Re: [asterisk-users] question about MeetMe performance.
Hi, I built a FARM OF ASTERISKs split into 3 geographically dispersed sites (for high aggregate bandwidth concerns). There were 60 machines in total. All of them Dual Xeon 3.0 with 2GB. [ This turned out to be way more CPU that I needed.] Each machine had 18~24 separate conference rooms. All rooms were available on all 60 machines. All 60 machines had 18 ~24 voice feeds (one into each conference. At peak loads every machine also had a maximum of 220 participants listening - - mixed among the different conferences available. I was listening DTMF codes from any listener to allow switching from room-to-room. But no audio allowed from the listeners. I designed very carefully to avoid any trans-coding from the point where the sound was first captured - - in the dirver's seat of a NASCAR race car. This was done for several years of live races broadcasts with amazing voice quality. At peak we had over 6,000 simultaneous callers listening. There. were no issues on Asterisk or SuperMicro machine performance. I had one major foul-up when 18,000 new call attempts came in within a 5 minute period and the regional telephone system of Southern California jammed up and dropped 10,000+ calls before the every reached the Asterisks. NOT AN URBAN LEGEND. This was deployed by a team of 3 guys using Asterisk 1.4.x in less than 30 days including purchase shipping of the SuperMicro 1-U machines, configuring the Asterisks, deploying to COLO sites, and cross connecting (via pre-existing SIP carrier gateways) with over 250 Nortel DMS switches in the mobile phone networks. ..mike.. At 02:45 AM 3/6/2009, you wrote: Content-Type: multipart/alternative; boundary==_NextPart_000_0038_01C99E37.E20D4480 Content-Language: fr hello, I will do a server to do a lots of conferences (MeetMe). I want to know that if I dont use a digum card, the limit of simultaneous calls is harder without a card than with a card ?if, yes, how harder is the limit? thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. ___ -- 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] Progress() and Proceeding()
At 07:28 AM 2/14/2009, Philipp Kempgen wrote: But OTOH Indicate progress or Indicate proceeding doesn't mean anything for the end user. 183 starts early media, 100 does not. Are there any situations where it makes sense to use either of these applications from the dialplan? Philipp Kempgen YES. In a large scale Farm-of-Asterisks application (thousand+ concurrent calls) where calls with a certain Tier 1 carrier requires 183 for some routes and not for others. Thus an adaptable dial plan and sip.conf. Unfortunately, the consumer should really care because it says a lot about their SIP provider(s) endeavors to provide reliable service - - - if we can adapt our Asterisks to their idiosyncrasies. RANT IMHO - - - because of the highly distributed [i.e. fractured-proxy ] design of the supposedly Tier-1 provider network. The carrier needs extra time and assistance to setup the RTP path with the real media port of another-carrier's point of origin. This usually indicates the calls are not organic calls from the carrier's own retail traffic but are 2nd or 3rd route calls collected from diverse wholesale sources [i.e. other carriers] where the RTP is probably not running over the Tier-1's own network. OUR OWN constant push to pass RTP over someone else's network must be controlled. We are deceiving ourselves (and our consumers) if we mistakenly believe that a recognizable Tier-1 brand name is synonymous with good VoIP service. /RANT ..mike.. ..mike.. ___ -- 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 and openser 1.4
At 07:32 AM 12/27/2008, fateme fatah wrote: Hi: Can asterisk 1.2 and openser 1.4 work togather ? Regards. Yes. I have done large deployments that where multiple SER (and OpenSIPS) are used for either inbound or outbound (supplier) proxy. with multiple ASTERISKS and Cisco 5400s. I have also put both on same linux box (5060 for Asterisk , 506X for SER) when necessary to meet technical challenges on interface with specific carriers. ..mike.. ___ -- 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] Conferencing Hardware
Go for it. ztdummy is not an issue. I have used ztdummy with 220 simultaneous participants in 18 different conference groups. At one time, I had 60 machines running simultaneously in a FARM all of which were carrying the same 18 conference groups with over 200 participants active on each machine. ..mike.. At 11:23 AM 9/28/2008, Gordon Henderson wrote: On Sun, 28 Sep 2008, Jim Boykin wrote: We plan to use asterisk for conferencing. As I understand, it requires either a separate hardware like x100p clone or ztdummy. What are the pro cons of x100p vs ztdummy. Any other hardware suggestions for conferencing? It should be able to handle few simultaneous conferences. I have one server which handles a few simultaneous conferences using just ztdummy - however there are rarely more than 4-5 participants and rarely more than 2 conferences on the go at any one time.. (2.5GHz AMD Semperon FWIW) Ztdummy using: ztdummy: Trying to load High Resolution Timer ztdummy: Initialized High Resolution Timer ztdummy: Starting High Resolution Timer ztdummy: High Resolution Timer started, good to go And zttest gets more 100%'s than not. Gordon ___ -- 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] Asterisk IVR Scalability
