Re: [asterisk-users] Where to download Junghanns ISDNguard software?
Nick Richardson a écrit : Hi list, I recently purchased an ISDNguard from Junghanns. It came with no software and there is no sign on their website or in any of their documentation where to download it. I have looked in http://www.junghanns.net/downloads/ and there is no sign of it there either. The only thing remotly close ther is isdnguard-asterisk-1.2.13.patch. Their documentation refers to /usr/sbin/ISDNguard. Where does one get this mysterious binary from? I have emailed their support a few times and get no response, needless to say I am NOT a happy customer. Can anyone help me with a download link? It is in their bristuff package: you'll have to pick res_watchdog and include it in your asterisk build. Regards, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Where to download Junghanns ISDNguard software?
Hi Nick, For using ISDNGuard, you can using res_watchdog from the bristuff patch. I attach for you a version who have patched and work for asterisk 1.2.24. It's an archive with a makefile with just res_watchdog. We have some clients who using this version from many month ago. Sylvain Le jeudi 04 octobre 2007 à 08:51 +1000, Nick Richardson a écrit : Hi list, I recently purchased an ISDNguard from Junghanns. It came with no software and there is no sign on their website or in any of their documentation where to download it. I have looked in http://www.junghanns.net/downloads/ and there is no sign of it there either. The only thing remotly close ther is isdnguard-asterisk-1.2.13.patch. Their documentation refers to /usr/sbin/ISDNguard. Where does one get this mysterious binary from? I have emailed their support a few times and get no response, needless to say I am NOT a happy customer. Can anyone help me with a download link? Thanks in advance.. - Nick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sylvain BOILY Proformatique - 67 rue Voltaire - 92800 Puteaux Tel. : 01 41 38 99 64 - Fax. : 01 41 38 99 70 Email : [EMAIL PROTECTED] - http://proformatique.com/ Vers un monde plus libre Proformatique est membre de l'ASS2L http://www.ass2L.org pf-asterisk-res-watchdog_0.1.1~svn1570.orig.tar.gz Description: application/compressed-tar ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tdm fxo module always busy
Hi Chawki, it is not uncommon that FXO or FXS modules do not work even if no error message is shown when wctdm module is loaded. Have you tried to replace/swap your FXO modules? Giorgio chawki hammoud wrote: Hi: I have a digium tdm04B. One fxo module always gives a faulty busy signal when a call comes in. I swaped places and re-compiled zaptel with no change. Is this a faulty fxo or configuration issue. Regards; Chawki Hammoud Looking for a deal? Find great prices on flights and hotels with Yahoo! FareChase. http://farechase.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday VOIP Users Conference 12:30 PM EDT
This Friday at 12:30 PM EDT We hope to hear more about Astricon and the 1.6 version. A UK legal professional, John Halton of Cripps Harries Hall LLP, joins us to discuss how the law is coming to terms with VOIP. We also expect a visit from Arick of IPKall about what's cooking with them. Most of all, we expect you, the community to share in this experience! For more info see: http://www.VoipUsersConference.org for conference-specific discussion: http://groups.google.com/group/VOIP-Users-Conference During the conferences we monitor IRC: freenode.net #voip-users-conference Thanks to Digium for their continued support. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multi tenant
Hi all, i just wanted to know if any one has done any multi-tenant version of the asterisk. thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tdm fxo module always busy
On 10/4/07, chawki hammoud [EMAIL PROTECTED] wrote: I have a digium tdm04B. One fxo module always gives a faulty busy signal when a call comes in. I swaped places and re-compiled zaptel with no change. Is this a faulty fxo or configuration issue. Hi, I suggest you post your related zaptel files here and why not call or email Digium support? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Infuriating problems: no dial tone, dropped calls, no voice: 1.2.13 and 1.4.11
Christian Weeks wrote: Hi I've had an asterisk setup for the past 15 months, based on the debian asterisk packaging. Until late August of this year, I had no problems once initial setup was complete- the system worked essentially flawlessly. Since August I have been having exceedingly infuriating intermittent problems that are causing me occasional periods of nasty trouble: 1. No Dial Tone. Every Sunday night at just prior to midnight ( about the last second before monday), the Dial tone ceases on all zap handsets. Investigation shows that the zaptel layer is not transmitting sound to the handset- ztmonitor shows it being sent, but nothing is arriving at the handset. (You can also see sound being rx from the handset). This problem has occured spontaneously at other times but the midnight thing is just plain odd (nothing is happening on that box at the time). 2. I tried to upgrade from the asterisk packages in debian etch (1.2.13) to sid (1.4.11). This added new problems to the mix. Suddenly, no voice was being offered through asterisk at all to any zap channel. I tracked it down to my use of T/t in the Dial strings- this was somehow preventing native bridging from occuring. This can be verified because other call routings (e.g. IAX-zap; SIP-zap) have the same problem (and cannot be corrected by removing a t/T because there's no native bridge). Reverting to 1.2.13 seems to have fixed this problem. (Is this some kind of sound path regression? Debug logging has shown nothing). 3. Finally, with 1.4.11 especially, the system seems to have been quite unstable, with asterisk crashing (and ringing every phone in the house incessantly- which my wife was NOT happy about), especially when two simultaneous calls overlap in some way (this hasn't crashed 1.2.13 as badly, but asterisk seemed to need a reset afterwards). I suspected a dodgy channel bank ( I had a really old eBay special for $20 which had timing problems from day 1 ) so I upgraded a little ( the Zhone - more expensive and not super, but at least it has firmware and a console for mgmt ). This has had some effect, but nothing has changed about the fundamental problem (1 above). Other hardware: the T1 interface card is the R1T1 from rhino, there is a Wildcard TDM400P REV I, with a single FXO port for the incoming line from the POTS. The computer itself seems quite fine, no sign of interrupt errors or other problems with the hardware (I ran a memtest and a cpuburn neither of which showed any issues). zttest shows nothing unusual (99.87% iirc over about 10 minutes). I am happy to share anything that will help resolve the issue- my feeble C skills in attempting some printf in the ast_channel_bridge command to see what was being chucked about pretty much failed entirely because the timing went badly off... Trying to chuck ast_log calls in there didn't work very well either :( Thanks Christian I think you should stick with 1.2 for now at least until all the bugs are worked out. Stability seems to be a real issue although many would have you believe otherwise. The other thing that jumps out is the exact timing of your issue in number 1. Random problems can be very difficult to track down but yours is consistent, so it should be pretty easy to find the culprit. Check your cron jobs. If you have something running at almost midnight on Sunday, that is probably your issue. Logs may be of help. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] extensions.conf vs. AEL
