[Asterisk-Users] Mark Spencer and John Maddog Hall visiting Toronto - come and join us
On April 21st, at 7:30 PM, Mark Spencer and John Maddog Hall[1] will be joining the Toronto Asterisk Users' Group[2], the Toronto Linux Users' Group[3] and the Ontario Asterisk and VoIP Enthusiasts Group[4] for an informal chat about Asterisk and The Open Source Telephony Revolution. If you are going to be in the Toronto area, we'd love to have you join us! When: 7:30PM-9:30PM, Thursday April 21st, 2005 Where: Room 1130 Bahen Centre, at the University of Toronto 40 St. George Street Toronto, ON Directions: The building is located at 40 St. George Street. This is just north of College Street on the west side of St. George and it is the next building north of the U of T Bookstore. There is a parking lot under the building and the entrance is on Huron Street just north of College. Huron is the street west of St. George. The nearest subway station is Queen's Park and it is a 5 minute walk west on College or take the streetcar two stops to St. George. There is a campus map available on the web at http://www.osm.utoronto.ca/map. If you require additional information, please contact Simon Ditner [EMAIL PROTECTED], or myself, Jim Van Meggelen[EMAIL PROTECTED] This event will be free of charge. [1] Jon Maddog Hall, of Linux International, http://www.li.org/who/bio.php?name=hall [2] Toronto Asterisk PBX Users Group, http://opensource.meetup.com/42/ [3] Toronto Linux Users Group, http://tlug.ss.org [4] Ontario Asterisk and VoIP Enthusiasts Group, http://uc.org/asterisk -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.9 - Release Date: 13/04/2005 ___ 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] Buying some Polycom IP300s
Dan Morin wrote: Sorry for the double post, I tried to paste and accidently sent the email I've been playing with Asterisk for a few weeks now, and I've gotten everything to work well with softphones, so I'm ready to move on to normal VoIP phones. I've been looking around and reading comments that people have had, and I was convinced that the Polycom IP300 was a great phone for a good price. But, then I ran into this page, which has been update in the last few days: http://www.voip-info.org/wiki-Polycom+SoundPoint+IP+500 The page in the wiki used to say that the person would not recomed Polycom phones to anyone. So anyway, I just want to make sure that the IP300 is a good choice. I don't want to get cheap phones that aren't business quality, since I do play on using them for my business after testing. Also, is the IP500 worth the extra money? What can it do that the IP300 can't. And finally, will the IP300 do ulaw encoding? The IP300 is a nice entry-level business phone. It does not have a speakerphone, and cannot handle PoE, but other than that it is excellent. It is more expensive than some of the fully-featured generic phones, but it also is built to a much higher standard, including a properly weighted handset and high impact plastic. If price is the main thing, then this phone might be a bit too expensive ($130-$150), but if quality (or even just the *feeling* of quality) is important, this phone will serve well. The IP500 is a similar phone with more line appearances, a higher resolution display, full handsfree (Polycom-quality) and PoE. The IP500 has been favorably compared to the Cisco 7940. Cheers, -- Jim Van Meggelen [EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.1 - Release Date: 01/04/2005 ___ 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] Are there online forums instead of this emailforum??
[EMAIL PROTECTED] wrote: Hi, I really regret bringing the subject up... I guess I hit some nerves so please accept my apology. I have adapted to using the mailing list (Mozilla Thunderbird with filters directing traffic a specific folder, and threading) and it works, not ideally, but it works. The search of goggle works but it would of been nice to have some sort of FAQ so that I didn't have to piss people off by asking about it. Thanks for all your help and again I apologize... Don't worry, you didn't piss anyone off. It's just that the crowd in here is pretty passionate. Kinda like a team of drunken Canadian hockey players; friendly, but you'd never know it cause they're so dangerous. Welcome to Asterisk-Users -- Jim Van Meggelen [EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.0 - Release Date: 31/03/2005 ___ 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] Comparison Charts
[EMAIL PROTECTED] wrote: I couldnt agree with you more Jim. Im realdy using Asterisk and agree 100% with what you say... I was asking for a comparison list with other PBX's because for example, for a customer, they have heard of Avaya and Cisco and they all are selling IP now... So In order to get your customer to trust Asterisk over those guys, you need to show him the diff. Between the two and some lists of the features on the others compared to Asterisk.. It kind of reminds me of the challenge in selling the Internet to management in the early 90s. The trick was getting them to think in a whole different way. In many cases, they bought into it simply because they had no choice. Businesses didn't fully get it until everyone was using it. You could compare Asterisk to other products, but that wouldn't show it in its best light. It might be better to explore some of the things that Asterisk can do that the other systems cannot. The VoIP part is a total red-herring - we've had VoIP for over 10 years; the real power is in the flexibility. Defining exactly what that flexibility is will in large part depend on your audience. Find out what excites them. Is it cost? Asterisk has a compelling story to tell. Is it standards-compliance? Asterisk again scores points. Flexibility? Yep. Open-source (or avoiding vendor lock-in)? You betcha! I would almost want to see a list of features that those other products had that could NOT be configured on Asterisk. Who really knows what the limits are? Ten years later we're still finding out new uses for the Internet. I imagine that ten years from now we'll still be adding features to open-source telephony . . . will we even call it telephony then? I'm betting no. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Van Meggelen Sent: Jueves, 10 de Marzo de 2005 12:17 a.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Comparison Charts [EMAIL PROTECTED] wrote: Guys. Anybody has a URL or some document with comparison charts with Asterisk's features against other PBXs? I would argue that what you ask is in some ways impossible. Asterisk is orders of magnitude more flexible than any other PBX you may have encountered, because it is more like a toolkit than a PBX. Whatever is missing can be built, so there's no list of features that can ever be considered complete. For people who are looking for a PBX that has a user-friendly interface and is easy to configure, Asterisk will tend to dissappoint. Where Asterisk shines is for those people who want to--need to--build their own PBX. People who are willing to do the work themselves; designing, testing, debugging, re-designing . . . Many of us believe that Asterisk is going to transform the telecommunication industry, but it won't do it because it has more features, it'll do it because it puts the control of the features list where it belongs: in the customer's hands. I would suggest that the best way to approach Asterisk is to have a list of things that you need your telephone system to do. Then, one-by-one, figure out how to handle each of those in Asterisk. When you are done, you may have a few that you couldn't find a satisfactory solution to. Those can typically be custom developed, and surprisingly, you will still probably come in at a lower cost than a closed, so-called full-featured proprietary system. What's more, as your needs grow, Asterisk can grow with you. Five years from now you won't need to hear oh sorry but that system is no longer supported. Want new functionality? Install it. Is the hard drive wearing out? Replace it. Need more CPU power? Migrate to a new chassis. Asterisk changes all the rules. Therfore, to understand it, you have to adopt a new way of thinking about telecom systems. Welcome to Asterisk! -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.7.1 - Release Date: 09/03/2005 ___ 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] Comparison Charts
[EMAIL PROTECTED] wrote: Guys. Anybody has a URL or some document with comparison charts with Asterisk's features against other PBXs? I would argue that what you ask is in some ways impossible. Asterisk is orders of magnitude more flexible than any other PBX you may have encountered, because it is more like a toolkit than a PBX. Whatever is missing can be built, so there's no list of features that can ever be considered complete. For people who are looking for a PBX that has a user-friendly interface and is easy to configure, Asterisk will tend to dissappoint. Where Asterisk shines is for those people who want to--need to--build their own PBX. People who are willing to do the work themselves; designing, testing, debugging, re-designing . . . Many of us believe that Asterisk is going to transform the telecommunication industry, but it won't do it because it has more features, it'll do it because it puts the control of the features list where it belongs: in the customer's hands. I would suggest that the best way to approach Asterisk is to have a list of things that you need your telephone system to do. Then, one-by-one, figure out how to handle each of those in Asterisk. When you are done, you may have a few that you couldn't find a satisfactory solution to. Those can typically be custom developed, and surprisingly, you will still probably come in at a lower cost than a closed, so-called full-featured proprietary system. What's more, as your needs grow, Asterisk can grow with you. Five years from now you won't need to hear oh sorry but that system is no longer supported. Want new functionality? Install it. Is the hard drive wearing out? Replace it. Need more CPU power? Migrate to a new chassis. Asterisk changes all the rules. Therfore, to understand it, you have to adopt a new way of thinking about telecom systems. Welcome to Asterisk! -- Jim Van Meggelen [EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.7.0 - Release Date: 08/03/2005 ___ 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] Where can I find all areacodes for USA (accountingpurpose)
[EMAIL PROTECTED] wrote: I need to setup all area codes for billing, but how can I do that easy for North America and Canada, where I have only one price anyway. Country code 1, and just exclude any Carribean nations that you need to handle differently. It'll be a much shorter list than trying to include every US and CDN NPA. Try nanpa.com This might help: http://www.nanpa.com/nas/public/npasInServiceByLocationReport.do?method= displayNpasInServiceByLocationReport -- Jim Van Meggelen [EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.7.0 - Release Date: 08/03/2005 ___ 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] Another Newbie Question
Callum McGillivray wrote: Hey all, My apologies if this sounds blindingly obvious, but am I correct in saying that I can use Asterisk to connect two extensions and make calls between them without needing an actual telephone line at all ? As I said, probably blindingly obvious but my techies have gone home for the evening and I was looking for an answer before I left. You could do that with two tin cans and a string! ;-P In all seriousness, the answer to your question is: yes, Asterisk can do that, and a whole lot more. Cheers, -- Jim Van Meggelen [EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.6.4 - Release Date: 07/03/2005 ___ 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] Survey: what's the best HTTPd/TFTPd/FTPd to serveup configuration files to sets
Thanks as always to everyone who provided feedback. It was most helpful! Regards, Jim. -- Jim Van Meggelen [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I would like to start a discussion centred around the various ways one might serve up configuration files from an Asterisk server (I know, it's better to use a secondary server for all this, but let's talk about a smaller system). The types of things being served would include: - Logo image for sets that support that - XML directory files - XML or raw text configuration files - what-all-else Seems to me that Apache is simply way too overpowered for all this, and thus would needlessly place load on the server. I have heard that khttpd is pretty lightweight, but its use seems to have been deprecated, and it does not appear to be actively maintained. Is TuX the way to go? As for tftpd and ftpd, I'm just not sure. Leightweight is the key, here. Thoughts, opinions, experiences? Thanks, Jim. -- Jim Van Meggelen [EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.6.2 - Release Date: 04/03/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Survey: what's the best HTTPd/TFTPd/FTPd to serve up configuration files to sets
I would like to start a discussion centred around the various ways one might serve up configuration files from an Asterisk server (I know, it's better to use a secondary server for all this, but let's talk about a smaller system). The types of things being served would include: - Logo image for sets that support that - XML directory files - XML or raw text configuration files - what-all-else Seems to me that Apache is simply way too overpowered for all this, and thus would needlessly place load on the server. I have heard that khttpd is pretty lightweight, but its use seems to have been deprecated, and it does not appear to be actively maintained. Is TuX the way to go? As for tftpd and ftpd, I'm just not sure. Leightweight is the key, here. Thoughts, opinions, experiences? Thanks, Jim. -- Jim Van Meggelen [EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.6.2 - Release Date: 04/03/2005 ___ 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] x101p + Nortel ATA2
[EMAIL PROTECTED] wrote: Hi, Does anyone have any experience connecting Asterisk to a Meridian system using an ATA2 and x101p? The basics work -- I can make outbound calls, receive inbound, and use flash to transfer calls, but certain things do not work, specifically with calls from internal extensions. - Does the Meridian/ATA2 pass any kind of callerid info? We do not have external callerid, but I'm not even getting extension numbers. Not a chance, sorry to say. - Calls involving an external line can use DTMF, but calls from another internal extension cannot. This is a problem for voicemail! I've tried *809 but it hasn't helped. Is this a limitation of the Meridian which won't pass DTMF internally? You want Long Tones. F808 - Calls from internal extensions do not detect hangup properly, external calls are ok. So if an internal extension calls to leave a voicemail, the recording goes on until I do a sofft-hangup from the CLI. There's no disconnect supervision on the ATA. The plan was to use Asterisk as a voicemail server, but those three issues make this setup completely useless for that! Yep. You need to try and find a VMI if you can. It's designed for voicemail integration (although you still won't get the CLID). Jim. -- Jim Van Meggelen [EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.5.1 - Release Date: 27/02/2005 ___ 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] SIP NOTIFY in stable branch?
[EMAIL PROTECTED] wrote: I didn't realize that the stable branch was never added to... So it will NEVER have any more features than it currently has??? 1.0 STABLE will never have any more features. 1.2 STABLE will be released in the next 3 to 6 months, and it will include all features that have been added since 1.0 was released. There will be a feature freeze on HEAD in the next month or so, which will become the basis of STABLE 1.2. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Thursday, February 24, 2005 1:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP NOTIFY in stable branch? Clay Reiche wrote: Does anyone know when the SIP NOTIFY feature from the CLI will be part of the stable branch? Is there any way I can install just that HEAD feature? Yes, I know when. Never :-) Stable means stable, no new features will be added. It's not a difficult feature to backport, but if you haven't worked inside chan_sip before it could take you quite some time. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.5.0 - Release Date: 25/02/2005 ___ 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] Introducing the Asterisk Realtime Architecture -ARA
[EMAIL PROTECTED] wrote: I've added an introduction article about the ARA on my web site http://www.voip-forum.com/ The same text is now also added to CVS head as README.realtime. On the same site, you will also find the news item about how we used Asterisk for a call from an airline jet above Greenland to Stockholm, Sweden. The world is getting smaller and more connected every day! /Olle Instead of ARA, how about naming it AURA - the Asterisk Unified Realtime Architecture. The English word Aura also conjures up great imagery: [from m-w.com] Main Entry: aura Pronunciation: 'or- Function: noun 1 b : a distinctive atmosphere surrounding a given source the place had an aura of mystery 2 : a luminous radiation : NIMBUS 4 : an energy field that is held to emanate from a living being [/m-w] The term RealTime conjures up images of RTOSes, not flexible database interaction. Giving it a friendly, easy to remember name would help to avoid confusion for newbies (and oldbies, for that matter). Cheers, Jim. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.5.0 - Release Date: 25/02/2005 ___ 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] Transfer a call ? Am I looking fortheflashcommand ?
[EMAIL PROTECTED] wrote:
Hello Jim,
thx for the answer..
Im happy I found someone that is using flash :)
It's not perfect, but it can be useful.
Am I right, if I transfer a call with flash, the line will be free
afterwards ?
Yep
Would you mind to past me how you did the flash part @the
extention file ? Also, If I use flash, do I have to setup
anything else or just @the extention file ?
Jere's the relevant section of my dial plan:
[macro-cell_user]
exten = s,1,Playback(transfer)
exten = s,2,Flash(zap/1)
exten = s,3,SendDTMF(${ARG1})
exten = s,4,Hangup()
Good luck!
Jim.
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Jim
Van Meggelen Gesendet: Freitag, 25. Februar 2005 05:57
An: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Betreff: RE: [Asterisk-Users] Transfer a call ? Am I looking for
theflashcommand ?
[EMAIL PROTECTED] wrote:
On Fri, 2005-02-25 at 00:50 +0100, Mateo Meier wrote:
Hey Guys
Im trying to forward a call with asterisk to a regular phone.
Something like I get a call on my regular phone, and he's trying
to reach some buddy of mine.. then I tell him wait a sec and push
Flash and get a other dialtone.. then I dial that other number
then hangup the phone, so the one that called will be connected to
where I dialed it to...
Some buddy of mine told me im looking for a function called flash
Only thing Im able to find is:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Flash
Im unsure how to use it now..
Let's say if I forward a call with asterisk as following: exten =
2,1,Dial(capi/720:07812345*,18)
How would I use the flash command to transfer that call above to 078
12345* ? I have no problem transferring a call, but when Im doing
this with the dial command (see above).. then my line will be busy
Been covered before, You can't do that on an analog line. Problem
comes from where you are and what flash would be working on at that
point. If you flash asterisk and get dialtone again, you are getting
the dialtone
from asterisk. At this point the only channel being worked is the one
you are on and flashing it won't help.
What you would need to do is get the other leg of the call to make
the flash.
It might be really handy to be able to specify the trunk to
flash() as an argument. I use flash in my dialplan to
transfer incoming calls to my cell phone when I'm out and
about - frees up the line and reduces attenuation caused by
an analog trombone. It'd be handy to be able to use it to
transfer terminated calls as well.
Of course if you where on a PRI link, you could do hairpinning,
ect or tromboning and get the call taken back by the PSTN and
transferred to the new number.
--
--
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Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 266.4.0 - Release Date: 22/02/2005
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RE: [Asterisk-Users] Transfer a call ? Am I looking for the flashcommand ?
