Re: [Asterisk-Users] just softphone
I'm trying to start with Asterisk, but I could not put 2 softphones to talk. The asterisk server rejects the connections always when I dial. May 17 07:49:22 NOTICE[1924]: Rejected connect attempt from 192.168.0.106 Try puting a permit=0.0.0.0/0.0.0.0 In the sip.conf for your two phones. BTW: your extensions.conf looks silly, you'll only be able to call test3 from test3. Busy most of the time ;-) Stefan Märkle ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Send flash through zap channel
OR you can try this: in features.conf: [applicationmap] zapflash=*3,callee,flash if you put any spaces in the above line, it will not work!!! in extensions.conf add this line right before the dial commands where you want this to work: exten = s,12, set(DYNAMIC_FEATURES=zapflash) Then *3 should flash the line. Thanx, so far this was what I missed. But ... Feb 22 09:27:13 WARNING[28084]: app_flash.c:101 flash_exec: Zap/1-1 is not an FXO Channel -- Hungup 'Zap/1-1' So the ISDN Bri (qozap) seems not to support flash. But hey, with an ISDN-Phone directly connected to the NEC PBX it works, so there seems to be a flash-equivalent in ISDN signalisation. Anyone got a clue how to signal flash over zaptel ISDN trunk? Stefan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Send flash through zap channel
Hi everyone, our setup includes a NEC PBX connected to our asterisk via bri lines. The NEC has a doorphone feature, which is just an extension that calls you when someone rings. When connected to this extensions, a 'flash' signalling opens the door. So now, i'd like to trigger this from asterisk, too, but unfortunately wasn't able to do so. Setup: asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1k, Quad-Bri Junghanns Card, Bris set on p2pte. What I tried and didn't work: * Using Flash() in dialplan - doesn't work since channel is Dial()-ed and doesn't allow applications at that very moment * Typing *0 on phone = zap channel doc says this should send flash, but doesn't seem to work in bridged scenarios (ZAP=*=ZAP or SIP=*=ZAP) * Typing # on phone = as of documentation, this sometimes emulates flash = not in my setup * Tried the above from snom sip phone, sip ata with analogue phone and flash-key, mobile phone called in via another zap channel = no difference between the incomings Has somebody any hints for me? Stefan -- Stefan Märkle Netpioneer GmbH Leitender Systemarchitekt Beiertheimer Allee 18 [EMAIL PROTECTED] 76137 Karlsruhe *** Besuchen Sie uns vom 09.03.- 15.03.2006 auf der Cebit 2006 in Hannover. Sie finden uns in Halle 3 auf Stand D31 als Mitaussteller der Imperia AG *** ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Large Asterisk Setup (~500 Concurrent Calls + Scalability)
List Members, Hi ! .. The goal of our new design is to offload the DSP to the Asterisk slave servers, then route the calls via IAX2 trunks to the Asterisk master server. The Asterisk master server will provide us with a centralized point for queuing, digital recording, and music on hold, as well as configuration, monitoring, and reporting. Configuration of the Asterisk slave servers would be limited to setting up extensions to terminate the incoming T1s and setting up IAX2 trunks to the Asterisk master server. These configurations would be rare, so the slave servers would be configured manually on the boxes themselves. Sorry, I don't have any concrete experiences with setups like this, but I'd see a big disadvantage of your design in your 'unscalable' master server that will have way too much to do. I'd consider dropping of the initialized calls back to the slave servers (e.g. SIP reinvite), so that the master server just does the SIP registering and the central dialplan for the setup. Additionally you could think about using a SER for your SIP clients, that would get SIP register load off your master too. I am not shure whether there are means to transfer a SIP=IAX call to the slave server where it would be replaced by SIP=ZAP termination but that would Certainly be the most effective solution, since these slaves could be handled Dynamically for scaling the installation. Just my 2 cents Stefan Märkle -- Stefan Märkle Netpioneer GmbH Head Software Architect Beiertheimer Allee 18 [EMAIL PROTECTED] 76137 Karlsruhe ___ 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] Conference solution for 100+ users