All this discussion is pointless. As pointless as the discussion of assembly versus high-level languages decades ago. Except most people rooting for extension.conf don't even have the technical and conceptual amplitude to understand what they are talking about... they just want some telephony system to make a quick buck, or save in their LD calls... A lot of Asterisk is technically and architecturally twisted, and spaghettied, and with many redundant ways of doing the same thing (in different stages of obsolescence, incompleteness, and (un)documented). At least AEL is a step in the right direction (even though it has to adapt itself to the ugliness that exists below..) BarZ Steve Murphy wrote: On Wed, 2007-10-03 at 09:33 -0600, Anthony Francis wrote: Eric ManxPower Wieling wrote: Let us not forget that AEL cannot be stored in a database therefore rendering you unable to utilize realtime. AEL converted into standard extensions.conf syntax in the dialplan. Doesn't this render having used AEL pointless? Absolutely not! Reasons to use AEL: 1. Several semantic checks are done on the AEL that are NOT done if you go straight to extensions.conf. We try to protect you... from yourself. 2. At least one security issue in USAGE is avoided by having AEL compile the corresponding code; as to how many more issues will automatically be handled via AEL in the future, is impossible to say. We'll see. If you keep coding via extensions.conf, be prepared to make corrections... if you do it in AEL, a restart of Asterisk will hopefully suffice, after AEL is updated. 3. Syntax errors are reported by AEL. It is pretty good at catching all omissions and commissions. Better than the extensions.conf parser is. For example, I don't know if we catch it now, but if you accidentally say extem = 3,... instead of exten = 3,... in extensions.conf, that line will silently be dropped. Sure, we could fix this, but to fix ALL possible problems will require an expensive rewrite of the config file parser, from the ground up. 4. You are insulated against any mods to extensions.conf; like the change to ',' instead of '|' in app arguments. No changes to AEL code are necessary. 5. In extensions.conf, you have to feed your dialplan to asterisk to find any problems. AEL provides the standalone parser, aelparse, so you can correct any problems BEFORE feeding it to a living asterisk. 6. AEL is easier to read, IF you take advantage of the ability to use tabs, etc. wisely. Especially for nested code. Staying away from goto as much as possible, and using the flow of control and looping statements will make your code easier to read, compose, and maintain in the future. It means fewer bugs in your code, and overall this all means lower cost. And higher profits. 7. Repetitious entry of extenname, priority, in your tabular extensions.conf can lead to subtle errors that could be hard to find, ESPECIALLY if you resort to using priority NUMBERS instead of n. And, if you ARE so foolish as to use just raw numbers, and you have to insert or delete a line or two, you have to renumber the remaining lines, and heaven help you if you make a simple error, and accidentally skip a number. 8. Work flow. Since aelparse allows you to dump the compiled dialplan in extensions.conf format, you can still use stuff like realtime. You can use this output against machines that don't even have pbx_ael loaded, then, and you should be able to use 1.4 compiled dialplans on 1.2 machines, as long as you are careful about what apps you call, and how you call them. 9. Easier to write code. Good Code. using Goto's in extensions.conf will allow you to do anything you need to do, but it also results in spaghetti style code. While the original author might be able to decrypt it, and maintain it, unless it's really well commented, the next guy to play with it, is going to have a hard time. Following the flow of control thru spaghetti can get your adrenalin flowing-- and side affects from strange cases and leakage in the spaghetti can make some devilishly hard to solve problems. Think of and treat extensions.conf like assembly code. Think of and treat AEL like a high(er) level language. For those who never did the computer science thing, I have just one piece of advise, and ignore this at your peril: your dialplan is a work of computer programming. It's software. If you don't treat it that way, and use good software methodologies, you'll pay your price. murf ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and
Re: [asterisk-users] No audio on Zap (T1/PRI) channels
Steve Edwards wrote: On Wed, 3 Oct 2007, Steve Totaro wrote: Kevin P. Fleming wrote: Steve Edwards wrote: [trunkgroups] trunkgroup = 1,24,96 spanmap = 1,1,0 spanmap = 2,1,2 spanmap = 3,1,3 spanmap = 4,1,1 You caused the behavior you are seeing by configuring your spanmap this way; you've got physical span #4 configured as the second span in the trunkgroup, so Zaptel will treat physical channels 73-95 as logical channels 1/1 through 1/23. If it were configured as the second span, shouldn't is be channels 25-48 rather than 1-23? voip-info was very unclear about this when I looked at it over a year ago. I finally got it working by trying different combinations in spanmap. Digium should have it's own wiki that is maintained by Digium. Voip-info is ok but much of it is old and or incorrect at this point. Qwest and I fiddled for a couple of hours. The channels are answered in ascending sequence and match between logical span/channel and zap/channel. Channels 0/24 (zap/24) and 1/24 (zap/48) are skipped because they are the primary and secondary D channels. Still no joy -- no audio. Here's zaptel.conf: # span 1 span= 1,1,0,esf,b8zs bchan = 1-23 dchan = 24 # span 2 span= 2,0,0,esf,b8zs bchan = 25-47 dchan = 48 # span 3 span= 3,0,0,esf,b8zs bchan = 49-72 # span 4 span= 4,2,0,esf,b8zs bchan = 73-96 # (end of /etc/zaptel.conf) and zapata.conf: [trunkgroups] trunkgroup = 1,24,48 spanmap = 1,1,0 spanmap = 2,1,1 spanmap = 3,1,2 spanmap = 4,1,3 [channels] context = block-ani echocancel = no echocancelwhenbridged = no echotraining= no group = 1 resetinterval = never signalling = pri_cpe switchtype = dms100 ; span 1 (1-24) channel = 1-23 ; span 2 (25-48) channel = 25-47 ; span 3 (49-72) channel = 49-72 ; span 4 (73-96) channel = 73-96 ; (end of /etc/asterisk/zapata.conf) Any more clues on where to look to find my missing audio? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 The wiki page is still as cryptic as ever. http://www.voip-info.org/wiki/view/NFAS As I said before, I just kept changing the values in spanmap until it worked. This was with Global Crossing. There are only a handfull of combinations to try. I did not use a backup D chan as I figured if one D chan went down on a trunk group, both probably would so that adds to your mix. Just keep fiddling with spanmap and testing calls. If you can get your provider to point DID a did to each individual span, it will make troubleshooting much easier without needing a tech on the phone. This is how I solved my issue. From the wiki: spanmap = 1,1,3 spanmap = 2,1,1 spanmap = 3,2,2 spanmap = 4,2,0 When we had it configured 0,1,2,3 any calls that came into spans 1 or 2 worked fine. But spans 0 and 3 did not pass any audio. Switching the logical span numbers fixed it. This is because zapspan #1 is suppose to be the one that the D-Channel is on. In our case that was actually logical span 3. Be sure to watch out for this, we were confused and took up a few hours of a GBLX tech's time to get it fixed. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Secondary Dialtone and selecting a specific line from Zap/g
Dear Mojo;
That is primary fine, but there are two issues looking
for help about them:
1) Based on your below example (dialing *4*18005551212
to select channel 4), the question is how to give
second dial tone just after dialing the *4*
(indicating the channel was captured)?