[EMAIL PROTECTED] wrote: On Fri, 2005-02-25 at 00:50 +0100, Mateo Meier wrote: Hey Guys Im trying to forward a call with asterisk to a regular phone. Something like I get a call on my regular phone, and he's trying to reach some buddy of mine.. then I tell him wait a sec and push Flash and get a other dialtone.. then I dial that other number then hangup the phone, so the one that called will be connected to where I dialed it to... Some buddy of mine told me im looking for a function called flash Only thing Im able to find is: http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Flash Im unsure how to use it now.. Let's say if I forward a call with asterisk as following: exten = 2,1,Dial(capi/720:07812345*,18) How would I use the flash command to transfer that call above to 078 12345* ? I have no problem transferring a call, but when Im doing this with the dial command (see above).. then my line will be busy Been covered before, You can't do that on an analog line. Problem comes from where you are and what flash would be working on at that point. If you flash asterisk and get dialtone again, you are getting the dialtone from asterisk. At this point the only channel being worked is the one you are on and flashing it won't help. What you would need to do is get the other leg of the call to make the flash. It might be really handy to be able to specify the trunk to flash() as an argument. I use flash in my dialplan to transfer incoming calls to my cell phone when I'm out and about - frees up the line and reduces attenuation caused by an analog trombone. It'd be handy to be able to use it to transfer terminated calls as well. Of course if you where on a PRI link, you could do hairpinning, ect or tromboning and get the call taken back by the PSTN and transferred to the new number. -- -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.4.0 - Release Date: 22/02/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Brainstorm: Running Asterisk as cool as possible - AKA solid state.
I would like to ask what people think the best way would be to build a low-power consumption, passively-cooled system. For example,one could use a fanless-Eden (mini-ITX/EPIA) system, but the loss of FPU power would limit performance. The obvious choice for FPU work are the Intels and AMDs, nut they're all power-hungry radiators. Is there something that can offer the quiet, power-savings of the EPIA, but some sort of respectable math capabilities? Transmeta was into that game, but I'm not sure if that's any help. This might work: http://mini-itx.com/projects/cluster/ But it'd take up a lot of space and look rather ugly. Hmm. It wouldn't be the first ugly PBX . . . How about AMD's Geode. Anyone have any experiences? Where my mind is at is building a system that could live in a harsh environment (like a typical key system closet). Passively-cooled systems can be sealed up much easier, and having no moving parts makes them easier to ruggetize. This is just a brainstorm, so all thoughts are welcome. TIA, Jim. -- Jim Van Meggelen [EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.4.0 - Release Date: 22/02/2005 ___ 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] A bit of a survey: What do do if youneedmorethan 4 C.O. lines
[EMAIL PROTECTED] wrote: If you have a TDM card already, buying a T1, channelbank, etc. to add a few lines is the stupidest thing I've heard of today. Not necessarily stupid, but certainly expensive. Have you looked into buying some cheap multiport ATAs? 2 port SIP/IAX2 ATA should be around $70-80? Yep, that's a possibility, but it's rather more kludgy than I'd like. (heck, the channel bank and T1 is more kludgy than I'd like - an integrated card would be my preference). -Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Van Meggelen Sent: Saturday, February 19, 2005 8:39 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] A bit of a survey: What do do if you needmorethan 4 C.O. lines Really? For five lines I need to buy all that hardware? Hmm. Well, I appreciate you taking the time to respond to my question. Regards, Jim. [EMAIL PROTECTED] wrote: Digium tech support recommends going with a t1 card and a channel bank. This is by far the simplest, cheapest and cleanest solution that I know of. Jon. On Friday 18 February 2005 09:21 am, Jim Van Meggelen wrote: Folks, In light of all the troubles people report when running more than one TDM400 card in a system, I wouldn't mind hearing what your solution of choice would be when having to connect 5 or more analog telco circuits to an Asterisk. I'll try and compile the answers together and get them into the Wiki, as I figure this could be useful knowledge for the community. TIA, Jim. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.1.0 - Release Date: 18/02/2005 ___ 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] A bit of a survey: What do do ifyouneedmorethan4 C.O. lines
[EMAIL PROTECTED] wrote: On February 20, 2005 09:25 am, Michael Giagnocavo wrote: Sorry, I understood the O.P. already had the hardware bought and installed and simply wanted to throw on an extra line. You understood correctly; Uh, nope. I've been unclear. It was a purely hypothetical situation. It could be a case of expansion, or a new system requiring more circuits than a single card supports. I think there's a gap. A channel bank tied into a T1 is kinda kludgy for such a small system. Technically sound, but kludgy. Any other system of that size wouldn't need all the integration gear. You'd just plug the lines in. I know, I know; this is Asterisk, and that means one has to be creative. Perhaps I already knew the answer before I asked the question . . . still, one can hope. But again even a TDM401P is $133 on Digium's site. Considering you could probably get 60% of the price of your original TDM404P ($200 is 60% of $337), then you're either spending $133 for another TDM401P (and all the hassle of trying to get two to work in a system) or $600 (my $800 estimate - the $200 you got for your old equipmen)... $600 != 8x $133, and $800 != 8x $537. Is there a place to buy brand new Adit600's with 5+ FXOs and a T1 card for $800? (I've looked on eBay, but that's not a reliable supply chain, and I have yet to see such a price for new equipment). Seems to me one is looking at more like $2000. Anyway I think he's got his recommendations :-) Sure do! Thanks. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.1.0 - Release Date: 18/02/2005 ___ 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] A bit of a survey: What do do if youneedmorethan4 C.O. lines
[EMAIL PROTECTED] wrote: On 20 Feb 2005, at 16:22, Jim Van Meggelen wrote: [EMAIL PROTECTED] wrote: If you have a TDM card already, buying a T1, channelbank, etc. to add a few lines is the stupidest thing I've heard of today. Not necessarily stupid, but certainly expensive. Have you looked into buying some cheap multiport ATAs? 2 port SIP/IAX2 ATA should be around $70-80? Yep, that's a possibility, but it's rather more kludgy than I'd like. (heck, the channel bank and T1 is more kludgy than I'd like - an integrated card would be my preference). -Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Van Meggelen Sent: Saturday, February 19, 2005 8:39 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] A bit of a survey: What do do if you needmorethan 4 C.O. lines Really? For five lines I need to buy all that hardware? Hmm. Well, I appreciate you taking the time to respond to my question. Regards, Jim. [EMAIL PROTECTED] wrote: Digium tech support recommends going with a t1 card and a channel bank. This is by far the simplest, cheapest and cleanest solution that I know of. Jon. On Friday 18 February 2005 09:21 am, Jim Van Meggelen wrote: Folks, In light of all the troubles people report when running more than one TDM400 card in a system, I wouldn't mind hearing what your solution of choice would be when having to connect 5 or more analog telco circuits to an Asterisk. I'll try and compile the answers together and get them into the Wiki, as I figure this could be useful knowledge for the community. TIA, Jim. If you already need 5+ lines, and expect any growth, ask your telco to quote for a T1 with 6 (or 8) channels enabled. It might not be as expensive as you'd think, and you get all the advantages of a digital circuit, plus an easy expansion route. I like the thinking; the challenge is often where in the world you are, and how much competition there is. Here in Ontario, T1's were generally priced such that fractional T1s hardly saved anything. There is more competition now, so prices are changing, but I still can't see frac T1 service competing with such a small number of analog circuits. I know there are places where such a thing could be had very competitively, so your advice is still good. Plus you avoid the (possible) need for a channel bank. Agreed. Thanks kindly for the reply. Regards, Jim. -- Jim Van Meggelen [EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.1.0 - Release Date: 18/02/2005 ___ 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] A bit of a survey: What do do if you need morethan 4 C.O. lines
Well, I appreciate everyone's input, and I'll give the matter some more thought. Just so no one stays up at night worrying, this is not a situation I am facing, it is simply a hypothetical scenario. As with so many things, there is always a trade-off between economy and functionality. The Adit 600 and T1 integration is certainly quality, but I have not been able find an economical way to do this (purchasing used equipment on eBay is fine for smaller deployments and lab gear, but not a very sound logistics strategy, and awfully difficult to explain to a customer). Again, thanks to everyone for their feedback. Regards, Jim. -- Jim Van Meggelen [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Folks, In light of all the troubles people report when running more than one TDM400 card in a system, I wouldn't mind hearing what your solution of choice would be when having to connect 5 or more analog telco circuits to an Asterisk. I'll try and compile the answers together and get them into the Wiki, as I figure this could be useful knowledge for the community. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.1.0 - Release Date: 18/02/2005 ___ 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] A bit of a survey: What do do if youneedmorethan4 C.O. lines
[EMAIL PROTECTED] wrote: Jim Van Meggelen wrote: Yep, that's a possibility, but it's rather more kludgy than I'd like. (heck, the channel bank and T1 is more kludgy than I'd like - an integrated card would be my preference). I haven't followed this thread closely but have you looked into the Voicetronix OpenSwitch cards? http://www.voicetronix.com.au/hda.htm I've looked at them, but never heard much about them. Is anyone using them? Can anyone give a comparison vs. the TDM400? -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.1.0 - Release Date: 18/02/2005 ___ 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] A bit of a survey: What do do if you needmorethan 4 C.O. lines
[EMAIL PROTECTED] wrote: On Sun, 2005-02-20 at 11:47 -0500, Jim Van Meggelen wrote: Well, I appreciate everyone's input, and I'll give the matter some more thought. Just so no one stays up at night worrying, this is not a situation I am facing, it is simply a hypothetical scenario. As with so many things, there is always a trade-off between economy and functionality. The Adit 600 and T1 integration is certainly quality, but I have not been able find an economical way to do this (purchasing used equipment on eBay is fine for smaller deployments and lab gear, but not a very sound logistics strategy, and awfully difficult to explain to a customer). This would be one of those cases where you keep a couple in stock and watch the ebay auctions when your stock goes low. You will find that your customers that are looking for the cheapest solutions possible will not baulk at used equipment. It is highly likely that they will price you against a used key system or pbx. Certainly keeping spares in stock is good advice, and I don't mind using pre-owned equipment if it's solid stuff (which I know the adit is). I'm going to think about this some. As for price, that's always the challenge. Thing is, the lowest price does not always win. Still, being able to keep costs low is always going to be an advantage. -- Jim Van Meggelen [EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.1.0 - Release Date: 18/02/2005 ___ 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] Digium TDM400P has RJ45 interface, how to connect to analog phone RJ11?
Title: Message Just plug it in. The RJ11 is narrower than the RJ48, but has the exact same connection mechanism. it'll fit perfectly (the centre two pins are the contacts) -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]Sent: February 20, 2005 1:51 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionCc: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Digium TDM400P has RJ45 interface,how to connect to analog phone RJ11? Hello, I bought a TDM400P, and intended to use it with my analog phone, which is RJ11 ofcourse. So, the question now, how do I plug in my RJ11 phone to the TDM400P card, which has an RJ45 interface? Also, since it's an 11B card, I also intend to bring in an analog line into the RJ45, so i am still left with the same questionhow do I go from the RJ11 standard analog to the RJ45 on the TDM400P card? I'd appreciate any response. thx chuks --No virus found in this incoming message.Checked by AVG Anti-Virus.Version: 7.0.300 / Virus Database: 266.1.0 - Release Date: 18/02/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.1.0 - Release Date: 18/02/2005 ___ 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] Segmentation fault {Writer given gnu-lashing}
[EMAIL PROTECTED] wrote:
I have no problem with Slackware,
Me neither. I learned Linux with Slack. Found it to be extremely
friendly. And that was 10 years ago. Last time I chacked, it was still
friendly (and not at all GUI, unless you want it served that way)
But when you are learning to drive a car you should first try
a Chevy with an automatic transmission first before strapping
on a 6 speed Ferrari.
Popular opinion holds that people who learn to drive standard first
generally end up being better drivers. And why wouldn't you want to
learn on a Ferrari since you can get one for free!?!
Humor helps in teaching and getting a person to step out of a
rut they are having a problem in and gives them a chance to
rethink what might be going on.
Ya, but humour should be dispensed carefully, lest offence be given.
Remember, my goal is to reduce the number of variables in the system.
The problem I see with Fedora is that you can install it successfully
without learning anything about Linux. Slackware is rather good for
learning Linux, because it is friendly and helpful, but still expects
you to make the decisions. I'd argue that a familiarity with the shell
is going to be essential for even a basic Asterisk install. It's not a
pre-qualifier so much as an essential skill.
LOL! You're just bored and are trolling for a holy war, eh? Well, I
guess we gotta shake off these Febraury blah's somehow.
GENTOO IS FOR WANNABE NEWBIES!!! (that oughta stir things up)
--
Jim Van Meggelen
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Andrew Kohlsmith
Sent: Sunday, February 20, 2005 1:39 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [Asterisk-Users] Segmentation fault {Writer
given gnu-lashing}
On February 20, 2005 01:11 pm, Race Vanderdecken wrote:
1. Why are you running on Slackware?
Are you trying to prove a point or just enjoy being frustrated?
Open Source is like Broad Spectrum Pesticide, it works but your
results may vary and you may end up killing your lawn.
Got a problem with Slackware? It works *very* well with Asterisk.
--
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Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 266.1.0 - Release Date: 18/02/2005
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RE: [Asterisk-Users] A bit of a survey: What do do if youneedmorethan4C.O. lines
[EMAIL PROTECTED] wrote: Jim Van Meggelen wrote: [EMAIL PROTECTED] wrote: Jim Van Meggelen wrote: Yep, that's a possibility, but it's rather more kludgy than I'd like. (heck, the channel bank and T1 is more kludgy than I'd like - an integrated card would be my preference). I haven't followed this thread closely but have you looked into the Voicetronix OpenSwitch cards? http://www.voicetronix.com.au/hda.htm I've looked at them, but never heard much about them. Is anyone using them? Can anyone give a comparison vs. the TDM400? I'm using a Voicetronix OpenLine4, and it works well under asterisk. Initially I had some echo problems, but Voicetronix support is excellent and solved them (I've just updated the wiki with the bal# values they gave me). I can't compare it to the TDM400, not having used one, but you can use multiple Voicetronix OpenSwitch 6 and 12 cards in one system without the interrupt problem of the TDM400. That sounds like the ticket, then. Thanks. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.1.0 - Release Date: 18/02/2005 ___ 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] A bit of a survey: What do do if you need morethan 4 C.O. lines
Really? For five lines I need to buy all that hardware? Hmm. Well, I appreciate you taking the time to respond to my question. Regards, Jim. [EMAIL PROTECTED] wrote: Digium tech support recommends going with a t1 card and a channel bank. This is by far the simplest, cheapest and cleanest solution that I know of. Jon. On Friday 18 February 2005 09:21 am, Jim Van Meggelen wrote: Folks, In light of all the troubles people report when running more than one TDM400 card in a system, I wouldn't mind hearing what your solution of choice would be when having to connect 5 or more analog telco circuits to an Asterisk. I'll try and compile the answers together and get them into the Wiki, as I figure this could be useful knowledge for the community. TIA, Jim. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.1.0 - Release Date: 18/02/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Monitoring a telco line for MWI through a TDM400 FXO
Folks, I've tried to find a reference, but I've had no luck, and would appreciate your thoughts: I'd like to be able to monitor a telco line for Message Waiting Notification, however I cannot figure out if this capability is available. Detecting either FSK or Stuttered Dial tone would serve, but I can't find anything in the list archives or Wiki with clues as to how this might be achieved. Any advice would be appreciated. Cheers, Jim. -- Jim Van Meggelen [EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.8 - Release Date: 14/02/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A bit of a survey: What do do if you need more than 4 C.O. lines
Folks, In light of all the troubles people report when running more than one TDM400 card in a system, I wouldn't mind hearing what your solution of choice would be when having to connect 5 or more analog telco circuits to an Asterisk. I'll try and compile the answers together and get them into the Wiki, as I figure this could be useful knowledge for the community. TIA, Jim. -- Jim Van Meggelen [EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.8 - Release Date: 14/02/2005 ___ 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] Monitor does not like variable subsitutions
You are using illegal characters in your file name.
See this line in your output?
ast_writefile: No such format 'wav|rec_to_448704386865_at_16022005-16'
It can't get past it because the colon is not a valid filename
character.
[EMAIL PROTECTED] wrote:
Hello,
I have been attempting to get the Monitor function to
accept a loal variable substitution in order to use
the same filename later in the same context. Monitor
does not appear to like it, as it attempts to use
wav|filename as the recording type, as opposed to just
wav.
Here is what I get if I just supply a filename
directly (it works fine):
--context-
exten = _9X.,3,Monitor(wav|recording|m)
--context-
--CLI-
-- Executing SetVar(SIP/3004-275c,
REC_FILE_NAME=rec_to_448704386865_at_16022005-16:54:10)
in new stack
-- Executing Monitor(SIP/3004-275c,
wav|recording|m) in new stack
-- Executing AGI(SIP/3004-275c, outbound.agi)
in new stack
--CLI-
Here is what I get when I attempt to to variable
substituion for the filename:
--context-
exten =
_9X.,2,SetVar(REC_FILE_NAME=rec_to_${EXTEN:1}_at_${DATETIME})
exten = _9X.,3,Monitor(wav|${FILENAME}|m)
--context-
--CLI-
-- Executing SetVar(SIP/3004-da21,
REC_FILE_NAME=rec_to_448704386865_at_16022005-16:56:35)
in new stack
-- Executing Monitor(SIP/3004-da21,
wav|rec_to_448704386865_at_16022005-16:56:35|m) in
new stack
Feb 16 16:56:35 WARNING[17028]: file.c:934
ast_writefile: No such format 'wav|rec_to_448704386865_at_16022005-16'
Feb 16 16:56:35 WARNING[17028]: res_monitor.c:154
ast_monitor_start: Could not create file
/var/spool/asterisk/monitor/m-in Feb 16 16:56:35
WARNING[17028]: res_monitor.c:300
ast_monitor_change_fname: Cannot change monitor
filename of channel SIP/3004-da21 to m, monitoring not
started-- Executing AGI(SIP/3004-da21,
outbound.agi) in new stack
--CLI-
I do believe that I had this working before (I am
running the CVS HEAD from yesterday).