See replies in text below: At 07:09 AM 8/31/2008, Sriram wrote: Hi My Scenario is to implement Asterisk in a Call center.. I;ve TE420 Digium card and plan to terminate 4 PRIs (E1) on it. I;ve 30 Agents inside..Since its a PRI i m not using any hardware echo cancellation module.The calls would first land on Asterisk and depending on the options would be transferred to the Agent. I've read lot of opinions on voip-info.org giving asterisk hardware dimensions. I would like to take a final call depending on your expert answers : Scenario : There would be 120 calls for sure during a 2 hour period of a day , rest of times it would be serving max. 50 calls. No matter how many calls come only 30 would be able to talk to agents rest would be listening to some files on the IVR or be involved in some polling..This is what the client wants as of now but he needs a scalable solution depending on traffic.. Queries : 1. For this initial setup Will a Dual processor (Xeon) with 2 Gigs of RAM with TE420 be able to handle the load ? All my agents would be using the softphones This is fine. Even more powerful that what I have used with 240 call legs (call in and extensions out) 2. What should be the ideal CPU load that i need to watch - may be if the load average crosses 6 or 7 - should i worry ? Not to worry unless you do a lot of media / codec transcoding. Keep everything the same at all times. 3. Even though i m adviced against AGI scripts (as they eat precious CPU cycles)- they seem very powerful and i m desperate to use them...Will the above setup get hampered in any way if i use them ? I have used them to answer IVR with up very complex scripting and interaction. Very little overhead noticed even with 50+ running at the same time. 4. Now scalability - If i want to increase the agents to 50 from 30 and add another PRI - what are the areas i should focus on - another machine ? or some additional RAM and processor ? I did the opposite. I put 60 astrisk units together in a farm to handle IVR front end to 6,000 simultaneous calls. With multiple CPUs, I gained 24*7 dependability I;ve been working all along on Dialogic but want to shift to Asterisk as it has lot of features and just fits in my needs (PBX + IVR in 1 box! ). Good move! ! ! ! Please advice Thanks in advance Sriram ___ -- 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] HP server and Meetme applications
THis machine is well OVER POWERED for the task you define. The greater issue will be the media bandwidth. I have operated with 220 simultaneous g711 participants in 18 ~ 20 different conferences. At 02:45 PM 8/11/2008, you wrote: Hi list 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 ? AyMaN ALMONTAHA .ICT 11 AUG 2008 -- Get Windows Live and get whatever you need, wherever you are. http://www.windowslive.com/default.html?ocid=TXT_TAGLM_WL_Home_082008Start here. ___ -- 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] Asterisk concurrent calls count
At 02:25 PM 5/18/2008, Tzafrir Cohen wrote: Please suggest a test environment IMHO, it is definitely NOT EASY to come up with a standardized test without some standardized network configurations and standardized load generation tools. It is even harder when a non-standard or niche application use is intended. For example: I had to run a bench mark with live traffic calls that entered by the PSTN and VoIP gateways into a PAY-FOR-PLAY farm of asterisks. I had to generate a minimum load of 5,000 simultaneous calls with a specific distribution of call durations and to continuously pump that traffic to attain specific objectives for rate-of-arrival on new calls. This required me to tie up 10,000 phone lines. 5,000 outgoing calls to specific numbers that would be terminated by specific carriers plus another 5,000 inbound VoIP and TDM lines to receive those calls. Reason eventually prevailed and we got the marketing program managers to understand what can be shown by a much smaller set of lines (3,000 total, 1,500 IN and 1,500 OUT). This was a 3% sample of the intended full scale loading rather than a 10% loading. I would not expect any generalized benchmark to even begin to address all of the non-Asterisk elements in this over-all system. Indeed, how could I even base any estimates for this based on generalized benchmarks for products optimized for mass-market PBX,IVR, or CALL-routing applications. I am using extreme examples to make the point that the Integrator-Reseller has that responsibility. For the non-extreme examples, the vendors conservative estimates for middle-of-the-road users of pre-prepared solutions are just fine. BTW: It required only two guys and one hour to setup and perform the test with Asterisks. It took several days of advanced negotiation to agree on the methodology with all concerned. This is a typical situation when you want to make sure the client knows enough to make a valid decision. ..mike.. ___ -- 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 concurrent calls count