2) How to let this second dial tone to be with a
frequency differs than normal tone when pickup the
handset to place a call?
3) How to let (assign) one of the button on my IP
Phone to be dedicating for a zap channel, so when I
select this button and do dialing for a number, then
call will be done via that specific zap channel.
Any help?
Regards
Bilal
-
It would be ugly, but you could prefix a zap channel
or group number
before the phone number to dial. Using groups for an
example:
exten = _*X*X.,1,Dial(ZAP/g${EXTEN:1:1}/${EXTEN:3})
exten = _*XX*X.,1,Dial(ZAP/g${EXTEN:1:2}/${EXTEN:4})
so dialing *4*18005551212 dials out over zap group
4...
bilal ghayyad wrote:
I need to select a line from the Zap group channel
using the SIP Phone (not FXO and not FXS ports).
ignorepat does not work?
Also, what is the method to let the second dial tone
has another tone frequency?
Regards
Bilal
Building a website is a piece of cake. Yahoo! Small Business gives you all the
tools to get online.
http://smallbusiness.yahoo.com/webhosting
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Secondary Dialtone and selecting a specific line from Zap/g
Dear Walt; Maybe I did not understand any thing from below :) - Are you talking about configuration to be done on the Telephone device is self or on the AVAYA server it self? If it is on the telephone device, so how you will give a second dial tone and you do not know if there is available channel :) - I am looking to have a second dial tone by doing such configuration at AVAYA server itself, and that to be used by all users of different IP Phones models (not link sys only). Can you help? Regards Bilal -- For another tone frequency for the outside dialtone, try putting this value [EMAIL PROTECTED],[EMAIL PROTECTED],[EMAIL PROTECTED];*(.4/0/1),10(*/0/2+3) in the Outside Dialtone field. It will give you a slight pause followed by a different dialtone frequency. On a Linksys/Siprua 941, that would be at the top of the Regional page. However, you won't hear any secondary dialtone unless you put a comma after EVERY initial '9' in the dialplan string for each line in use. On a 941, that would be at the bottom of the Ext 1 and Ext 2 pages of the web interface. I suggest the dialplan string of: (*xx|[1-7]xx|9,[3469]11|98|99|9,[2-9]xx|9,11|9,[2-9]xx|9,1[2-9]xx[2-9]xx|9,011xxx.) - Walt Joyce Eric ManxPower Wieling wrote: I can't help you with that. I only wanted to point out that ignoreopat is not what you need. On Polycom SIP phones you continue dialtone by placing a , in the phone's dialplan. SIP phones have their own internal dialplan that is not part of Asterisk's dialplan. You would have to check the docs for your phone. Not all SIP phones can continue dialtone. bilal ghayyad wrote: I need to select a line from the Zap group channel using the SIP Phone (not FXO and not FXS ports). ignorepat does not work? Also, what is the method to let the second dial tone has another tone frequency? Regards Bilal Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. http://sims.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configuration files inside SQLite3
Hi Mark, Am Mittwoch, den 03.10.2007, 11:15 -0500 schrieb Mark Michelson: GNUbie wrote: Hello all, Is it possible to store, read and write configuration files in an SQLite3 database instead of using the configuration files inside the /etc/asterisk/ directory? If it is then can you point me to the right documentation on how to do this or probably hints on how to do this? Thank you in advance. GNUbie It is possible to store configuration files in any relational database which has ODBC compatibility. Thus, sqlite qualifies. If you are using trunk, you won't even need to use ODBC, because Asterisk has native support for sqlite. Are You shure the native support of asterisk is for SQLite3 as the original poster asks for? AFAIK * supports SQlite (Version 2, not 3), which has a completely different API. Karsten Wemheuer ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] About Megaco
Hi all, I've been searching for a while and haven't found if asterisk supports already or if it's going to support h.248. thanks Eve ¡Sé un mejor ambientalista! Encuentra consejos para cuidar el lugar donde vivimos. http://telemundo.yahoo.com/promos/mejorambientalista.html ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] About Megaco
On Oct 4, 2007, at 8:39 AM, Steve Totaro wrote: Try searching using MGCP which is what Megaco evolved into. http://www.voip-info.org/wiki-Asterisk+MGCP+channels Thanks, Steve Totaro Too bad the MGCP channel isn't the full implementation. /b ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] extensions.conf vs. AEL
On Thursday 04 October 2007 07:07:47 Barzilai Spinak wrote: All this discussion is pointless. As pointless as the discussion of assembly versus high-level languages decades ago. As one of the main architects, I don't find this discussion pointless. My personal opinion of AEL is that it's coming along nicely, but it's still not up to the point where I would consider using it for most dialplans. That day will come, and I'm working with Steve Murphy to ensure that it does. One thing that you did not see in the language wars of yesteryear was of the assembly language changing in subtle ways, to make development in the higher level language easier or more consistent, as is the case with AEL and extensions.conf. Except most people rooting for extension.conf don't even have the technical and conceptual amplitude to understand what they are talking about... they just want some telephony system to make a quick buck, or save in their LD calls... This seems like a rather harsh indictment, when it really comes down to the fact that writing in extensions.conf works today, and while AEL does work to a certain extent, many people would rather not have to rewrite their dialplans every time an architectural flaw is found in AEL that limits what they can do; ergo, they write their stuff in extensions.conf until the point where AEL becomes more trusted. A lot of Asterisk is technically and architecturally twisted, and spaghettied, and with many redundant ways of doing the same thing (in different stages of obsolescence, incompleteness, and (un)documented). As a maintainer and architect, I would very much like to hear specific criticisms on how you think this could be improved. We try to deprecate specific functionality that doesn't work correctly or which could be expressed in better ways, which allows users of the system to transition away from those expressions to better methods over a period of time, instead of immediately at an upgrade; we believe this facilitates adoptions and upgrade processes. At least AEL is a step in the right direction (even though it has to adapt itself to the ugliness that exists below..) All high level languages have to adapt themselves to the ugliness below. That is part of what makes them high-level languages. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Line State
Hello all, I need a little help to check the state of the line from asterisk on aa TDM400P because when the telco lines goes down, asterisk get that line for outgoing calls. There is a way to check it out? And when all lines are busy to do outgoing calls how can i do to callback the people that call when a line is free? Thanks Alejandro González Grupo Gestión 4384-0660 www.grupo-gestion.com.ar [EMAIL PROTECTED] --- --- RI 9000-1069 Sistema de Gestión de Calidad Certificado por IRAM Norma ISO: 9001-2000 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.15.24/592 - Release Date: 18/12/2006 01:45 p.m. attachment: image001.gif___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] About Megaco
Floyd wrote: Hi all, I've been searching for a while and haven't found if asterisk supports already or if it's going to support h.248. thanks Eve Try searching using MGCP which is what Megaco evolved into. http://www.voip-info.org/wiki-Asterisk+MGCP+channels Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi tenant
