be necessary to refresh the lead table by pumping in new data. This is not really feasible in situations with a very, very large amount of leads. It creates yet another scheduled, recurring potential point of failure (the sync job). The data simply may not be possible to load to keep pace with the dials. It consumes bandwidth and resources and time. If a company does most of its lead management, qualification and other applications of business rules in its own database (which may or may not be something Linux/open-source based, btw!), that database becomes a redundant middle storage layer for the data since the goal is simply to get it into ViciDIAL. But then, some persistent information about the leads must be stored, as a lot of companies tend to call leads back and follow up... etc, etc, etc. In my experience, this led to attempts to bifurcate the ViciDIAL lead table and join on some other custom tables containing needed supplemental information. Then it was discovered that this will require more code changes (many of redundant, duplicated statements) in ViciDIAL than miles in the width of Heaven. That's usually the point at which hats were thrown down and, Damn it, we might as well just write something ourselves was said. Anyway, my real point here is that I think relatively easy, turn-key integration and customisation on a software and data interchange level may prove to be the most significant issue in determining the success and future of ViciDIAL, at least if my experience is any indication. (And what my experience suggests is that almost nobody uses it straight out
Re: [asterisk-users] Experience with Vicidial
Hello, I am the creator of VICIDIAL(which makes me a bit biased) and there are over 700 installations that we know of in use in over 70 countries around the world right now. It is stable and has a very high uptime percentage in properly installed and scaled systems, and we have done installations of VICIDIAL for companies from 1 to 300+ seats. VICIDIAL is GPL and free, and there is a very active community support forum. There are also paid support plans available. Hope that helps, MATT--- On 7/16/08, Ein Bielaczyc [EMAIL PROTECTED] wrote: I have a small customer looking to update their aged telemarketing system. I ran across astGUIclient and Vicidial (http://astguiclient.sourceforge.net/) during a Google search and was wondering if anyone had any experiences to share, positive or negative. Thanks. -- Ein Bielaczyc [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Experience with Vicidial
Hello, You can try posting on the VICIDIAL forums asking for feedback from average users: http://www.eflo.net/VICIDIALforum/index.php MATT--- On 7/16/08, Ein Bielaczyc [EMAIL PROTECTED] wrote: On Thu, Jul 17, 2008 at 1:35 AM, Matt Florell [EMAIL PROTECTED] wrote: Hello, I am the creator of VICIDIAL(which makes me a bit biased) and there are over 700 installations that we know of in use in over 70 countries around the world right now. It is stable and has a very high uptime percentage in properly installed and scaled systems, and we have done installations of VICIDIAL for companies from 1 to 300+ seats. VICIDIAL is GPL and free, and there is a very active community support forum. There are also paid support plans available. Hope that helps, MATT--- While I respect and appreciate your biased opinion, I was hoping for more input from objective users. Thanks :-) -- Ein Bielaczyc [EMAIL PROTECTED] NOTICE: This E-mail (including attachments) is covered by the Electronic Communications Privacy Act, 18 U.S.C.2510-2521, is confidential and may be legally privileged. If you are not the intended recipient, you are hereby notified that any retention, dissemination, distribution or copying of this communication is strictly prohibited. Please reply to the sender that you have received the message in error, then delete it. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] US T1 Hangup Detection
Is there any way you could get a cut-sheet from Verizon. I know they are difficult to work with, but it would help to see for sure if your circuit is indeed Loop-start. You could always try EM_wink or EM immediate and see if there is any change. MATT--- On 7/8/08, Daniel Hazelbaker [EMAIL PROTECTED] wrote: Date: Mon, 7 Jul 2008 16:48:00 -0400 From: Jason Aarons \(US\) [EMAIL PROTECTED] Digital ISDN used Q931 messages. You should get a disconnect message from telco on the d-channel 23. I am pretty sure it is a T1 and not a PRI. I did try configuring it as a PRI and it started spewing all kinds of errors and completely stopped working. Date: Mon, 07 Jul 2008 16:55:27 -0400 From: Doug Lytle [EMAIL PROTECTED] Daniel Hazelbaker wrote: We are in the process of preparing to move our Asterisk server to a Digital T1 interface card instead of a analog card (via an Adtran which is now connected to the T1). I did a preliminary test the other A T1 or a PRI? Just make sure we're on the same page. Also, show us your zaptel and zapata.conf Again, I am pretty sure T1. It is a Verizon Flex-Grow package, which they list as expandable up to 24 voice channels. That and I tried configuring as a PRI and it harfed. The Adtran box we use now is configured as: Timing Mode Network Format ESF Line Code B8ZS Equalization0 dB CSU LpbkEnable Rx Sensitivity Auto Right now with Asterisk mostly working (it answers calls, dials out, etc. just doesn't detect hangup) my /etc/zaptel.conf is: # # Span Configuration # ~~ span=1,1,0,esf,b8zs span=2,0,0,esf,b8zs # # Channel Configuration # ~ fxsks=1-24 fxoks=25-48 loadzone = us defaultzone=us --CUT-- /etc/asterisk/zapata.conf: [channels] usecallerid=yes callerid=asreceived cidsignalling=bell cidstart=ring callprogress=yes# I have turned this off too ;- ; ; Define telco channels in rotary, these should be answered ; like a normal incoming call. ; context=bridgeNEC usecallerid=yes signalling=fxs_ks group=1 ; Part of ZAP group 1 channel = 1-9 context=incoming channel = 12 ;- ; ; Telco line, computer dialup, needs to be routed to output line. ; group=2 usecallerid=no channel = 10 ; PSTN attached to Span1:Port10 ;- ; ; Telco line, construction trailer fax, needs to be routed. ; group=3 usecallerid=no channel = 11 ; PSTN attached to Span1:Port11 ;- ; ; ADTran lines, used for outgoing to analog devices ; context=incoming group=4 usecallerid=no signalling=fxo_ks channel = 25-36 --CUT-- For context, the bridgeNEC context just dials out one of the ADTran lines to our existing NEC system, but the incoming context starts our menu-system, which was also not detecting hangups. I have also tried using loopstart and groundstart signalling, doesn't seem to make a difference. I am pretty well stumped myself. I need to call the telco about the caller id not working to verify that it is still turned on, but I figure I might as well wait so that if I need to ask them about the signalling I can know all the questions to ask at the same time. Thanks, Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Who has the best call recording solution!
Hello, We have done all sorts of customized recording archiving solutions like this with both Asterisk and VICIDIAL. Some of them housing millions of recordings that are stored on archive servers and are available through web-form for download instantly. We have also worked with programs like OrecX that are extremely flexible and offer a user interface for file access and management as well as live monitoring. All of the high-volume recording solutions we have installed use separate archive servers to store the recordings. MATT--- On 6/18/08, Mark Hamilton [EMAIL PROTECTED] wrote: Hi guys, So, I was wondering this morning as to who might have the best recording solution implemented. When I say best, I mean how they record, convert it to some low-diskspace-consuming format, and then leave it there, until a web-app requests it, and then it's changed to wav or mp3 and then lets it download, etc. Either that or someone records, then pushes off the recordings to a 'recordings server', then when someone requests to listen to it on the box that was recorded, it pulls the relevant recording from the 'server', converts it and allows it for download? Something like that.. you get the drift. Basically, I'm looking to record different queues that are hosted. But do not want to compromise too much diskspace, yet want to make it available for download through some web-app for listening (wav or mp3). Thanks, Mark. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] World Cheapest Predictive Dialer!
