Re: [Asterisk-Users] mpg123 or asterisk

Steve Totaro wrote: Please let us know your results. I cannot really test this in production system since it is a $16,000/hr call center. I was using madplay but it was crashing and creating zombie processes, I figured native was not the way to go since all of the different audio streams.

Re: [Asterisk-Users] mpg123 or asterisk

Erick Perez wrote: Are there any good scripts to stress test MoH? I want to test this machine for 1000 "calls" on hold. Steve Totaro wrote: When I say high, I mean 1,000+ calls. Erick and Steve, You both speak of Asterisk systems capable of handling 1,000 calls. I currently have a Dell

Re: [Asterisk-Users] Turning AAAH into a call-center

Kevin Savoy wrote: Speaking as one of those call centers we are looking at doing a turn over to Asterisk from our Nortel systems and are doing it ourselves. We've looked at a lot of packages from Fonality, Signate, Aheeva and others and none fit our needs. Each has good aspects but none have all

Re: [Asterisk-Users] Dumping queue_log to MySQL

Kevin Savoy wrote: Anyone have a working solution for this? I played with the demo that came with QueueMetrics to see how they were doing it and it was working for a bit but now somehow every night it stopped. Perl and Tail are still running on the server but the information is not dumping to

Re: [Asterisk-Users] (Semi-OT) QoS Question FTP Living with Asterisk

Steve Totaro wrote: Not sure if cheesy is the right word. Sound solution may be a better adjective. Adding two NICs, one to each machine and connecting them directly via crossover cable on a totally separate network may be my best solution. No FTP traffic would even hit the NIC or the netwo

Re: [Asterisk-Users] Power over Ethernet (PoE) switch recommendations

We're using Cisco Catalyst 3560 Series 48 port PoE switches. So far, *they just work*. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users maili

Re: [Asterisk-Users] still some moh troubles

>> Bart van Daal wrote: >> Hi, >> >> After following the suggestions on the mailing lists and the wiki I'm still >> experiencing >> choppy moh. The song plays but with frequent noise parts. >> >> - I'm using asterisk 1.2.4 on our production server and 1.2.7 on the test >> server. >> - na

Re: [Asterisk-Users] Unable to allocate socket: Too may open files

Stefan Günther wrote: Hello, we are curently benchmarking an asterisk system 1034 sip users are logged into this system and the test software is trying to establish 400 concurrent calls. In the CLI I see the following messages: Apr 19 14:20:51 WARNING[4045]: rtp.c:911 ast_rtcp_new: Unable t

Re: [Asterisk-Users] Asterisk hyperthreading compiling.

Wai Wu wrote: Hi, Anyone know how to compile asterisk for a hyperthreaded processor? Thnx Wai, Asterisk makes heavy use of threads, so all that's required to take advantage of HyperThreading is an SMP kernel. The kernel itself will take care of scheduling the threads on the different v

Re: [Asterisk-Users] call center running Asterisk -soundquality-critical!

Wai Wu wrote: I did not install soxmix in my linux box. If you having issues with mixmonitor, you can put both legs of the call into a conference and record the conference Wai, How are you mixing the leg files? Do you run a process that moves them to a remote box with soxmix installed? Y

Re: [Asterisk-Users] call center running Asterisk -sound quality-critical!

> Wai, > > Please explain how "the in and out channels are mixed first before > they are written to the disk" using "monitor with no mixing onto the > scsi drive." I'd love to implement this on our system to cut in half > the I/O associated with Monitor(). > > Also, what bug does MixMonitor() hav

Re: [Asterisk-Users] Music on hold problem

Don Pobanz wrote: I believe I am using MP3. My musiconhold.conf file looks like this [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 Daniel and Don, Try switching to native MOH. You'll eliminate the decoding of the MP3s and the host of problems that come along with using mpg123.

Re: [Asterisk-Users] playback soundfile in memory

Akpome Akpoguma wrote: I want to playback sound file loaded in memory not from a file...is this possible? Akpome, If the sound file is being played more than once, there is a good chance that this is already happening. At one point, our production system had 100 calls in queue.