Hi List, Hi! 1-Skype-like softphone for *. is there any? None that I know of. But IAX isn't bad in most of the firewalled environments, give it a try. It only has to get a udp socket open for an outbound connection (may well be NAT-ed) and to receive the answer packets back. 2-Just do audio streaming and have the customers use windows media player. (I dont know how to do this) This would mean exactly the same prerequisites as an iax-based solution as the media stream (usually udp) has to be received by the media players. One technique that circumvents this is using HTTP/1.1 streaming which may or may not work through an application level http-proxy. 3-Use some kind of Softphone with VPN... Again, if you are able to do an outside connect through the firewall (as with openvpn which uses udp or with ipsec which uses ip), you can also do some other things by this means (e.g. iax). 4- Do Softphone---Port 80--- SER---Asterisk w/meetme. Only reason for this might be an application level http-proxy that allows for outbound 'connect' calls, since I don't think you want to encapsule SIP in HTTP, do you?. And for the outbound 'connect' method, port 443 might be a better choice for your port number, but you have to choose a protocol that uses TCP and only one single socket for this to work. Maybe using iax over some sort of UDP-in-TCP tunnel could work (like zeebeedee). Whatever solution I come up with MUST allow anybody to listen in assuming nobody can change firewalls. Any one has already done this? Any feedback will be much appreciated. We're working on similar problems, so if you come up with a perfect solution, please let me know. Also, if you are interested in a commercial solution feel free to contact me off-list. Stefan Märkle -- Stefan Märkle Netpioneer GmbH Head Software Architect Beiertheimer Allee 18 [EMAIL PROTECTED] 76137 Karlsruhe, Germany ___ 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] ISDN X-Over
David J Carter schrieb: Hi all, I have just been reading an article on the asterisk-doc site about ISDN X-Over cables. The article mentioned the converting of an NT1 to make this possible, has anybody got the information required to modify a BT NT1? Or any information on the BT NT1. As far as I know, you just need a second ISDN card and a X-cable. No mods to the NT1 are needed. To build such a cable, just swap the outer pair with the inner pair. 3 - 4 4 - 3 5 - 6 6 - 5 In Addition, you need bus termination: - Use a device that terminates the bus * Your NT1 does that (often already enabled with dip-switches inside), just plug the x-over to the nt1 as well as your other device(s) * Use a multiplug that terminates the bus - or just put in two 100 ohm resistors, one between 45 one between 36. Extremely easy and cheap if you cut a cable in half, rewire them using a screw joint, where you can also put the resistors in. Depending on the device you want to connect, you might need bus powered, THEN you really need your NT1 (or a card that provides this feature). Link to PBX4Linux - documentation has already been posted. Greets Stefan -- Stefan Märkle Netpioneer GmbH Leitender Systemarchitekt Beiertheimer Allee 18 [EMAIL PROTECTED] 76137 Karlsruhe ___ 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] IAX2 Softphone
For all the peoples that wanted to test my windows IAX2
phone, I've put it up on a server where it can be downloaded.
Wow! That is a nice piece of work especially for a primer version.
My first tests looked very promising: no problems at all ...
The phone can be used mostly with the keyboard :
I liked that very much. Some Softphones allow dialing but no hangup or no #/*
on the keyboard = unuseable with keys only.
All comments (good or bad) are welcome
I've just one headache with your software: It's another piece of closed source
(just as firefly is, and they are a bunch of months ahead of you).
Please consider donating the phone to the community by releasing it under an
approved open source license.
You might keep your commercial interests in mind by dual licensing it, so
customizations you do for customers of yours won't necessarily fall under the
open source license.
I think a whole bunch of developers would be glad to join you in the future
development of this client when source is opened, and hey, who never wanted to
be the project manager of a successful open source project? ('wave' to mark ;-)
http://www.marccharbonneau.com/asterisk/mediaxphone.php
As I said: nice piece of work and at the moment the softphone of my choice on
W2k instead of firefly or purtel iaxphone I used before.
