[asterisk-users] Really Big Queues
Hi, How do you folks handle really large queues (350+ simultaneous callers) in your Asterisk PBXes? We're going to be bringing in around 16 PRIs' worth of inbound callers, doing skills-based routing, and queuing them up for approximately 200 agents. What's the best way to handle all of these callers? We want to record the calls and we'll probably use the ramdisk method that has been discussed on this list. Here's some ideas that I'm considering: Idea #1: Use servers with (2) Digium 4-port PRI cards, running Asterisk, as media gateways. From here, send calls to 2 or more Asterisk queue servers. For each incoming call, run an AGI on the media gateways that determines which queue server is least loaded. Send this incoming call to the queue server over an IAX2 trunk. The problem with this method is that the queues are not unified; if one queue server suddenly has available agents, queued callers on the other queue server cannot be (easily?) transfered to the server with available agents. Also, running an AGI for each incoming call is lame and slow. Idea #2: Use 3com VCX V7122 media gateways to terminate the PRIs and send the calls to a load balanced pair of SER proxies. These proxies will somehow keep track of the state of the Asterisk queue servers and redirect the incoming calls to the least loaded (most available) queue server. The problem with this method is that, by using SIP, we'll probably see higher interrupt load on the Asterisk queue servers. Additionally, I'm not a SER expert yet and I have no idea how to get SER to monitor the state of the Asterisk queue servers. As with Idea #1, the queues are also not unified, which sucks. Idea #3: ??? (profit!) Do you fine folks have any ideas or suggestions? thanks, Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium TE407P vs. Sangoma A104d
Sorry for the old thread revival...I bought three Sangoma A104 cards to use as T1 (not PRI) data cards in an OpenBSD router. I was disappointed to find out that trunking is not supported with this configuration. I contacted Sangoma and was told that they would look into it but I haven't heard back from them since. Sangoma has chapped my ass a bit because of this. I'm sitting on $4500 of useless hardware. Anybody want to by some A104s? ;) My advice: go with Digium. Chris On 12/4/06, Michael Collins [EMAIL PROTECTED] wrote: Has anyone had experience with one or both of these cards? I'm in a position where I might need to recommend one over the other. I've read everything that I can find online, so now I'd like to hear of personal experiences. Everything I read on both cards is 5 stars! Awesome! It Rocks! They both seem to have similar capabilities, similar pricing, etc. Could those of you who have seen these in action please give us some feedback? I'm interested in anything that might help me decide, be it warranty info, vendor responsiveness, etc. Thanks! -MC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cyberdata paging speakers - anyone use them?
For our stores, it would be nicer to have some kind of device that automatically mutes our music before playing input from the Asterisk pager. We already have a store full of speakers, no reason to duplicate them. Has anybody heard of such a thing? On 7/21/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: POE sip speaker http://www.cyberdata.net/voip/voip-speaker.html Anyone use these? How well do they work? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MoH from Sound Card: Does it actually work?
Hi, I've followed the instructions on the Wiki for pulling music-on-hold from my sound card's line input. It doesn't work, however. MoH starts and immediately stops. Apparently, I'm not the only person having this problem. I'm thinking that maybe arecord(1) is not sending the right kind of audio to Asterisk. To test things, I took my ast-playlinein script (mentioned in the wiki) and piped it to aplay(1). By doing this, I was able to hear the line input over my speakers, so the sound card and ALSA *are* working properly. Ideas, anyone? thanks, Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music on Hold from Soundcard
Alex, Did you ever get an answer to this or figure it out? I'm having the exact same problem. In fact, I can run the ast-playlinein and pipe it to aplay and hear the sound over the sound card just fine. I have a suspicion that Asterisk does not like the format that arecord is spitting out or something along these lines. Any ideas, anyone? thanks, Chris On 5/5/06, Alex Robar [EMAIL PROTECTED] wrote: Hi everyone, Sent this out previously, but it didn't seem to show up. My apologies if this is a duplicate! I've been trying to get MoH to work from the line-in on my soundcard, but as of yet have had no success. I found this script that should allow for it to happen: http://www.sineapps.com/news.php?rssid=722 The script, when run as the asterisk user, works properly and streams sound to stdin. I can use arecord to record wavs which playback fine. But when Asterisk starts MoH it stops it immediately afterwards with no explanation. Has anyone gotten this to work? Or does anyone have any ideas on how to debug why MoH stops immediately after starting? Thanks in advance! Alex Robar -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF detection and Sangoma cards
On 7/12/06, El Flynn [EMAIL PROTECTED] wrote: Are you only having this problem for call parking? Any issues when the caller is navigating an IVR? We're not running an IVR on this particular system. Here's the strange thing: the DTMF is not coming from the inbound caller but rather, the agents who are using Polycom SIP phones and trying to park the calls. I'm not sure how Asterisk's DTMF detection works but in this instance, despite what I said earlier, I'm starting to think that the Sangoma card is not involved. It gets even stranger: when I call into the system using a Polycom phone (over POTS, on a different PBX in a different state), the agent can park my call. When I call in with my cell phone, the agent cannot park the call. Strange! Any ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server redundancy