Hello, If you have a PRI-T1 in the USA, then you can set outgoing CallerID with just about any carrier. MATT--- On 6/17/08, Mark Hamilton [EMAIL PROTECTED] wrote: How can they even set such 1234567890 callerIDs anyway? For example, our inter/intra state calling depends a lot on the callerIDs. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Florell Sent: June 13, 2008 8:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] World Cheapest Predictive Dialer! Hello, I am not suggesting that the USA's laws exist outside of the USA, I can imagine the horrible problems that would cause in the rest of world. I wanted to point out that if you are using this service and doing business in the USA that you could face penalties for not following the law. According to the FTC, both companies(the scrubber and the client) are guilty of breaking the laws of the USA. If you are calling the USA and need to use this company's FTC DNC list filtering services then you may have USA-based operations of some kind. In such cases it is important to note that companies have been fined millions of dollars and have been shut down in the USA for violating these regulations. I am well aware of the fact that companies based outside of the USA routinely call-blast the USA with auto-dialers that send out callerIDs such as 1234567890 and do no filtering against the USA FTC DNC lists. A large portion of these companies are doing lead-generation for USA-based companies, and over the years a lot of those USA-based companies have been shut down for the activities of their lead suppliers. MATT--- On 6/13/08, Dean Collins [EMAIL PROTECTED] wrote: Yep it's funny how few people on this list realize that the usa's borders and laws stop 50 miles off the coast. It's also surprising how few Americans realize that a company incorporated internationally (Pakistan in this instance) even if owned as a subsidiary of a USA parent doesn't have to follow the laws of the USA but actually falls under the jurisdiction of the laws they are incorporated under. I'm not saying this is good or bad, 'm just saying that as 'asterisk' people we should be smart enough to play the laws that suit us to our advantage, if you think that the Global 1000 companies don't then you are kidding yourself. Besides we have the advantage in that almost everything we do can be virtual in most instances. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, 13 June 2008 7:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] World Cheapest Predictive Dialer! My guess is that they are outside of the FTC's jurisdiction. Thanks, Steve T On Fri, Jun 13, 2008 at 6:15 AM, Matt Florell [EMAIL PROTECTED] wrote: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Center
Hello, We have set up dozens of call centers, some using Asterisk Queues and the rest using VICIDIAL Call Center Suite. What you want can easily be accomplished with an average server and Asterisk Queues with not too much effort using standard Asterisk configuration features. we have set up a small 7-seat inbound call center like this for a client on a P4 1.6GHz PC and it has worked great for the last 3 years. Thanks, MATT--- On 6/16/08, Sherwood McGowan [EMAIL PROTECTED] wrote: broadband Voice wrote: Is anyone using Asterisk as a call center. I want to be able to set it up for my office line, when calls come in after 7:00pm Est want a recording to says the office is closed and have about 5 phones that I want to use as an agent. Can anyone share their implementation? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users There's a ton of us on here who have installations in call centers. What would you like to know? I currently have a Dual AMD64 2.4Ghz (Dual Cores on each) with 4GB RAM running a Tormenta 2 and a Digium 407. Two T1s going to a PRI, 12 FXO channels in a Rhino modular channel bank (all on the Digium card), and 2 24 port adtran total access channel banks running on the Tormenta. The Adtrans drive the 40 analog phones for the sales floor, and we have 25 SIP phones. All phone conversations are recording by Asterisk and are converted from GSM to Speex post-call by speexenc. We also run PostgreSQL and Apache on the same system to serve CDRs with links to recordings. Anything else you'd like to know? -- Sherwood McGowan VoIP / Telecom Solutions [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT How Digium Saved My Bacon!
Hello, I guess I am one of the lucky few to have one of these handy screwdrivers and it saved me when my son(aged 2) somehow locked himself in a bedroom and couldn't unlock the door. The door knob needed a very small slotted screwdriver to twist-unlock the door and the Digium tweeker(which was also in my pencil cup) saved my bacon as well that night :) Any chance of more of these being handed out at Astricon this year? Thanks, MATT--- On 6/16/08, Mark Hamilton [EMAIL PROTECTED] wrote: Now you're just trying to get us all jealous, Steve. No good. But I'd like that screwdriver! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: June 16, 2008 8:41 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT How Digium Saved My Bacon! I had a laser pointer and power point controller device but the Digium logo rubbed off after a week I do have a t-shirt though Thanks, Steve T On Mon, Jun 16, 2008 at 8:36 PM, Andrew Kohlsmith (lists) [EMAIL PROTECTED] wrote: On June 16, 2008 07:22:18 pm Mark Hamilton wrote: How come he has it, and he's in Paris! I'm in Toronto, and I don't have it? Yeah, me too. I even got a mention in the book, but no screwdriver? :-( -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] World Cheapest Predictive Dialer!
Hello, This looks an awful lot like an advertisement for a commercial product, which is only allowed on the biz list. Which you already posted this message to this week. I'm kind of confused. How do you get cheaper than free? Are you paying people to use your dialer? One other thing, it is illegal to scrub leads for a company against the USA FTC DNC lists unless those companies have paid the FTC and registered to have access to those leads, do you verify FTC registration before offering this service? MATT--- On 6/13/08, Muhammad Zulqarnain [EMAIL PROTECTED] wrote: Dear User! Although this email is intend for asterisk-business list however this might be useful for asterisk-user as well. Global IT Vision is proud to announce the World Cheapest Predictive Dialer. TeleRep Performance Optimizer Predictive Dialer is Hosted Web Based solution for Call Centers (with FREE DNC Scrubbing for US) that works from any where in the world with virtually unlimited agents. It's a prepaid pay as you go service. You just pay for the calls you make as our system allows you to add your own TRUNK so you can make calls to anywhere in the world with your own terminator. Pricing are as low as 0.014c/minute plus you will also save hundreds of $$$ with free DNC Scrubbing by using our hosted service. FEATURES LIST: · Free DNC Scrubbing for US Call Centers · Web Based Live Administration · Distributed Virtual Call Center · Campaign Management · Campaign Start/Stop Scheduling · Multiple Campaigns at a time · Agent Login from home · Press 1 for Live Transfer · Support from 1-1000 users · No Minimum Commitment · Pricing as low as 0.014c/minute · Use Your Own Carrier · No Dedicated Hardware/Software Required · Free Phone/Email Support · Live up-to-minute statistics Before starting TeleRep Performance Optimizer Predictive Dialer Solutions, our team along with Global IT Telecom Ltd A British Company amassed 7 years of experience building first class, mission-critical voice and Internet applications for large and small corporate clients. Our solution resides in a Tier-1 data center and employs the latest in voice and Internet technology to ensure security, redundancy, and the highest quality of service. Please contact [EMAIL PROTECTED] for more details! Thanks Regards, Muhammad Zulqarnain Email: [EMAIL PROTECTED] MSN: [EMAIL PROTECTED] http://www.gitv.pk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] World Cheapest Predictive Dialer!
Hello, I am not suggesting that the USA's laws exist outside of the USA, I can imagine the horrible problems that would cause in the rest of world. I wanted to point out that if you are using this service and doing business in the USA that you could face penalties for not following the law. According to the FTC, both companies(the scrubber and the client) are guilty of breaking the laws of the USA. If you are calling the USA and need to use this company's FTC DNC list filtering services then you may have USA-based operations of some kind. In such cases it is important to note that companies have been fined millions of dollars and have been shut down in the USA for violating these regulations. I am well aware of the fact that companies based outside of the USA routinely call-blast the USA with auto-dialers that send out callerIDs such as 1234567890 and do no filtering against the USA FTC DNC lists. A large portion of these companies are doing lead-generation for USA-based companies, and over the years a lot of those USA-based companies have been shut down for the activities of their lead suppliers. MATT--- On 6/13/08, Dean Collins [EMAIL PROTECTED] wrote: Yep it's funny how few people on this list realize that the usa's borders and laws stop 50 miles off the coast. It's also surprising how few Americans realize that a company incorporated internationally (Pakistan in this instance) even if owned as a subsidiary of a USA parent doesn't have to follow the laws of the USA but actually falls under the jurisdiction of the laws they are incorporated under. I'm not saying this is good or bad, 'm just saying that as 'asterisk' people we should be smart enough to play the laws that suit us to our advantage, if you think that the Global 1000 companies don't then you are kidding yourself. Besides we have the advantage in that almost everything we do can be virtual in most instances. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, 13 June 2008 7:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] World Cheapest Predictive Dialer! My guess is that they are outside of the FTC's jurisdiction. Thanks, Steve T On Fri, Jun 13, 2008 at 6:15 AM, Matt Florell [EMAIL PROTECTED] wrote: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] World Cheapest Predictive Dialer!