Al, Randy, (and others): What Al calls one very weak area for Asterisk is IMHO a difference in market perceptions. Asterisk is positioned for CPE - PBX - Appliance market which needs feature-rich appeal and mass-market focus. Using asterisk for large scale does not mean that I have used it as a large scale PBX. Indeed, the FARM approach that we will be discussing on Friday 23rd is for very-large scale deployments with a reduced-feature-set focus. Simply put, this is not on Digium'a program for broad market push. Rightfully, Digium is expecting it's distribution channels to push it CPE-PBX and mass market solutions. So it is up to the thousands of Asterisk consultants to be aware of these techniques and to serve the much smaller number of clients (mostly VoIP network operators) who need to deploy very large scale networks. Indeed, I am now working on a design now that supports 100,000+ simultaneous participants in an application specific deployment. In this scale of telephony application, the issues of IP bandwidth and PSTN carrier access points are much more difficult to manage than anything related to the Asterisk platform. If this is your interest, then drop in http://voipusersconference.org The context of the discussion is NON-COMMERCIAL. I have no product or service for sale. I am just discussing a different approach to using Asterisk. ..mike.. At 09:42 AM 5/16/2008, randulo wrote: http://voipusersconference.org On Fri, May 16, 2008 at 1:59 PM, Al Baker [EMAIL PROTECTED] wrote: this is one very weak area for *. There is NO ANSWER. Hi, There have been a couple of threads on this subject this week, so I'd remind everyone that next Friday's VoIP Users Conference is about *large scale* asterisk: After many requests, we finally have someone to talk on large scale implementation of VoIP systems with asterisk. Using a farm of Asterisk and Digium cards, tens Of Thousands of simultaneous calls can be made and Mike Trest has offered to take it all apart for us to look inside. More about Mike Trest: http://www.mike.trest.com/ ___ -- 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 concurrent calls count
At 11:44 AM 5/16/2008, you wrote: Yes, you could probably add 2 or 3 or 10 or 15 to the number of calls that a particular machine could handle, but from a support perspective, it doesn't matter how many the machine could theoretically handle, it matters how many it could handle in the particular installation in a supportable configuration (those are all those pesky variables we've been talking about). Absolutely! Right On! Tell it like it is!And many other cryptic encouragements. With very large scale deployments, I have a set of numbers available in my head that work well to predict how many machines will be needed for a particular application but I wind up being surprised by non-predictable rate of arrival issues. Since most of my deployments are tied with Television and other promotional support, a single reference by the on-screen (or on-radio) commentator, and the phones are instantly flooded with thousands of new call setup requests. Indeed, one such incident in a NASCAR race with 13M viewers, produced 18,000 new calls within two minutes. The rate of arrival of new calls was dispersed to a farm of 60 Asterisk in three widely separated regions of the US. However, approximately 15,000 calls were actually dropped on the PSTN / SS7 network before ever reaching three dispersed Asterisk farms. Those farms were being fed inbound calls by a network of 250+ Nortel switches with millions of subscribers. However, the Los Angeles area PSTN network access facility had only 900 spare channels available in that two minute period. Meanwhile, every asterisk answered every call and joint the callers into appropriate conference groups until every single available port was fully occupied. This illustrates that such issues of call capacity exist completely apart from the Asterisk or whatever machine is used for implementation. So everyone should not be surprised by it depends kinds of answers to the question of concurrent call counts. This application was so far off the typical product specifications that nothing published by Digium or anyone else could anticipate those surprises that come when you least expect. ..mike.. ___ -- 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] RTP and Sip Provider