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RE: [Asterisk-Users] Monitor does not like variable subsitutions
OK, forget I said that. Wrong side of my brain.
Still, it is funny that it truncates the filename at the colon.
These lines are suspicious:
CLIast_writefile: No such format
'wav|rec_to_448704386865_at_16022005-16'
Where'd the :56:35 on the end go? Also, why is is trying to set the
format to wav|rec_to_4487 Shouldn't it just be wav?
Almost as if it sees the : as the first delimiter instead of the |
Here, it's trying to name the file using the third argument instead of
the second.
CLIast_monitor_start: Could not create file
/var/spool/asterisk/monitor/m
CLIast_monitor_change_fname: Cannot change monitor filename of channel
SIP/3004-da21 to m, monitoring not started
I don't know if this is too kludgy, but it might help to determine if
the : is buggering it up.
Cut(hrs=DATETIME,:,1)
Cut(mins=DATETIME,:,2)
Cut(secs=DATETIME,:,3)
REC_FILE_NAME=rec_to_${EXTEN:1}_at_${hrs}_${mins}_${secs}
Monitor(wav|${REC_FILE_NAME}|m)
(I didn't test this, so there might be typos in there, but you get the
idea . . .)
Where ya go from there, I couldn't say.
[EMAIL PROTECTED] wrote:
You are using illegal characters in your file name.
See this line in your output?
ast_writefile: No such format
'wav|rec_to_448704386865_at_16022005-16'
It can't get past it because the colon is not a valid
filename character.
[EMAIL PROTECTED] wrote:
Hello,
I have been attempting to get the Monitor function to
accept a loal variable substitution in order to use
the same filename later in the same context. Monitor
does not appear to like it, as it attempts to use
wav|filename as the recording type, as opposed to just wav.
Here is what I get if I just supply a filename
directly (it works fine):
--context-
exten = _9X.,3,Monitor(wav|recording|m)
--context-
--CLI-
-- Executing SetVar(SIP/3004-275c,
REC_FILE_NAME=rec_to_448704386865_at_16022005-16:54:10) in new
stack -- Executing Monitor(SIP/3004-275c,
wav|recording|m) in new stack
-- Executing AGI(SIP/3004-275c, outbound.agi)
in new stack
--CLI-
Here is what I get when I attempt to to variable
substituion for the filename:
--context-
exten =
_9X.,2,SetVar(REC_FILE_NAME=rec_to_${EXTEN:1}_at_${DATETIME})
exten = _9X.,3,Monitor(wav|${FILENAME}|m)
--context-
--CLI-
-- Executing SetVar(SIP/3004-da21,
REC_FILE_NAME=rec_to_448704386865_at_16022005-16:56:35) in new
stack -- Executing Monitor(SIP/3004-da21,
wav|rec_to_448704386865_at_16022005-16:56:35|m) in
new stack
Feb 16 16:56:35 WARNING[17028]: file.c:934
ast_writefile: No such format
'wav|rec_to_448704386865_at_16022005-16'
Feb 16 16:56:35 WARNING[17028]: res_monitor.c:154
ast_monitor_start: Could not create file
/var/spool/asterisk/monitor/m-in Feb 16 16:56:35
WARNING[17028]: res_monitor.c:300
ast_monitor_change_fname: Cannot change monitor
filename of channel SIP/3004-da21 to m, monitoring not
started-- Executing AGI(SIP/3004-da21,
outbound.agi) in new stack
--CLI-
I do believe that I had this working before (I am
running the CVS HEAD from yesterday).
--
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Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 265.8.8 - Release Date: 14/02/2005
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RE: [Asterisk-Users] Monitor does not like variable subsitutions
[EMAIL PROTECTED] wrote:
Hello,
There is no colon the filename below.
Exactly. But there *is* (or rather *was*) in the filename you told it
you wanted to write.
Where'd it go?
--- Jim Van Meggelen [EMAIL PROTECTED] wrote:
You are using illegal characters in your file name.
See this line in your output?
ast_writefile: No such format
'wav|rec_to_448704386865_at_16022005-16'
It can't get past it because the colon is not a
valid filename
character.
[EMAIL PROTECTED] wrote:
Hello,
I have been attempting to get the Monitor function to
accept a loal variable substitution in order to use
the same filename later in the same context. Monitor
does not appear to like it, as it attempts to use
wav|filename as the recording type, as opposed to just wav.
Here is what I get if I just supply a filename
directly (it works fine):
--context-
exten = _9X.,3,Monitor(wav|recording|m)
--context-
--CLI-
-- Executing SetVar(SIP/3004-275c,
REC_FILE_NAME=rec_to_448704386865_at_16022005-16:54:10)
in new stack
-- Executing Monitor(SIP/3004-275c,
wav|recording|m) in new stack
-- Executing AGI(SIP/3004-275c, outbound.agi) in new stack
--CLI-
Here is what I get when I attempt to to variable substituion for
the filename:
--context-
exten =
_9X.,2,SetVar(REC_FILE_NAME=rec_to_${EXTEN:1}_at_${DATETIME})
exten = _9X.,3,Monitor(wav|${FILENAME}|m)
--context-
--CLI-
-- Executing SetVar(SIP/3004-da21,
REC_FILE_NAME=rec_to_448704386865_at_16022005-16:56:35)
in new stack
-- Executing Monitor(SIP/3004-da21,
wav|rec_to_448704386865_at_16022005-16:56:35|m) in new stack
Feb 16 16:56:35 WARNING[17028]: file.c:934
ast_writefile: No such format
'wav|rec_to_448704386865_at_16022005-16'
Feb 16 16:56:35 WARNING[17028]: res_monitor.c:154
ast_monitor_start: Could not create file
/var/spool/asterisk/monitor/m-in Feb 16 16:56:35
WARNING[17028]: res_monitor.c:300
ast_monitor_change_fname: Cannot change monitor
filename of channel SIP/3004-da21 to m, monitoring not
started-- Executing AGI(SIP/3004-da21,
outbound.agi) in new stack
--CLI-
I do believe that I had this working before (I am
running the CVS HEAD from yesterday).
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RE: [Asterisk-Users] Setting a Forward to an external number on yourphone
[EMAIL PROTECTED] wrote:
Hi!
Maybe I have just been looking on the wrong pages but there is a
question that is very important for me. I already studied some
Demo-Dialplans and made some basic experiences with Asterisk.
But what I
need to find out is how I can handle this.
I am leaving my office and I want to tell asterisk to forward
calls now
to my mobile phone by just hitting a key (on my IP-Phone) or
by using a
special key-sequence.
How can this be handled because I need to change the dialplan
based on
some information coming from a device attached to a channel.
When back in my office I hit the key again and the calls are
now routed
to my IP-Phone (or ISDN-Phone on zap-channel) again.
With IP-Phones I can imagine just unregistering the phone and
having a
dialplan with a fallback-option or something like that. But what if I
want to tell asterisk to forward calls from now on to a
number I want to
manually add just for today (hitting a key, entering the new target
number and that's it). where can I find some information on
how to make
this feature available.
There's probably a whole lot of ways this could be achieved. It's kind
of a cookbook type thing: more than one recipe.
What I've been working on is a way to change where zero-out or main
menu timeouts. I want to be able to do this while on the road or in the
office, so I built it into the dial plan.
All the authentication issues aside, I need to set a variable that
defines where I want calls to go. Additionally, I want the state of this
variable to survive a restart.
What I've been messing with is using a file-based semaphores to do this.
Perhaps it's kludgy, but it works. Also, there's no database work
required: it all happens in the dialplan.
There's a lot more work needed here - I just hacked this together. Works
pretty well though, it's just not very friendly.
I wouldn't intall this at a customer without some more work, but mostly
because I'd want it a bit more friendly.
Anyhow, I have not tested this exact one, but it's similar to what I've
got so far. Enjoy:
[global]
MyCell=18005551212
Reception = SIP/YourPhone
; These are your semaphores. Because they are files they will survive a
restart
#include /var/lib/asterisk/remote_context
#include /var/lib/asterisk/remote_exten
[incoming]
; somewhere you'll need to put the exten that sends you to [presence]
; Make sure this is secure, but you'll probably want to be able to
; access it extenally
exten = 4321,1,Goto(presence,s,1)
[local_sets]
exten = 1000,1,Macro(exec_set,${EXTEN},${Reception})
[remote_sets]
exten = 2000,1,Macro(cell_user,${MyCell})
[presence]
;just record a basic prompt so you know what's up
exten = s,1,SayDigits(${ATDT_EXTEN}) ; raw, but it'll tell you where
it's going
exten = s,2,Background(prompt_for_choice)
this sets calls to go to one place
; first we write the semaphore
exten = 1,1,System(echo ATDT_CONTEXT=remote_sets
/var/lib/remote_context)
;then we set the value of the variable
exten = 1,2,SetGlobalVar(ATDT_CONTEXT=remote_sets)
exten = 1,3,System(echo ATDT_EXTEN=2000 /var/lib/remote_exten)
exten = 1,4,SetGlobalVar(ATDT_EXTEN=2000)
exten = 1,5,Goto(presence,s,1)
this sets calls to go to somewhere else
exten = 2,1,System(echo ATDT_CONTEXT=local_sets
/var/lib/remote_context)
exten = 2,2,SetGlobalVar(ATDT_CONTEXT=local_sets)
exten = 2,3,System(echo ATDT_EXTEN=1000 /var/lib/remote_exten)
exten = 2,4,SetGlobalVar(ATDT_EXTEN=1000)
exten = 2,5,Goto(presence,s,1)
you can have as many of these as you need
exten = 3,1,System(echo ATDT_CONTEXT=[some_context]
/var/lib/remote_context)
exten = 3,2,SetGlobalVar(ATDT_CONTEXT=[some_context])
exten = 3,3,System(echo ATDT_EXTEN=[some_exten]
/var/lib/remote_exten)
exten = 3,4,SetGlobalVar(ATDT_EXTEN=[some_exten])
exten = 3,5,Goto(presence,s,1)
[cleanup]
; hang up the call and whatever else
exten = s,1,Playback(goodbye)
exten = s,2,Hangup()
;;; MACROS ;;;
[macro-exec_set]
exten = s,1,Dial(${ARG2},20) ; Ring the interface, 20
seconds maximum
exten = s,2,Goto(s-${DIALSTATUS},1); Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send
to voicemail w/ unavail announce
exten = s-NOANSWER,2,Goto(default,s,1) ; If they press #,
return to start
exten = s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to
voicemail w/ busy announce
exten = s-BUSY,2,Goto(default,s,1) ; If they press #,
return to start
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as
no answer
exten = a,1,VoicemailMain(${ARG1}) ; If they press *, send
the user into VoicemailMain
[macro-cell_user]
; This will only work if your external line supports link transfer
exten = s,1,Playback(transfer)
exten = s,2,Flash()
exten = s,3,SendDTMF(${ARG1})
exten = s,4,Hangup()
--
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Checked by AVG
RE: [Asterisk-Users] Nortel i2004
[EMAIL PROTECTED] wrote: Stefan Gofferje wrote: Hi folks, has anybody knowledge about the Nortel i2004? Nortel calls it Internet Phone. I'm curious, which protocols it may understand... I just came back from a Nortel roadshow and was told it's H.323. That's one of the things that's so fascinating about Nortel - they themselves often have no clue how their products work. Nortel is desperately trying to pretend their VoIP products are standards compliant. What they fail to mention is that you have to buy a Nortel gateway product of some sort to actually achieve this. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.8 - Release Date: 14/02/2005 ___ 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] Nortel i2004
[EMAIL PROTECTED] wrote: Carl Sempla schrieb: On Monday, 14 February, 2005 02:30 : George Cohn [EMAIL PROTECTED] wrote: Stefan Gofferje wrote: Hi folks, has anybody knowledge about the Nortel i2004? Nortel calls it Internet Phone. I'm curious, which protocols it may understand... I just came back from a Nortel roadshow and was told it's H.323. Nop. This phone uses a proprietary protocol except for the audio part (RTP G711, G723 and G729). Some work are in progress right now for an UNISTIM support in asterisk. Got even more curious. Nortel's website also says something about H.323. I sent an inquiry to them - will post the result here if anybody's interested... I've worked with their IP phones for many years (have two right here in my lab). They are not, nor ever will be, standards compliant. If Nortel is promising such, ask them to outline EXACTLY what hardware is needed to make that happen. There WILL be a gateway of some sort involved. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.8 - Release Date: 14/02/2005 ___ 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] Nortel i2004
[EMAIL PROTECTED] wrote: Nortel i2004 spec is available here. http://www.nortelnetworks.com/products/library/collateral/i2004.pdf Documentation implies H.323 and MGCP interoperability. I'd love to know where to get the firmware for that. I am not sure of the nature of the relationship, but the Nortel phones are very similar, at least in appearance, to IP and analog phones sold by Aastra. That's because Nortel sold their analog phone division to Aastra, so some of the common plastic bits went too (the handsets, for example, look identical). -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.8 - Release Date: 14/02/2005 ___ 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] Cannot reset an IAXy box!!!
[EMAIL PROTECTED] wrote: asterisk-users-bounces at lists.digium.com wrote: Hi everyone, I was working yesterday and after I provide my IAXy box it loose any network comunication, the link light (green) is on and the activity light (orange) when the power is turned on it does nothing, but when I pickup the phone connected to the box, this light start blinking once per second. I've use ethereal to sniff a bit and I found that the box keeps asking to broadcast the MAC address for the IP of the asterisk server, the server answer but the IAXy miss it and keeps asking forever. A detail of the capture file of ethereal is attached to this message in plain text. The reset button does nothing (I've read that this button is just a cosmetic button here: http://lists.digium.com/pipermail/asterisk-users/2004-November /074909.html ). Any body has an idea to solve this issue??? Try re-programming the IAXy. That often fixes these types of problems. Well, I try to re-provisioning my IAXy box but it has no IP at all as I can see in the Ethereal capture: Sender MAC address: 00:0f:d3:00:0a:f0 (Digium_00:0a:f0) Sender IP address: 0.0.0.0 (0.0.0.0) Hmm. Any chance it's trying to get another IP address from DHCP? (from your description it doesn't sound like it, but who knows - that 0.0.0.0 address is kinda suspicious). And if I try to use the 0.0.0.0 IP, the loopback interface answer the request and no provisioning is made. Yeah, I can't see that working. 0.0.0.0 isn't really an address; more like a lack of one. That's what's got me wondering about DHCP. I'm still stuck, if I'm missing something obious please tell me what it is. Those IAXys are pretty quirky when it comes to configuration. You might need to run this one by Digium support. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.7 - Release Date: 10/02/2005 ___ 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] Cannot reset an IAXy box!!!
[EMAIL PROTECTED] wrote: Hi everyone, I was working yesterday and after I provide my IAXy box it loose any network comunication, the link light (green) is on and the activity light (orange) when the power is turned on it does nothing, but when I pickup the phone connected to the box, this light start blinking once per second. I've use ethereal to sniff a bit and I found that the box keeps asking to broadcast the MAC address for the IP of the asterisk server, the server answer but the IAXy miss it and keeps asking forever. A detail of the capture file of ethereal is attached to this message in plain text. The reset button does nothing (I've read that this button is just a cosmetic button here: http://lists.digium.com/pipermail/asterisk-users/2004-November /074909.html ). Any body has an idea to solve this issue??? Try re-programming the IAXy. That often fixes these types of problems. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.7 - Release Date: 10/02/2005 ___ 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 charge for Asterisk installations andongoing support?