Not sure who complains, but it has happened before. the first case was in 2006 when Phase One Marketing who was fined by the FTC for indirectly acquiring the FTC DNC list from another entity. MATT--- On 6/13/08, Steve Totaro [EMAIL PROTECTED] wrote: I suppose if they are properly scrubbing (not the legal definition, but the practical definition of removing people that are on the DNC lists), then who is going to complain? Thanks, Steve T On Fri, Jun 13, 2008 at 8:19 AM, Matt Florell [EMAIL PROTECTED] wrote: Hello, I am not suggesting that the USA's laws exist outside of the USA, I can imagine the horrible problems that would cause in the rest of world. I wanted to point out that if you are using this service and doing business in the USA that you could face penalties for not following the law. According to the FTC, both companies(the scrubber and the client) are guilty of breaking the laws of the USA. If you are calling the USA and need to use this company's FTC DNC list filtering services then you may have USA-based operations of some kind. In such cases it is important to note that companies have been fined millions of dollars and have been shut down in the USA for violating these regulations. I am well aware of the fact that companies based outside of the USA routinely call-blast the USA with auto-dialers that send out callerIDs such as 1234567890 and do no filtering against the USA FTC DNC lists. A large portion of these companies are doing lead-generation for USA-based companies, and over the years a lot of those USA-based companies have been shut down for the activities of their lead suppliers. MATT--- On 6/13/08, Dean Collins [EMAIL PROTECTED] wrote: Yep it's funny how few people on this list realize that the usa's borders and laws stop 50 miles off the coast. It's also surprising how few Americans realize that a company incorporated internationally (Pakistan in this instance) even if owned as a subsidiary of a USA parent doesn't have to follow the laws of the USA but actually falls under the jurisdiction of the laws they are incorporated under. I'm not saying this is good or bad, 'm just saying that as 'asterisk' people we should be smart enough to play the laws that suit us to our advantage, if you think that the Global 1000 companies don't then you are kidding yourself. Besides we have the advantage in that almost everything we do can be virtual in most instances. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, 13 June 2008 7:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] World Cheapest Predictive Dialer! My guess is that they are outside of the FTC's jurisdiction. Thanks, Steve T On Fri, Jun 13, 2008 at 6:15 AM, Matt Florell [EMAIL PROTECTED] wrote: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] World Cheapest Predictive Dialer!
You are correct, a company that is outside of the USA does not fall under the laws of the USA. I said that myself. I also said that a company that is INSIDE of the USA or has operations INSIDE of the USA is subject to the laws of the USA. This includes companies that are based in the USA that use lead generation company that are outside of the USA. The company that is doing lead generation outside of the USA will not get shut down. The company that they are doing lead generation for INSIDE of the USA can get shut down for the activities of the company OUTSIDE of the USA because they are acting on their behalf. This can still be a problem for the non-USA company because they might not get paid for their lead generation activities if the USA-based client of theirs is shut down. There are many instances of this happening. A recent one was last year where a company called Ameriquest was fined $1 million for violation of the DNC through it's affiliates, some of which were off-shore lead generation companies. The company shut down because of this fine. MATT--- On 6/13/08, Dean Collins [EMAIL PROTECTED] wrote: A large portion of these companies are doing lead-generation for USA-based companies, and over the years a lot of those USA-based companies have been shut down for the activities of their lead suppliers. MATT--- Source please? I'm calling bullshit. If an incroporated entitiy outside of the USA makes international calls into the USA they do not fall under this law regardless of the purpose of the calls. Cheers, Dean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon question: four or five tracks?
Hello, I would recommend that if you do add another tech track that you spend a great deal of effort trying to make sure that sessions that would appeal to similar audiances are not done at the same time. This has happened a few times in past Astricons and it's always a tough choice for attendees that are interested in both talks to choose between them. To this end, I might suggest even video-recording the presentations to be replayed at night during the conference(or possibly on the web) so attendees can see what they missed if they were unable to sit in on a presentation. One other suggestion I might make is that after 6PM I think there might be a benefit from loosly structured BOF or discussion sessions. There is only so much Red Bull and Alcohol you can drink in the code zone. I quickly organized two after-hours discussion sessions during last year's Astricon and actually had a few dozen people involved in each one, it would be great if this could be done on a larger scale and officially organized. Thanks, MATT--- On 6/12/08, John Todd [EMAIL PROTECTED] wrote: We're busily churning away at creating the Astricon (http://www.astricon.net/) talk track this year, and it's been delayed by a problem that we've never had in years past: too many high-quality talk submissions. Not a bad problem to have, but still a problem. We have four tracks on the schedule: 1) Business Track - this relates to things like creating business models around Asterisk, technologies that embed aspects of Asterisk into their platforms, discussions of open source in the marketplace, and new technologies that can be added to Asterisk for specific application delivery reasons, among other topics. 2) Technology Track - Intro/Intermediate - Topics here range from basic introductions to Asterisk as far as feature sets and capabilities, and even into the moderately challenging topics of introductions to embedded systems and case studies. 3) Technology Track - Advanced - This includes more advanced implementation studies, protocol topics, new Asterisk features (LUA, for example), and inner workings of various Asterisk and third-party components. 4) Technology Track - Call Center/Large Scale - More case studies here but focused on large-scale systems. Carrier issues such as call recording, conferencing, clustering, and call center topics. We have had an overwhelming number of top-notch technical submissions for talks this year, which has been GREAT. Last year, we heard that there was a desire for even more technical tracks, so this year will fulfill that need. But we're stuck - we have way more topics than we have slots in the 4-track schedule, and so we've hit an impasse. We've had to start looking at cutting some really interesting topics because we simply don't have the space in the schedule. This is a terrible position, and so we're looking for what we can do to fix the problem. The obvious choice is Well, why don't you add a fifth track? So that is why I'm putting this message out. It's possible for us to add a fifth advanced technical track, but that would mean that there would be at any one time FIVE talks happening, four of which would be technical, and three of which would be classified as advanced. It will certainly be the case that there are overlapping areas of interest. Even with a fifth track, we are STILL going to have to turn down a few of the requests in the queue because of lack of slots, and at this point extending the conference another day is a very difficult option due to the hotel scheduling which is done far in advance. We also had some feedback from years past that a two-day conference seemed to suit everyone's schedules better, so this may be some unintended consequences from the compression. Our question to the community is: Is it too much to have 5 talk tracks at Astricon? Our initial instinct is Go ahead and do it but this does sound like a question that should be posed to the people who will attend. Your opinion would be valued if you could take the time to reply, but please try to summarize at the top of any replies with a Yes or No (even if you have more things to say) so I can keep a bit closer eye on the reply volumes. Feel free to reply on or off list. JT -- -- John Todd [EMAIL PROTECTED] Asterisk Open Source Community Director ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe Limits