HI, In sip.conf you need only the call SETUP ip address. The RTP may come from anywhere . It is NOT assured that it is just another port on the same IP address. Therefore, be careful you do not block the RTP port ranges in a firewall. Google voip-info for more information about RTP port ranges and Asterisk. Exactly where the RTP is will be established by details passed on the INVITE when a call is setup (either direction). ..mike.. At 02:48 PM 5/3/2008, you wrote: Hello all, I need to configure a new provider to complete calls to us, the provider gave to me 2 different ip address, one is the default host and another to RTP server, so far as i knew the rtp server should be the same address but different ports, anyway i think i´m completelly wrong about it.. someone could tell me how can i configure in asterisk this connection in sip.conf? Thanks, Chet ___ -- 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 generic sounds
Suggested General Recording Phrases: This number is . . . restricted by customer request . . . For call screening that allow customer to setup restrictions on selected inbound calls. ALERT ! . . . For preface to emergency notifications like reverse-911 This is . . . your requested notification . . . To change or cancel . . . your notifications . . . For generic notification applications. ___ -- 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 for larg
Hmmm, IMHO this is a fundamental SIP architecture issue. To meet my understanding of distribution, this would required a proxy function before the call answer() on the Asterisk. If , in an ideal world, this proxy function were to be in the path before answer(), the proxy would need added intelligence to examine the INVITE and deduce from it's content the need for IVR or VoiceMail. This is an Active Call Director functionality. IMHO, this well within the capabilities of several SIP proxy packages available today. Once answered, the Asterisk will remain in the call path. If that is ok in your need for distribution, you can push the call onward via any Asterisk dialplan extension to be serviced by a whole farm of other Asterisks for specific chores. ..mike.. At 07:03 AM 4/25/2008, you wrote: Does anybody know how to off-load an Asterisk Box so that to distribute its functions like IVR and VoiceMail or its PTSN gateway function into different servers? in this case , will the installation of Asterisk on each server differe and how these different servers will interact as a single logical -vs physical- server? thx alot ___ -- 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] need examples of asterisk and mysql integration
At 02:58 PM 4/23/2008, you wrote: Re performance hit. I actually re-wrote one of my frequently used AGI in C and even set the STICKEY-BIT to avoid reloading the static text portions. I noticed slightly lower disk activity level (but the perl file was probably in memory cache too). So keeping it memory resident with stickey-bit probably was not really needed. But I did see a drop in over-all memory usage. Alas, none of these 'savings' were big enough to matter. I notice same performance level regards number of simultaneous calls - - - but that too was moderated by eternal load balancing distribution to multiple Asterisk gateways. For a slightly different twist. I re-designed the AGI to be only a message-passing function that talked via SOCKET to a always running service. I let that service do all the heavy lifting DB work and the AGI itself only asked questions and got replies. Again, no measurable difference in performance. So, I summarize to say that with over 6,000+ simultaneous call on 60 different Asterisk boxes, there was no difference in OPERATIONAL performance that the human operators could measure. Since any one box was running a balanced distribution, the large peaks of inbound calls did cause a problem either. The CPU cores (dual core Xeon) were seeing about the same utilization levels with lots of idle time left over. Probably because I was using more lower cost boxes spread about the US rather than fewer large boxes with a greater number of calls per box. As several folks have commented, everyone should write your AGI in whatever language you feel is best for your long term support needs because that is where your real costs savings are to be found.Getting a few more cycles from the CPU by squeezing the programming is not high on my list of priorities. IMHO, My suggestion to use perl AGI still stands as good advice on any current technology machine. ..mike.. ___ -- 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] Can I roll my own E911?
At 01:17 PM 4/22/2008, you wrote: My question would be - is this actually compliant with the FCC E911 regulations applicable to VoIP providers? IMHO and EXPERIENCE before FCC, this arrangement is NOT compliant Reason: multiple subscribers using the same number neither number nor name nor location can be tied to a specific subscriber at the time of use. You will need waivers on file from each client that acknowledges the E911 service is verbal contact only and will not have a fixed location or subscriber name associated with the number that is seen by the E911 service provider. ___ -- 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] Can I roll my own E911?