Re: [Asterisk-Users] Company List

>> Bruce wrote: >> >> >> The question was raised by a CFO who is looking at Asterisk if there is a list of >> companies using Asterisk. I have not found one yet, has anyone seen anything like >> this I can give him. > > Curt Shaffer wrote: > > I have not but if you find one, please pass it on b

Re: [Asterisk-Users] call center running Asterisk -sound quality-critical!

>>> Matt Roth wrote: >>> >>> These statements seem contradictory. I know of no way (short of a >>> custom patch) to tell Monitor() to mix the in and out legs prior to >>> writing them to disk. On the other hand, MixMonitor() does just that >>

Re: [Asterisk-Users] call center running Asterisk - sound quality-critical!

Wai Wu wrote: > You got to be kidding about 53 calls being recorded at sametime is an > issue. I have done at least twice as many on my dual xeon 3.4Ghz system > and had no problem as clients like to record every call that goes > through the system. Nope. We took our system to MCI's development

Re: [Asterisk-Users] call center running Asterisk - sound quality - critical!

>>On 4/10/06, Dov Bigio <[EMAIL PROTECTED]> wrote: >> >>Hi, >> >>I am using Asterisk for a call center on a Dual Xeon machine.. >> >>I currently have >> >>109 active channels >>53 active calls >> >>Every body is complaining about quality and cpu is around 80% idle. >> >>Is there any tuning I can d

Re: [Asterisk-Users] Re: update - 512 Simultaneous Calls with DigitalRecording

Boris Bakchiev wrote: The simplest solution and the one already implemented in linux, tmpfs. It would be best to allocate 4-8GB to tmpfs on /tmp and let the kernel do the work it was designed to do. And you would not be limited to PCI bus speeds. The DDR2800 is about 12GB/sec. Some would say "ov

Re: [Asterisk-Users] Re: update - 512 Simultaneous Calls with Digital Recording

Luki wrote: Has anyone seen these solid state "Drives" from gigabyte yet? - http://www.pcper.com/article.php?aid=224&type=expert&pid=3 Interesting device. Looks like the burst throughput is right on par with good drives, but you have better sustained throughput and obviously near zero lat

Re: Fwd: [Asterisk-Users] update - 512 Simultaneous Calls with Digital Recording

Erick Perez wrote: How much RAM disk is needed or are you using for your current needs? We're planning to do something like this. But I can't figure proper dimensioning. Erick, We are using Asterisk to handle our inbound call center operations. There are currently 158 leg files (produced by

Re: [Asterisk-Users] update - 512 Simultaneous Calls with Digital Recording

Matt Florell wrote: Matthew, thanks for your feedback and advice. what I actually experienced was the complete breakdown of Asterisk at around 60 concurrent recordings without it (the reality). The drive for saving your voice recordings is the same as your OS (Asterisk)? What do yo

Re: [Asterisk-Users] Too many open files

Raymond Chen wrote: Dear all, we have encounter problem when starting asterisk in the foreground, "asterisk -gc" with more 100 SIP calls concurrently. we have set ulimit to the highest value. still has this problem. Is this the problem keeping asterisk in the foreground or this is a

Re: [Asterisk-Users] queueue recording and what to do next

Anton Krall wrote: Guys, if you define recording on queues.conf and also define a monitor_filename var on your dialplna, you can record a queue call but, isthere a way to do something with the file after the call ends? I need to move the file to some other place but I cant find where to define a

Re: [Asterisk-Users] RE: Need More Simultaneous Voice Channel Capacity on Asterisk

Tadepalli, Hari K wrote: (OK - sorry for a 3rd attempt. I see that my message came up with no line breaks in the first two attempts). We are testing Asterisk (1.2.5, configured for an IP PBX) for the number of simultaneous multiple VoIP calls supported. Whenever we increase the number of SI

Re: [Asterisk-Users] 1.2.6 doesn't use mpg123?

>Matt, > >Before switching our system from the rawplayer method to native MOH, I >consulted Kevin Fleming. He said the impact on the system would be "not >much, more memory usage though." > > >Right now we have seventy ca

Re: [Asterisk-Users] 1.2.6 doesn't use mpg123?