[EMAIL PROTECTED] wrote: Hi, I'm just wondering how other people cost/charge for * boxes they have installed in small businesses? Asterisk is such a complex beast and is capable of so much it seems equally complex to figure out a charging model!? I agree. I'm contemplating installing a pair of servers at a small company with two separate locations, one main aim is to provide a IAX link between the two but I'm sure they will want other features!? Once they find out what it can do, they maight want all kinds of things. Renting vs purchasing? Support contracts? etc? Absolutely! Does anybody have any pearls of wisdom about this topic they are happy to share? There was a discussion about this on the -biz list. That's probably the better venue, as this list is more technically oriented. It's a good question, though: every system is like a custom app, so pricing will have to take into account how much development is going to be required. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.4 - Release Date: 01/02/2005 ___ 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] TDM400 stopped working
[EMAIL PROTECTED] wrote: On Tue, 1 Feb 2005, David Brodbeck wrote: -Original Message- From: Matthew Laird [mailto:[EMAIL PROTECTED] I excitedly installed my TDM dev kit earlier this weekend, installing asterisk and all the kernel drivers to make it work. And it did, it was fantastic. I then reboot the machine, and even after doing a modprobe wctdm, I get the following: == Parsing '/etc/asterisk/zapata.conf': Found Jan 31 13:34:27 WARNING[342]: chan_zap.c:793 zt_open: Unable to specify channel 1: No such device or address Did you remember to run ztcfg after loading the module? You have to do it every time or the channels won't be configured. Yes I did. It's even in /etc/modules.conf as a post-install action for the module. Hmm, found the problem, I just manually ran it again (I did last night) specifying the configuration file well that's annoying. I have zaptel.conf in /etc/asterisk along with the other configs, ztcfg looks in /etc So, why does it expect the file somewhere else from all the other asterisk configuration files? There's a very good reason for that: The zaptel driver is for the Linux operating system. Asterisk simply makes use of the driver. Theoretically, any software can use the zaptel driver, so it is appropriate to place its configuration file in /etc, with other, similar driver configuration files. The zapata.conf file, on the other hand, is specific to Asterisk. It tells Asterisk how to use the zaptel driver. Asterisk - /etc/asterisk/zapata.conf Linux - /etc/zaptel.conf Or, hierarchically: [Asterisk] (/etc/asterisk/zapata.conf) ^ | v (/etc/zaptel.conf) [Linux] Not well explained by the documentation, perhaps, but logical nonetheless. Cheers, Jim. -- Jim Van Meggelen [EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.3 - Release Date: 31/01/2005 ___ 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] Canadian Content: Telus and Shaw...
[EMAIL PROTECTED] wrote: Are there any VOIP lobbyist groups in Canada? Well, he's not exactly *in* Canada, but there's Jeff Pulver: http://www.crtc.gc.ca/ENG/transcripts/2004/tt0922.htm He seems to be about all we've got, thus far. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brandon Patterson Sent: Monday, January 17, 2005 2:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Canadian Content: Telus and Shaw... The CRTC is the biggest joke in the world. Ten people sit on their ass making decisions for 30+ million and no one has ever done anything to remove them from power. Heck even HBO is illegal in Canada. Why? Because they few that run the country want the many to remain their captive audience. We can go on all day about the CRTC - they represent the interests of the large companies only. Branson Patterson If they will do it, you are welcome to write the letter to CRTC and other governmental agencies for uncompetitive behavior. I think it should work. All the Best! Sergey. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.2 - Release Date: 28/01/2005 ___ 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] Nortel -- Asterisk--------Asterisk
[EMAIL PROTECTED] wrote: I am looking at setting up the following configuration and any help/input/comments before signing the PRI contracts will be greatly appreciated. PRI Tampa PRI Sarasota PRI -- Nortel BCM--Asterisk--Asterisk--- I would like to link the Nortel BCM to * using the a digital trunk card. Your diagram is a bit confusing to me. Still, the use of PRI in the BCM is a good plan. You've avoided using the highly unstable VoIP functions in the BCM. PRI support on the BCM is based on the Norstar architecture, which is very stable. So that basic design philosophy should serve you well. The BCM will continue to service the Tampa location, the * box would simply be used to pass extensions over the PRI to another * server in Sarasota and for a few SIP clients in other locations. Both servers will require two T100P cards. The Tampa server requiring one for communication with the BCM and one for the PRI to Sarasota. The Sarasota server will have one for communicating with the PRI to Tampa and one for accessing the PRI for local lines. That should work well. One thing you may want to consider is fronting the BCM with the Asterisk, as the Asterisk dial plan is far more flexible. I really could do away with the * server in Tampa but I figured with an * solution being so much cheaper then another proprietary system the added redundancy for any SIP clients would be worth the extra money, Never mind the money, Asterisk is technically superior to the BCM in every way on the VoIP and OS side of things. The only place the BCM might be able to argue a slight advantage is the wealth of key system features it offers on its sets. Whatever you do, avoid using the BCM for VoIP - you *will* regret it. Bottom line? The BCM has a limited future, whereas Asterisk *is* the future! plus Tampa has twice the data rate capacity of Sarasota already. I am really looking to achieve the following: 1. Reliability Asterisk - BCM is famously unstable (as is its operating system - Windows NT4.0). 2. Call quality The BCM may offer a slight advantage on the legacy links, but Asterisk is no slouch in this regard. 3. Cost effectiveness Asterisk - The BCM is closed, obsolete and expensive 4. Redundancy Asterisk - Linux has far more redundancy options than BCM ever will. Thank you in advance for your input. Good luck. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.2 - Release Date: 28/01/2005 ___ 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] Varion - Digium compatible cards
[EMAIL PROTECTED] wrote: Hi community, does anyone out there made some experience with Varion (www.govarion.com) based E1/T1 cards ? I have no hands on experience with them, but they are based on the open-source Tormenta card: www.zapatatelephony.org With Asterisk, I think any hardware manufacturer needs to be evaluated not just on their products, but also on what they contribute to the Asterisk community in general. Does Varion submit apps or patches? Documentation? Community support? Something to consider. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.2 - Release Date: 28/01/2005 ___ 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] Asterisk HEAD - Stable schedule?
[EMAIL PROTECTED] wrote: does anyone know when current HEAD is scheduled to stabilise? Is there a plan, or is it still some time in the future? I believe I saw an announcement recently that it will start stabilizing in February, with the goal of releasing 1.1 on the six-month anniversary of the 1.0 release. When was this? 1.2 is the next planned release of STABLE: http://dev.asteriskdocs.org/index.php/Main_Page -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.2 - Release Date: 28/01/2005 ___ 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] Minimum Setup
Dave Morrow wrote: Hi all, I have asterisk installed and working just fine with a couple of Cisco IP Phones. I am now ready to pilot connectivity to PSTN and am wondering what hardware would be recommended to make minimum connectivity to the public telephone network. Minimum would be any device that gives you an FXO port. Then you'd just order (or connect to) an analogue circuit. I am think ISDN as I would like a few external lines to be accessible. ISDN? Well, a PRI-ISDN would give you 23/30 channels, so that'd certainly be plenty. Not exactly minimal, though. If you are talking BRI-ISDN, then you'd get 2 channels per BRI loop. The value and availablilty of BRI is very dependant on where you are in the world. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.2 - Release Date: 28/01/2005 ___ 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] PRI for Data and Voice
David Norton wrote: Hi, Currently I only have 1 PRI which I am using for dial-in customers. The line is connected to a Portmaster3. I have never used more than 10 concurrent channels. The calls can be both analog or ISDN. It would be a waste to order another PRI for my Asterisk box. Is there any way of splitting a PRI into 2 PRIs of 15 channels each, or plugging the PRI into the * box and it send the data calls to the portmaster, or handles them itself? Any advice would be much appreciated I betcha Sangoma has something that'd do this for you. They've been supporting T1 data on Linux for years, and they're recently added zapata to their list of open-source drivers. Give them a shout, they love this kind of stuff. Cheers, Jim -- Jim Van Meggelen [EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.2 - Release Date: 28/01/2005 ___ 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] Nortel -- Asterisk--------Asterisk
[EMAIL PROTECTED] wrote: Jim Van Meggelen wrote: [EMAIL PROTECTED] wrote: [snip] Your diagram is a bit confusing to me. Still, the use of PRI in the BCM is a good plan. You've avoided using the highly unstable VoIP functions in the BCM. Sorry, it got a little jumbled No worries. I got the gist of it. PRI support on the BCM is based on the Norstar architecture, which is very stable. So that basic design philosophy should serve you well. [snip] That should work well. One thing you may want to consider is fronting the BCM with the Asterisk, as the Asterisk dial plan is far more flexible. The Nortel is set up and runs well enough to serve the one location. I am a strong believer in if it isn't broke don't fix it. Any short comings that we have experienced with the BCM, I hate to say it, but people have gotten used to it. As you said. If it ain't broke . . . Never mind the money, Asterisk is technically superior to the BCM in every way on the VoIP and OS side of things. The only place the BCM might be able to argue a slight advantage is the wealth of key system features it offers on its sets. Whatever you do, avoid using the BCM for VoIP - you *will* regret it. Too late! I have 32 i2004's currently running off of it. Please don't even let me get started on how hard it is to get a clean call through the system even with QOS and dedicated links. LOL! Did you try putting the sets on their own LAN? They don't seem to play nice with other stuff on the network, but on their own it can be tolerable (a Nortel switch can sometimes help as well). It's all that proprietary thinking that leads them to think they can get away with not testing in multi-vendor environments. Bottom line? The BCM has a limited future, whereas Asterisk *is* the future! plus Tampa has twice the data rate capacity of Sarasota already. I am really looking to achieve the following: 1. Reliability Asterisk - BCM is famously unstable (as is its operating system - Windows NT4.0). We experienced the VOIP phones sporadically losing connection to the BCM. The ethernet link on the BCM was locking up for some reason. Have you got all your patches up to date? There's a new one every week, it seems. The only way to do a hard reset on a BCM it to literally pull the power plug out of the back of the machine. To turn it on you plug it back in. Mind you these are Nortel directions. When they first told me that I thought they were joking. The problem there is the MSC card. That card is actually a totally independant Norstar KSU that's been wired onto a PCI board. The reason you have to yank the power on the box is that it's the only supported way of rebooting that card (it gets its power from the PCI bus, but it has it's own CPU - M68030 - and operating system - WindRiver pSOS). There are some pretty interesting utilities on the hard drive of the BCM (there's one called resetksu.exe in the \Program Files\Nortel Networks\Voice Platform folder), but they are neither supported nor documented. For the most part it's a mess and when you see the number of OS2- DOS-type CMD/BAT files Nortel has handling critical system scripts you will want your money back. Most of the Nortel support folks barely understand Windows, PCs and Norstar, never mind this beast they've kludged together. My Dream? Format the hard drive of the BCM and install Asterisk. I've had Linux running on a BCM chassis several times (back when I had access to a lab with several BCMs). Not PCM telephony, though (at least not integrated with the network). The trick'd be getting an API from Nortel for the MSC card. Not much chance of that. Rumour has it that BCM 4.0 is going to be Linux/PPC-based. Won't take long before someone ports Asterisk to it. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.2 - Release Date: 28/01/2005 ___ 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] Newbie
[EMAIL PROTECTED] wrote: Hello, I am new here. I am also somewhat new to telephony and IT, however, I am technically adept. That's probably the most important thing with Asterisk. I work for a small but growing non-profit in Colorado and I wear many hats. Because I am the only one who has a clue about computers, I am the default IT person. Our communications needs are growing and we are checking out phone system -- particularly Asterisk. As I understand it, for a relatively low initial hardware investment (PC, TDM4 card, phones), I can get into this VoIP system, use it with our current computer network and will have the power and flexibility of a much more expensive system. You've got it. My concern is that I know nothing about Linux or Asterisk. I can troubleshoot most any problem on Mac or Windows (even networking where I have little experience) but I fear I will be over my head with this and it will consume all my time. It will require you to learn a thing or two, no doubt about it. Once you get it running, it should not require endless tweaking or babysitting (although you may fall in love and become addicted to tweaking your Asterisk box :-) I understand there will be a time commitment (more at first and less for maintenance) but as my primary duties are elsewhere, I cannot afford to put in too much time on this project. That's going to be a tough one to evaluate. You could end up with more work than you expect (especially if your skills do not grow in sync with your schedule). Having said that, Linux and Asterisk are both labours of love, so they are not cruelly complex - they actually tend to make a lot of sense, once you understand the methodology. There's no wierd math or arcane syntax here - everything is in plain English for the most part. Finally, my questions: 1) How much time should I expect to become familiar with Linux considering I am only interested in it at this point for the Asterisk installation (however, I have been looking for an excuse to learn it)? If you are technically adept (i.e. familiar with Windows, PC hardware, Networking) you can get Linux going without too much trouble. Linux is really quite simple, it's just that it does so many things there's a lot to learn - this can sometimes make it seem complex. To learn Linux properly you will want to learn it as an Administrator, not a User. Many books will attempt to mislead you into believeing that Linux is actually the GUI desktop. This is guarantted to confuse you. I first learned about Linux by reading O'Reilly's Running Linux (still in print ten years later). Better yet, look on Amazon and find out what Linux Administration book is most popular with newbies. Read the reviews and you'll get a feel for what makes sense to you. Warning: stay away from too friendly Linux books. Most of them only teach you about the GUI. This is no good. If the book has too many pictures of the desktop, avoid it like the plague. You need to learn Linux as an Administrator, not a User. This is crucial. 2) How much time should I expect it to take for me to configure Asterisk for a basic installation? Once we have dial-tone and can make/receive calls, I will add/configure features, voice mail, etc. That is a tough question to answer, as it's impossible to evaluate your existing skills, not to mention your ability to aquire more. 3) I know what specs they say to run Asterisk, but realistically, what minimum specs would you recommend for the PC? Grab a current generation P4, install it on a current generation MoBo (Intel chipset, or possibly NForce), and give it a gig of RAM. That'll serve ten or twenty users quite easily, even if you want to do some fancy processing (conference calls and such). Keep in mind that there are certain best practices with respect to telecomunication equipment you should bear in mind: - A power-conditioned UPS will provide stable, clean power, and provide some run time during black outs and brown outs. - Spending a lot of money on a high-end server may not be required, but do not buy junk either. Proven, brand name parts for all your components is highly recommended. - Build a proper home for your system. Treat it like the mission-critical server it is. - Devise a back up strategy that ensures that you can restore all your critical data. This can be fairly simple if you have tolerance for some downtime. 4) How much should I expect to pay initially for the basic hardware (PC, card, phones, book)? Any recommendations for providers? Buy the PC from a trusted local source. Our current phone company is offering us a full, fractional T1 with four lines for $368/mo and a $250 install with a three-year contract. Does this sound like a good deal? I'm hoping I can get the initial installation and configuration done in 30-40 hours over two weekends and a few evenings. Does this sound reasonable? It is certainly not unrealistic, if you can learn the skills you need fast
RE: [Asterisk-Users] Asterisk HEAD - Stable schedule?
[EMAIL PROTECTED] wrote: Jim Van Meggelen wrote: I had always thought that the difference between STABLE and HEAD *was* the stability. Not so much stability in terms of code, but more in terms of features and compatibility. Is that not the same numbering philosophy that applies with Linux? It used to be, yes. However, saying 1.1.x applies to HEAD is irrelevant if there are never any releases made from HEAD. Version numbers are only applied to releases, and if the next release that is made is a release candidate for STABLE, then it will be 1.1.0-rc1 or something like that. I guess I had this picture in my head of some incrementing number associated with HEAD. 1.1.[what day is it?] Having even/odd version numbers does make sense if releases are made from the development branch, so that they can be tracked separately from the stable branch. The bottom line is going to be something that provides useful information to help in selecting your version of choice. The even=stable, odd=unstable is nice if you're familiar with the terminology. Otherwise I suppose it doesn't really matter. Cheers, Jim. -- Jim Van Meggelen [EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.4 - Release Date: 25/01/2005 ___ 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] Asterisk HEAD - Stable schedule?
[EMAIL PROTECTED] wrote: Roy Sigurd Karlsbakk wrote: does anyone know when current HEAD is scheduled to stabilise? Is there a plan, or is it still some time in the future? I believe I saw an announcement recently that it will start stabilizing in February, with the goal of releasing 1.1 on the six-month anniversary of the 1.0 release. I seem to recall somewhere that the thinking is that we'll be going with Linux-type nomenclature, where the even-numbered releases are STABLE, and the odd are HEAD. So 1.0.x STABLE will become 1.2.0 STABLE, and 1.1.x HEAD will be continued as 1.3.0 HEAD Could be wrong, but it'd make sense. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005 ___ 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] Asterisk HEAD - Stable schedule?
[EMAIL PROTECTED] wrote: Jim Van Meggelen wrote: So 1.0.x STABLE will become 1.2.0 STABLE, and 1.1.x HEAD will be continued as 1.3.0 HEAD Could be wrong, but it'd make sense. It would make sense if we ever did unstable releases from HEAD, but we don't do that currently :-) I had always thought that the difference between STABLE and HEAD *was* the stability. Not so much stability in terms of code, but more in terms of features and compatibility. Is that not the same numbering philosophy that applies with Linux? -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005 ___ 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] Any experience with Sangoma cards?
[EMAIL PROTECTED] wrote: Hi, I am considering A101/102/104 cards for my asterisk installations. Has any of you used these or any Sangoma cards in such environment? Any thoughts? How do they stack up against Digium cards? Any input would be greatly appreciated. robert The Sangoma cards are very well built and very well supported. They've been in business since 1984, and are commited to Linux and open source. The T1 cards they have for Asterisk are ASIC-based, which means that they can program their card to be whatever they want it to be. For Zaptel compatibility, they actually dumbed-down their cards - the hardware is capable of a lot more. Expect to see exciting things from Sangoma with respect to open source telephony. Cheers, Jim. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005 ___ 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] PSTN and Asterisk
Title: Message www.voip-info.org www.asteriskdocs.org -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vassili GontcharovSent: January 24, 2005 5:13 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] PSTN and Asterisk Hi quys, I look for a solution for interconnection beetwen PSTN and VoIP. My application have to treat few protocols comming from PSTN lines and mixing data , dtmf and voice. Can I use Asterisk for : PSTN-- Asterisk (converting analog call to IP) -- MyApplication( translation protocolsand do some workswith incomming data) What hardware I can use for this? Do use Asterisk G.711 protocol? Thanks Vassili --No virus found in this incoming message.Checked by AVG Anti-Virus.Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005 ___ 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] Anyone know where a good source of mailing liststats might be found?