Yes, if you have signed waivers, you may operate without fear of FCC. Just be sure to have physical paper in a file somewhere for each client in the event of an audit. And, this will also satisfy you legal advisors to avoid liability in lawsuit by a consumer towards you for any crazy reason if they think that you provided inadequate E911 service and failed to advise the consumer regards proper use and expectations.. But if you do get those waivers, it is compliant in principle, on that basis, right? -- 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] need examples of asterisk and mysql integration
Hi, I suggest you look at writing a PERL agi program to handle all of the MYSQL / DB access and just pass variables between your CONTEXT/dialplan. I have done a lot of these things. You can get PERL examples for DBI and use one of provided agi scripts as a prototype. ..mike.. At 04:13 PM 4/22/2008, you wrote: I'm presently working on a project to build a scheduling system accessible by both web and phone. on the web side one can query what items are available when by using the time or the item as a key then reserve for an available time slot. reservations may also be modified by the user that made them or an admin. Where may I find examples of doing similar things with asterisk? all I've been able to find thus far is examples of how to store call detail records and voicemail using a database. Thanks in advance, Eric P.S. Has anyone already built an asterisk/web based scheduling system like this? ___ -- 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] do cards just instantly go bad
At 12:52 PM 4/14/2008, Jerry Geis wrote: Hi - Been using a TE205P for a number of months - no issues. Today I was talking to someone and I heard click No more phone service. I still have data service on this T1 line. (partial phone) zttool reports the SPAN as OK. calls are not coming in or going out. Does a card just go bad like that? How can I tell if the card is bad? I was expecting/hoping to see something other than OK on zttool. Its reporting OK but still no calls. Probably not the card. More likely the T1 provider. Contact your carrier. Ask them to put a T1 level trouble ticket. Ask that you be on the phone when the tester is bringing the T1 down and up. Quite often, all that is needed is to BOUNCE the circuit from the switch end. ..mike.. ___ -- 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??
-Original Message- I'd be interested in sections like Rolling out a new server or How we maintain all the little configuration files without losing our sanity. Hi, I will contribute my 2-cents on how I maintained consistency on a large application with 64 + Asterisks that all had to have the same config and links back to a central DB. Whenever we needed a new machine, we just We had a master source location.with a master image We cloned the hard drive with linux dd copy of master image boot the new machine with this disk assign appropriate IP address perform some sanity checks prior to shipping Send either disk or full machine to remote COLO for physical install. After the machine came on line, it would have enough configuration to join the other members of the farm of asterisks. For intermediate updates, we used SSL-DSA keys between the master master image machine and each of the 64+ remotes. We would wrote our own script and gave it a list of each machine on which to perform the particular steps. When it was launched, we just went out to lunch or home at night while the remotes were updated. This application had as many as 6,000 simultaneous call running and we wrote the scripts such that each remote were placed in a take no calls status by the script so we did not kill any active traffic. We found that no canned package was useful to do this because each maintenance cycle was addressing a different part of the overall configuration and had slightly different commands that were needed. Any good script writer can do the same for what you described. Regards, ..mike.. ___ -- 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] E911 Recommendations?
At 03:05 PM 4/14/2008, Doug wrote: Anybody have recommendations for a reliable, good valued, E911 provider? In my experience, the most reliable service for me has always been associated with commercial PSTN number providers. When it comes to consumer line service, you want E911 to always work correctly as a human life is often at risk. We use a $.$$ per-call pass-thru via a major US carrier. We pass the service along to our wholesale DID clients. Even though it costs, it is such low usage, that very few of our clients pass it along to their retail consumers. I will look with interest at other responses to your question. ..mike.. ___ -- 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] Next Move - Hosting