Matt wrote: Right now we have seventy calls waiting in queues (all with native MOH) and 120 calls connected to agents. The box is jumping between 50%-60% idle. "ps auxm" shows 241 threads for Asterisk, but none of them take more than 0.8% CPU. Personally, I wouldn't mind seeing an option that

Re: [Asterisk-Users] 1.2.6 doesn't use mpg123?

Matt wrote: Ok.. see it... so now my question is which should I use? Obviously a hold system using ulaw for hold files is going to use less CPU, but is it more stable to have Asterisk playing the sound files? Especially since it has to start a seperate stream for every on hold person? Se

Re: [Asterisk-Users] update - 512 Simultaneous Calls with Digital Recording

On 4/3/06, Isaac Xiao <[EMAIL PROTECTED]> wrote: Hi All, In previous mail lists, people talked about a solution to record large amount of simultaneous calls. And then it seems that RAM disk solution was the best choice due to the I/O bottleneck of Hard disk (System). Please find the prev

Re: [Asterisk-Users] Re: Asterisk in production as a fax server, anyone?

Craig, Please correct the date on your machine. Your emails stick to the top of the list because they have a date of 6/30/2006. Thank you, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --Bandwidth and Coloc

Re: [Asterisk-Users] Asterisk & SMP: Is irqbalance Redundant on 2.6 Kernels? - Resolved

; Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer Matt Roth wrote: Asterisk users, I posted the following email to the Fedora users list <https://www.redhat.com/archives/fedora-list/2006-March/msg04154.html> and it got no responses, so now I'm

Re: [Asterisk-Users] Asterisk & SMP: Is irqbalance Redundant on 2.6 Kernels? - Resolved

quot; Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer Matt Roth wrote: Asterisk users, I posted the following email to the Fedora users list <https://www.redhat.com/archives/fedora-list/2006-March/msg04154.html> and it got no responses, so now I'm

[Asterisk-Users] Asterisk & SMP: Is irqbalance Redundant on 2.6 Kernels?

Asterisk users, I posted the following email to the Fedora users list and it got no responses, so now I'm calling on your expertise. Please take a look at it and share your knowledge on the subject with me. Additionally,

Re: [Asterisk-Users] VERY IMPORTANT(TREAT WITH URGENCY)

I think this is "The Last Starfighter" of Asterisk. If you solve this problem in a timely manner, expect to be taken away by aliens to help them develop their VOIP networks. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer Andrew D Kirch wrote: Andrew D Ki

Re: [Asterisk-Users] music on hold without mpg123

Lenz wrote: Hello list, after the last time that mpg123 wen ballistic on our production system, we decided to skip mp3 playback altogether and to go for raw files. After half an hour playing with mpg123 and sox parameters in order to translate a mp3 file to a wav file that can be streamed

Re: [Asterisk-Users] Asterisk programmer needed

Please take this discussion to the Biz list. Thank you, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer Bob McDowell wrote: Just curious, but what does it pay??? Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On

Re: [Asterisk-Users] Brainstorming dual-core and Asterisk

Jim Van Meggelen wrote: Let me run something that's been floating about in my noggin by everyone: Given that Asterisk does not make use of dual core CPUs or dual processors... Jim, That statement bothered me, because we are running Asterisk on a multi-processor system to help accomplish our

Re: [Asterisk-Users] Lowering Server Load

All, Just a quick update on our progress with the RAM disk solution for digitally recording large numbers of calls via Monitor.  We are currently recording approximately 80 - 100 concurrent calls to the PCM format on our production server.  We also have over 220 dynamic agents logged into 10 q

Re: [Asterisk-Users] Milliwatt Analyzer available

Roger, Thank you very much for this valuable contribution. In my opinion, this is a great candidate for asterisk-addons. Sincerely, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer Roger Schreiter wrote: Hi, some days ago we discused here the need for

Re: [Asterisk-Users] Re: MOH native files

Tomislav Parčina wrote: sox out.wav -r 8000 out.gsm I have problem with this command. It runs fine, but when I play that file it is twice long as it should be and double slow as it should be. So wav file that was 2 min long becomes 4 min long gsm file. How can I fix that? Tomislav,

Re: [Asterisk-Users] Analyzer for Milliwatt

Andrew Kohlsmith wrote: What is being discussed here is basically what I was planning on doing for an automatic VOIP quality check. Using miliwatt and analyzing it for pop/jitter/etc as well as sending other known waveforms and comparing what was received to what was expected and coming up wi

Re: [Asterisk-Users] mpg123 alternative?