For redundancy on the PRI side, we plug our PRIs into Redfone Networks' foneBRIDGE boxes. They've worked quite well for us. As others have stated, use short registration periods combined with some HA software to handle your SIP redundancy. You might also look into load-balancing SER proxies. On 7/10/06, Alejandro Acosta [EMAIL PROTECTED] wrote: Hi all, I have read a lot about * server redundancy, however I still don't know how to do it. Can any of you give me any advise? For example, I've read about ranchnetworks appliances but don't know if it will solve my problems. As you may guess, I need to have two servers with the same information (including configuration, cdrs and logs). Of course, If a server fells down I would like the IP Phone to register with the other server. Can I do that?, do I need a third server?, other appliances? Alejandro Acosta, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Parking breaks suddenly
Hi, We're using Polycom IP501 SIP phones (app version 1.6.4.0043) with Asterisk 1.2.9.1. I set up call parking last week and for a while, it worked great. It stopped working yesterday, all of the sudden. What happens is that when the phone user dials #999 (our parkext), the call does not get parked and the caller hears the DTMF. Actually, they don't hear the DTMF, they hear a popping noise as the keys are pressed. The configuration files have not been changed since call parking was initially enabled. I'm running a console with -vvv and I don't see any errors reported. Any ideas? Thanks... Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF detection and Sangoma cards
Hi, I posted earlier about Call parking breaks suddenly. I believe that I have narrowed this down to a problem with DTMF detection and the Sangoma A101 card that we use. Earlier, DTMF detection was not working at all. Then, I set 'relaxdtmf=yes' in zapata.conf and it works...sort of. When I call into the PBX from a digital desk phone, Asterisk is able to detect the agent's DTMF and parks the call as requested. However, if I dial in from my cell phone, the agent's DTMF is not detected and the caller (me) hears the DTMF on the lines. Does anybody have any ideas? thanks, Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] soekris hadware
Google and voip-info.org will have answers to all of your questions.On 5/17/06, Jonathan Gonzalez [EMAIL PROTECTED] wrote:Hi group,i'm brand new and i would like to ask about soekris hardware. I read along the web but i have some doubts that i think can be solved here.My question are the following:[...] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail wav49 format problem
On 5/19/05, Michael Stahl [EMAIL PROTECTED] wrote: May 19 13:48:51 WARNING[7860]: Not a wav file 49 May 19 13:48:51 WARNING[7860]: Unable to open fd on /cygdrive/e/pbx/voicemail/default/2460/INBOX/msg.wav I'm seeing the same thing. I'm running -HEAD, checked out earlier this afternoon, on Mac OS X. Here are the symptoms: 1) I can leave voicemail messages without a problem 2) The file is present in the directory: % ls -l /var/spool/asterisk/voicemail/bikeworld/102/INBOX/ -rwx-- 1 asterisk asterisk 5066 May 31 18:39 msg.WAV -rw-r--r-- 1 asterisk asterisk 251 May 31 18:39 msg.txt 3) The wav file is e-mailed to the recipient, who can play them without a problem 4) When the user dials the voicemail extension/application to listen to the message, I see the following console messages when Asterisk attempts to play the file: May 31 18:35:27 WARNING[21838]: format_wav.c:135 check_header: Not a wav file 49 May 31 18:35:27 WARNING[21838]: file.c:416 ast_filehelper: Unable to open fd on /var/spool/asterisk/voicemail/bikeworld/102/INBOX/msg.wav May 31 18:35:27 WARNING[21838]: file.c:802 ast_streamfile: Unable to open /var/spool/asterisk/voicemail/bikeworld/102/INBOX/msg (format ulaw): No such file or directory == Spawn extension (sipphones, , 2) exited non-zero on 'SIP/102-f008' At this point, the voicemail app hangs up on the user. Some possible factors: - I'm running Asterisk as a non-root user. /var/spool/asterisk and its subdirectories *are* owned by this user - I'm running -HEAD and I'm running on a Mac. Case sensitivity issue maybe? - In my voicemail.conf, I have 'format=wav49' because the default settting of 'format=wav49|gsm|wav' yielded voicemail files that were nothing but loud static Does anybody have any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail wav49 format problem
On 5/31/05, Christopher Snell [EMAIL PROTECTED] wrote: May 31 18:35:27 WARNING[21838]: format_wav.c:135 check_header: Not a wav file 49 - In my voicemail.conf, I have 'format=wav49' because the default settting of 'format=wav49|gsm|wav' yielded voicemail files that were nothing but loud static OK, setting 'format=wav' is a temporary workaround. Could there be some problem with writing wav49 files in -HEAD? Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Open Source Billing Software
On Tue, 29 Mar 2005 09:53:03 +1000, Rod Bacon [EMAIL PROTECTED] wrote: What I would like to know is has anyone found an open-source billing platform that performs basic billing functionality (pre/post) from RADIUS and/or Asterisk CDR and is written (well-written) in either PHP or PERL. What features and functionality is needed for such a system? I've been thinking about using Perl to write LaTeX source files, which can then be compiled into pretty PostScript and PDF paper bills or plain text that can be sent out by e-mail. Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HOW-To write an AGI
On Thu, 17 Mar 2005 15:18:02 +0800, Ronald Wiplinger [EMAIL PROTECTED] wrote: I tried wiki, but I got too many pages (I think all of them), ...as answer. I want to write an agi. I need a HOW-TO, is there anything available? It depends. What language do you want to use? Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