Hi Mujtaba, We have a multi-tenant version of our Asterisk based management and end-user software called Thirdlane PBX Manager. You can see a demo of a single-tenant version on our web site http://www.thirdlane.com/pbxmanager.htm the multi-tenant adds tenant and DID management, and allows to partition Asterisk to manage independent tenants with their own administrators, extensions, routes, queues, etc Please contact me off list for more information. Best regards, Alex Alex Epshteyn Third Lane Technologies, LLC http://www.thirdlane.com _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mujtaba Mahmood Sent: Thursday, October 04, 2007 2:14 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Multi tenant Hi all, i just wanted to know if any one has done any multi-tenant version of the asterisk. thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: [asterisk-dev] chan_h323 and chan_oh323 compatibilities
-- Forwarded message -- From: Tzafrir Cohen [EMAIL PROTECTED] Date: Oct 4, 2007 12:56 PM Subject: Re: [asterisk-dev] chan_h323 and chan_oh323 compatibilities To: [EMAIL PROTECTED] Hi On Thu, Oct 04, 2007 at 11:46:30AM -0300, Caciano Machado wrote: I'm receiving a lot of warning messages from my Asterisk 1.2.5/chan_oh323 every time it establishes a channel with other Asterisk 1.4.2/chan_h323. Questions about such matters should go to the asterisk-users mailing list . -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Caller ID Info
Hi Asterisk Users, I was wondering why a call that is received from Asterisk shows a caller ID 'Unknown' . So here is the scenario, 'A' calls 'Asterisk'. 'A' joins a conference on 'Asterisk'. 'Asterisk' calls 'B'. 'B' gets joined to the same conference also. 'B' somehow receives the caller ID 'Unknown' and not the number of 'Asterisk', is this a feature not supported in Asterisk or is there a problem in my network ? Any hints or suggestions would be really helpful ? Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial-Chain interrupted by Operator Called Party not reachable Messages
Hi, I have the following problem: I want asterisk to dial a chain of n-numbers until somebody picks up the line. I am using Digium E1 Hardware (zaptel) for dialing out. Dialing a Chain is basically no problem, I use somwthing like: dial(no1,50) dial(no2,50) dial(no3,50) However, If no1 is not reachable, for example it is a mobile and switched off, then some automatic Operator-Voice from the Mobile-Telco says forever: This number is currently not reachable and this means asterisk thinks the call was succesfull and does not continue with the the other numbers. Does somebody has an idea how I can distinguish those Operator Voices from real calls ? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using PHP to reload extensions
On Wed, Oct 03, 2007 at 09:10:58PM -0500, Moises Silva wrote:
If you are running the script from a web server, the script gets
executed with the web server process permissions, hence, probably does
not have access to /var/run/asterisk.ctl.
You can give permissions to your web server, or better yet, dont
execute the command using shell_exec, better open a socket connection
to the Asterisk manager and execute Action: Command
Command: extensions reload
Not that, in essense, this permits the web server's user to control
Asterisk as well - the web server's user must be able to read the
password from somewhere.
The only real benefit is if you can limit the permissions you give to
that specific manager user. But there's a limit to ohw useful this can
be. Even write=command alone allows changing the dialplan ('dialplan
add' / 'dialplan remove') and running an arbitrary command as the
asterisk user (originate a call to the application System).
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] need suggestion
Dear All, my client wants a asterisk pbx with 30 FXO 30 FXS analogue ports, please suggest if sangoma A400 is a good option for that. Also please suggest server hardware. regards, Umair ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Supermicro PDSME+ and TE110P [ ref:00D36mPe.50033qy57:ref ] NEW CASE 22828
I never thought very useful to search in this http://www.digium.com/en/docs/misc/compatibility_notes.php page, as it looked rather static for holding such compatibility issues. You proved me I was wrong. Taking this e1000 driver issue as an example, this http://sourceforge.net/projects/e1000 project seems to provide a solution. Reading http://www.digium.com/en/docs/misc/compatibility_notes.php I would still have a doubt about the way to go. Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question FXO Port
Hello list I am new in this list. Before I wrote this email, i search with google and in the list arichves for the question. I look for a possibility to install FXO ports not over RJ11 Ports. I will install the Ports by LSA+ Patch panel. Someone an idea ore link? Thanks for help. Bye MZ PS: Perhaps you look on my blog http://two-weeks-fun.blogspot.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Setting caller id value on outgoing calls using .call files
Hi all, I was looking at a way to add the caller id to the outgoing calls (which are made using .call files) using asterisk. Any ideas how to do this ? Currently I get 'Unknown' number displayed on my phone when asterisk makes an outgoing call. Also using something like this is not working as it still displays unknown number. I want set the callerid on the 1.call which is made. exten = _.,4,Set(CALLERID(all)=Joe 911) exten = _.,5,system(cp /var/spool/asterisk/1.call /var/spool/asterisk/outgoing/) Any suggestions how to do that. Thanks a lot. Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting caller id value on outgoing calls using .call files
Thanks guys. No need to reply. I got my answer from someone. On 10/4/07, Arpit Mehta [EMAIL PROTECTED] wrote: Hi all, I was looking at a way to add the caller id to the outgoing calls (which are made using .call files) using asterisk. Any ideas how to do this ? Currently I get 'Unknown' number displayed on my phone when asterisk makes an outgoing call. Also using something like this is not working as it still displays unknown number. I want set the callerid on the 1.call which is made. exten = _.,4,Set(CALLERID(all)=Joe 911) exten = _.,5,system(cp /var/spool/asterisk/1.call /var/spool/asterisk/outgoing/) Any suggestions how to do that. Thanks a lot. Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Caller ID Info
Thanks a lot guys. I got my answer from someone. :) On 10/4/07, Arpit Mehta [EMAIL PROTECTED] wrote: Also what are the ways if any to set this DNIS or RDNIS information ? Regards Arpit On 10/4/07, Arpit Mehta [EMAIL PROTECTED] wrote: Hi Asterisk Users, I was wondering why a call that is received from Asterisk shows a caller ID 'Unknown' . So here is the scenario, 'A' calls 'Asterisk'. 'A' joins a conference on 'Asterisk'. 'Asterisk' calls 'B'. 'B' gets joined to the same conference also. 'B' somehow receives the caller ID 'Unknown' and not the number of 'Asterisk', is this a feature not supported in Asterisk or is there a problem in my network ? Any hints or suggestions would be really helpful ? Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Caller ID Info
Also what are the ways if any to set this DNIS or RDNIS information ? Regards Arpit On 10/4/07, Arpit Mehta [EMAIL PROTECTED] wrote: Hi Asterisk Users, I was wondering why a call that is received from Asterisk shows a caller ID 'Unknown' . So here is the scenario, 'A' calls 'Asterisk'. 'A' joins a conference on 'Asterisk'. 'Asterisk' calls 'B'. 'B' gets joined to the same conference also. 'B' somehow receives the caller ID 'Unknown' and not the number of 'Asterisk', is this a feature not supported in Asterisk or is there a problem in my network ? Any hints or suggestions would be really helpful ? Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail/dtmf not working?