Soner Tari wrote: I vote for the raw file format, due to the reasons listed here: http://www.orderlyq.com/asteriskqueues.html Of course you need to convert all mp3 moh files to raw format manually, but it's easy as described there. We were using the rawplayer method on our server, but it end

Re: [Asterisk-Users] How do I install speex for asterisk?

Jesus, If you recompile Asterisk and still have problems, take a look in "/codecs/Makefile". It'll tell you where Asterisk expects to find stuff in order to trigger the building of the speex-related objects. If the build goes as planned, the "/codecs" directory will contain three speex-rela

Re: [Asterisk-Users] Disabling SELinux in FC3 - good or bad

Zach A wrote: Hi all, I had problem running MySQL on FC3 and what I found from googling was that SELinux should be disabled to make MySQL work n FC3. Now I am concerned about Asterisk, is it a good idea to disable SELinux. Or is there any other way to make MySQL work without disabling SELinux?

Re: [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update ornot?

Keep in mind that if you want to run Asterisk Business Edition, RedHat Enterprise 3 or Fedora Core 3 are currently required in order to receive full technical support. My options were narrowed down further by the amount of RAM in our production server. It has 20GBs, and all of the documentati

Re: [Asterisk-Users] Help Debugging Dropped Call Audio - Add'l Info

Martin, Please follow the "Steps to Reproduce" in my bug report and post your results back to the list. If you're comfortable with adding the ast_log() statements to the ast_read() and ast_write() functions located in channel.c, it would really help to show that we are experiencing the same

Re: [Asterisk-Users] Help Debugging Dropped Call Audio - Add'l Info

replicated the problem, as well. Thanks, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer Matt Roth wrote: List users, Below is a bug report documenting Asterisk dropping call audio at very low loads (1 call). I have personally reproduced it on th

[Asterisk-Users] Help Debugging Dropped Call Audio

List users, Below is a bug report documenting Asterisk dropping call audio at very low loads (1 call). I have personally reproduced it on three separate machines, multiple network architectures (including a 48-port Cisco Catalyst 3560 POE switch dedicated to an Asterisk server and two Snom 3

Re: [Asterisk-Users] Pops in Call Recordings Tied to Dropped Audio in Calls

Kevin P. Fleming wrote: On 2.6 kernels, ztdummy can use either the kernel ticks (jiffies) for timing, or the hardware realtime clock (RTC). By default it uses the RTC when built against kernel headers for 2.6.13 or newer; it can be manually configured to use the RTC for older kernels. The RTC

Re: [Asterisk-Users] Pops in Call Recordings Tied to Dropped Audio in Calls

Kevin P. Fleming wrote: Matt Roth wrote: We have no hardware timing device on the box (no Zap hardware) and are using the 2.6 kernel as the timing source. Digium tech support told us this is better than ztdummy, which we were using before. We experienced the same problems then, as well

[Asterisk-Users] Pops in Call Recordings Tied to Dropped Audio in Calls

n of the same problem. Im newish to Asterisk, but I thought RTP only came into play on SIP/IAX/MGCP calls ... So the fact that I seem to have the problem when calling from a CO Trunk line (well, Inbound PRI) into a digium 4 port PRI card means it couldn't be RTP related? just frame related?