Thanks to everyone who provided feedback. [EMAIL PROTECTED] wrote: Folks, I'm curious to know how the volume of Asterisk-Users rates as far mailing lists go. This list sees over 200 messages per day, which has GOT to put it in the top 5%, doesn't it? I'd love to know if anyone has knowledge of any organization that might maintain such stats. Regards, Jim. -- Jim Van Meggelen [EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone know where a good source of mailing list stats might be found?
Folks, I'm curious to know how the volume of Asterisk-Users rates as far mailing lists go. This list sees over 200 messages per day, which has GOT to put it in the top 5%, doesn't it? I'd love to know if anyone has knowledge of any organization that might maintain such stats. Regards, Jim. -- Jim Van Meggelen [EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005 ___ 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 Synchronization
Title: Message I tested this as well and found the same thing. I have submitted a bug report: http://bugs.digium.com/bug_view_page.php?bug_id=0003394 That one's severe enough that I have flagged it as a major. On a busy system, this bug could cause all kinds of messages to go missing, with no indication that anything had gone amiss . . . "Did you get my mesage?" "No, I never got any message" "Well I left one" "Are you sure?" [battle ensues] Good catch! -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stojan Sljivic - PametSent: January 21, 2005 5:32 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [Asterisk-Users] Voicemail Synchronization Hi, Ihave stress tested the Asterisk Voicemail. We have encountered problem with simultaneous calls that are sent to the same mailbox. It occurred that several calls were writing to the same file. It seems that there is a synchronization issue in the Voicemail application. Did someone else find this issue? What would be the solution/workaround for it? Regards, Stojan Sljivic --No virus found in this incoming message.Checked by AVG Anti-Virus.Version: 7.0.300 / Virus Database: 265.7.1 - Release Date: 19/01/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.1 - Release Date: 19/01/2005 ___ 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] G.729? Worth it?
[EMAIL PROTECTED] wrote: Hi All, For a small installation using ITSPs via DSL is G.729 a worthwhile exercise? I have G.729 capable SIP phones and my ITSPs cupport the codec so I could go end-to-end without transcoding. What's call quality like compared to G.711, GSM or iLBC? Low bandwidth Low CPU utilization Best audio quality Pick any two. G.729 is an extremely low bandwidth codec, offering surprisingly good quality. Unfortunately, it is also a CPU hog. Generally speaking, you should avoid the use of compressed codecs unless bandwidth is a concern. Especially with Asterisk, the performance penalty due to transcoding is a high price to pay. On the LAN, I'm not able to see any reason to use anything but G.711. In the WAN, it depends on how much bandwidth you need for voice, versus what you have available. Keep in mind that the type of trunking you use plays a role. For example, although I haven't done the math, it is entirely possible that 20 trunked channels of IAX2/G.711 would use less bandwidth than 20 discrete SIP/G.729A connections between the same endpoints. Much of the bandwidth used by packetized voice is overhead, so a lower bitrate codec does not yield a linear improvement in bandwidth. There are other considerations as well, such as how well a codec handles a lossy link. This can be very important to the percieved quality of the connection, which, to the user, is far more important. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.0 - Release Date: 17/01/2005 ___ 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] G.729? Worth it?
[EMAIL PROTECTED] wrote: Low bandwidth Low CPU utilization Best audio quality I think you might want to clarify that Best audio quality is in relation to other highly compressed codecs. Certainly my (albeit limited) experience is that g711 is much more clear than g729. Compared against gsm, for example, however, the audio quality is quite good What I was saying was that you can only have two of the three. It's an old adage about compromise. G.711: Low Nadwidth: NO Low CPU utilization:YES Best Quality: YES (as far as telephony codecs go) G.729 Low Bandwidth: YES Low CPU Utilization:NO Best Quality: NO -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.0 - Release Date: 17/01/2005 ___ 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] CAS voice signalling?
[EMAIL PROTECTED] wrote: According to CarrierAccess, the Adit 600 uses CAS for voice signalling. What is this? This should not be a problem for Asterisk? Does the Asterisk server need to reencode CAS into aLaw when going to Euro ISDN? Try this: http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a 00800e2560.shtml Cheers, Jim. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.13 - Release Date: 16/01/2005 ___ 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] Any interest in a Canadian Asterisk mailing list?
[EMAIL PROTECTED] wrote: On January 17, 2005 01:47 am, John Sellens wrote: Just on the off chance that Canadian Asterisk users might be interested in a place to discuss topics specific to the great white north (sources, services, telcos, etc.), I created the asterisk-canada mailing list: I know as a Canadian I'm not interested in a list Just for Canadians -- It's just fragmenting the help available for very little benefit. I do, however, appreciate the thought. I don't think the idea is to be just for Canadians, but more as a forum for topics that relate to Asterisk in the Canadian environment. A very relevant example is the CRTCs deliberations on VoIP, which may have huge repercussions to Canadian Asterisk users, but is hardly relevant to the international version of the Asterisk-Users list. Bell and TELUS bashing might also be popular topics :-) I do agree that any subject that is not specific to the Canadian experience should remain in the international list. We are an international community; therein lies our power. Anyhow, I signed up, and am planning to start a thread about the CRTC VoIP deliberations (and the generous act performed there by Jeff Pulver), something I wouldn't feel was appropriate on Asterisk-Users. Time will tell how many topics there are to discuss. The way I see it, a Canadian mailing list will be no different than our country itself: visitors will always be welcome. -- Jim Van Meggelen [EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.13 - Release Date: 16/01/2005 ___ 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] simple over view of the process
[EMAIL PROTECTED] wrote: Hello All, Please forgive the lack of understanding as of yet but I have been trying to follow the mailing list messages over the last few days and would like to know if someone could wither point me into the right direction or possibly give me a brief overview of the complete process. Start here: www.asteriskdocs.org Then read this: http://www.digium.com/handbook-draft.pdf Ultimately, here is where the most information can be found: www.voip-info.org/wiki-Asterisk When you've done all that, the Users list and IRC will be a great place to come and brainstorm. Basically, I see that the Asterisk PBX systems can run on linux and seems to offer the engine base that is needed for the SIP clients to connect. For pure SIP, you may want to look at SER. Asterisk is not as powerful on the SIP side of things, but is overall more powerful due to it's support of all the major voice standards (both legacy and VoIP). It's an incredible engine, but it comes with a price: there is a lot to learn. Spend a few hours reading, get a Linux system you can play with, download it, and take the time to play. Don't know Linux? You WILL suffer. Learn Linux first (gotta crawl before . . . ) Additionally, it seems that the various hardware (of which I have no idea) if installed into the server will allow the SIP clients to communicate with analog lines. Asterisk can act as a gateway, yes. What inexpensive hardware is need to set up a basic system? As a learning exercise, Digium's development kit is how many get their start. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.13 - Release Date: 16/01/2005 ___ 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] Any interest in a Canadian Asterisk mailing list?
Gyrion, Larry M. wrote: I believe the US and Canada use the same methods for voice services, maybe we could make it a North America list serv instead. Just some thoughts here I see the value of regional lists primarily as relates to non-technical items such as local service providers, regulatory issues, and so forth. For the technical stuff, I much prefer the single list. I don't mind at all reading about POTS lines in the UK, or the joys of ISDN in the EU. Maybe one day the sheer volume of messages will make that untenable, but for now I love the international flavour of the list. As VoIP becomes ubiquitous, the value of regional lists might be less and less relevant. Jim. -Original Message- From: Jim Van Meggelen [mailto:[EMAIL PROTECTED] Sent: Monday, January 17, 2005 11:42 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Any interest in a Canadian Asterisk mailing list? [EMAIL PROTECTED] wrote: On January 17, 2005 01:47 am, John Sellens wrote: Just on the off chance that Canadian Asterisk users might be interested in a place to discuss topics specific to the great white north (sources, services, telcos, etc.), I created the asterisk-canada mailing list: I know as a Canadian I'm not interested in a list Just for Canadians -- It's just fragmenting the help available for very little benefit. I do, however, appreciate the thought. I don't think the idea is to be just for Canadians, but more as a forum for topics that relate to Asterisk in the Canadian environment. A very relevant example is the CRTCs deliberations on VoIP, which may have huge repercussions to Canadian Asterisk users, but is hardly relevant to the international version of the Asterisk-Users list. Bell and TELUS bashing might also be popular topics :-) I do agree that any subject that is not specific to the Canadian experience should remain in the international list. We are an international community; therein lies our power. Anyhow, I signed up, and am planning to start a thread about the CRTC VoIP deliberations (and the generous act performed there by Jeff Pulver), something I wouldn't feel was appropriate on Asterisk-Users. Time will tell how many topics there are to discuss. The way I see it, a Canadian mailing list will be no different than our country itself: visitors will always be welcome. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.13 - Release Date: 16/01/2005 ___ 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] Any interest in a Canadian Asterisk mailing list?
[EMAIL PROTECTED] wrote: On January 17, 2005 11:42 am, Jim Van Meggelen wrote: Anyhow, I signed up, and am planning to start a thread about the CRTC VoIP deliberations (and the generous act performed there by Jeff Pulver), something I wouldn't feel was appropriate on Asterisk-Users. Time will tell how many topics there are to discuss. I think that's more than relevant here. Hmmm. I'd have a hard time appreciating regulatory challenges in regions that I don't have involvement with, so I assume that the reciprocal is generally true as well. Granted, I shouldn't presume that everyone thinks as I do, but people tend to have a limited interest in subjects that don't directly affect them. I figure technical discussions in Asterisk-Users are nearly always of some value, while regulatory peculiarities mostly are not. Having said all that, I want to stress that I do not feel that I am right and you are wrong. I not only respect your opinion, but am open to the possibility that you are correct in your assertion. For me, however, the separate list seems to have value. It is not because I want to start a Canadian Club(tm), but more that I would love to talk about certain subjects of interest to Canadian Asterisk users, and somehow feel that Asterisk-Users is not the right venue for it. It's not a matter of right or wrong; more like personal taste, really. Regards, Jim. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.13 - Release Date: 16/01/2005 ___ 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] Any interest in a Canadian Asterisk mailing list?
[EMAIL PROTECTED] wrote: On January 17, 2005 01:57 pm, Jim Van Meggelen wrote: I'd have a hard time appreciating regulatory challenges in regions that I don't have involvement with, so I assume that the reciprocal is generally true as well. Granted, I shouldn't presume that everyone thinks as I do, but people tend to have a limited interest in subjects that don't directly affect them. I figure technical discussions in Asterisk-Users are nearly always of some value, while regulatory peculiarities mostly are not. I agree with you on this, but then again I put up with so much on the -users list that I really don't care to listen to, especially the same tired discussions over and over, that I figure some regulatory babble might be a welcome change. LOL. I hadn't thought of it that way. Little vignettes amidst the commercials? That and for the half dozen or so Canuck-only threads I can think of off the top of my head it's hardly worth putting together an entire mailing list. Just because the volume isn't there? That might be a good thing, ya know - have a list with, say, one or two messages a day, on average. Having said all that, I want to stress that I do not feel that I am right and you are wrong. I not only respect your opinion, but am open to the possibility that you are correct in your assertion. hahaha don't worry I've got a pretty thick skin and actually enjoy being proven wrong... makes me a better person. It was a policy at our company that any new product implementation would always require Technical Support be involved until several engineers, technicians and installers were comfortable with it. I hope I always remember the lessons learned from getting new products, and having to develop training and implementation practices. I'd figure I was an expert on the ones that went smoothly, but then six months later if someone would call me with problems, I'd discover that I didn't know a thing. As for the ones that gave nothing but trouble, I'd fear them, assuming I didn't know what I was doing, until it became clear that more often than not, I knew more about them than the manufacturer did. The fewer the troubles, the less knowledge retained; the more painful the task, the more god-like skills were obtained. It took me many years to figure this out - I used to think it should be the other way around. Seems that making mistakes is actually a fantastic (albeit uncomfortable) way to learn. I sometimes wonder if I unconsciously muck things up at first as a rite of passage. Nobody knows a thing so well as those who can expertly break it. For me, however, the separate list seems to have value. It is not because I want to start a Canadian Club(tm), but more that I would love to talk about certain subjects of interest to Canadian Asterisk users, and somehow feel that Asterisk-Users is not the right venue for it. Well as I said there's so much fluff in -users already that until we really become a nuisance I don't see the harm in it at all. Half of the stuff belongs on -biz anyway and a full third of the remaining in asterisk-newbies-who-refuse-to-research-before-posting... a dozen or so threads of stuff that makes no sense to anyone outside of Canada wouldn't even register as a blip on the screen. You're probably right. Would it be considered trolling to start a thread on Cleaning Maple Syrup off of Dial Pads, or Wiring your Moose for Wi-Fi? Besides my personal bet is that the CRTC is going to just mimic what the FCC does anyway (c.f. UL vs CSA, wireless regulations, etc., etc.) that it would be great to see what the various big boys stateside have to say and how they interpret what's going on with the FCC and VOIP regulation. It's not a matter of right or wrong; more like personal taste, really. Agreed. Does that mean I'm right and you're wrong? ;-P -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.13 - Release Date: 16/01/2005 ___ 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] Any interest in a Canadian Asterisk mailing list?
[EMAIL PROTECTED] wrote: On January 17, 2005 04:47 pm, Jim Van Meggelen wrote: LOL. I hadn't thought of it that way. Little vignettes amidst the commercials? Exactly -- It's precisely why I hang around on linux-elitists and a couple other oddball lists... a good 90% of what's there is crap but man when something good comes by... wowza. It sure can be time consuming trying to keep up, though. Just because the volume isn't there? That might be a good thing, ya know - have a list with, say, one or two messages a day, on average. True, but that's why I like looking at sineapps now and again -- they sometimes focus on things that I've not even seen, it's interesting reading... but I slug it out on the list well, just to slug it out. :-) I think we are all a bit masochistic on this list . . . all 10,000 plus of us . . . It was a policy at our company that any new product implementation would always require Technical Support be involved until several engineers, technicians and installers were comfortable with it. I hope I always remember the lessons learned from getting new products, and having to develop training and implementation practices. ... Seems that making mistakes is actually a fantastic (albeit uncomfortable) way to learn. I sometimes wonder if I unconsciously muck things up at first as a rite of passage. We have a similar policy here and it really helps people understand why things are done a certain way when they have to field some of the customer calls themselves. right and wrong take on new nuances that they would have otherwise been oblivious and even belligerent towards. Nobody knows a thing so well as those who can expertly break it. That sounds very close to As soon as you make something idiot-proof along comes a better class of idiot. :-) One of my favorites is (to paraphase Douglas Adams): It's easy to be blinded to the essential uselessness of these things by the sense of satisfaction you get from making them work at all Would it be considered trolling to start a thread on Cleaning Maple Syrup off of Dial Pads, or Wiring your Moose for Wi-Fi? Let's not forget the weekly tooques and telephony segment, and a review of the best block heaters for your wi-fi fones. Snoms and Snowmobiles? Ice Phishing? Tim Hortel? -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.13 - Release Date: 16/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Canadian Content: Telus and Shaw...
Kim Lux wrote: I called Telus before Christmas requesting some sort of VOIP connection. Here is what I learned: a) the guy I was talking to never heard of * That'll change. b) they didn't think there was any way that a PC could perform the duties of a PBX He was probably thinking of the Nortel BCM. It's a PC pretending to be a PBX, and yeah, it ain't up to the task. c) they told me they didn't have any VOIP connections, but then told me that they would supply and connect Nortel PBXs using H323 Yep. The name of the game is customer lock-in. d) they would not supply me with an H323 connection for *. More to the point, they don't have the processes in place to make it happen. I don't have time to discuss this in detail, I just thought I'd share it based on the chat in the CDN list discussion. We are going with babytel. I'll advise how that works when it is up and running, hopefully next week. I don't see an option to connect Asterisk to them. BTW: Shaw is supposed to start supplying VOIP on a separate network from their high speed network. Here is the news clip: http://www.canoe.ca/NewsStand/CalgarySun/News/2005/01/14/898082-sun.html I find this interesting because several people have told me they are using Shaw's high speed Internet service as the backbone of their VOIP system. (Extreme is supposed to work even better.) I wonder if Telus is going to block the SIP ports on their ADSL network I'm wondering what the CRTC is planning with respect to VoIP. Whatever they do, they'll probably miss IAX entirely, so no worries. ? I wonder if Shaw will ? (Telus presently blocks the SMPT port so that you MUST you their mail server.) I wonder if shaw or telus people lurk on this site. If you mean people that work for those companies, then sure (but for the most part they love it the same as any of us). If you mean decision-makers, I would say generally no. Asterisk is not yet on anyone's radar scope, so it won't be considered a threat. That'll happen when it starts stealing market-share. Jim. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.13 - Release Date: 16/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Any interest in a Canadian Asterisk
[EMAIL PROTECTED] wrote: Quoting [EMAIL PROTECTED]: Would it be considered trolling to start a thread on Cleaning Maple Syrup off of Dial Pads, or Wiring your Moose for Wi-Fi? Let's not forget the weekly tooques and telephony segment, and a review of the best block heaters for your wi-fi fones. Oh, we're gonna have a good time next Thursday. We need to get Molson Canadian to sponsor us and find Bob Doug for the event? Who needs those hosers, eh? By the way, eh. It's hard to get the moose to cooperate. When you put the parabolic antenna on his antlers you have to ride him backwards when you're leaving your cabin, eh. That's nuthin'! Try gettin him on the snowmobile, eh? Or makin' him sit still in the drive-through at Tim Hortons. I'm tellin' ya . . . -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.13 - Release Date: 16/01/2005 ___ 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] Hardware issues
[EMAIL PROTECTED] wrote: Since i need to run at least 30 FXS-channels, what hardware should I use? Both Motherboard etc and FXS-hardware? There seems to be only 4-channel cards from digium. Then I would need at least 8 cards. 8 FXS-cards + 1 E1-card. What motherboard handles this? And to they all need unique IRQ's? You'll probably want to look into channel banks. http://www.voip-info.org/wiki-Asterisk+Channel+Bank -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.11 - Release Date: 12/01/2005 ___ 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 present a dialtone to a dial-in user?