Have a nice day and greets from Germany
Stefan
--
Stefan Märkle Netpioneer GmbH
Leitender Systemarchitekt Beiertheimer Allee 18
[EMAIL PROTECTED] 76137 Karlsruhe
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Re: [Asterisk-Users] spandsp on FC3
Nathan C. Smith [EMAIL PROTECTED] wrote: If I start Asterisk from the command line (usually, asterisk -c or asterisk -vvvcp ) I can receive faxes and mailtofax sends them to me OK. If I start the asterisk service (service asterisk start) that uses safe_asterisk, faxes appear to be received, but what actually gets sent to me is about a postage stamp PDF or in some cases the PDF just has errors open file error. I'm guessing this is a timing or processor type issue but I'm not sure what the big difference is. Hi Nathan, my tip would be file permissions or paths mangled. Does your service script run as the same user as when you try starting in a normal shell? Check permissions on the spool dir, perhaps make it world-writeable and retry as a first test, if that works you can narrow problems down. Just my 2 cents Stefan Märkle ___ 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] SMS - how to send one
Hi Stefan co, I second the request for some config info. I've read everything I could find on this and it wasn't enough. I'm still missing some basic concept. As I seem to be the third 'mee too'-cry for SMS-help, may I beg you for updating the asterisk+SMS page in the wiki with a basic working configuration? That would be very useful for us 'still-fighting' people ;-) Stefan -- Stefan Märkle Netpioneer GmbH Leitender Systemarchitekt Beiertheimer Allee 18 [EMAIL PROTECTED] 76137 Karlsruhe ___ 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] SysMaster and GPL Violation
to see anything come of it. Just a little note that Sysmaster is packaging up Asterisk in their product and not giving a notice with the product with an offer to the source of the the GPL software they use inside their products. They even lie and tell people such as myself outright that it's not Asterisk. Seems like strong evidence for a GPL violation. The Linux netfilter team was very active in the past few months to find license infringements with their software (Routers with iptables based firewalls). They won all their cases (mostly before they came to a court), usually gaining a publishing of the modified source code under GPL and a 'donation' to one of the open source foundations or their project team. So if one of the asterisk core team members wants a contact, Harald Welte was the one who fought this iirc (see http://gnumonks.org/users/laforge for contact details). Stefan ___ 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] SysMaster and GPL Violation
From: Matt Riddell [EMAIL PROTECTED] what's about slashdot? Matteo Yup. 1. The strings prove it is Asterisk 2. They deny it is 3. We have multiple sources who can confirm this So, let's all post the article on GPL violation to Slashdot. All in favour? No I'm not. One of the asterisk regulars (doesn't have to be Mark but can) should write to them and give them a hint about what 'the project' expects from them and why (the proves etc.). Something like a GPL publication of their code and small to medium donation to the asterisk project. Set them a deadline for reaction (7 to 14 days). Furthermore the only one who can claim a GPL violation is a buyer of their software, since only the direct customer has to be informed about sourcecode availability. Give them one more chance to react before slashdotting them. Since I suspect they will not, have your slashdot fun afterwards (in a week or two) ;-) Just my 2 cents and a claim for communication culture Stefan ___ 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
[Asterisk-Users] Re: quasi-skype channel for Asterisk?
But I'm not aware of such a converter. See my previous mail of the thread (which I quote below) indicating the URL. The devises are exactly the skype converter to FXO/FXS. http://www.pcphoneline.com/skype Sorry for not reading your Mail completely. So with this thing and an FXO Card in your asterisk box, it should be possible to connect the two worlds, yes. BUT. I don't see a real reason to connect two probably ethernet-equipped boxes via FXO/FXS to transport VOIP-Traffic. When the api is openly available and legally usable to connect phone lines to skype, there should be no problem with a skype-api to iax2 or a skype-api to sip converter on the Windows box which then connects to your asterisk via IP. But since the FXO/FXS solution seems to be instantly available, maybe the solution is the one to use right now. Stefan ___ 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
[Asterisk-Users] Re: quasi-skype channel for Asterisk?