Hi, Your question is appropriate because you are asking about the best design approach. Although it does have economic issues related. First, let us dispatch the economics because they will impact your technical approach. The real issue here is not cost of additional E1 but exposure to fairly large liabilities for Trunk/International calls on the same E1 as your company. If your shared customer is late on payment, your company will need to pay the E1 provider for all calls in any circumstances. This could lead to cut off of your companies service if you do not pay. Now the technical. Avoid the MITEL unless it has features that you cannot manage on the Asterisk - - - which are very few if any ! ! ! IF YOU ARE GOING TO REMAIN E1 CARRIER BASED: I suggest you separate your clients onto a different E1 where the phone numbers assigned can be easily tracked separately by your Asterisk billing. If your clients simultaneous calls justify the cost. Put each client on a separate E1. IF YOU ARE GOING TO ELECT VOIP CARRIER BASED: I suggest you select a SIP provider who can issue local numbers to you and that you manage the clients on a pair of asterisks with shared service. You can, with experience, deal with the over-lap extensions quite nicely since you are an IT guy.That pair of asterisks can both be served by a decent internet connection. Depending on the class of machine(s) you use, you can support 200 ~ 400 simultaneous calls. YES, pay greatest attention to the billing software you select because that will be you biggest black hole which can pull all your energy into it depths to resolve billing matters with your customers. NO, I cannot tout one billing system over another. Perhaps someone else will do that via private EMAIL because that part of this discussions is NOT for this list. Best wishes. ..mike.. At 06:09 AM 4/4/2008, you wrote: I posted this to asterisk biz but didn't get a reply.. I didn't want to offend anyone being that this is kind of branching into hosting, and maybe outside of the remit of this list. Hi Been lurking on the user list for a while but I have some what of an immediate requirement and I'm wondering if you can suggest the best solution (if mines a rubbish idea) I have been testing Asterisk as a bolt on to our Mitel 3300.. its been doing some softphones for users abroad, etc and I'm happy with the fact I want to progress to a full system. However during this testing phase 2 customers of mine (I'm a IT Service Provider) have ask for some managed, collocated small business servers, which include the requirement for me to host their phones. No Problem I thought, I'm well on the way to this anyhow. So I'm thinking (although not tried it) that if I got my Asterisk box running for my company (E1 card for outside link) I would AIX the hosted PBX for the customers to my PBX to allow them to make outgoing calls. I would get my teleco to provide phone numbers for them and also get my PBX to redirect that number to the hosted PBX. Is this correct so far? Or should I keep their system separate on another E1? Or should I forget my PBX and push their incoming / outing calls out to a SIP / AIX provider on the net and wash my hands of it? Also I know you can run multi context on one host BUT can they also run the same extension numbers? Or would I have to let one company have 401-410 and the next company have 411 to 420, etc, etc (I'm guessing that's the case) And lastly.. Call accounting.. Certainly found a lot of good info about certain call accounting applications but as anyone got any good feedback about one they personally use.. Id like to keep it GNU / Open Source / Free while I build myself up.. Although I don't want to compete with the big boys, Id like to think I could get 10-20 or so customers co-located. Cheers Tim This message is sent in confidence for the addressee only. Unless specifically stated, the contents are not to be disclosed to anyone other than the addressee. Unauthorised recipients must preserve this confidentiality and should please advise the sender immediately of any error in transmission. The views an opinions expressed in this e-mail message are the sender's own and do not necessarily represent the views and opinions of NS Optimum Ltd. Although this e-mail and attachments are believed to be free of any virus or other defects which may affect any computer or IT systems into which they are received, no responsibility is accepted by NS Optimum Ltd for any loss or damage arising in any way from the receipt or use thereof. Place of registration: England, Registered Office: Jenton Road, Sydenham Ind Est, Leamington Spa, Warwickshire, CV31 1XS, Registered No 3018839 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz This
[asterisk-users] Quick Help, Anyone? EMERGENCY
Hi, I have a disk crash on an 2006 vintage Asterisk box that has a g729 license from Digium. I have been able to re-install from media on the same chassis. Reactivation is in progress. Good so far. . . HOWEVER, I cannot locate my g729 files for the 'digits' portion of the sounds. I only need the ten digits 0.g729 1.g729 . . . 9.g729 Can some kind person zip them up and email attachment to me via [EMAIL PROTECTED] or give me a pointer of where to get them in next few minutes! REALLY URGENT. Thanks ..mike.. ___ -- 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] Quick Help, Anyone? EMERGENCY RESOLVED
Wow! Fast response. Thanks. I am back up now. ..mike.. ___ -- 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] does the meetme module still require an external timing source?