Hello, We routinely run meetme with over 140 ULAW channels connected to 70 meetme rooms with no issues on an Intel Core 2 Quad core CPU. The major factor in capacity would be your CPU and RAM capacity. If you have at least a base-level P4 you don't need to worry about 12 participants. MATT--- On 6/8/08, Adrian Marsh [EMAIL PROTECTED] wrote: I've got to agree.. I've never given it much thought either... All of my calls are SIP/IAX based, coming in from the PSTN from a peer like that too.. I've never tracked the total number of conference users... But I'll bet we've hit at least 10.. And I've never seen the CPU go above 10%.. And that's on a really low powered (2Ghz, 1Gb ram, Dell 745) box. But it will be setup-specific.. So I would look at your CPU and memory stats, and run some tests and monitor that.. A. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John covici Sent: 08 June 2008 16:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MeetMe Limits 12 people is nothing -- I do 20 regularly -- however you may want to have them come in as muted or tell them to mute themselves, because the latency can cause very severe echoes if they are on a speaker phone or cell phone. on Sunday 06/08/2008 Sam([EMAIL PROTECTED]) wrote Actually I think they will all be calling in using regular pstn phones and cell phones. Sam Al Baker wrote: The 2 big questions are: -Are all participants using QoS end to end ? -Are all of them using the SAME CODEC. As the amount of Transcoding goes up,the work on the * box goes up and can be a problem. Sam wrote: I am thinking about using my asterisk server to host a conference withabout 12 other people from around the USA. Bandwidth issues aside, willthis work or will all the different latencies cause issues? Yea I know,I could just try it and find out but it is going to take alot of timeto get everyones schedule to line up, I don't want to go through thetrouble if I will just be disappointed. Thanks, Sam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe Limits
Hello, The load is usually quite high because this is VICIDIAL inbound call center traffic with full Asterisk-based recording. On a system with 70-80 Meetme rooms running with 2 participants each doing full Asterisk-based recording in each Meetme room the loadavg stays between 2.00-4.00 on a Quad-core Intel core 2 Quad processor with 4GB RAM. I have three systems like this in place at different call centers and the load is consistent for all three of them. Usually we put less load on a single server, but these were inbound-only scenarios which is less load than outbound. MATT--- On 6/8/08, Steve Totaro [EMAIL PROTECTED] wrote: Matt, Could you share the CPU usage, memory, and load average in the scenario you describe? What type of ULAW channels (SIP,DAHDI,IAX), or does it not matter? Thanks, Steve Totaro On Sun, Jun 8, 2008 at 5:29 PM, Matt Florell [EMAIL PROTECTED] wrote: Hello, We routinely run meetme with over 140 ULAW channels connected to 70 meetme rooms with no issues on an Intel Core 2 Quad core CPU. The major factor in capacity would be your CPU and RAM capacity. If you have at least a base-level P4 you don't need to worry about 12 participants. MATT--- On 6/8/08, Adrian Marsh [EMAIL PROTECTED] wrote: I've got to agree.. I've never given it much thought either... All of my calls are SIP/IAX based, coming in from the PSTN from a peer like that too.. I've never tracked the total number of conference users... But I'll bet we've hit at least 10.. And I've never seen the CPU go above 10%.. And that's on a really low powered (2Ghz, 1Gb ram, Dell 745) box. But it will be setup-specific.. So I would look at your CPU and memory stats, and run some tests and monitor that.. A. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John covici Sent: 08 June 2008 16:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MeetMe Limits 12 people is nothing -- I do 20 regularly -- however you may want to have them come in as muted or tell them to mute themselves, because the latency can cause very severe echoes if they are on a speaker phone or cell phone. on Sunday 06/08/2008 Sam([EMAIL PROTECTED]) wrote Actually I think they will all be calling in using regular pstn phones and cell phones. Sam Al Baker wrote: The 2 big questions are: -Are all participants using QoS end to end ? -Are all of them using the SAME CODEC. As the amount of Transcoding goes up,the work on the * box goes up and can be a problem. Sam wrote: I am thinking about using my asterisk server to host a conference withabout 12 other people from around the USA. Bandwidth issues aside, willthis work or will all the different latencies cause issues? Yea I know,I could just try it and find out but it is going to take alot of timeto get everyones schedule to line up, I don't want to go through thetrouble if I will just be disappointed. Thanks, Sam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com
Re: [asterisk-users] MeetMe Limits
Forgot to address your second question. DAHDI, that's a good one :) The channel type doesn't seem to matter. One has all agents on Zap channels through channelbanks with all calls coming in over IAX and monitoring done through SIP. One has all SIP agents with all calls coming in over SIP trunks, and another has SIP agents with calls coming in over Zap T1 channels. MATT--- On 6/8/08, Matt Florell [EMAIL PROTECTED] wrote: Hello, The load is usually quite high because this is VICIDIAL inbound call center traffic with full Asterisk-based recording. On a system with 70-80 Meetme rooms running with 2 participants each doing full Asterisk-based recording in each Meetme room the loadavg stays between 2.00-4.00 on a Quad-core Intel core 2 Quad processor with 4GB RAM. I have three systems like this in place at different call centers and the load is consistent for all three of them. Usually we put less load on a single server, but these were inbound-only scenarios which is less load than outbound. MATT--- On 6/8/08, Steve Totaro [EMAIL PROTECTED] wrote: Matt, Could you share the CPU usage, memory, and load average in the scenario you describe? What type of ULAW channels (SIP,DAHDI,IAX), or does it not matter? Thanks, Steve Totaro On Sun, Jun 8, 2008 at 5:29 PM, Matt Florell [EMAIL PROTECTED] wrote: Hello, We routinely run meetme with over 140 ULAW channels connected to 70 meetme rooms with no issues on an Intel Core 2 Quad core CPU. The major factor in capacity would be your CPU and RAM capacity. If you have at least a base-level P4 you don't need to worry about 12 participants. MATT--- On 6/8/08, Adrian Marsh [EMAIL PROTECTED] wrote: I've got to agree.. I've never given it much thought either... All of my calls are SIP/IAX based, coming in from the PSTN from a peer like that too.. I've never tracked the total number of conference users... But I'll bet we've hit at least 10.. And I've never seen the CPU go above 10%.. And that's on a really low powered (2Ghz, 1Gb ram, Dell 745) box. But it will be setup-specific.. So I would look at your CPU and memory stats, and run some tests and monitor that.. A. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John covici Sent: 08 June 2008 16:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MeetMe Limits 12 people is nothing -- I do 20 regularly -- however you may want to have them come in as muted or tell them to mute themselves, because the latency can cause very severe echoes if they are on a speaker phone or cell phone. on Sunday 06/08/2008 Sam([EMAIL PROTECTED]) wrote Actually I think they will all be calling in using regular pstn phones and cell phones. Sam Al Baker wrote: The 2 big questions are: -Are all participants using QoS end to end ? -Are all of them using the SAME CODEC. As the amount of Transcoding goes up,the work on the * box goes up and can be a problem. Sam wrote: I am thinking about using my asterisk server to host a conference withabout 12 other people from around the USA. Bandwidth issues aside, willthis work or will all the different latencies cause issues? Yea I know,I could just try it and find out but it is going to take alot of timeto get everyones schedule to line up, I don't want to go through thetrouble if I will just be disappointed. Thanks, Sam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL
Re: [asterisk-users] Disable sending CNAM over facility for 2bct
I think this is a bit beyond the average users-list question. There are very few people who do 2BCT and it was quite difficult to get anyone to help last year when I was trying to get it working on NI2 in libpri. I'm not really sure how to go about what you are asking, but I would suggest getting on the IRC channel for Asterisk and asking around there. Also if you can somehow get a hold of Matt Fredrickson(who is a very busy guy) at Digium, he could probably figure this out in a matter of minutes. MATT--- On 6/6/08, Remi Quezada [EMAIL PROTECTED] wrote: Hey, Is there a way I can disable sending cnam over the facility message when I am performing a two b-channel transfer? Thanks, Remi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error Counters on PRI Circuit
Hello, In Asterisk you can type zap show status to at least show you some basic error information: CLI zap show status Description Alarms IRQbpviol CRC4 wanpipe1 card 0 OK 0 0 0 wanpipe2 card 1 OK 0 0 0 wanpipe3 card 2 OK 0 0 0 wanpipe4 card 3 RED0 0 0 I also just want to mention that on Sangoma T1/E1 cards you get several tools for looking at the errors and statistics for each circuit with the wanpipe drivers. Here are some examples of the info you get: -- w3g1: AFT OPERATIONAL STATISTICS -- Number of frames transmitted: 17 Number of bytes transmitted: 136 Transmit Throughput: 0 Transmit frames discarded (length error): 0 Transmit frames realigned: 0 Number of frames received: 17 Number of bytes received: 136 Receive Throughput: 0 Received frames discarded (too short): 0 Received frames discarded (too long): 0 Received frames discarded (link inactive): 0 HDLC link active/inactive and loopback statistics Times that the link went active: 17 Times that the link went inactive (modem failure): 16 Times that the link went inactive (keepalive failure): 0 link looped count: 0 And there is a lot more information in there that helps to debug and monitor troubled circuits. MATT--- On 5/20/08, Joe Pukepail [EMAIL PROTECTED] wrote: Is there a way to see error counts on the T1 of a PRI? Hooked up to asterisk via a digium TE122. Looking for something to make sure I'm not getting any CRC, framing or other errors on the T1. Using asterisk 1.4.19 and zaptel 1.4.10 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk dropping around 2% of ALL calls ever since we moved to EM_W signalling?