Title: Message Is there a dialtone recording? I'm thinking : exten = s,1,background(dial-tone-recording) exten = s,2,goto(,s,1) I have not tested this, but I can't see why it wouldn't work. The only problem would be the lack of a timeout (and possibly a small challenge getting a recording of an appropriatedial tone). Or, you could have a recording that was the length you wanted before going to timeout. Just brainstorming here, so any reasonably civilized feedback is welcome. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Silviu HerchiSent: January 13, 2005 11:16 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [Asterisk-Users] How to present a dialtone to a dial-in user? Hello, Heres what Id like to do: call my Asterisk box from a phone, hangup after a few rings, then Asterisk calls me back and presents a dialtone, than I can dial any valid number in the context the call originated. Ive done it with CAPI (thanks to the script on http://www.junghanns.net/asterisk/page14.html), Id like to do it with H323. Problem is, how to present a dialtone to the user and wait him dialing? Thank you, Silviu --No virus found in this incoming message.Checked by AVG Anti-Virus.Version: 7.0.300 / Virus Database: 265.6.11 - Release Date: 12/01/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.11 - Release Date: 12/01/2005 ___ 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] Ring Voltage Supplied by Wildcard TDM400P REV E/F AUTO FXS/DPO
[EMAIL PROTECTED] wrote: Hi; I'm trying to connect a TDM400P with an FXS module to a Valcom V-9940 Paging adaptor. This port on the TDM400P was connected to a 2500 Set and was working I just re-connected it to the Valcom (which is known to work on a Telco POTS line) and its not picking up. The Valcom docs say it need a minimum of 75 Volts at 20-30 Hz to recognize a call... So the question is what ring voltage does the FXS modules on a TDM400P put out? Ultimately, that depends on how much current is being drawn, but my multimeter reports about 70V on a 6 foot line cord. Generally, I'd expect an FXS to put out more like 90-110V (that's what my TalkSwitch supplies). -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.10 - Release Date: 10/01/2005 ___ 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] Ring Voltage Supplied by Wildcard TDM400P REV E/F AUTO FXS/DPO
Peter Svensson wrote: On Tue, 11 Jan 2005, Jim Van Meggelen wrote: [EMAIL PROTECTED] wrote: I'm trying to connect a TDM400P with an FXS module to a Valcom V-9940 Paging adaptor. This port on the TDM400P was connected to a 2500 Set and was working I just re-connected it to the Valcom (which is known to work on a Telco POTS line) and its not picking up. The Valcom docs say it need a minimum of 75 Volts at 20-30 Hz to recognize a call... So the question is what ring voltage does the FXS modules on a TDM400P put out? Ultimately, that depends on how much current is being drawn, but my multimeter reports about 70V on a 6 foot line cord. Generally, I'd expect an FXS to put out more like 90-110V (that's what my TalkSwitch supplies). The ring voltage can be configured. Try setting the module parameter boostringer when the wctdm module is insmod / modprobed. Which is accomplished in 1.0.x by the following: # modprobe wcfxs boostringer=1 And in CVS HEAD (I assume, as I haven't tested this) by: # modprobe wctdm boostringer=1 Who, exactly, Boo Stringer is has not yet been determined. Boo Radley? NO, Boo Stringer. (Sorry for the pun, folks, but I will now never forget this parameter name). -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.10 - Release Date: 10/01/2005 ___ 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] Ring Voltage Supplied by Wildcard TDM400P REV E/F AUTO FXS/DPO
Rich Adamson wrote: I'm trying to connect a TDM400P with an FXS module to a Valcom V-9940 Paging adaptor. This port on the TDM400P was connected to a 2500 Set and was working I just re-connected it to the Valcom (which is known to work on a Telco POTS line) and its not picking up. The Valcom docs say it need a minimum of 75 Volts at 20-30 Hz to recognize a call... So the question is what ring voltage does the FXS modules on a TDM400P put out? Ultimately, that depends on how much current is being drawn, but my multimeter reports about 70V on a 6 foot line cord. Generally, I'd expect an FXS to put out more like 90-110V (that's what my TalkSwitch supplies). According to the Silicon Labs spec sheet, the 3210 chip can supply ringing signals of up to 88v peak or more, enabling the ProSLIC to drive a 5 REN ringer load across loop lengths of 2000 feet or more. The multimeter will be measuring rms voltage, not peak. Adding boostringer=1 to the modprobe parameters convinced my DMM to display the 89V promised. But yeah, it's not a true RMS DMM I'm using (I suspect a typical non-true RMS DMM would tend to be biased towards 60Hz and thus give odd AC readings at different frequencies - I noticed that the detected voltage would start at ~89V, but then drop as the ring phase of the cycle continued - I'm figuring it's aliasing somehow) -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.10 - Release Date: 10/01/2005 ___ 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] GSM adapter for analog telephone - connect withfxo or fxs to Asterisk
[EMAIL PROTECTED] wrote: Hi, I have Siemens combiset - it can gateway GSM phone to normal analog phone. It has output where I can connect regulat analog phone. That would be an FXS connection. How can I connect to combiset with Asterisk - via fxo or fxs ? You would need to connect the FXS port on your combiset to an FXO port on your Asterisk. Cheers, Jim -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.9 - Release Date: 06/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Toronto?
Anyone in the Toronto area interested in getting together to share notes and swap war stories? -- Jim Van Meggelen [EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.9 - Release Date: 06/01/2005 ___ 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] Toronto?
Well, that'd be quite a trip for a cup of coffee! But you'd be very welcome for sure! Jim. Wojciech Tryc wrote: well, I am in Ottawa...only 50mins by air :) Wojtek - Original Message - From: Leif Madsen [EMAIL PROTECTED] To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, January 08, 2005 12:37 PM Subject: Re: [Asterisk-Users] Toronto? On Sat, 8 Jan 2005 05:40:02 -0500, Jim Van Meggelen [EMAIL PROTECTED] wrote: Anyone in the Toronto area interested in getting together to share notes and swap war stories? I'm in Oakville, right across from Sheridan College. So I guess I can be considered part of the GTA at least. But you already knew that :) Leif Madsen. http://www.leifmadsen.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.9 - Release Date: 06/01/2005 ___ 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] TDM400P - Problems
César Davi Ávila do Nascimento wrote: Hi All I've bought a TDM400P and need some help with configuration. Can you tell me what to do ? I've tried to install and the message below has appeared: [EMAIL PROTECTED] asterisk]# modprobe zaptel [EMAIL PROTECTED] asterisk]# modprobe wcfxo /lib/modules/2.4.22-1.2115.nptl/misc/wcfxo.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.22-1.2115.nptl/misc/wcfxo.o: insmod /lib/modules/2.4.22-1.2115.nptl/misc/wcfxo.o failed /lib/modules/2.4.22-1.2115.nptl/misc/wcfxo.o: insmod wcfxo failed Try to modprobe for wcfxs or wctdm. Wcfxo is for the old card. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.8 - Release Date: 03/01/2005 ___ 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] Qs about FXO/FXS cards
[EMAIL PROTECTED] wrote: This has been an interesting discussion. I'll chime in with my experience here. I have two servers. One with the cheapest motherboard and athlon processor I could find on Newegg.com. The other is a 1999 era motherboard with a Via C3 processor, again a bargain basement special. The Athlon system has a decent power supply - 400+ watt, the Via has a very generic PS that came with the case - 300 watt tops. On both system I have TDM cards, the Athlon has a 4 port FXS and two x100p's, the Via has a 2 port FXS. Both systems are in production if you could call it that because they handle little traffic - home/hobby systems. I have had no problems at all with the tdm cards or Asterisk. I occassionally lose my network on the Athlon machine - I chalk that up to the fact that I'm currently sharing an IRQ with two ethernet cards and an X100P. I'm thinking of ditching the two x100p's in my Athlon machine for a TDM card with FXO modules to free up a slot and hopefully the burdened IRQ. Based on what I'm reading here I probably should think *really hard* about that. If I may, I'd like to ask you some general questions about the environment these systems are running in. - How are these systems powered and grounded? - Are the lines feeding the FXO cards coming from the PSTN, or are they being fed by a PBX or similar? (basically, how long is the loop between the card and whatever is feeding it?) You are successfully running systems that many would tell you to expect problems with. The TDM400 FXO modules are generally agreed to be an improvement over the X100P, so if you are having no troubles now, it is entirely plausible that migrating to TDM400-based FXOs will work for you as well. Unfortunately, there is no way of guaranteeing that, and it's your money, so I can't advise you much more than that. Frankly, what is most interesting is the fact that your systems are trouble-free. Certainly if you were to ask if such systems could be put into production, you would probably be advised not to expect much. There seem to be a lot of variables with these TDM400s. Cheers, Jim. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.7 - Release Date: 30/12/2004 ___ 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] Qs about FXO/FXS cards
Dorn Hetzel wrote: On Sat, Jan 01, 2005 at 07:23:58PM -0500, Jim Van Meggelen wrote: [...] What if, for example, the TDM400 issues were a cumulative thing? If you had over 6dB of attenuation on the PSTN loop, coupled with greater than 5V potential on the neutral-ground of your elecrical receptacle, compounded by a cheap power supply, exascerbated by a Via-chipset, would you not be virtually guaranteed some strange behaviour? But if your PSTN was -3dB, your electrical feed derived from a power conditioner, your power supply manufactured by PC Power Cooling, and a ServerWorks chipset-based MoBo, would your system always be faultless? Can you recommend any favorite motherboards? That is the million dollar question. Chipsets and MoBos seem to change so fast that I've lost confidence in my ability to make sense of it all. The Intel and ServerWorks chipsets are generally well regarded; Via chipsets are almost universally avoided for audio work. The linux-audio-dev folks seem willing to give nVidia's nForce chipsets a chance. In general, I would avoid PC-class motherboards, and go with server-class motherboards. That being said, the ultimate goal would be to find a way to build a reliable Asterisk system on *any* half-decent motherboard. Personally, I'm of the mind that power (the power supply, the AC being supplied to the system, and grounding) plays as much of a role as the motherboard does, but that is a working theory only. I wonder if clean power on a lousy MoBo might serve as well as dirty power on a quality MoBo. If one reads about power quality issues, the symptoms of dirty power sound suspiciously similar to the kinds of problems people are having with their analog Asterisk cards. I'm also wondering about the TDM400s ability to handle PSTN loops at the extreme limits. Since those TDM cards were probably developed largely in a lab environment, the telco lines would have been simulated with a channel bank or C.O. simulator. What happens when the lines fall out of certain limits? Annenuation, loop current, and longitudinal imbalance are all factors that proprietary PBXs are able to correct within fairly wide limits - but they do have limits (a Norstar, for example, tends to have trouble pulling dial tone when attenuation exceeds 7dB). Has the TDM400/FXO been similarly optimized? It must have limits; what are they? I think what we are all looking for is some empirical evidence of what conditions cause the biggest problems. Is it the TDM400 that is to blame (either hardware or drivers)? Or is it Linux? PC Hardware? Telco Lines? Electrical? Grounding? A combination of some or all of those factors? No one seems to know for certain. Where much frustration comes from is the fact that a typical PBX simply does not suffer from these troubles. We've come to expect that our telecom equipment handles these little noises for us (so much so that we're suprised to find that these are genuine engineering issues). With Asterisk, some of the responsibility for correcting those problems falls to us, the system designers. Unfortunately, a comprehensive engineering methodology for analog devices on Asterisk does not yet exist. It's all kind of hit-and-miss. Cheers, Jim. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.7 - Release Date: 30/12/2004 ___ 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] Qs about FXO/FXS cards
Truman Beal wrote: One thing to look at is the proximedy to the powersupply of your audio devices. Some mobos have their chipset integrated in very closely to their power supply pins. With an unclean power source the fluxuations would be enough to add some of the white noise which would give you the whine. Excellent examples of that are on the really small all in ones. Sometimes power source it self may be in quesiton, and a cheap ups with nothing else on it would probably solve that one quickly. I have to warn people about cheap UPS units. They have to be a *POWER-CONDITIONED* UPS to gain the power quality benefits. Many *cheap*UPS*units*do*not*clean*up*the*power!* Some even make it worse. If it doesn't say power conditioned, it isn't; it's nothing more than a battery back up (you will not like what these cheap UPS units do to your input power when running on batteries). For eample, APC Smart-UPS models provide power conditioning. ABC Back-UPS do NOT. Run a Back-UPS on battery only and check out the harmonics on the output - ouch! It's AC, but it is NOT a sine wave - it's a square (some of them deliver a stepped square or pseudo-sinusoidal waveform). Not clean at all. Just remember that power conditioners and UPSs are not the same thing. You can buy stand-alone power conditioners (PowerVar and OneAC make these), stand-alone UPSs (many cheaper UPS units are this type), or UPSs with built in power conditioners (PowerVar, OneAC, and APC). Know what kind of UPS you are buying. If you see them trying to tell you something about any kind of protection, without actually saying the magic words power conditioned, you cannot be sure that your UPS is giving you clean power. Addionally, you may want to see if your power supply offers any sort of shielding. Mylar can offer a small amount of shielding, but some power supplys won't cool effectively, and the results of that are similar to fireworks- (see tom's hardware guide to power supplies)- Spend the extra money and get PC Power Cooling (no, they don't pay me, they are simply the best - no contest). Alternatively, you find some cases offer no shielding to the outside with their plastic constructs. A good quality case may be in order, and if you are racked, you may want to make sure one of the other machines in the rack may not be grounding out onto the rack, thus causing additional headaches. Noise could be getting in from all over, eh? For my setup, I'm not using the onboard stuff in favor of an old sb live. What's the SB live for? Console? I't moved of to a slot farily away from the digium board , and too close for my comfort to the networking cards. I gain fairly good quality, and not enough white noise for me to really pick it up. In my house, there are a few spots where power isn't the cleanest, so I'm certain that I'm getting at least some noise, despite every machine is fed from their own ups :) [here he goes again . . .] Are they power-conditioned UPSs? Hope that helps- T Jim Van Meggelen wrote: Dorn Hetzel wrote: On Sat, Jan 01, 2005 at 07:23:58PM -0500, Jim Van Meggelen wrote: [...] What if, for example, the TDM400 issues were a cumulative thing? If you had over 6dB of attenuation on the PSTN loop, coupled with greater than 5V potential on the neutral-ground of your elecrical receptacle, compounded by a cheap power supply, exascerbated by a Via-chipset, would you not be virtually guaranteed some strange behaviour? But if your PSTN was -3dB, your electrical feed derived from a power conditioner, your power supply manufactured by PC Power Cooling, and a ServerWorks chipset-based MoBo, would your system always be faultless? Can you recommend any favorite motherboards? That is the million dollar question. Chipsets and MoBos seem to change so fast that I've lost confidence in my ability to make sense of it all. The Intel and ServerWorks chipsets are generally well regarded; Via chipsets are almost universally avoided for audio work. The linux-audio-dev folks seem willing to give nVidia's nForce chipsets a chance. In general, I would avoid PC-class motherboards, and go with server-class motherboards. That being said, the ultimate goal would be to find a way to build a reliable Asterisk system on *any* half-decent motherboard. Personally, I'm of the mind that power (the power supply, the AC being supplied to the system, and grounding) plays as much of a role as the motherboard does, but that is a working theory only. I wonder if clean power on a lousy MoBo might serve as well as dirty power on a quality MoBo. If one reads about power quality issues, the symptoms of dirty power sound suspiciously similar to the kinds of problems people are having with their analog Asterisk cards. I'm also wondering about the TDM400s ability to handle PSTN loops at the extreme limits. Since those TDM cards were probably developed largely in a lab environment, the telco
RE: [Asterisk-Users] Configuration details for Asterisk interactionwith Vocal
Title: Message
Please put
this in the Wiki (www.voip-info.org). It
is valuable information and will reach a wider audience
there.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]Sent: January 3, 2005 1:52
AMTo: asterisk-users@lists.digium.comSubject:
[Asterisk-Users] Configuration details for Asterisk interactionwith
Vocal
I have seen a number of people in this newsgroup asking for information
regarding asterisk interworking with Vocal. I was able to configure Vocal and
Asterisk so that calls originating from vocal can land on an extension in
Asterisk. I would like to share this info with the group
The scenario that I tested was as follows.