Sorry, I don't have that kind of skill as well as the time for development. So, back to my original question, I'll use the way I proposed if that is possible. If skype can be one of the channels, that is an advantage in holding conference call over the Internet, I think. Hi, There was a long thread about integrating asterisk and skype about 4 weeks ago. The big Pro is the user base skype has the big con is the proprietary license skype uses and that it does nothing that asterisk and dundi can't do themselves better. I do not understand what you mean by 'phone jack'. If you plan to use microphone-in/speaker-out of your skype computer to the opposite jacks on your asterisk box, you will be able to transmit the voice channel in both directions using either chan_oss/chan_alsa or another voip software on the asterisk box. But you have no means to establish a skype channel with this solution as the dial or answer functions only exist in the skype software or the api mentioned above. If there really is a converter (box or software) from skype to 'real' phone jacks (isdn or FXO/FXS or SIP/H323), you are certainly done, as these interfaces have proven that they support voice AND signalling ;-). But I'm not aware of such a converter. Just my 2 whatsoever Stefan ___ 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
[Asterisk-Users] Re: ISDN-Problem with Quadbri behind Tenovis
Hello kietlak and others, Any clues to what happens here? Seems the communication asterisk=Tenovis does not work. And why is the cause not handled in chan_zap? Send your zaptel.conf and ANZG output from AOGD. Sorry, but i only managed apprenticeship on all those 3-letter-acronyms, the 4-letters are way out of reach for me ;-) I don't know what you mean by anzg and aogs, but here are zaptel and zapata Config as well as the bri debug stuff from the failing conversation between asterisk and the tenovis pbx. Identical setup works on german telco (Deutsche Telekom Anlagenanschluss) lines plugged to the four BRIs. Hope someone can point me to a direction where to look further, as we seem stuck right now. Stefan Märkle bri debug span 1 asterisk*CLI bri debug span 1 Enabled debugging on span 1 -- Registered 'stefanm' (AUTHENTICATED) at 10.13.253.154:4569 Protocol Discriminator: Q.931 (8) len=43 Call Ref: len= 2 (reference 39/0x27) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 01 82] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Preferred Dchan: 0 ChanSel: B2 channel ] [6c 0c 21 80 37 32 31 39 32 30 36 30 31 32] Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '7219206012' ] [70 05 81 36 39 35 31] Called Number (len= 7) [ Ext: 1 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '6951' ] [7c 03 80 90 a3] IE: Low-layer Compatibility (len = 5) [7d 02 91 81] IE: High-layer Compatibility (len = 4) -- Making new call for cr 39 -- Processing Q.931 Call Setup -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 108 (cs0, Calling Party Number) -- Processing IE 112 (cs0, Called Party Number) -- Processing IE 124 (cs0, Low-layer Compatibility) -- Processing IE 125 (cs0, High-layer Compatibility) Protocol Discriminator: Q.931 (8) len=7 Call Ref: len= 1 (reference 167/0xA7) (Terminator) Message type: CALL PROCEEDING (2) [18 01 8a] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0 ChanSel: B2 channel ] -- Executing Answer(Zap/2-1, ) in new stack -- Accepting call from '7219206012' to '6951' on channel 0/2, span 1 Protocol Discriminator: Q.931 (8) len=11 Call Ref: len= 1 (reference 167/0xA7) (Terminator) Message type: CONNECT (7) [18 01 8a] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0 ChanSel: B2 channel ] [1e 02 81 82] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ] -- Executing MP3Player(Zap/2-1, /usr/share/asterisk/sounds/pioneer.mp3) in new stack Protocol Discriminator: Q.931 (8) len=41 Call Ref: len= 2 (reference 39/0x27) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [6c 0d 21 80 30 37 32 31 39 32 30 36 30 31 32] Calling Number (len=15) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '07219206012' ] [70 05 80 36 39 35 31] Called Number (len= 7) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '6951' ] [7c 03 80 90 a3] IE: Low-layer Compatibility (len = 5) [7d 02 91 81] IE: High-layer Compatibility (len = 4) -- Processing Q.931 Call Setup -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 108 (cs0, Calling Party Number) -- Processing IE 112 (cs0, Called Party Number) -- Processing IE 124 (cs0, Low-layer Compatibility) -- Processing IE 125 (cs0, High-layer Compatibility) Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 39/0x27) (Originator) Message type: DISCONNECT (69) Oct 27 16:05:07 WARNING[1088519088]: chan_zap.c:6902 zt_pri_error: PRI: XXX Missing handling for mandatory IE 8 (cs0, Cause) XXX Oct 27 16:05:07 WARNING[1088519088]: chan_zap.c:8128 pri_dchannel: Hangup REQ requested
[Asterisk-Users] ISDN-Problem with Quadbri behind Tenovis
Hello everyone, We try to establish a * voicemail system behind a Tenovis (soon to be avaya) Integral 55 with Junghanns quadbri card in the * server. The Tenovis has 4 bri ports configured in nt ptp (edsi 61) which we connected to the quadbri (te, ptp) card. Signaling in one direction seems to work as the asterisk receives a call and seems to answer, but the Tenovis pbx never understands this and switches to 'unreachable' after a short while of ringing. Also, dialing out from an iax-phone via the zap channel results in a ringing signalled in the iax phone but no traffic to the Tenovis (level 2 indicator is alight in tenovis, but d-channel indicator stays dark). We use the bri-stuff-0.1.0-rc4a package from junghanns.net which means asterisk CVS-HEAD-08/13/04. The error asterisk shows when Tenovis dials in: -- Executing Answer(Zap/2-1, ) in new stack -- Accepting call from '7219206012' to '6951' on channel 0/2, span 1 -- Executing MP3Player(Zap/2-1, /usr/share/asterisk/sounds/pioneer.mp3) in new stack Oct 27 19:45:34 WARNING[1088519088]: chan_zap.c:6902 zt_pri_error: PRI: XXX Missing handling for mandatory IE 8 (cs0, Cause) XXX Oct 27 19:45:34 WARNING[1088519088]: chan_zap.c:8128 pri_dchannel: Hangup REQ requested on unconfigured channel 255/255 span 1 Oct 27 19:45:34 WARNING[1088519088]: chan_zap.c:8061 pri_dchannel: Hangup requested on unconfigured channel 255/255 span 1 Oct 27 19:45:39 WARNING[1088519088]: chan_zap.c:8061 pri_dchannel: Hangup requested on unconfigured channel 255/255 span 1 Any clues to what happens here? Seems the communication asterisk=Tenovis does not work. And why is the cause not handled in chan_zap? Stefan -- Stefan Märkle Netpioneer GmbH Leiter Knowledge Center Beiertheimer Allee 18 [EMAIL PROTECTED] 76137 Karlsruhe ___ 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
AW: [Asterisk-Users] Integrating an old PBX with Asterisk
Hi all, Hi Marco, I was thinking about integrating an old PBX with Asterisk and I was wondering some possible configurations. You didn't mention the number of lines your PBX uses, but think of a third scenario: Install an asterisk with twice the number of BRI/PRI-Ports your current PBX has. Connect half of them to your carrier, the other ones to your old PBX (Some sort of proxy scenario, isn't it?). Pro: - You don't have to change a single configuration option in your old PBX if you don't want to or are not able to ;-) - Full integration of asterisk users with old PBX users (internal calls, voicemailbox on asterisk even for ole PBX users etc.) - Very flexible configuration. For example let asterisk use 3-digit extensions while switching prior 2-digit extensions to the old PBX Con: - Costs depending on your setup (Quad-Bri Junghanns costs about 600 Euro, Octo-Bri about 1000 don't know about PRI Cards or other vendors as Digium) - Risk management. Asterisk becomes SPOF for whole telephony system (at least for inbound/outbound calls) In our company we will probably setup your scenario 2 for integration of some voip-Users and voicemail functionality. Just my 2 cents, Stefan -- Stefan Märkle Netpioneer GmbH Leiter Knowledge Center Beiertheimer Allee 18 [EMAIL PROTECTED] 76137 Karlsruhe ___ 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
AW: [Asterisk-Users] Large Enterprises using asterisk