Hi,
I am setting up an asterisk server for testing purposes and cannot get
voicemail to work at all.
My host OS is Linux From Scratch 6.3 and the asterisk software versions
I built are zaptel-1.4.5.1 and asterisk-1.4.12.
I am using the Ekiga softphone on my Ubuntu desktop machine. My asterisk
server and client phone are on different computers but are on the same
LAN, i.e. no NAT.
I have an echo test extension which works fine. But when I try to call
into my voicemail account using 8100, I do not hear the first Playing
'vm_password' message (although do I hear the subsequent messages). And
any numbers I enter on the ekiga keypad do not seem to be recognised by
asterisk (I enabled debug rdp and rfc2833 packets appear to be
transmitted and received). The softphone is set up to use RFC2833 for DTMF.
I have a single x100p card installed which seems to work - to a
fashion... Incoming calls are answered and the greeting is heard, but
the line hangs up instantly the message finishes. (A different problem
which I will investigate seperately unless someone has a quick answer).
Outgoing calls seem to be O.K. An lsmod of my system reveals the following:
===
Module Size Used by
zttranscode 6280 0
wcfxo 9760 0
zaptel186660 6 zttranscode,wcfxo
crc_ccitt 1792 1 zaptel
===
Below is a typical call log (I *am* typing 1234 on the ekiga keypad
during this call) and my extension, sip and voicemail.conf files. Anyone
got any suggestions?
Messages on log
=
-- Executing [EMAIL PROTECTED]:1] Answer(SIP/100-081d9478, ) in
new stack
-- Executing [EMAIL PROTECTED]:1] VoiceMailMain(SIP/100-081d9478,
100) in new stack
-- SIP/100-081d9478 Playing 'vm-password' (language 'en')
-- Incorrect password '' for user '100' (context = default)
-- SIP/100-081d9478 Playing 'vm-incorrect' (language 'en')
-- SIP/100-081d9478 Playing 'vm-password' (language 'en')
-- Incorrect password '' for user '100' (context = default)
-- SIP/100-081d9478 Playing 'vm-incorrect' (language 'en')
-- SIP/100-081d9478 Playing 'vm-password' (language 'en')
-- Incorrect password '' for user '100' (context = default)
-- SIP/100-081d9478 Playing 'vm-incorrect' (language 'en')
-- SIP/100-081d9478 Playing 'vm-goodbye' (language 'en')
== Auto fallthrough, channel 'SIP/100-081d9478' status is 'UNKNOWN'
=
extension.conf
;exten = $name,$priority,$application()
[globals]
ALANL=SIP/100
OUTBOUNDTRUNK=Zap/1
FWDNUMBER=867*** ; My FreeWorldDialup Number
FWDCIDNAME=Alan Lord ; My CLI
FWDPASSWORD=**
FWDRINGS=${ALANL} ; Phone to ring
FWDVMBOX=1000 ; Voice Mail Box (not yet setup)
[zap_incoming] ; Channel defined in zapata.conf
exten = s,1,Answer( )
exten = s,2,Set(TIMEOUT(digit)=5)
exten = s,3,Set(TIMEOUT(response)=30)
exten = s,4,Background(vm-enter-num-to-call)
exten = s.5,Wait(5) ;Try to stop line hanging up straight away - failed
exten = t,1,Goto(s,2) ; Repeat s,2 if no input from caller
exten = i,1,Playback(pbx-invalid)
exten = i,2,Goto(s,2)
exten = 100,1,Dial(${ALANL},10)
exten = 100,2,VoiceMail(u100)
exten = 100,102,VoiceMail(b100)
exten = 100,3,Hangup()
[internal]
include = outbound-local
; My ekiga SoftPhone
exten = 100,1,Dial(${ALANL},,r)
;Outbound to FreeWorlDialup
exten = _393.,1,SetCallerId,${FWDCIDNAME}
exten =
_393.,2,Dial(IAX2/${FWDNUMBER}:[EMAIL PROTECTED]/${EXTEN:3},60,r)
exten = _393.,3,Congestion
; Local echo test
exten = 611,1,Answer()
exten = 611,2,PlayBack(demo-echotest)
exten = 611,3,Echo()
exten = 611,4,PlayBack(demo-echodone)
exten = 611,5,Hangup()
; Manage Voicemail
exten = _8XXX,1,Answer()
exten = _8XXX,2,VoiceMailMain(${EXTEN:1})
; Outbound via PSTN
[outbound-local]
exten = _9XX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _9XX,2,Congestion()
exten = _9XX,102,Congestion()
exten = 999,1,Dial(${OUTBOUNDTRUNK}/999)
exten = ,1,Dial(${OUTBOUNDTRUNK}/999)
[fromiax] ; IAX trunk from Alan B defined in iax.conf
;TBD
[fromiaxfwd] ;IAX Trunk from FWD
exten = ${FWDNUMBER},1,Dial(${FWDRINGS},20,r)
exten = ${FWDNUMBER},2,Voicemail,u${FWDVMBOX}
exten = ${FWDNUMBER},102,Voicemail,b${FWDVMBOX}
=
sip.conf
=
[general]
srvlookup=yes
disallow=all
allow=alaw
allow=ulaw
allow=gsm
dtmfmode=auto
[100]
type=friend
callerid=Alan Lord
secret=**
qualify=yes ; Qualify peer is no more than 2000 ms away
nat=no ; This phone is not natted
host=dynamic ; This device registers with us
canreinvite=no ; Asterisk by default tries to redirect
context=internal ; the internal context controls what we can do
mailbox=100 ; Voicemail Box
===
Re: [asterisk-users] Cisco 7940G licensing with asterisk
There are other Gigabit SIP phones from Nortel and Avaya, if my memory serves me right. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.12 and Asterisk-addons 1.4.3 released
Just forget it about the 1.2 mantra, it's not going to happen. Focus your energy elsewhere. Lot's of bug fixes are good. Even Cisco comes out with regular bug fixes for IOS. Open source just makes things more visible. -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 02, 2007 3:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.4.12 and Asterisk-addons 1.4.3 released Don Pobanz wrote: the Asterisk release contains a large number of bug fixes for all parts of Asterisk. I am thankful to see the amount of fixes that have gone into this release. However, seeing this many fixes does not give me a warm fuzzy feeling that we won't see a lot more fixes in the near future. So are bug fixes good or bad? ;-) And more importantly, will any of the remaining bugs bite me? Branch 1.4 has one important to us feature that 1.2 does not and that is the queue autofill option. Because of this one feature, I have been wanting to switch to the 1.4 branch for some time. We have a backup system that I will be using for testing. If all goes well, we will move to the 1.4 branch. I hope many others are doing the same so the stability of 1.4 can be improved to the point where no one is concerned. Thanks to all the developers for improving an already great product! Don Pobanz Another reason to call for a 1.2 spoon or fork! Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.722: ast_channel_make_compatible failure