Re: [Asterisk-Users] Skips and Pops in Call Recordings - channel.c Analysis

List users, I've traced the writing of the leg files to two functions in channel.c: ast_write() ast_read() They both contain similar code, so I'm going to limit my analysis to one of them. If I'm misunderstanding anything or am flat out wrong, please don't hesitate to correct me. Your input i

Re: [Asterisk-Users] Skips and Pops in Call Recordings

s bloated Linux Distro, something like Slackware, just to see if that had any effect. And make sure you use the megaraid2 linux drivers. MATT--- On 12/13/05, Matt Roth <[EMAIL PROTECTED]> wrote: Matt Florell wrote: >Hello, > >Need some more information here: >-

Re: [Asterisk-Users] Skips and Pops in Call Recordings

ms to think that the issue is scaling (some resource contention that causes a bit of audio to be unavailable when the write occurs). I see their point, but given our hardware and the current call volume I'm not completely sold on it. Could it be a configuration issue (file handles, inte

[Asterisk-Users] Skips and Pops in Call Recordings

List users, I'm using the Monitor application to record calls. Most of the recordings are audible, but contain skips accompanied by a popping sound. Sometimes they are isolated, sometimes they appear in groups. Call quality is excellent and seems unaffected by whatever is causing this prob

[Asterisk-Users] Realtime Replication of a Single File

List users, Please provide me with tips on how to replicate a single file to a separate machine as changes are made to it. I would prefer a method that reacts to file modifications (ie. FAM/gamin) as opposed to timed loops/polling (cron + rsync). I'd also like to avoid NFS altogether. Keep

[Asterisk-Users] AstManProxy Segmentation Faults

List users, I am experiencing segmentation faults in AstManProxy. If anyone could help me identify their source, it would be appreciated. The pertinent information is below. Please let me know if you need any more. Asterisk Version Asterisk ABE-A.2-beta AstManProxy Version

[Asterisk-Users] MOH: Most Efficient Method

List Members, What is the most efficient method of providing music on hold? We are currently using the rawplayer method documented here: - Rawplayer MOH - http://www.voip-info.org/wiki-Asterisk+mpg123+faking+it I am also aware of the following alternatives: - Native MOH - http://www.voip-

[Asterisk-Users] Agent/Queue Scalability (Formerly: UPDATE - 512 Calls...)

you so much for all of your help. I hope I can reciprocate in the future. On 10/4/05, *Matt Roth* <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote: > We do not use Asterisk Queues or Agents becaus

Re: [Asterisk-Users] UPDATE - 512 Calls w/ Dig Rec: NFS Setup and Benchmarks

Matt, Well, almost all of the reports, database schema and other code is all fully GPL'd and available on the project site for download: http://astguiclient.sf.net I guess I should've realized it was open source. Thanks for the link, we'll download everything and take a look for ourselves.

Re: [Asterisk-Users] UPDATE - 512 Calls w/ Dig Rec: NFS Setup and Benchmarks

2005, at 22:54, Matt Roth wrote: List members, It has been a while, but I once implemented a simple shared database over NFS, so dredging my memory produced the following: Future Plans and Unresolved Issues == I wrote Windows software for another project that

[Asterisk-Users] UPDATE - 512 Calls w/ Dig Rec: NFS Setup and Benchmarks

List members, My previous post "SUCCESS - 512 Simultaneous Calls with Digital Recording" documents using a RAM disk to eliminate the I/O bottleneck associated with digitally recording calls via the Monitor application. By recording directly to a RAM disk I was able to maintain good call quality

Re: [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording

Waldo, Thanks for the information. If you don't mind answering: are you guys developing this solution for your internal needs (meaning serving UAs from within your enterprise) or are you planning on offering services to the public? This solution is being developed for our internal needs. It

Re: [Asterisk-Users] Will Digium Wildard work with PCI-X or PCI Express

discontinuing the Wildcard series - that would be there whole product line! In particular I am looking at the Wildcard TDM 400P series of cards.. Thanks Matt Roth wrote: Don't bank on it. We were going to use a Wildcard as a timing source on our Dell PowerEdge 6850 and the BIOS didn

Re: [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording

Hi everyone, This is just another attempt to address everybody in one place and consolidate the thread. Zoa, - I think its the best you can do. If something pops into your head, even if it's off the wall, don't hesitate to share i