Hello, I have quite a bit of experience with EM Wink T1s, and I have seen the problem you describe twice. In both cases it was either the carrier's equipment or the wiring somewhere between the carrier shelf and your equipment. In one case it was water in the line that would seem to cause the problem after it rained, and the other case was bad carrier equipment at their shelf, once they moved it to another port on another shelf the problem disappeared. Good luck, MATT--- On 5/15/08, Sherwood McGowan [EMAIL PROTECTED] wrote: Alright guys and gals, I'm a little lost, I'm primarily a SIP/IAX based guy, and have ended up with a Zap installation. Everything was fine with our old provider when we were using PRI, but the new provider screwed up on provisioning and we've been temporarily stuck with a pair of EM Wink T's. Ever since then, we've been dropping 1-2% of all calls (in or out) and even more strange, when a call gets dropped, a phantom call was being generated on the incoming side, but only by Asterisk, the T providers (Qwest) say they have no records of those calls. So, my question to you is, has anyone else dealt with a EM Wink T before using Asterisk, if so did you experience problems similar to this, and finally, if so how did you deal with it? Here's an outline of our system specs: Dual 2.3Ghz Athlon 2GB RAM Asterisk 1.4.16 (Tried 1.4.19 as well) Zaptel 1.4.10 51 Zap phones connected via SEPARATE TE407 and channel bank 2 EM_W T1's connected via TE407 25 SIP Phones All calls are being recorded by the Monitor() application, there is no timeout on the dial command, I can find NOTHING in the system config that would instruct Asterisk to dump the call. I have spoken with the Qwest technicians who have pulled their call records, and they report that we disconnected the call Any ideas, thoughts? I've reviewed the verbose (full setting, writing to file) and see that the far end channel disconnects, and then the near end goes into TIMEOUT. I've watched full debug output as well, from file, cannot find ANYTHING... Thanks for any help, Sherwood McGowan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Number of meetme conferences
Hello, The capacity greatly depends on the rate of calls entering and leaving those conferences. I see that you do call center systems so I would guess that the rate would be fairly rapid. It is really something you have to test and see. Using VICIDIAL in performance testing mode I have gotten to over 100 conferences on a similarly equipped server with a very rapid call turnover rate. MATT--- On 5/15/08, Wai Wu [EMAIL PROTECTED] wrote: Hi all, What is maximum number of three party conferences can a quadcore 3GHz system can handle? All the parties a setup with G.711 codec. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk queue cluster
Hello, You can cluster queues across several servers with VICIDIAL. We have clients with hundreds of seats taking in hundreds of lines across multiple Asterisk servers, and the calls are distributed to agents on all systems. MATT--- On 5/9/08, Vieri [EMAIL PROTECTED] wrote: --- Vieri [EMAIL PROTECTED] wrote: Is there a way of coherently setting up a clustered queue? Does anyone have examples/workarounds/links? I guess this isn't easy to implement, at least in current Asterisk versions (* 1.6?). I think Yate2 may have support for clustered queues but it's still alpha. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk queue cluster
Hello, VICIDIAL is completely separate of Asterisk queues and does not use them at all. It uses a database to keep track of all calls and agent availability and when a call comes into the inbound AGI script the system looks for the next agent in line no matter what server they are on and will send the call to them. The Asterisk Queues code is not really written to handle calls in this way and when you get to hundreds of calls in the queue there are a lot of issues that can come up with Asterisk Queues. When I started coding VICIDIAL 5 years ago I tried to use app_queue but was frustrated by how it kept changing from version to version, as well as the limitations of it, and (at the time) the fairly serious bugs. So I built VICIDIAL around AGIs, manager interface daemons and agents in meetme rooms. This allowed for extra flexibility as well as the ability to run the same codebase on many different version of Asterisk. There is a slight extra load cost for these advantages, but the ability to cluster many Asterisk servers together greatly overrides that problem in my opinion. MATT--- On 5/9/08, Steve Totaro [EMAIL PROTECTED] wrote: Matt, Is there any module or code that would allow this functionality without using VICIDIAL? I have been able to have about four hundred agents on a single box, that is not a problem (ULAW SIP only, no TDM). For distributing queues, I just use the queue timeout value set to a low threshold. In addition, I use a database to keep track of the queues to load balance. Can you elaborate more on what VICIDIAL does that my method does not. Thanks, Steve Totaro On Fri, May 9, 2008 at 8:38 AM, Matt Florell [EMAIL PROTECTED] wrote: Hello, You can cluster queues across several servers with VICIDIAL. We have clients with hundreds of seats taking in hundreds of lines across multiple Asterisk servers, and the calls are distributed to agents on all systems. MATT--- On 5/9/08, Vieri [EMAIL PROTECTED] wrote: --- Vieri [EMAIL PROTECTED] wrote: Is there a way of coherently setting up a clustered queue? Does anyone have examples/workarounds/links? I guess this isn't easy to implement, at least in current Asterisk versions (* 1.6?). I think Yate2 may have support for clustered queues but it's still alpha. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk queue cluster
On 5/9/08, Philipp Kempgen [EMAIL PROTECTED] wrote: Matt Florell schrieb: I built VICIDIAL around AGIs, manager interface daemons and agents in meetme rooms. Sounds a bit scary. Doing everything in MeetMe rooms just doesn't feel right IMO. the ability to cluster many Asterisk servers together greatly overrides that problem in my opinion. Agreed. I was unsure about using app_meetme at first, but it has proven extremely reliable and allows for things that just can't be done in app_queue like multi-party calls, DTMF macros, blind monitoring without using chanspy, playing of pre-recorded audio prompts to all parties in the session, raising/lowering/muting audio volume on any channel in the session and a few other features. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk queue cluster
On 5/9/08, Steve Totaro [EMAIL PROTECTED] wrote: On Fri, May 9, 2008 at 10:25 AM, Matt Florell [EMAIL PROTECTED] wrote: On 5/9/08, Philipp Kempgen [EMAIL PROTECTED] wrote: Matt Florell schrieb: I built VICIDIAL around AGIs, manager interface daemons and agents in meetme rooms. Sounds a bit scary. Doing everything in MeetMe rooms just doesn't feel right IMO. the ability to cluster many Asterisk servers together greatly overrides that problem in my opinion. Agreed. I was unsure about using app_meetme at first, but it has proven extremely reliable and allows for things that just can't be done in app_queue like multi-party calls, DTMF macros, blind monitoring without using chanspy, playing of pre-recorded audio prompts to all parties in the session, raising/lowering/muting audio volume on any channel in the session and a few other features. MATT--- I think app_bridge will eventually eliminate meetme. Meetme seems like such a hack. Yes, we are VERY MUCH looking forward to when Asteirsk 1.6 and app_confbridge are stable :) MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Predictive dialer - which one would you recommend?