A call was originated from extn. 1001 on Vocal and the call was made to
land on extension 12456 hanging from an asterisk pbx registered with vocal.
Vocal was configured such that, if the dialed digits are 8500, the call will
be routed to asterisk to which will handle the call on extension 12346 hanging off asterisk
Configuration on Vocals side
All calls made to 8500 are forwarded to asterisk. This is achieved by
adding the following entry to the
dial plan of Vocal using the provgui as mentioned below:
Add a new entry in the dial plan as shown below
Key
-
^sip:8500
Contact
-
^sip:8500@ip-address:port
ipaddress:port is where the asterisk pbx is listening for sip
messages.
Also make a ummy entry for extension 8500 in Vocal (so that vocal
thinks this is an extn. Connected to vocal).
Configuration Changes in
asterisk
---
A register entry as shown below is added to the [general] section of
the sip.conf file in the asterisk
server
register =8500:[EMAIL PROTECTED]/12346
This would register the extension 8500 with the vocal server using the
vocal tag. All calls received from vocal would terminate in extension
1246.
The following is the information contained in the vocal tag.
[vocal]
type=friend
; either "friend" (peer+user), "peer" or "user"
callerid=Test 1 12346
host=10.117.4.236
; we have a static but private IP address
port=5065
This indicates that the calls will be received from the Vocal server
running on host 10.117.4.236 on port 5065 (port where Marshall server is
running).
The following details are included in the extensions.conf file so that
calls originating from vocal can be answered by the extension
12346.
exten = 12346,1,NoOp(.call for .${EXTEN})
exten = 12346,2,Dial(SIP/${EXTEN},60,tr)
exten = 12346,3,Congestion
The following is the configuration information for the extension 12346
(Xlite UA client running on Windows) in the sip.conf file.
[12346] ; X-Lite client 12346
type=friend
secret=blah
auth=md5
nat=no ; we assume clients are not behind NAT
host=dynamic ; and have dynamic IP addresses
reinvite=no ; if so, we need to make them
canreinvite=no ; always go through Asterisk
qualify=1000
dtmfmode=inband
callerid="Test 1" 12346
disallow=all
allow=gsm ; add whatever other codecs we fancy
context=test1 ; use a context that exists ;-)
A corresponding entry for the context test1 is required in
extensions.conf to complete configuration of extension 12346
[test1]
exten = _[123456789],1,NoOp(.call for ..${EXTEN})
exten = _[123456789],2,Dial(SIP/${EXTEN},60,tr)
exten = _[123456789],3,Congestion
This should be suffecient for you to land all calls originating from
Vocal with dialed digits 8500 to land on extn 12346 in asterisk.
Tip:
To ensure that asterisk has correctly registered with vocal just run
the show sip registry from the asterisk console to show all the peers in the
network
Confidentiality Notice
The information contained in this electronic message and any
attachments to this message are intendedfor the exclusive use of the
addressee(s) and may contain confidential or privileged information.
Ifyou are not the intended recipient, please notify the sender at
Wipro or [EMAIL PROTECTED] immediatelyand destroy all copies of
this message and any attachments.
--No virus found in this incoming message.Checked by
AVG Anti-Virus.Version: 7.0.296 / Virus Database: 265.6.7 - Release Date:
30/12/2004
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Checked by AVG Anti-Virus.
Version: 7.0.296 / Virus Database: 265.6.7 - Release Date: 30/12/2004
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RE: [Asterisk-Users] Qs about FXO/FXS cards
[EMAIL PROTECTED] wrote: On Sat, 2005-01-01 at 16:14 -0700, Michael Welter wrote: Since you asked, and since I'm well into this bottle of Merlot on New Year's day: 1. Power alarms. WTF does that mean? Wish I had some support docs. 2. On bootup, Excessive leakage module x, ProSLIC failed Auto Configuration. Again, WTF? Reboot and it's ok. But, just a reboot after driving 100+ miles to the client site is not a good option. 3. On bootup, a LED won't light. When zapata gets to it, it can't find the channel. Usually means a complete power cycle to get it to work. Those first 3 all sound like you have a problem with power supply and consistency. You don't mention what modules you have in the cards, but I bet you have FXS ports and have too light of a power supply for the job. Oddly, the maximum power requirements of the TDM400 (fully loaded with 4 FXS modules) is 20W. That'd have to be a pretty weak power supply (or heavily loaded chassis) to have problems drawing that power. Still, I agree that the power supply is a suspect. I'd want to know who makes the power supply, which model it is, and whether that model has a good reputation. An electrically noisy power supply could cause the kinds of anomalies described. So could a faulty supply, of course. More important to my mind is the overall quality of the power feeding the system. Is a dedicated electrical circuit employed? Isolated, insulated grounding conductor right back to a separately-derived source? Power conditioner? So many of the problems people are having with the TDM cards sound like power-quality issues, one has to wonder. I don't mean that as a panacea, because the TDM400 troubles seem to go beyond any one issue. It's merely one thing that might bear looking into. It'd be nice to see some statistics on not only what percentage of TDM400 users are having problems, but also what kind of environment they're in. I'd want to know about the elctrical environment, manufacturer and model of each system component (power supply and motherboard especially). I'd also like to get a report from a circuit analysis performed on the PSTN loop. I realize that much of this would be impossible to get, but one of the most important steps towards solving a bug is being able to identify the conditions which cause it. So far that data is not known, which is a large part of the reason the problem is not getting fixed - no one knows exactly what is causing the troubles - we just have symptoms. What if, for example, the TDM400 issues were a cumulative thing? If you had over 6dB of attenuation on the PSTN loop, coupled with greater than 5V potential on the neutral-ground of your elecrical receptacle, compounded by a cheap power supply, exascerbated by a Via-chipset, would you not be virtually guaranteed some strange behaviour? But if your PSTN was -3dB, your electrical feed derived from a power conditioner, your power supply manufactured by PC Power Cooling, and a ServerWorks chipset-based MoBo, would your system always be faultless? With enough data, we could really start to hone in on this animal. 4. A TDM card that isn't recognized at all. DOA. 5. Impedience matching to eliminate humm? I'm calling Matt on Monday, and hopefully he'll RMA these cards. I hope that everyone that has a life is out enjoying the New Year. http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.7 - Release Date: 30/12/2004 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.7 - Release Date: 30/12/2004 ___ 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] IAXy reliability issues
[EMAIL PROTECTED] wrote: Paul Fielding wrote: Hmmm I could certainly see that being the issue. If it is the issue, though, then I think it's something that needs to be addressed. In my opinion, Digium needs to address it, as well as the whole provisioning via cli thing. I know Asterisk itself is a CLI oriented piece of software, but the more one can do do decrease configuration timing and issues the better off one is. I think it would be a benefit to allow the IAXy to be programmed via web interface. For that matter, from what I can tell via my own experimentation, it appears that you cannot use DNS to define the asterisk server to it. This is bad, since it means that if the IP of the asterisk server changes, you need to directly reprovision *all* of your IAXy devices For a new product, it has potential, hopefully these things will be addressed The IAXy does not have the CPU, RAM, or Flash to be able to add any significant features. I think it has 4k or RAM and 4k of Flash. Well, that certainly limits it's useful future. A neat toy, with limited market potential. I'd certainly like to hear about it's successor, then, because any kind of IAX-based ATA is something that would seem to have a future with Asterisk. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.6 - Release Date: 28/12/2004 ___ 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] zapata.conf not being parsed by *
You need two files to make the zapata stuff work: /etc/zaptel.conf /etc/asterisk/zapata.conf The first one (/etc/zaptel.conf), configures the Linux driver for the hardware. In theory, you could have another application other than Asterisk using the zaptel driver. The second one (/etc/asterisk/zapata.conf) contains the information Asterisk needs in order to use the zaptel driver. Asterisk uses /etc/asterisk/zapata.conf Linux uses /etc/zaptel.conf Also, did you run modprobe and ztcfg? The zaptel driver won't light up until you give it the spark. [EMAIL PROTECTED] wrote: I am running * 1.0.3 for some reason when I start * is does not appear to be parsing my zapata.conf file. I do not see any errors * just does not seem to know to look for zapata.conf. I am unable to use my FXO card to make calls or receive calls. I have been able to configure SIP to work correctly. Any help would be greatly appreciated, I spent most of last night searching for an answer. Thanks Jerry ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.6 - Release Date: 28/12/2004 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.6 - Release Date: 28/12/2004 ___ 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] IP Phone recommendations?
[EMAIL PROTECTED] wrote: Hey gang, I'm looking at escaping from a Nortel Meridian CISC system to Asterisk/Digium/SIP phones. I'm currently in the testing and proof of concept phase. I'm going to need a SIP phone and don't want to re-purchase and have orphans around. I've got a few different IP phones in my lab (including a C7960), I'm currently loving my Polycom 300 - a solid phone for the price, and everyone says the 500 and 600 are even better. I think I'll be going with Polycom for my customers until the next best thing comes along. I looked at the Snom phones when I was at Astricon, and while they may be technically great, the problem I had with them is that they are not weighted properly. If you've ever yanked your phone off the desk you'll understand the need for a proper ballast. The handsets feel cheap too, because they're too lightweight. Still, from everything I've read you'll certainly want to try one out. Also, the Snom 220 seems to be the best bet as a reception phone, especially if you want a busy lamp field on your swithboard. The Cisco phones are great, but it's hard to stomach paying an extra $100-$300 for that little drawing of the Golden Gate Bridge they put on all their products. One of the exciting things about standards-based telephony is that you can mix and match your phones. It's the same as analog sets; the agony is in the sheer number of choices available. We currently run Nortel 7310 phones and they work great. I'm sort of overwhelmed by all of the different IP phones. I was hoping some folks would share what they have found. My primary goal is to replicate the 7310's features and to allow room for growth in the future with telephony applications. One of the big differences between the Norstar and the Asterisk is that the Norstar is a key system, the Asterisk is a PBX. If you completely replace the Norstar your users will will no longer have access to line status on their phones; that is all handled behind the scenes. Also, you will not get busy lamp field, which means you won't be able to monitor who is on the phone (there are ways of doing this in Asterisk, but it's not as intuitive to implement). Finally, the Norstar has hundreds of easy to use features; each one you'll want to keep will need to be carefully hand-crafted in the dial plan. Our primary driver is configurability and features that we can get in Asterisk, that we can get without a lot of money from Nortel. Nortel sure has fallen behind. Even the VoIP stuff they have does not work well, and is barely standards-compliant (if at all). Namely- Voicemail, telecommuting workers on the pbx, better call handling, better automation. I'd like to be able to integrate smart features like directory and call handling to the handset, but I'll freely admit I'm just starting out. My initial goal is to just to get onto Asterisk and get it working. I'll worry about cool stuff later. I think you'll be wise to leave the Nortel KSU in place for a bit. That way you can introduce new features to the users without them also having to learn new phones. There are challenges either way. Our integration and migration plan is as follows: If anyone has some suggestions or pointers I'd love to hear them. 1. Test and evaluate Asterisk with TDM400 with 1 FXO/FSO port each. 2. Configure Asterisk to be the primary PBX and slave the Nortel Meridian system to it using a second TDM400. This avoids immediate replacement of all handsets. Will allow immediate access to features such as Voicemail. 3. Overtime, upgrade desk phones to IP phones. When all phones are replaced, decommission Nortel and sell on Ebay. :) Are you using calling line ID? The problem here is that you have two systems that will each need to wait two rings before answering. The Asterisk will need two rings to get the caller ID, and then it'll take two more to pass the same CLID on to the Norstar. [PSTN]==(2 rings for CLID)==[Asterisk]==(2 rings for CLID)==[Nor*] Make sure you put an autoattendant in the middle, to ensure your callers don't have to wait too many rings before some indication that there's a system at the other end. Also, there is some danger of echo if you put the Asterisk in the middle. You'll want to be patient with this, as it may take a bit of tweaking to sort out. IMPORTANT: Make sure your Asterisk and Nortel are grounded to the same point. Best way to achieve this easily will be to plug them into the same electrical outlet. You do NOT want voltage potentials on the analog loop between the * and Nor*, believe me. The fact is, analog is a technology that really doesn't lend itself well to integration. It can be made to work, but callers and users will have to deal with a lot of extra rings. Also, transfers and the like will involve hookswitch flashes and such. I'm not saying avoid it, just be aware of the need to manage user expectations. One possible way to handle this would be to configure the system so
RE: [Asterisk-Users] Ouch, part reset, quickly
[EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Saturday, 4 December 2004 1:44 AM To: Ed Greenberg; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Ouch, part reset, quickly On Fri, 2004-12-03 at 09:09 -0800, Ed Greenberg wrote: I have exactly this problem. When it happens, I lose access to some FXS ports and get Geiger counter style clicking on the FXOs. I just opened a ticket with Digium on the subject, but given what I just read, perhaps I should not have high hopes. Is it possible that your PSU isn't up to the task? If you aren't running a 400 or 500 watt PSU, I would be suspect of the PSU. That error message was attributed to not getting enough power before they put a power plug on the board itself. Now you know you aren't getting strangled by the PCI bus, but it still might not be enough power if you PSU isn't up to snuff to hold the power stable and high enough. -- Steven Critchfield [EMAIL PROTECTED] Interestingly enough, the data sheet does not give any info. on what are power requirements for this board. Does anyone have this data? Regards Garry Taylor I requested this information from Digium and was provided with the following: Max Power Draw per TDM400 (Watts): 20 2 Watts per FXO 5 Watts per FXS They also told me each FXS port would support a REN (ringer equivalence number) of 5.0, which means that you should be able to to ring five of the old electromechanical telephones simultaneously off of each FXS port on the card - 20 of them in all. Does anybody know what the wattage is for a REN? Cheers, Jim. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.2 - Release Date: 20/12/2004 ___ 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] Ouch, part reset, quickly
[EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Saturday, 4 December 2004 1:44 AM To: Ed Greenberg; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Ouch, part reset, quickly On Fri, 2004-12-03 at 09:09 -0800, Ed Greenberg wrote: I have exactly this problem. When it happens, I lose access to some FXS ports and get Geiger counter style clicking on the FXOs. I just opened a ticket with Digium on the subject, but given what I just read, perhaps I should not have high hopes. Is it possible that your PSU isn't up to the task? If you aren't running a 400 or 500 watt PSU, I would be suspect of the PSU. That error message was attributed to not getting enough power before they put a power plug on the board itself. Now you know you aren't getting strangled by the PCI bus, but it still might not be enough power if you PSU isn't up to snuff to hold the power stable and high enough. -- Steven Critchfield [EMAIL PROTECTED] Interestingly enough, the data sheet does not give any info. on what are power requirements for this board. Does anyone have this data? Regards Garry Taylor I requested this information from Digium and was provided with the following: Max Power Draw per TDM400 (Watts): 20 2 Watts per FXO 5 Watts per FXS They also told me each FXS port would support a REN (ringer equivalence number) of 5.0, which means that you should be able to to ring five of the old electromechanical telephones simultaneously off of each FXS port on the card - 20 of them in all. Does anybody know what the wattage is for a REN? Here's some technical information about REN. Not sure how useful it is to the discussion, or whether it answers my question. http://www.powerdsine.com/Support/FAQ/#ringers_faq -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.2 - Release Date: 20/12/2004 ___ 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] Ouch, part reset, quickly
[EMAIL PROTECTED] wrote: On December 21, 2004 08:29 am, Jim Van Meggelen wrote: 5 Watts per FXS They also told me each FXS port would support a REN (ringer equivalence number) of 5.0, which means that you should be able to to ring five of the old electromechanical telephones simultaneously off of each FXS port on the card - 20 of them in all. Does anybody know what the wattage is for a REN? Well given what you just provided, I'd say the wattage for a REN is 1 Watt, minus whatever the card itself uses which would be minimal. :-) That was my thinking exactly, except that I was concerned that 20W just didn't seem like enough oomph to drive twenty of those old mechanical ringers, and I wanted to determine if the official value for a REN actually _was_ 1Watt. Part of me says to drop it, and part of me says that since these TDM cards are causing so much trouble (with no one figuring out what's wrong), every crazy idea is worth at least a look. Cheers, Jim. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.2 - Release Date: 20/12/2004 ___ 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] Ouch, part reset, quickly
[EMAIL PROTECTED] wrote: On December 21, 2004 08:29 am, Jim Van Meggelen wrote: 5 Watts per FXS They also told me each FXS port would support a REN (ringer equivalence number) of 5.0, which means that you should be able to to ring five of the old electromechanical telephones simultaneously off of each FXS port on the card - 20 of them in all. Does anybody know what the wattage is for a REN? Well given what you just provided, I'd say the wattage for a REN is 1 Watt, minus whatever the card itself uses which would be minimal. :-) REN really only applies to the old style ringers. The newer electronic ones are basically sensors only, and as such, are wy lower then that. I know it. But if an FXO says it can provide 5 REN, then it has to be able to handle 5 of those 2500 sets, yes? And since the TDM400 has 4 ports, that means that you should be able to drive 20 electromechanical ringers simultaneously off one card, yes? I know, I'm a sadist - who would really want to do that? But if it says REN 5 _per_FXS_port_, then I have this burning desire to know (yes, Rich, I'm the kid who stuck my finger in the electrical socket because I just _had_ to know -- did it twice, as a matter of fact). And then I contemplate getting twenty 2500 sets (what's that, about 400lbs?) and connecting them up to my precious lab system, sending ringing to all of 'em, and either smelling smoke, or causing everyone to think the fire alarm is going off! And . . . success or failure, what exactly have I achieved? It's all academic, and slightly masochistic . . . -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.2 - Release Date: 20/12/2004 ___ 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] TDM120 card?