Re: [Asterisk-Users] Will Digium Wildard work with PCI-X or PCI Express

Just correcting myself. The 3 PCI-X slots are one 64-bit 133 MHz and two 64-bit 100 MHz. Matt Matt Roth wrote: Don't bank on it. We were going to use a Wildcard as a timing source on our Dell PowerEdge 6850 and the BIOS didn't see it. Depending on the PCI-X slot I insta

Re: [Asterisk-Users] Will Digium Wildard work with PCI-X or PCI Express

Don't bank on it. We were going to use a Wildcard as a timing source on our Dell PowerEdge 6850 and the BIOS didn't see it. Depending on the PCI-X slot I installed it in, sometimes the box wouldn't even boot. For perspective the 6850 has 4 PCI-e slots, and 3 PCI-X slots (one 64-bit 133 MHz,

Re: [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording

agine that a single file being written is much more efficient and more disk-friendly.. perhaps I don't know what the heck I'm talking about but at least in my mind the theory sounds better than the current 'stream-to-file' method employed by asterisk. Matt Roth wrote

Re: [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording

All, This message has generated a lot of responses, so I'm going to address each of them here in an attempt to consolidate the thread. Matt, - I'm very interested in the specifics of your setup. - How much space is on the RAM disk?

Re: [Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording

to storing the digital recordings on a RAM disk. Please shoot holes in this setup if you see any weaknesses.  Better today than on our go-live date. Thanks, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer Patrick wrote: On Tue, 2005-09-20 at 18:37 -0400,

[Asterisk-Users] SUCCESS - 512 Simultaneous Calls with Digital Recording

List users, Over the last few days we have been working with MCI's development lab to test our Asterisk setup. We were using a piece of hardware called an Abacus 5000 that is capable of creating and terminating thousands of SIP calls. Initially, we could not get past 64 simultaneous digitall

Re: [Asterisk-Users] Cisco AS5400 Configuration as a SIP Peer - URGENT

Leandro, Thanks for the response. Please let me know when you get your wiki page up. I can be contacted at <[EMAIL PROTECTED]> (remove the NOSPAM from the address). I've compiled a list of messages from Asterisk-Users that may be helpful to anyone else trying to configure a Cisco gateway:

[Asterisk-Users] Cisco AS5400 Configuration as a SIP Peer - URGENT

List users, It's been a while since I've posted here, but I've been hard at work pushing toward our large scale Asterisk goal and keeping up with this list can be a full time job by itself (I have19,543 unread list messages!!). This Friday, September 16th 2005, my team will be at the MCI Deve

[Asterisk-Users] Asterisk Hardware Architecture Group

List members, Does anyone have an interest in forming a hardware architecture group? It seems that Asterisk is so tightly linked to specialized hardware and its corresponding architecture that developing the software alone is insufficient for its adoption to large scale applications. Thank you, M

Re: [Asterisk-Users] T1 Technology and VoIP Gateway Primer

Callum, Matt, is this similar to the idea that you have for your project ? Similar, except we are looking to have a single Asterisk server attached to the Gateway for centralized queuing, reportings, call recoring, etc. We are a call center, so having everything in a single environment is a h

Re: [Asterisk-Users] Asterisk Hardware Recommendation

Does anyone have experience with using NAS (http://en.wikipedia.org/wiki/Network-attached_storage) or SAN (http://en.wikipedia.org/wiki/Storage_area_network) for this application? Matthew Roth http://voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian Daniel Salama wrote: Sure. I

Re: [Asterisk-Users] Asterisk Hardware Recommendation

Thanks Daniel, We may end up replicating your tests in order to confirm some of your results. I don't know if it will be anytime soon, because we don't have the hardware yet. Regardless, I will share my results with the list. Anyone out there have any ideas on why the NFS mount affected call q

Re: [Asterisk-Users] Re: Re: T1/DS1/ISDN PRI

David Josephson, Not off-base, but you haven't made it all the way home yet. This is another layer of the puzzle, and again we are not talking about apples and apples here. "Circuit switched" means that there is a (real or virtual) circuit that takes data on an input port and delivers it to an