On 5/6/08, Asterisk [EMAIL PROTECTED] wrote: Hi guys, I would like to ask you, if any of you has any experiences with the predictive dialers available for Asterisk? Are open source predictive dialers such as VICIDIAL Dialer any good? Which one would you recommend for a ca. 45 seat call center where most of the agents work on both inbound/outbound and are already using their own CTI software (so the predictive dialer software will be an appendix to the existing system and should be integrated with 3rd party software reasonably easy). Thanks, Alex I think that VICIDIAL is pretty good(full disclosure, I wrote it). It is currently in use at over 700 companies in over 70 countries around the world and is available in 9 languages. 45 seats blended are no problem for VICIDIAL, we even have some installtions running that are over 300 seats. As for integration with 3rd party software, we have integrated with many different kinds of web-based and client/server applications for our clients. Could you explain a little more exactly what functions that you would want the call center software to perform, and what functions you want your existing application to perform? Thanks, MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hyperthreading and multicore
On 4/28/08, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Sat, Apr 26, 2008 at 11:15:42AM +1000, Rob Hillis wrote: Two dual core processors would should four processors - each processor has two virtual processors for a total of four. I *think* Rob wrote that; *please*, people, turn your HTML off on mailing lists? :-) Two dual-cores don't have *any* virtual processors; all four cores are real. The processors that are virtual are the ones on HT Pentiums. Cheers, -- jra Multi-core processors on the Intel side are typically much faster and run cooler than P4/Xeon Hyperthreading processors. Also, I have heard HT processors explained this way, on an HT processor it's like running 2 virtual processors at 70% of the specs of the processor with HT turned off. It's not really like that in all situations, but overall it has held pretty much true for me in most non-Asterisk situations. Asterisk didn't benefit much from having HT enabled on a P4 with HT capability. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manual Wardialer
Hello, Sequential auto-dialing like this is pretty much illegal in the USA. The FTC has specific regulations against this as well as several states. Obligatory Simpsons reference: http://www.internerd.com/frink.retired/frinkv.3/inventions/at5000-2.gif http://www.snpp.com/episodes/4F01.html MATT--- On 4/27/08, Arthur [EMAIL PROTECTED] wrote: some people use a war dialer to provide call centers with numbers for their campaigns ... if number called rings the number is valid if it doesn't its invalid discarded. i wonder if that is legal .. its basically a scan of the network for valid numbers (that is potential buyers). i once was contacted by a company who offered this kind of service but i didn't trust them ... the numbers without names is ugly. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manual Wardialer
On 4/27/08, Andreas van dem Helge [EMAIL PROTECTED] wrote: So I can't dial my own number blocks for auditing? I do this manually right now dial 1 number, dial another on and on it gets very tedious and sometimes you loose your place. Approx every 2 months per number. The companies using these numbers have very specific reasons for requiring these audits, but franky I don't think its needed. AFAIK in my state doing that is legal because: 1) Its not telemarketing 2) its with the intent to communicate (if someone answers an 3) its for a legit business purpose, so its not harassment 4) The owner of the numbers (my company) and the users of the number (the clients) have expressly authorized this, although the law does not mention authorization I think this would be justification enough. I am not familar with any FTC / federal regulations since we don't telemarket I didn't think they were relevant but you do remind me when anything crosses a state line it can usually be considered interstate commerce... any resource you might have for interstate phone calling laws? I was thinking VCDial too... let me give that a try I've always wanted to mess with it anyways. I think I could load all the number ranges at one time also instead of doing one range at a time like I was thinking. And yes this is not war dialing because I looked up the definition and it seems war dialing is just scanning for modems, which is not the case here. If you expressly have permission then you can pretty much do anything you want. The regulations banning sequential auto dialing are primarily to prevent fishing for numbers with no intention of contacting people that first time you are calling them. It sounds like you are doing service level verification calling, which is very different from war dialing. Especially since you control the numbers you are calling. On Sun, Apr 27, 2008 at 9:23 AM, Matt Florell [EMAIL PROTECTED] wrote: Hello, Sequential auto-dialing like this is pretty much illegal in the USA. The FTC has specific regulations against this as well as several states. Obligatory Simpsons reference: http://www.internerd.com/frink.retired/frinkv.3/inventions/at5000-2.gif http://www.snpp.com/episodes/4F01.html My servers generally don't have built in legs or otherwise any way to automatically relocate itself :) Wouldn't that be handy in if there was a power or data outage? :) MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quality problems with ISDN PRI
On 4/25/08, Jared Smith [EMAIL PROTECTED] wrote: On Fri, 2008-04-25 at 18:48 +, Arthur wrote: I still hope someone would enlighten us by his experience in doing call recordings without recording to RAM Drive. I can't speak for Steve's solution (as I'm not sure exactly what he's doing) but I could take a stab in the dark and guess that he's capturing the audio at the network layer (on a completely different box than Asterisk is running on) and recording it from there. But that's just a guess... To address several points: OrecX (http://www.orecx.com/) can do call recording outside of the Asterisk core using several different methods depending on your needs and channeltypes. In fact even with Sangoma TDM cards you can capture audio at the kernel level and send the audio as RTP streams very efficiently(3% CPU load for 92 channels) to an OrecX server on your network. It must be mentioned that setting up Orecx with retrieval might be a little complex for some Asterisk users, especially if you are recording a large amount of calls, or are recording on more than one Asterisk server, and if you choose this route you would do well to hire an experienced consultant(or contact Oreca directly) to do the install for you. As far as Asterisk-based recording, writing to a RAM drive(or tmpfs) is about your only option if you are planning on doing more than 50 concurrent recordings, if you are using Asterisk it is a viable and tested solution. I have several client systems that are recording well over 50 calls concurrently on a daily basis this way. If you will be recording directly to hard drives with any frequency or volume I would strongly recommend NOT using standard IDE or SATA hard drives, they burn up and fast. Use a caching SCSI drive controller with some high quality SCSI drives and you can record to those drives for years even at 40 concurrent channels recording all day every day. Hope that helps, MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk really good??
On 4/14/08, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Sun, Apr 13, 2008 at 04:39:39PM -0700, Steve Edwards wrote: The shell script approach has the advantage of light weight. I do a minimal Centos 5 install and wget a single script which does everything -- configures the network, installs packages (OpenSER, Asterisk, Zaptel, Libpri, MySQL), adds users, and configures everything from services to timezone. I may stick with it, but it's getting a bit combersome and am interested in what has worked for others. Noted. Our solution may not help you all that much; I gather that with the exception of one small chunk of one file, all our boxen are configured exactly the same. It is actually two small chunks of two small files in Asterisk and one line in the vicidial conf file, and that's about it for unique server configurations, everything else is pretty much the same. We did recently add a custom backup utility to our SVN for VICIDIAL(AST_backup.pl) that will backup all conf files, agi, sound and other files(optionally web files and mysql DB and my.cnf backup) and tar/gz them then send to FTP server. This has worked well for multi-server backups for a couple of our clients so far and it will be included with the next release of VICIDIAL. The idea behind the script is to create a very simple hot-spare solution where all you have to do to replace a running machine is change the IP address of the spare server and un-tar/gz the file on a base-installed system and it will take the place of the failed machine within minutes. We haven't had to use it in production in this capacity yet, but it has worked in testing. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk really good?? (added, What is ViciDial??)
On 4/14/08, Eugen Soare [EMAIL PROTECTED] wrote: I'm glad so much has been sent about on the thread I create (bloated ego head :) ) It has gotten my curiosity up. What is VICIDIAL? Is it Public Domain? Pay for Software? What's it all about? (not looking for all the features, maybe I should put my understanding of it's functions and people can correct me.) It seems to be a software product that can handle call centers, be they in coming our out going calls. Has modules to take credit cards / and is customizable so that added functionality can be written. This is been very interesting! es Hello, VICIDIAL is call center software for Asterisk. It is designed around Asterisk, not compiled into Asterisk. VICIDIAL takes a different approach to the call center application from how Asterisk inbound Queues/Agents does it, since it uses Meetme rooms to house the agents allowing for more consistency across versions of Asterisk as well as a lot more flexibility in terms of features. The agent web interface is an AJAX application that will run well in most modern web browsers on computers with a PIII 500MHz or higher. With VICIDIAL you can do inbound/outbound/blended call handling and there are all sorts of features for call handling and agent functions. The latest VICIDIAL release is GPLv2, but for future major releases we are moving to the AGPLv2. VICIDIAL is free as in cost and speech. There are currently well over 400 companies using VICIDIAL in over 40 countries(unconfirmed survey results show over 700 company users, with over 17,000 seats total) and the agent interface is available in 9 languages. Hope that helps. For more info go to: http://astguiclient.sourceforge.net/vicidial.html MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk really good?? (added, What is ViciDial??)
On 4/14/08, Eugen Soare [EMAIL PROTECTED] wrote: Matt. Thanks for the reply and Link. That should get me started looking at that. Unfortunately, coming from the Nortel world. It may take some time to get up to speed on things. The hardest part (as I see it) is getting hardware/software instructions on setting up and then maybe connecting to someone elses box to play around with the integration of different sites. This looks like a good Fall/Winter project. Need to remodel the basement now. Anyway, I think that's a little off list. :) oops. It looks like there is a link on the web-page of the link that you sent, that provides a startup from scratch! COOL! Thanks again. Eugen Ah yes, my monster SCRATCH_INSTALL document :) If you run into any problems, please check out our very active VICIDIAL Forums: http://www.eflo.net/VICIDIALforum/index.php Good luck! MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk really good?? (added, What is ViciDial??)