Hi, I've never run against a commercial PBX that didn't need maintenance. Acknowledged. VM hard drives fail, ... Asterisk is every bit as stable as the old-gen KSUs and PBXs. There are big differences. As I know of no other PBX that uses 'consumer' hardware, asterisk is also struggling with problems in the underlying Hardware. And even Software. The NEC/Nitsuko PBX I bought for our company 4 years ago has still no problems beeing extended. Try this with a four year old version of Linux underneath your asterisk. Don't get me wrong, ist also a BIG advantage of the * solution to always be able to upgrade to HEAD without 'licensing' upgrades. Also the hardware upgrades to a decent PC platform is a LOT cheaper than any hardware upgrades commercial PBXs offer. BUT. What are the key points if asterisk wants to make it into many offices and companies? I sorted them in the order they mattered to me: 1. Reliability. Triple nine is not enough as for almost every company, phones are the vital wire to the outside world. Here asterisks suffers major disadvantages to 'closed' solutions commercial PBXs offer. This is due to its complexity, due to its speed of development and change and due to its underlying Hard- and Software. 2. Reliability. ;-) a second aspect of reliability goes further than the 'technical' reliability I already mentioned. If a CIO in a company decides that asterisk is beeing rolled out as a telephony solution, then he has to be absolutely convinced, that the project and the development of asterisk is guaranteed for multiple years (continuity). Or else he buys Cisco or IBM, nobody is fired because of buying Cisco or IBM, you know this topic. 3. Extendability. Here is big point number one for asterisk. No PBX I know has even such an impressing static list of features. So nothing to say about the dynamic changes in evolving features. 5. Ease-of-use. Asterisk with its configuration files has managed to stay manageable despite of the huge feature list. So no problem from operator side. For the consumer this is quite different, because system phones have a whole lot of functions (and function keys etc.) that a 'vanilla' phone isn't able to manage, even as asterisk provides all of that functions. So here the proprietary vendors score a point. Innovaphone for example has the solution that ONE hardphone is guaranteed to work perfectly with their VOIP-PBX since the protocol extensions it uses are supported, * should think about something alike ... 4. TCO. Asterisk with the underlying structure of PC hardware and open source operating system has a big advantage here. Also system and network-administrators with Linux or BSD knowhow are a common must-have for many companies so we don't suffer any disadvantage here. I am CIO of a small company (~80 phones) where the actual PBX has reached its limits. We decided against Asterisk because of the two 'reliability' points. PC Hardware fails. Asterisk has Bugs (I had several crashes with voicemail functions, perhaps codec-related as it only happened with some VOIP-Softphones) and Asterisk and Linux evolve too quick to follow without remarkable manpower. I would very much like to rethink our decision next year, but at the moment, I would not rollout a vital infrastructure based on asterisk (we will continue our asterisk evaluation, but only fore some teleworkers). GNU/Linux has the same problem in many companies. The big distributors as Red Hat or SuSE address this by guaranteeing continuity in there 'enterprise server' products, Debian does the same with the 'stable' branch. So my proposal is, that with the 1.0 version let a distribution form AND an attitude of 'stability and continuity' in that branch do the job of convincing managers (not more than one major release in two years). And let an evolving base of features in the HEAD branch convince the technicians. But this means a LOT of back- and forward porting of patches and managing of developers, who tend to only fight on the 'bleeding edge', especially when not being paid for their work. So this is something only Mark or Digium could achieve or a person/team that has full support by them. Just my 2 Euro-Cents, and sorry for the long text, hope somebody made it to the end ... ;-) Stefan Märkle -- Stefan Märkle Netpioneer GmbH Leitender Systemarchitekt Beiertheimer Allee 18 [EMAIL PROTECTED] 76137 Karlsruhe ___ 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