Re: [Asterisk-Users] Asterisk Hardware Recommendation

Daniel, Could you expand upon your experience recording to an NFS mounted drive. We are looking to use a TDM-VoIP gateway to route 16+ spans to a single Asterisk server. We were hoping to Monitor using the following scheme: - Monitor application executed on Asterisk server (no 'm' flag) - Pick a

Re: [Asterisk-Users] T1 Technology and VoIP Gateway Primer

Michael, Have you decided which PSTN-VoIP gateway you'll use? Not yet, but our preference is a Cisco gateway. Lucent, Quintum, and AudioCodes also make TDM-VoIP gateways. Prior to purchasing any hardware, our entire layout will be posted to this list in detail for review. Matthew Roth http://

[Asterisk-Users] T1 Technology and VoIP Gateway Primer

Asterisk Users / Asterisk Biz List Members, About a week ago I cross-posted a message titled "Large Asterisk Setup (~500 Concurrent Calls + Scalability)" to Asterisk-Users and Asterisk-Biz. For reference, the threads generated by that message are archived at the following locations: http://lis

Re: [Asterisk-Users] Large Asterisk Setup (~500 Concurrent Calls + Scalability)

Callum, We are very early in the research phase of this project, so I only know the names of a few manufacturers and a couple of models that we are looking at. They are: Cisco (AS5400) Lucent (TNTs) AudioCodes (Mediant 2000) Quintum A thread on this list titled "VOIP Gateways & Asterisk" has ju

Re: [Asterisk-Users] Digium Quad Span Cards

Initially, I believed that the limitation was the PCI bus, but I was mistaken. There is a lot of confusion surrounding this issue, and it would be great if someone stepped forward with a concrete answer. That said, here's what I've learned about the issue through my research. We started off a

Re: [Asterisk-Users] Large Asterisk Setup (~500 Concurrent Calls + Scalability)

ution, but not scalable over WANs since the audio streams congest the WAN during busy periods. - Daniel On Apr 22, 2005, at 9:10 PM, Brian Roy wrote: On 4/21/05, Matt Roth <[EMAIL PROTECTED]> wrote: Daniel, I would be interested to hear if anyone knows of a method to completely offload

Re: [Asterisk-Users] Problems with app_dbodbc.c

Matt, app_dbodbc.c may be deprecated, but what you described is a common problem. As per this thread: http://lists.digium.com/pipermail/asterisk-dev/2004-June/004809.html if you are running make in one of the Asterisk source directory's subdirectories (such as /usr/src/asterisk/apps/) you need

Re: [Asterisk-Users] Large Asterisk Setup (~500 Concurrent Calls + Scalability)

Daniel, I would be interested to hear if anyone knows of a method to completely offload the Monitor command from the master server. It is the missing piece of the puzzle to optimizing the digital recording process. I'm assuming that the CPU usage you are referring to would be incurred due to c

Re: [Asterisk-Users] Error in starting asterisk

Angelo, Looks like an error in zapata.conf (/etc/asterisk/zapata.conf). Without seeing the file, I'd guess that you're missing a context somewhere. It would be helpful if you posted the contents of the file to the list. Matthew Roth http://www.voip-info.org/tiki-index.php?page=Running%20Asterisk%

Re: [Asterisk-Users] Large Asterisk Setup (~500 Concurrent Calls + Scalability)

I just wanted to make everyone aware that I cross-posted my original message to the Biz list. You may want to check out the responses there, too. It looks like the entire Asterisk slave server pool in my diagram (http://home.comcast.net/~mroth01/LargeAsteriskSetup.gif) can be replaced by a Vo

Re: [Asterisk-Users] Large Asterisk Setup (~500 Concurrent Calls + Scalability)

Daniel, So far our digital recording client is a box on a diagram. I haven't digitally recorded a single call through Asterisk yet, but this is a learning process so I'll share what I know in hopes that I can help people out and have any of my mistakes corrected. As I understand it, digital re