[EMAIL PROTECTED] wrote: hi any chance of making asterisk support these? http://voipstore.atacomm.com/shops/ViewItem.aspx/27934028032-3835624908 8.htm According to the manufacturer, they already do: http://www.ipvolution.com/ Cheers, Jim. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.0 - Release Date: 17/12/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to increase the performance?
[EMAIL PROTECTED] wrote: Rich Adamson schrieb: In the past I had problems with the audio over sip. Then I tried the -p Option and increased the memory. Now it is better but not perfect. Are there any more possibilities to increase it more? By now I'm using a P-II/333. Could a completely hand optimized kernel (I use 2.6.) help a bit? There's no way to answer your question with any degree of reasonable truth as you haven't mentioned they type of phones, type of pstn interface, which codecs, etc, etc. Okay. My server has got: - One Phonejack Lite - One X100P Clone - 256mb Memory - P II/333 - Linux 2.6.5 - Debian Woody - Asterisk 1.0.1 - Codecs: GSM, ulaw, alaw - ADSL 1000kBit/s Downstream, 128kBit/s Upstream That upstream bandwitch will need to be managed carefully. If you're using G.711, one channel would be using roughly 80kbit of your upstream. Who has the most quality complaints: you, or the people you are talking to? Calls from or to the pstn are completely okay. Calls over SIP aren't. Calls over IAX couldn't be tested at the moment. Can you make a SIP connection directly to the box? No LAN, no WAN, just a crossover cable between your SIP phone (soft or hard) and your Asterisk system? That'll give us some idea of whether the problem is network or server-based. Jim. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.0 - Release Date: 17/12/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: Meetme with video???
[EMAIL PROTECTED] wrote: Hi Noah, I have been contacted by 2 people but nothing so far. If you want to add $500 please email me your details and I'll add it to the wiki to co-ordinate this. I agree I'm really surprised why no one has shown more of an interest in video calls on asterisk yet. It is certainly worthy of discussion, but perhaps not really that surprising; video has been the next big thing in telephony since the 50s. I think the price of entry has always scared folks off, not to mention that the kind of bandwidth and horsepower required by video makes audio seem a piece of cake. But I think the real barrier with video is the cultural environment. We have certain expectations when we're seeing video: years of television has trained us to expect a certain level of production. This means that video connections come with certain cultural expectations that audio does not. Please understand that I'm not saying video is bad, or useless, or unimportant, or whatever. I'm just observing the fact that the industry has been promising that video will be the standard in communication for over fifty years, and people still love their telephones. Consider: I could have a business conversation with you right now over the telephone, but as I'm still in my underwear, a video conversation would not be fun for you at all! The interest is there, but perhaps everyone's got enough to do with audio to get into worrying about video just yet. It's not that no one wants it, it's more a matter of priorities. Again, I'm not trying to put a value judgement on it, merely to speculate on why the interest is not as high as one might expect. Regards, Jim. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller Sent: Friday, December 17, 2004 9:05 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RE: Meetme with video??? I wonder if there is an application available, what would allow me to have a conference call (meetme) with video. Nope, AFAIK there's nothing yet. There is a bounty of $2000 for this functionality: http://www.voip-info.org/tiki-index.php?page=Asterisk+bounty+Meet+Me+vid eo+conferencing You can add to this bounty, if you want. I'm trying to convince the money people at my company that we should add $500 to this. BTW: Is anybody working on this? -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.0 - Release Date: 17/12/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to increase the performance?
[EMAIL PROTECTED] wrote: Jim Van Meggelen schrieb: [EMAIL PROTECTED] wrote: - ADSL 1000kBit/s Downstream, 128kBit/s Upstream That upstream bandwitch will need to be managed carefully. I know. If you're using G.711, one channel would be using roughly 80kbit of your upstream. Who has the most quality complaints: you, or the people you are talking to? I don't know. I guess its equal. Since the problems occur even when I'm using GSM the bandwith shouldn't be the problem. Not as much, but if you have any other outbound traffic over that connection, it WILL still be a factor. Have you got a firewall/router that provides any QoS features? Calls from or to the pstn are completely okay. Calls over SIP aren't. Calls over IAX couldn't be tested at the moment. Can you make a SIP connection directly to the box? No LAN, no WAN, just a crossover cable between your SIP phone (soft or hard) and your Asterisk system? No. My phone is connected via the Phonejack in the server that runs asterisk. But ... I could try to setup a software client that I connect to the server. That might help to determine if the problem is in the Asterisk box or the network. If you can make a local SIP client talk to your Phonejack cleanly, and yet have problems with network connection, that could be a factor. Also, I recommend getting an account on IAXtel or FWD, and connecting a softphone directly to those networks, and then into your system. That'll stress your outgoing bandwidth, and potentially provide all kinds of interesting information about what works and doesn't. Cheers, Jim. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.0 - Release Date: 17/12/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hardware based DSP
[EMAIL PROTECTED] wrote: Thank you all for your explanations related to my question. I have one follow-up question though. When I said that all dsp related stuff has to be handled by software within asterisk, I was thinking of conferencing at the time. I mean in order to be able to conference a sip session with a PSTN call, it would have to be handled by software, even if both the channels had hardware dsp capabilities. Right ?? Hmmm. Ultimately, yes, I suppose there's always going to be some of that. Where the DSP work gets really intensive is when what is going into the DSP is quite different from what's coing out. If a hardware device existed for Asterisk to transcode all channels into a common format, so that internally the same codec was being used, then the CPU would have relatively little DSP work to do when connecting them together. I don't know if this would be a problem in other areas, so what I'm saying is more of a brainstorm than anything I think needs to happen. If you are dealing with just a single channel, then the driver may handle codec/echo cancelation stuff with hardware help (??) That sounds pretty much correct. I know there's been some talk about using the powerful floating-point capabilities of 3d video cards to do transcoding; other ideas are in the works as well. As an aside, what is the best way to go about learnig about the aritecture of asterisk, other than using the source ?? There simply isn't enough documentation in that regard. At least not that anyone's found in a single repository . . . other than the wiki, of course. The Wiki pages are great, but I have not (yet) found any info about asterisks architecture itself (in depth that is). Nor has anyone, to my knowledge (other than reading the source). Some linked websites / blogs provide good info on some topics, but is there a good high level design doc available anywhere ? The Asterisk Documentation Project has that very thing as one of it's goals. It's a big job, and there are few of us, so it's not happening as fast as everyone would like. There is so much to write, and so little time . . . It would stop people like me asking so many basic questions in the -dev list. There's no doubt that more documentation is needed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Old posts and the ability to search...
[EMAIL PROTECTED] wrote: Gents, Just a passing thought... is there any reason why the ability to search the past posts on here isn't switched on? Just wondered, since it makes much more sense to be able to search the old archives if you have a problem, rather than ask the same question again and again... Use Google and type in site:lists.digium.com [whatever you are looking for] Just replace the part in the brackets with whatever you wanted to search for. If I wanted to search for zapateller, I'd type in site:lists.digium.com zapateller Google would then restrict its search to lists.digium.com. You can do the same for the wiki: site:www.voip-info.org [search term(s)] Works with any site, actually. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Total newbie here looking to do a VoIP conferencecall?
[EMAIL PROTECTED] wrote: I am looking to help out my company find a more budget conscious but reliable way to hold conference calls between 5+ people. 4x a month we hold several hour long conference calls during non-business hours. All of the employees have high speed internet. Currently we dial up an ATT conf using regular analog phones. And pay handsomely for the privilege, no doubt. I don't have a great grasp as to what Asterick is capable of, but my thoughts were that perhaps with VoIP telephone lines (either hooked up to the company's network or just using a 3rd party VoIP provider such as Packet8, which is whatI have for personal use) and an Asterick server, that we could setup a VoIP conference bridge. Asterisk can certainly do this for you. Can someone enlighten an unknowledged as to whether or not this is possible, and if so, how might it be done? Would the Asterick server need X number of VoIP lines? I.e. If there's 10 participants, it'd need 10 VoIP lines? Asterisk is a complicated animal, and to walk you through it could take days/weeks/months. This is probably the route you need to take: 1) Make sure you are familiar with Linux administration. That knowledge, while not exactly essential, will certainly save you a boatload of confusion and misery. 2) Read about Asterisk. Look at: -The Digium handbook --(http://www.digium.com/handbook-draft.pdf) -The Documentation Project --(http://www.asteriskdocs.org) -The Wiki --(http://www.voip-info.org/wiki-Asterisk) 3) Set up a Linux server and play. Specifically, work towards getting a conference going. Along the way you will learn what you need to know to make a decision. 4) Evaluate what you've learned, and make a decision as to whether this makes sense for you. Asterisk can do what you want, but it has a steep learning curve - it's more of a toolkit of telephony functions than anything. Picture walking into Home Depot and asking what can I build?. The answer is just about anything, but in reality, it depends on budget, experience, tools, resources, regulations, and so on. Welcome to Asterisk! Be careful, or you'll get addicted! Cheers, Jim. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Total newbie here looking to do a VoIP conference call?
[EMAIL PROTECTED] wrote: Sorry for the misspelling... Thanks for the replies. I will set it up and start playing. This is all very exciting. I've been using VoIP as my primary phone but this is going a bit further. At the office we have a T1 that is probably fairly dead after hours. Supporting 5-10 users should be fine I'd imagine. I've read 1 VoIP connection uses about 64kbps or 8KB/s? That's a tough one. The bandwidth of a VoIP connection is a combination of the bandwidth used by the codec itself, plus the overhead. In some cases, encapsulation can mean that each VoIP packet is as bad as 50% overhead or more. Generally the overhead will be between 10Kbps and 16Kbps (although I might be off by a few K - these are ballpark figures). The codecs range from 5.3Kbps all the way up to 64Kbps. So a VoIP connection could use between 15K and 80K per channel. I believe IAX can be more efficient, as it eliminates a lot of overhead by combining multiple channels into trunks. It is a very interesting VoIP protocol. So... Asterisk can act as a SIP server... Yeah. It's probably best decribed as a gateway, although it can be just about anything you need it to be. For heavy, pure SIP work you'd want to look at SER (SIP Express Router), and use Asterisk as a PSTN/non-SIP gateway, or an application server (voicemail and such). My packet8 dta310 adapter has the SIP server hardcoded into it. If I could change that, I could use that? If it's SIP, it can talk to Asterisk. But since it can't be modified, we'd have to purchase SIP adapters for each employee... Something like this? http://store.sipphonestore.com/ I guess we need one where I can dynamically define the SIP server. If your existing phone system has a T1, you might consider putting the Asterisk in front of it, like this: [PSTN]---[Asterisk]---[PBX] That'd allow you to keep your existing phones running and gradually migrate your users to Asterisk. The Wiki is full of fun stuff. Read, experiment, and hang out with us here. Cheers, Jim. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Calculating required bandwidth
[EMAIL PROTECTED] wrote: --On Thursday, December 16, 2004 3:59 PM -0500 Jim Van Meggelen [EMAIL PROTECTED] wrote: I've always found Newton's Telecom Dictionary to be a great reference. It's not too technical, packed with humour, and very comprehensive. I have a very old copy of this, so went off to Amazon to see about a new one. I discovered that that a 2005 edition (21st edition) will be available in February, so I'm going to wait. That reminds me of last January, when I said to myself The 2004 version is coing out in a few weeks, I'll wait. I just picked mine up last week :-) Regards, Jim. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How expensive are the different codecs?(Regarding CPU time)
[EMAIL PROTECTED] wrote: Antony Stone schrieb: On Wednesday 15 December 2004 21:26, Michael Vogel wrote: Is it a little bit too much for such a machine? What could be the bottleneck? CPU? Memory? Interrupts? My advice would be to whack in a load more RAM - basically, try to get the poor little thing so it doesn't need to use swap. That will make a big difference to performance. I just doubled the memory, now I have 256mb and I am using - by now - zero bytes for swap ;-) The values at show translation doesn't change. And they change only a little bit when I unload the baycom_ser_hdx-module that generates three times more interrupts than the wcfxo-module. Hmmm. I propose that we make your system the list guinea pig! If we can get that one tweaked, there's no telling what alse we can do with Asterisk! Have you tried running Asterisk at pseudo-realtime priority? (asterisk -p) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How expensive are thedifferent codecs?(Regarding CPU time)
[EMAIL PROTECTED] wrote: Jim Van Meggelen schrieb: Hmmm. I propose that we make your system the list guinea pig! If we can get that one tweaked, there's no telling what alse we can do with Asterisk! I'm not really sure that I completely get the meaning of this sentence. (Which can be because of the fact that its 06:47 AM or that I'm no native speaker or both ;-)) But I guess you meant that we could try to test everything that is known to work - including some voodo - and to see if it works? ;-) You understand. I was kidding a little bit, but yes, I am also wondering just what things can be done to get a slower machine to work as well as possible. Have you tried running Asterisk at pseudo-realtime priority? (asterisk -p) That helps in one way: At the moment my system is doing its morning routine. That means it makes a tar archieve of my /home directory to my backup drive. With the -p option the show translation-values are equal to the values when my system is idle. And otherwise not? I guess this option could help me a lot regarding the sound problems I got sometimes. Yes, it might help a lot. Cheers, Jim. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How expensiveare thedifferent codecs?(Regarding CPU time)
[EMAIL PROTECTED] wrote: Jim Van Meggelen schrieb: You understand. I was kidding a little bit, but yes, I am also wondering just what things can be done to get a slower machine to work as well as possible. Okay. So lets try. What are you running in terms of a kernel or distro? Have you tried running Asterisk at pseudo-realtime priority? (asterisk -p) That helps in one way: At the moment my system is doing its morning routine. That means it makes a tar archieve of my /home directory to my backup drive. With the -p option the show translation-values are equal to the values when my system is idle. And otherwise not? Otherwise - without the -p option - the system had values of 400ms (and higher) converting speex when it wasn't idle. Now the value is constantly at about 210. Nice. The system is now giving Asterisk the priority it needs. Don't forget to change that in your rc.local, or wherever you're starting Asterisk from. I guess this option could help me a lot regarding the sound problems I got sometimes. Yes, it might help a lot. I will see when doing some calls over sip (there I had the most problems). Maybe at the evening. Now I have to breakfast, shower and go to work. And I need to go to bed! Cheers, Jim. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Calculating required bandwidth
[EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: I was posed this question: A T1 set up for voice carries 24 conversations on a circuit that is 1.544 megabits/second. Right? Yes and no. If the T1 is channelized, then yes. If it's a PRI circuit, then it has only 23 channels to carry voice, as the 24th channel is used for the D-channel (signalling channel). Only if you're in the US. We have 30 + 1 :-) Nope. I'm in Canada. And what you are referring to is an E1, not a T1. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Calculating required bandwidth
[EMAIL PROTECTED] wrote: I was posed this question: A T1 set up for voice carries 24 conversations on a circuit that is 1.544 megabits/second. Right? Yes and no. If the T1 is channelized, then yes. If it's a PRI circuit, then it has only 23 channels to carry voice, as the 24th channel is used for the D-channel (signalling channel). PRI is superior, because it offers far more flexible use of the circuit, and provides far more information (like CallerID). Well, if you set that T1 up to carry data and run a link between two IP networks over it, how many SIP conversations could it be expected to carry? How about IAX? Interesting question. I'll tell you this, it won't have so much to do with the 24 channels as it will with how efficiently the circuit is used. When you run data on a T1, all of the pipe is treated as one big channel by the upper layers. The 24 timeslots are all still there, but the network doesn't have any knowledge of them. How would one extend this calculation to varying bandwidth circuits and various VOIP protocols (MGCP, SCCP and H323 come to mind)? Each network layer (think of the OSI model) will add overhead, so the calculation has to take into account how the data (in this case, the voice packets) is encapsulated at each layer. Of the protocols, IAX would probably utilize the circuit most efficiently, due to it's trunking. Naturally, the codec you use will be another key factor. Rather than asking for a full education here, can somebody point me at a suitable practical reference? Of course, if somebody wants to actually post the answer that'd be fine too :) I've always found Newton's Telecom Dictionary to be a great reference. It's not too technical, packed with humour, and very comprehensive. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