Well I know. My problem is, that the phone offering g722 could do alaw as well. I expected asterisk should just chose alaw for the communication - no transcoding is necessary then... Please help. Thanks, Ondrej Kevin P. Fleming wrote: Ondrej Valousek wrote: [Sep 20 10:14:32] WARNING[30706]: chan_sip.c:2963 sip_call: No audio format found to offer. Cancelling call to phone3 Asterisk 1.4 does not have G.722 transcoding, only passthrough support. It can connect G.722 channels together, and record or playback G.722 audio files, but that is all. The information contained in this e-mail and in any attachments is confidential and is designated solely for the attention of the intended recipient(s). If you are not an intended recipient, you must not use, disclose, copy, distribute or retain this e-mail or any part thereof. If you have received this e-mail in error, please notify the sender by return e-mail and delete all copies of this e-mail from your computer system(s). Please direct any additional queries to: [EMAIL PROTECTED] Thank You. Silicon and Software Systems Limited. Registered in Ireland no. 378073. Registered Office: Whelan House, South County Business Park, Leopardstown, Dublin 18 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting caller id value on outgoing calls using .call files
Arpit Mehta wrote: I was looking at a way to add the caller id to the outgoing calls (which are made using .call files) using asterisk. Any ideas how to do this ? Add Callerid: Name 123 to the call file. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail/dtmf not working?
Alan Lord wrote: sip.conf = [100] type=friend callerid=Alan Lord secret=** qualify=yes ; Qualify peer is no more than 2000 ms away nat=no ; This phone is not natted host=dynamic ; This device registers with us canreinvite=no ; Asterisk by default tries to redirect context=internal ; the internal context controls what we can do mailbox=100 ; Voicemail Box Try adding: dtmfmode = rfc2833 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting caller id value on outgoing calls using .call files
On 10/4/07, Arpit Mehta [EMAIL PROTECTED] wrote: I was looking at a way to add the caller id to the outgoing calls (which are made using .call files) using asterisk. Any ideas how to do this ? Currently I get 'Unknown' number displayed on my phone when asterisk makes an outgoing call. Add a CallerID: whatever line to your callfile. http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out has a reference of the callfile contents. -- j. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting caller id value on outgoing calls using .call files
On Thu, 2007-10-04 at 16:10 -0400, Arpit Mehta wrote: exten = _.,4,Set(CALLERID(all)=Joe 911) exten = _.,5,system(cp /var/spool/asterisk/1.call /var/spool/asterisk/outgoing/) You need to set the Caller ID in the call file itself. The sample call file (sample.call) in the Asterisk source contains the following line: Callerid: Wakeup Call Service (555) 555- Hopefully that gets you going in the right direction. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Caller ID Info
Arpit Mehta wrote: Thanks a lot guys. I got my answer from someone. :) Not that I'm interested in this specific issue, but usually it won't hurt to share your solution with the other list members (and archives). Doesn't apply to this case maybe. Cheers, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] #modprobe wctdm or #modprobe zaptel
Hi list; I need to run the command modprobe wctdm and whenever I write it, then it gives me the following message: FATAL: Module wctdm not found FATAL: Error running install command for wctdm So, do I have to run that command from specific path? Or what is the problem? Any help? Regards Bilal Looking for a deal? Find great prices on flights and hotels with Yahoo! FareChase. http://farechase.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using PHP to reload extensions
No, because then asterisk would be presented three arguments: '-rx',
'extensions', and 'reload' -- as 'extensions' is not a command by
itself, and the 'reload' appears superfluous to asterisk, this would not
work as desired.
Asterisk needs to be presented two arguments - the first is '-rx', the
second is extensions reload (needs additional quoting to contain the
space) which is actually a parameter to the '-x' switch just used.
$output = shell_exec(asterisk -rx 'extensions reload')
is right.
Generally, the difference between single quotes and double quotes is
that with double quotes, PHP is allowed to make $variable substitution
while with single quotes, it is not.
Mojo
Lee Jenkins wrote:
I'm not a PHP guy, but shouldn't the double quote be surrounding the
entire shell command like this?
$output = shell_exec('asterisk -rx extensions reload');
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail/dtmf not working?