On 4/14/08, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Mon, Apr 14, 2008 at 01:54:22PM -0400, Matt Florell wrote: With VICIDIAL you can do inbound/outbound/blended call handling and there are all sorts of features for call handling and agent functions. The latest VICIDIAL release is GPLv2, but for future major releases we are moving to the AGPLv2. VICIDIAL is free as in cost and speech. I noticed you had gone Affero. Could you expand on that decision, if you have a moment? What's the difference between the two licenses, did you consider GPLv3, and what's your situation on contributed code? We finally decided we would be going to AGPLv2 for our next major release due to a few hosted service providers out there that were altering the code to VICIDIAL, offering VICIDIAL hosted and not contributing their changes back to the project. And under the GPL they have every right to do this as long as the code is not installed on a client-owned machine or transferred to a client. This is known as the GPL-ASP-loophole. AGPL just closes that loophole and says that any customer of a hosted service like that has the right to the source code too. We have not done enough research on GPLv3 yet to want to move to it, and a lot of other GPLv2 projects are staying put as well for the time being. As for contributed code, we require a statement of this is my code and the project can use it and redistribute it from the author. Nothing very detailed at the moment because there are not many code contributors and the project is entirely GPL-based and is not dual-licensed. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk really good?? (added, What is ViciDial??)
On 4/14/08, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Mon, Apr 14, 2008 at 02:47:12PM -0400, Matt Florell wrote: On 4/14/08, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Mon, Apr 14, 2008 at 01:54:22PM -0400, Matt Florell wrote: With VICIDIAL you can do inbound/outbound/blended call handling and there are all sorts of features for call handling and agent functions. The latest VICIDIAL release is GPLv2, but for future major releases we are moving to the AGPLv2. VICIDIAL is free as in cost and speech. I noticed you had gone Affero. Could you expand on that decision, if you have a moment? What's the difference between the two licenses, did you consider GPLv3, and what's your situation on contributed code? We finally decided we would be going to AGPLv2 for our next major release due to a few hosted service providers out there that were altering the code to VICIDIAL, offering VICIDIAL hosted and not contributing their changes back to the project. And under the GPL they have every right to do this as long as the code is not installed on a client-owned machine or transferred to a client. This is known as the GPL-ASP-loophole. AGPL just closes that loophole and says that any customer of a hosted service like that has the right to the source code too. Ok; that's what I *thought* Affero's change was, but it's kind of hard to tell from the actual license... Yes, we had to read it several times ourselves, the version we have in our SVN trunk is what we settled on since there are several different text formats of the AGPL license floating around. We have not done enough research on GPLv3 yet to want to move to it, and a lot of other GPLv2 projects are staying put as well for the time being. I'm not really fond of it myself. I don't know enough about it at the moment to be fond of it or not myself. As more people move to it and it's provisions are tested I will hopefully be able to move from neutral to one side or the other at some point. As for contributed code, we require a statement of this is my code and the project can use it and redistribute it from the author. Nothing very detailed at the moment because there are not many code contributors and the project is entirely GPL-based and is not dual-licensed. Yeah; I was just worried about someone getting pissy about your relicensing from GPL to AGPL. Not that I expect it or anything... :-) I am fairly surprised that I have not heard a single negative comment about it from any members of our VICIDIAL community or anywhere else. There are actually other web-based projects that are moving to it as well(which is how I originally heard about it) and since it became an official OSI-approved Open Source License along with the special provisions that GPL made allowing for AGPL compatibility, there are more people talking about it in the last few months. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk really good?? (added, What is ViciDial??)
On 4/14/08, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Mon, Apr 14, 2008 at 03:24:02PM -0400, Matt Florell wrote: As for contributed code, we require a statement of this is my code and the project can use it and redistribute it from the author. Nothing very detailed at the moment because there are not many code contributors and the project is entirely GPL-based and is not dual-licensed. Yeah; I was just worried about someone getting pissy about your relicensing from GPL to AGPL. Not that I expect it or anything... :-) I am fairly surprised that I have not heard a single negative comment about it from any members of our VICIDIAL community or anywhere else. Well, I'm not, actually... the people who *like* the GPL (that's, y'know, everyone except Trixter :-) would be more inclined to like AGPL, I would think; it merely extends the letter to better reflect the spirit -- which a lot of people think GPl3 does *not* do... There are actually other web-based projects that are moving to it as well(which is how I originally heard about it) and since it became an official OSI-approved Open Source License along with the special provisions that GPL made allowing for AGPL compatibility, there are more people talking about it in the last few months. If OSI approved it don't *they* then have the official language? Yes and no, they have the official language for AGPLv3, but not AGPLv2 which is the actual license that they first approved on March 12th. I can't find the exact version 2 draft that they approved, since it seems that they moved immediately to post version 3 on their website and just skipped version 2. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk really good??
On 4/10/08, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Thu, Apr 10, 2008 at 05:49:26PM -0400, Al Baker wrote: Please share more about this. What/How are you clustering the boxes ? Is this all VOIP or TDMF front and VOIP for agents in back ? What kind of Boxes ? What O/S What tools are you using to monitor this big-azz mother ? What, Matt? You haven't already talked about this here? :-) My new job is Matt Florell's old job, where VICIdial got started. I haven't counted the boxes lately, but I think there are 14 with quad-T cards in them, separate boxes for MySQL and Apache. Our architecture is FXS T-1 channel banks for the agent phones, usually 1 + 3 IXC spans per box, though we turned up a box a couple weeks ago with 3 channel banks, and no spans. All TDM; the only VoIP is the IAX trunks hauling load-balance calls around. And just the usual VICIdial tools, mostly, though I'm fixin to deploy either Big Sister or Nagios. Of course I have talked about it here, 3 years ago:) I even gave a presentation about it at Astricon in 2005: http://eflo.net/presentations/Astricon2005_matt_florell_PDF.pdf It is a bit dated(as are some of the servers there) but it is a good description of how that system was originally set up. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium T1 Card Crashing Server (Dell 2950)
Hello, It might not be Digium's fault, I ran into similar problems with Dell 2950 servers and other PCIexpress cards. I even went so far as to have several components replaced by Dell on one of the affected servers to no avail. After many months of banging my head against a wall I stumbled across the following posts on the Trixbox forums: http://www.trixbox.org/forums/trixbox-forums/open-discussion/acpi-default-install-2-4-0 http://www.trixbox.org/forums/trixbox-forums/open-discussion/tb-2-4-crashing-asus-amd-and-new-dell-server-spec After talking to some computer engineers at a few companies I learned that It seems Dell does not have very good quality control on the power control chipsets that they use and so on some machines you have to disable acpi(or enable it) at the kernel level. If you do not set it correctly, when the power saving functions trigger there is a higher likelyhood that an error will occur leading to a kernel panic. This is most likely the same problem so take a look at the forum postings and try disabling/enabling acpi in your grub startup. Of course it could be something else entirely, but this problem does seem to be common with Dell 2950, and this did fix the problem for me on more than one Dell 2950. MATT--- On 4/10/08, broadband Voice [EMAIL PROTECTED] wrote: We're using PAE Kernel. On Thu, Apr 10, 2008 at 4:30 PM, Michael L. Young [EMAIL PROTECTED] wrote: BUG: soft lockup detected on CPU#1! [c044b2a4] softlockup_tick+0x96/0xa4 [c042e214] update_process_times+0x39/0x5c [c04196ff] smp_apic_timer_interrupt+0x5b/0x6c [c04059bf] apic_timer_interrupt+0x1f/0x24 . You don't happen to be running a XEN Kernel are you? I saw this problem while running CentOS 5.1 XEN kernel and if you search their bug tracking system you will see some reports about this bug. A search on google revealed some possible solutions. This was the first thought that came to my mind when I saw this. Regards, Michael L. Young (elguero) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] interrupting MOH
Hello, We achieve this using an AGI script in the VICIDIAL project for our version of inbound queues. You start MoH then when you stream a sound to the channel it will stop MoH then after the sound is done you start MoH back up again. Probably a bit more involved than what you want, but it dose work well for us. MATT--- On 4/2/08, Atis Lezdins [EMAIL PROTECTED] wrote: Sorry for top-posting, but seems everyone on this thread did so. Also that would be my suggestion for now - call queue with periodic-announce. However i see that this would make nice architectural improvement - allow inject sound files into MoH stream. This would be useful for example in call queues - to inject all the queue announcements into MoH directly, rather than play them while blocking further queue actions. Regards, Atis On Wed, Apr 2, 2008 at 4:11 AM, Andreas van dem Helge [EMAIL PROTECTED] wrote: I think that's still a better idea than using a dump the caller into meetme hack and is actually what I was going to suggest. If you want something simpler than a queue then inject the sounds into the moh already. On Tue, Apr 1, 2008 at 3:09 PM, Rob Hillis [EMAIL PROTECTED] wrote: You may be able to achieve the desired result using queues rather than Dial statements. Overkill perhaps, but it's the only way I can think to implement it at the moment. John Millican wrote: Tilghman Lesher wrote: On Tuesday 01 April 2008 05:14:25 Pete Kay wrote: I am hoping someone can help me out on this. I want to be able to interrupt MOH every X seconds after the DIAL command is executed. The interrupt greeting is something like please wait while we transfer your call. How can I do that? Within the DIAL options, I can't see any announce frequency or options that can help. Could anyone please tell me how that function can be accomplished? The only way to do that currently is to implement the prompt within the MOH stream itself. Just off the top-o-my head(YMMV), couldn't you create a meetme and play hold music into the meetme and then also play the prompt into the meetme at the same time without interrupting the hold music? This would obviously not work for high load but... JohnM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No audio on Sangoma A104.