[Asterisk-Users] Large Asterisk Setup (~500 Concurrent Calls + Scalability)

List Members, I am involved in the process of designing a large Asterisk setup for a call center. A graphical overview of our tentative design can be found here: http://home.comcast.net/~mroth01/LargeAsteriskSetup.gif Originally, we planned to implement this design by purchasing one multi-proc

Re: [Asterisk-Users] RE: Re: a simple question

Weiming, At the Asterisk CLI the "Show Version" command will print a string similar to the following: Asterisk CVS-v1-0-04/14/05-13:17:05 built by [EMAIL PROTECTED] on an i686 running Linux At the Linux command line, "asterisk -V" will print a string similar to the following: Asterisk C

Re: [Asterisk-Users] Help compiling zaptel in Debian

Manuel, Here are a couple suggestions: 1) I see that your modules directory is "/lib/modules/2.4.20-686", while mine is "/lib/modules/2.4.19". Maybe your kernel source directory also has a "-686" appended to it. Verify the location of your kernel source and adjust the ln statements accordingly

Re: [Asterisk-Users] Upgraded now Asterisk won't start

Paul, Noloading modules via modules.conf is a band-aid to your problem. Each time you upgrade to the latest CVS, you are likely to run into similar problems requiring more noloads. I'm assuming that you were upgrading a working installation of Asterisk when this problem occurred (if not, it l

Re: [Asterisk-Users] spandsp and cvs head

Here is a web page that I found useful when I ran into the same spandsp patch problem. http://www.linuxjournal.com/article/1237 I had no experience with patches, but after reading that I was able to update the patch to work with my Makefile. Matthew Roth http://voip-info.org/tiki-index.php?page

Re: [Asterisk-Users] Installed ztdummy, Asterisk doesnt work anymore

Michel, Try this: First: cd /usr/lib/asterisk/modules/ rm *.so Then: Recompile Zaptel, then recompile Asterisk. You may need to recompile anything else that put loadable modules into that directory, such as asterisk-addons, as well. Does anyone out there know why "make clean" doesn't take care

Re: [Asterisk-Users] Help compiling zaptel in Debian

Manuel, This is from my Wiki page on running Asterisk on Debian/GNU Linux. Build and Install Zaptel Zaptel provides support for Digium hardware. The following steps can be followed to build and install Zaptel. 1. Create symbolic links to the new kernel's source files by issuing the following co

Re: [Asterisk-Users] Undefined symbol in res_features & Others

First: cd /usr/lib/asterisk/modules/ rm *.so Then: Recompile Asterisk. You may need to recompile anything else that put loadable modules into that directory, such as asterisk-addons, as well. Does anyone out there know why "make clean" doesn't take care of this task? I spent a few hours dealin

[Asterisk-Users] Enhanced Queue App Revisited

In July of 2003, this message was posted asking about an enhanced queue app for Asterisk. http://lists.digium.com/pipermail/asterisk-users/2003-July/014965.html I am evaluating Asterisk for use in a 200+ seat call center, and I need to know whether any advances have been made in its queue repor

[Asterisk-Users] Asterisk Hardware Requirements for a 50-100 Seat Call Center

I am looking for estimated hardware requirements for running a 50 to 100 seat call center off of a single Asterisk server. The Asterisk server will have one quad T1 card installed (probably a Digium TE410P) with two T1s connected. The OS is Debian GNU/Linux (woody) with a custom 2.4.xx kernel

[Asterisk-Users] Digium T1 Card Questions

I have a couple of questions about Digium's T1 cards, such as the TE410P. Any answers would be greatly appreciated. 1) Do they support standard T1s or are they ISDN-only? 2) Do you know of anyone offering support for configuring T1s for Digium cards, and if so at what cost? Thanks, Matthew Rot

[Asterisk-Users] Best Headsets for a Call Center Environment

I'm looking for suggestions as to the best multimedia headsets for a call center environment. A few considerations: 1) USB headsets are preferable, because they don't require a soundcard. 2) Omnidirectional microphones are problematic, because they pick up too much background noise. Thanks, Mat

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