At 11:10 10/4/2007, Alan Lord wrote: Hi, I am setting up an asterisk server for testing purposes and cannot get voicemail to work at all. My host OS is Linux From Scratch 6.3 and the asterisk software versions I built are zaptel-1.4.5.1 and asterisk-1.4.12. I am using the Ekiga softphone on my Ubuntu desktop machine. My asterisk server and client phone are on different computers but are on the same LAN, i.e. no NAT. I have an echo test extension which works fine. But when I try to call into my voicemail account using 8100, I do not hear the first Playing 'vm_password' message (although do I hear the subsequent messages). And any numbers I enter on the ekiga keypad do not seem to be recognised by asterisk (I enabled debug rdp and rfc2833 packets appear to be transmitted and received). The softphone is set up to use RFC2833 for DTMF. I have a single x100p card installed which seems to work - to a fashion... Incoming calls are answered and the greeting is heard, but the line hangs up instantly the message finishes. (A different problem which I will investigate seperately unless someone has a quick answer). SIP Info? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.722: ast_channel_make_compatible failure
Ondrej Valousek wrote: My problem is, that the phone offering g722 could do alaw as well. I expected asterisk should just chose alaw for the communication - no transcoding is necessary then... That is not how Asterisk works, and is well known in the community as something that users would like to see changed, but has not yet been done. Asterisk negotiates the codecs (formats) for each call leg pretty much independently of the others, so if a G.722 endpoint initiates the first call leg, and the destination call leg cannot accept G.722, and there is no transcoder available, then the call will fail. If the non-G.722 endpoint initiates the first call leg then the call will likely go through, which is somewhat unfortunate :-) -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange Call problems on some numbers
I am having some really strange problems calling from 2 asterisk boxes of mine. One is version 1.2.22 the other 1.2.18. The problem is identical on both boxes. When I try to call certain numbers (8006375410, for instance) the call rings and rings and rings. Eventually the receiving end will pick up in the middle of an IVR as if I had been connected for some time already. When I call this number from a cell phone it connects normally. When I call this number from a sip phone connected to an asterisk box running 1.4 and out through a pri it connects normally. When I call it from these other two asterisk boxes out through whatever provider(my pri box, voip providers, whatever) it exhibits this behaviour.. Does anyone know what might be causing this? It's turning int a significant problem. Thanks Steve Glaus Peachnet Communications. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange Call problems on some numbers
Mail Lists wrote: I am having some really strange problems calling from 2 asterisk boxes of mine. One is version 1.2.22 the other 1.2.18. The problem is identical on both boxes. When I try to call certain numbers (8006375410, for instance) the call rings and rings and rings. Eventually the receiving end will pick up in the middle of an IVR as if I had been connected for some time already. When I call this number from a cell phone it connects normally. When I call this number from a sip phone connected to an asterisk box running 1.4 and out through a pri it connects normally. When I call it from these other two asterisk boxes out through whatever provider(my pri box, voip providers, whatever) it exhibits this behaviour.. Does anyone know what might be causing this? It's turning int a significant problem. Thanks Steve Glaus Peachnet Communications. Do you have an r in your outbound dial statement. If so, you will hear ringing until you get an answer event rather than the actual audio on the line. Some companies pass a little audio before technically answering. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] #modprobe wctdm or #modprobe zaptel
bilal ghayyad wrote: Hi list; I need to run the command modprobe wctdm and whenever I write it, then it gives me the following message: FATAL: Module wctdm not found FATAL: Error running install command for wctdm So, do I have to run that command from specific path? Or what is the problem? Any help? Regards Bilal Did you compile and install zaptel? Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial-Chain interrupted by Operator Called Party not reachable Messages
Christoph Adomeit wrote: Hi, I have the following problem: I want asterisk to dial a chain of n-numbers until somebody picks up the line. I am using Digium E1 Hardware (zaptel) for dialing out. Dialing a Chain is basically no problem, I use somwthing like: dial(no1,50) dial(no2,50) dial(no3,50) However, If no1 is not reachable, for example it is a mobile and switched off, then some automatic Operator-Voice from the Mobile-Telco says forever: This number is currently not reachable and this means asterisk thinks the call was succesfull and does not continue with the the other numbers. Does somebody has an idea how I can distinguish those Operator Voices from real calls ? This is one of the VERY few times the r option to Dial will be helpful. Dial(no1,50,r) etc. As long as the call is not answered (and the telco does not answer when they play that message) the r option will hide the audio the telco is sending. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Secondary Dialtone and selecting a specific line from Zap/g
The problem here, if it's a problem, is that it IS a POTS dialtone and
not asterisk's. So you can dial only what the telco lets you dial.
Al lists wrote:
Here is how i overcome this problem,
ignorpat = 9
exten = 9*,1,Dial(ZAP/1/w)
press 9* from your handset and after 1 second you have POTS line dial
tone on your phone,
On 10/3/07, *Mojo with Horan Company, LLC* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
It would be ugly, but you could prefix a zap channel or group number
before the phone number to dial. Using groups for an example:
exten = _*X*X.,1,Dial(ZAP/g${EXTEN:1:1}/${EXTEN:3})
exten = _*XX*X.,1,Dial(ZAP/g${EXTEN:1:2}/${EXTEN:4})
so dialing *4*18005551212 dials out over zap group 4...
bilal ghayyad wrote:
I need to select a line from the Zap group channel
using the SIP Phone (not FXO and not FXS ports).
ignorepat does not work?
Also, what is the method to let the second dial tone
has another tone frequency?
Regards
Bilal
No, ignorepat is for FXS ports (FXS ports use FXO
signaling). Also,
ignorepat does not apply to SIP phones, because SIP
phones provide
their
own dialtone, not a dialtone provided by Asterisk.
Al lists wrote:
Correction, on FXO port not FXS,
second, read his email first:
Also, how it will be possible to assign an
dedicated
line (connected to FXO) to an
button on the Polycom IP Phone or Broadtel IP Phone,
so if user select that button
then he will be sure that his outside call will be
via
that specific line.
Just assign a key on your phone to dial that
extension, and you will
have
dial tone on selected line,
then as a traditional PBX you can send any digits to
your provider.
On 10/1/07, Eric ManxPower Wieling [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
wrote:
ignorepat continues dialtone after a leading digit
has been dialed
on
FXS ports. How does ignorepat help this guy?
Al lists wrote:
ignorpat is your friend
On 9/30/07, Tzafrir Cohen
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
On Sun, Sep 30, 2007 at 02:34:01AM -0700, bilal
ghayyad wrote:
Dear List;
How can I place a call via Zap/g1 (group) but
need to
determine the line (FXO port)
that will go via it?
Simply don't use groups. Use channels directly.
To dial via the
specific
Zaptel channel NN, use Zap/NN
Am I missing anything?
Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now
(it's updated for today's economy) at Yahoo! Games.
http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation Provided by
http://www.api-digital.com-- http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange Call problems on some numbers
Also, so TFN's do not answer the line. Airline TFN are famous for doing this. Arick Davis -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, October 04, 2007 2:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Strange Call problems on some numbers Mail Lists wrote: I am having some really strange problems calling from 2 asterisk boxes of mine. One is version 1.2.22 the other 1.2.18. The problem is identical on both boxes. When I try to call certain numbers (8006375410, for instance) the call rings and rings and rings. Eventually the receiving end will pick up in the middle of an IVR as if I had been connected for some time already. When I call this number from a cell phone it connects normally. When I call this number from a sip phone connected to an asterisk box running 1.4 and out through a pri it connects normally. When I call it from these other two asterisk boxes out through whatever provider(my pri box, voip providers, whatever) it exhibits this behaviour.. Does anyone know what might be causing this? It's turning int a significant problem. Thanks Steve Glaus Peachnet Communications. Do you have an r in your outbound dial statement. If so, you will hear ringing until you get an answer event rather than the actual audio on the line. Some companies pass a little audio before technically answering. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting caller id value on outgoing calls using .call files
On Thu, Oct 04, 2007 at 04:10:09PM -0400, Arpit Mehta wrote: Hi all, I was looking at a way to add the caller id to the outgoing calls (which are made using .call files) using asterisk. Any ideas how to do this ? Currently I get 'Unknown' number displayed on my phone when asterisk makes an outgoing call. Also using something like this is not working as it still displays unknown number. I want set the callerid on the 1.call which is made. exten = _.,4,Set(CALLERID(all)=Joe 911) exten = _.,5,system(cp /var/spool/asterisk/1.call /var/spool/asterisk/outgoing/) Two things: 1. You use a copying. Which means: - Create /var/spool/asterisk/outgoing/1.call - Start writing content from original to new - close that 1.call If Asterisk catches the call file in the middle of copying, you have a problem. 2. _. may behave unexpectedly, I guess. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using PHP to reload extensions
Hello,
I am trying to use PHP to reload the extensions in an Asterisk
installation. I keep getting this error:
Easiest way without compromising security or changing permissions. Use
the AMI.