Are you using 64bit Linux? Do you have more than 4GB of RAM? Have you contacted Sangoma support? MATT--- On 3/23/08, Alex Balashov [EMAIL PROTECTED] wrote: Alex Balashov wrote: I'm running kernel 2.6.19 (tried 2.6.24.3 but had to downgrade as wanpipe stuff would not compile), zaptel 1.4.9.2, and wanpipe 3.3.2. And Asterisk 1.4.18.1. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
On 3/19/08, Benny Amorsen [EMAIL PROTECTED] wrote: Matt Florell [EMAIL PROTECTED] writes: But seriously, several of my clients use SIP exclusively, passing tens of thousand of calls a day on Asterisk 1.2.X with no issues. I have noticed that the load is slightly lower for SIP-only in 1.4, but I have not noticed any stability issues revolving around SIP on 1.2.X. No hung calls? Our 1.2.x customer PBX's are drowning in channel.c: Avoided deadlock for '0x91dbee8', 9 retries!. Of course you can just ignore the hung calls if you want, but they mess up hint state and prevent graceful restarts. 1.4.x fixes it. I will say that we did notice some SIP issues with older 1.2 releases, but on the current 1.2.24+ releases we really haven't had many problems, and we do not have hung channels. I should mention that most of these installations have all phones on a LAN and almost none of the calls are native SIP-bridged since they go through meetme rooms which might account for why we do not see problems like this. As for 1.4.X we are moving closer to putting a live production machine on it, just a few more weeks of testing like we have had for the last month, and I should be convinced of it's stability. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
I would suggest upgrading to at least 1.4.18. I was able to run it for about 2 weeks and almost one million calls before I could get it to crash, and the 1.4.19RC2 seems to fix even more of the locking issues as well. I know a lot of these problems still existed under 1.4.17. MATT--- On 3/18/08, Patrick [EMAIL PROTECTED] wrote: On Tue, 2008-03-18 at 07:04 -0400, Al Baker wrote: Could you clarify what you mean by a Dead Locked Channel ? That is not a term I am familiar with used in context to channels, databases yes, channels ??? A channel got locked but never unlocked causing all sorts of funky behavior. It's a bug. The developers have fixed a ton of these deadlocks in 1.4 so it's usually a good plan to try the latest and greatest version to see if the problem goes away. I'm not very familiar with queue setups but Doug Lytle's advice sounds like a plan. And try 1.4.19-rc2 to see if the deadlock problem persists. If it does then please file a bug so it can be looked at. Regards, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
On 3/18/08, Ben Willcox [EMAIL PROTECTED] wrote: A million calls sounds good, but 2 weeks, not so good. It's a bit disappointing to me that crashing /ever/ is acceptable, I had always had the understanding that asterisk was supposed to be rock-solid. I suppose it's some consolation that its not just me that has problems! Thanks for all the input. I think short term I will restart asterisk daily, then the action plan is to revert back to Debian Etch, and then install asterisk 1.4.18 from source, and hopefully this will improve things. Keep in mind that my tests go from 0 to 400 calls in about 1 minute then they keep that volume for several hours, and I kept running them for two weeks, and about 6 hours into the last test is when it crashed. I should mention that 1.2.26.2 is what I still use on all of my production servers and they will go for months without a crash. As for rebooting nightly or weekly, that is something we do on a lot of our high-volume servers just to be safe. When pushing Asterisk to high concurrent call volumes it is a good idea to give it a fresh start every day if you can. If Asterisk is being used as a standard office PBX it should be able to run for months with no crashes. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
On 3/18/08, Benny Amorsen [EMAIL PROTECTED] wrote: Steve Totaro [EMAIL PROTECTED] writes: I will probably continue this train of thought (1.2.X is more production ready) until these threads stop popping up on the list. I think you're being too kind to 1.2.x. It has numerous problems, most especially with locking in chan_sip. 1.4.x is a HUGE improvement. Who uses chan_sip? Long live IAX! :) But seriously, several of my clients use SIP exclusively, passing tens of thousand of calls a day on Asterisk 1.2.X with no issues. I have noticed that the load is slightly lower for SIP-only in 1.4, but I have not noticed any stability issues revolving around SIP on 1.2.X. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in the call center - how do you do it?
Hello, I have many clients(from 10 to 300 seats) running VICIDIAL for call centers, both inbound and outbound(and blended). I also have acouple clients that have over 100 agents using Asterisk Queues for inbound only. One of them wrote a little web page that integrated with their timeclock application that logs the agents in on Asterisk when they clock into the system. MATT--- On 3/5/08, Kev S [EMAIL PROTECTED] wrote: I was going to ask the same thing today as i am looking for better and more efficient ways to run a call centre using asterisk! Look forward to some responses. Kev -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ade Vickers Sent: Thursday, 6 March 2008 8:27 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Asterisk in the call center - how do you do it? Hi folks, If you are running a call centre (large or small) using Asterisk, I'd be interested to know how you log your agents in out: E.g. - Do you use AgentLogin (to force calls onto the agents, perhaps)? - Do you still use AgentCallbackLogin? - If you use AddQueueMember, are you - running it through the agent's phones (i.e. in the dialplan)? - through a manager interface some software (homebrew or otherwise)? - Do you allow agent hot-desking? - if so, how do you determine which agent is logged in at which desk at what time? - how do you deal with authentication, or don't you bother? It'd also be useful if you could tell me what version of Asterisk you're using. And, finally, a pure fishing expedition: - What kind of reporting (if any) do you currently get out of the Asterisk, and are you happy with it? The reason I'm asking this stuff is because since 2003 I've been working on an ACD reporting product for Nortel Meridians (and, more recently, Avaya and Cisco systems, although that's all early days); and I'm thinking that as Asterisk gains a toe-hold in the call centre market, there maybe a market for this reporting tool for Asterisk users too. The only downside is I just know that my client (who owns the IPR) will never allow the s/w to be opensourced, or even available for free :( But I guess I shouldn't be too unhappy, as it puts the bread butter on my table too... All the above said - I should add that I'm a complete convert to Asterisk, use it daily (albeit at a fairly low simplistic level), e.g. I've only just got around to using a queue on my main POTS line, so I can login at any of the 4 Asterisk boxes I use around Europe, without having horridly complicated dialplans... Many thanks in advance for any responses, Ade. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.21.4/1312 - Release Date: 04/03/2008 21:46 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by Mail Call antivirus software, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