At 01:13 PM 4/1/2008, you wrote: Is app_conference stable now? I've never made it through a thousand calls without a crash. (With a busy call center this doesn't take all that long.) -HJC I have deployed a MEETME conference bridge based on a FARM of asterisks with 6,000 conference ports active using basic meetme() with a very complex IVR front end that we wrote in perl for the customer specific needs. Still in use after 3+ years. Hundreds of thousands of total participants and millions of minutes later, still running. Very happy with results. ___ -- 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] Langugae issue
Ayman, One solution is to write an AGI scrip to parse the number and read back in Arabic semantic order. for the last two digits and for certain special numbers like 11 , 100 , 1000, ... .I must bring out my old Arabic language books to do this myself, but if you will share the language files with the asterisk group, then I will make an example AGI for you that we can share with the list. If you are agreeable, let us continue EMAIL messages privately until we have something working that we can share with the list. ..mike.. At 09:20 AM 3/30/2008, aymen warfalli wrote: Hi list I add new directory for Arabic voices support and I 'd translated all the English voices files into Arabic , with language = ar ,and it is working fine ,except some problems in saying the number order ,because the Arabic structure is quite different for numbers ,where in English language we can say twenty two while the order should be two and twenty ,so please if you can guide me how to change the setting to do that . regads Ayman -- Watch Cause Effect, a show about real people making a real difference. http://im.live.com/Messenger/IM/MTV/?source=text_watchcauseLearn more. ___ -- 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] Langugae issue
Tzafrir, Anselm, and others. Thanks for your comments on my suggestion to Ayman. As one who is familiar, but not-native speaker with Arabic, Hebrew, and several other classical Semitic family languages, it would require much more time to try to fit those into the linear structure of SAY.CONF than to deal with it in a directly parsed manner. I can say the same for some Asian languages too. The results would recognized but would not be culturally acceptable. OFF TOPIC COMMENTS: I am constantly amazed at cross-language translations that try to follow the western language standards in computerized applications. Historically, the use of numbers came relatively late to western languages. While I am proud to be an American (as well as a computer-geek), I have crossed the multi-lingual multi-cultural barriers many years back! END OFF TOPIC COMMENTS. ..mike.. At 11:23 AM 3/30/2008, you wrote: On Sun, Mar 30, 2008 at 05:16:00PM +0200, Anselm Martin Hoffmeister wrote: Am Sonntag, den 30.03.2008, 09:54 -0400 schrieb Mike Trest - Personal: Ayman, One solution is to write an AGI scrip to parse the number and read back in Arabic semantic order. for the last two digits and for certain special numbers like 11 , 100 , 1000, ... .I must bring out my old Arabic language books to do this myself, but if you will share the language files with the asterisk group, then I will make an example AGI for you that we can share with the list. If you are agreeable, let us continue EMAIL messages privately until we have something working that we can share with the list. ..mike.. Dear Mike, for me it seems that this is what say.conf is good for: http://svn.digium.com/view/asterisk/branches/1.6.0/configs/say.conf.sample?revision=105596view=markup (which seems to be considered the new format). Perhaps it would be better to implement Arabic there than by means of an AGI script. Be sure to check with the developers wether this will be relevant for Asterisk 1.4 or if you need to go with 1.6 SVN to benefit. say.conf works nicely for some languages. I was not able to make something useful enough with its syntax for Hebrew, and from the little I know of Arabic syntax, it will share the same problem. One basic problem is that there's no gender-form parameter anywhere in the interface. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] Had it with Dell Garbage
Steve, I have fielded several hundred Asterisk and related VoIP boxes. I buy SuperMicro 1-U units mostly. I have also used their larger units with RAID and a full load of ULTRA SCSI (for MySql application). I like these because, after bad experience with DELL/COMPAQ/HP/IBM compatibility issue, the supermicro systems always load and work with all of the Fedora kernels will just with their RAID controllers. ..mike.. ___ -- 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] Pattern matching....
[746][704][048]
[At 01:21 PM 2/21/2008, you wrote:
On Thu, 2008-02-21 at 12:44 -0500, Michael Munger wrote:
Will this work to match any number from the 770,404, or 678 area
codes?
_[404|770|678]NXX
If this wonât work, is there a pattern that will do this?
No, it won't work, there's no '|' for alternative matches, and no parens
available for grouping, either. And your usage of char sets is off.
Try something like this:
_404NXX
_770NXX
_678NXX
as three separate extensions.
If you REALLY want to keep that as one extension, then you could:
_NXXNXX = {
Set(areacode=${EXTEN:0:3})
if ('${areacode}' = '404') {
things to do here
} else if ('${areacode}' = '770') {
things to do here
} else if ('${areacode}' = '678') {
things to do here
}
OR, you could do it this way, also:
_NXXNXX = {
Set(areacode=${EXTEN:0:3})
switch(${areacode})
{
case 404:
things to do here
break;
case 770:
things to do here
break;
case 678:
things to do here
break;
}
This is, of course AEL code, and this stuff would be inside a context
construct...
murf
Yours,
Michael Munger, dCAP
404-438-2128
[EMAIL PROTECTED]
Attachment encrypted? click here.
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Software Developer
Digium
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Re: [asterisk-users] Asterisk Scalability
This is standard stuff. I have switch over 200 simultaneous with g711 on a 1-U, Xeon-DualCore @ 3.0 using RH versions of Linux. Even higher with pass thru (no-transcoding) on g729. ..mike.. At 07:54 AM 2/8/2008, Femi wrote: This will be closed service provider network with own VoIP phones and gateways so we can assume that there is no transcoding Femi -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: 08 February 2008 12:15 To: Asterisk Users Subject: Re: [asterisk-users] Asterisk Scalability Femi wrote: Does anyone have data on the switching capacity of Asterisk based on the hardware? I need to know what type of hardware would be required to switch 100 simultaneous calls as opposed to 1000 or 1 calls, no TDM just SIP to SIP VoIP calls That largely depends on whether you need to do transcoding and between which codecs, etc. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ -- 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] Multi-SPAN (4xE1) Zap Group (Outbound)
Thanks to all who replied privately as well! ..mike.. At 03:41 PM 1/7/2008, you wrote: Mike Trest - Personal wrote: Hi, Can someone point me to a zapata.conf example that will create a single DIAL OUT group including all 4 spans on a TE4XXP? Try: group=0,1 channel = 1-15,17-31 group=0,2 channel = 32-46,48-62 group=0,3 channel = 63-77,79-93 group=0,4 channel = 94-103,110-124 This allows you to use group 0 to dial out over all 4 spans, but each span still has it's own group that you can use to troubleshoot. You can break this down even further if you need. ___ --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] Bugs??
When similar problem occurred, I traced the issue to remote GSM gateway with poor protocol stack. The asterisk was doing exactly what it was supposed to do. The IMMEDIATE work around we used was to put maximum call timer into extensions.conf exten = s, 6,Set(TIMEOUT(absolute)=3660) This gives one hour+one minute. With average call duration below 30 minutes this worked quite well for our GSM traffic purposes. You set to any value appropriate to your traffic. ..mike.. Currently I am running 120 VoIP SIP channels on my asterisk server but each day 2, 3 calls got hanged in asterisk, and on asterisk CLI show channels showing us as call UP but in real there is no call. When asterisk restarted the hanged calls removed from CLI with very high duration which damaged our billing system and customers accounts goes in high negative. First I tried to get call info from asterisk mysql CDR using billsec field but the same result then I create PERL AGI to get duration from ANSWEREDTIME and same result. Please have a look to the following URL which I put the result of show channel channelname you can see the DIALSTATUS=CONGESTION but Elapsed Time: 20h54m16s which really strange and out of my mind to control such as call. http://www.emafone.net/bugs.html Please advice us if it is Bug and solved in some ver or its need some configuration to avoid this issue. This is in both ver of asterisk 1.2 and 1.4 Regard, Looking for last minute shopping deals? http://us.rd.yahoo.com/evt=51734/*http://tools.search.yahoo.com/newsearch/category.php?category=shoppingFind them fast with Yahoo! Search. ___ --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] Multi-SPAN (4xE1) Zap Group (Outbound)
Hi, Can someone point me to a zapata.conf example that will create a single DIAL OUT group including all 4 spans on a TE4XXP? One friend says to change the group number all to 1 on all 4 spans. Another suggestions says it is possible to have these unique groups (1-4) and to combine all 4 into a single group 5. I like the second suggestion best. Can you guide me to the correct changes for my current zapata.conf? The 4 spans are stand alone E1/PRI trunks (Not NFAS). The CURRENT channel and group statements are: ;Span 1group=1 channel = 1-15,17-31 ;Span 2group=2 channel = 32-46,48-62 ;Span 3group=3 channel = 63-77,79-93 ;Span 4group=4 channel = 94-103,110-124 Thanks, ..mike.. ___ --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