1. Download phpagi (Just google it)
2. Use it to connect to the Manager interface
3. Use it to issue:
Action: Command
Command: reload
Here's the two functions I've been using. Add a new user to
manager.conf with the appropriate permissions and this should work
fine.
Hope that helps,
-Michael
function extensions_reload() {
return asterisk_exec(dialplan reload);
}
function asterisk_exec($cmd, $output=NULL) {
$token = md5(uniqid(rand()));
$errno = 0;
$errstr = 0;
$fp = fsockopen(localhost, 5038, $errno, $errstr, 20);
if (!$fp) {
return 1;
}
fputs($fp, Action: login\r\n);
fputs($fp, Username: newusername\r\n);
fputs($fp, Secret: newpassword\r\n);
fputs($fp, Events: off\r\n\r\n);
usleep(500);
fputs($fp, Action: COMMAND\r\n);
fputs($fp, command: $cmd\r\n);
fputs($fp, ActionID: $token\r\n\r\n);
usleep(500);
$out = fread($fp, 38000);
while(strpos($out,--END COMMAND--)==0) {
$out .= fread($fp, 38000);
}
fclose ($fp);
$out = substr($out, strpos($out, ActionID));
$out = substr($out, strpos($out, \n) + 1);
$out = substr($out, 0, strpos($out, --END COMMAND--) - 1);
$output = $out;
return 0;
}
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Where to download Junghanns ISDNguard software?
What if you don't use or want to use bristuff? We use Digium PRI cards and don't need any of the BRIstuff On 10/4/07, Sylvain Boily [EMAIL PROTECTED] wrote: Hi Nick, For using ISDNGuard, you can using res_watchdog from the bristuff patch. I attach for you a version who have patched and work for asterisk 1.2.24. It's an archive with a makefile with just res_watchdog. We have some clients who using this version from many month ago. Sylvain Le jeudi 04 octobre 2007 à 08:51 +1000, Nick Richardson a écrit : Hi list, I recently purchased an ISDNguard from Junghanns. It came with no software and there is no sign on their website or in any of their documentation where to download it. I have looked in http://www.junghanns.net/downloads/ and there is no sign of it there either. The only thing remotly close ther is isdnguard-asterisk-1.2.13.patch. Their documentation refers to /usr/sbin/ISDNguard. Where does one get this mysterious binary from? I have emailed their support a few times and get no response, needless to say I am NOT a happy customer. Can anyone help me with a download link? Thanks in advance.. - Nick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sylvain BOILY Proformatique - 67 rue Voltaire - 92800 Puteaux Tel. : 01 41 38 99 64 - Fax. : 01 41 38 99 70 Email : [EMAIL PROTECTED] - http://proformatique.com/ Vers un monde plus libre Proformatique est membre de l'ASS2L http://www.ass2L.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No audio on Zap (T1/PRI) channels
Steve Totaro wrote: Steve Edwards wrote: I have 12 T1's going into 3 servers, 4 in each into Digium, Inc. Wildcard TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02) cards. Each group of T1's have the primary D on 24 and the secondary D on 96. The first server (ts20) and the last server (ts22) can playback demo-congrats fine. The middle server (ts21) cannot -- just dead air. If I call via ZAP, dead air. If I call via IAX, I hear the file. I copied /etc/zaptel.conf, /etc/asterisk/*, /var/lib/asterisk/sounds/demo-congrats.gsm from ts20 to ts21 -- no joy. I have seen this in my system log file: Oct 2 18:41:49 WARNING[7477]: chan_zap.c:8087 zt_pri_error: [Span 0 D-Channel 0] PRI: !! Got reject for frame 95, but we have nothing -- resetting! I'm running asterisk-1.2.24, asterisk-addons-1.2.7, libpri-1.2.5, zaptel-1.2.20.1. show channel zap/?, zap show channel ? appear identical between working and non-working systems both on-hook and off-hook. Any clues or clues where to start looking? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 Double check both zaptel.conf and zapata.conf and also call the telco to make sure they have they have the same NFAS scheme on all T1s setup correctly. Sometimes (let's face it, alot of times, the provider messes something up). Also check that all of your T1 cables are plugged into the correct T1 port. I have made that mistake myself when doing 28 T1s off a T3. I got dead air just as you described. Yes, if you are running NFAS, getting dead air on a call is a symptom of not having the logical span identifier correctly corresponding to the physical span you have plugged in (spanmap option in zapata.conf, IIRC). -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk status in Debian
Hello, This is a update on the current status of Asterisk in Debian. Apologies for the really long mail, it is targetted both to users and maintainers :) I'm Ccing asterisk-users as a one-time thing; users that are interested can subscribe to our list[1] for updates to prevent noise on a non-Debian list. Please Cc pkg-voip-maintainer on replies. 1: http://lists.alioth.debian.org/mailman/listinfo/pkg-voip-maintainers sarge/etch status - About a month ago, I fixed all the long-standing knownn vulnerabilities in both sarge/oldstable (1.0.7) and etch/stable (1.2.12). The updates are present security.debian.org a Advisory has been released (DSA-1358[1]), thanks to Debian's Security Team. These updates are fixing CVE-2007-1306, CVE-2007-1561, CVE-2007-2294, CVE-2007-2297, CVE-2007-2488, CVE-2007-3762, CVE-2007-3763 and CVE-2007-3764 (...). 1: http://www.debian.org/security/2007/dsa-1358 lenny status 1:1.4.11~dfsg-4 has been recently uploaded to unstable. The previously mentioned block by the openh323 dependency which currently fails to build in unstable (binutils bug: #440015) has been workaround-ed (by having less strict shlibs in openh323) From our POV, it's a good candidate for lenny/testing. However: - it depends on perl and net-snmp versions that are not present in testing and are not in a shape to be there; we'll need new versions from the respective teams. - asterisk needs to go together with yate because of a shared libpri dependency. However yate is being blocked[2] by gtk+2.0. - more importantly, asterisk produces an Internal Compiler Error of GCC 4.2 on hppa (#445336). Until it builds successfully there, it cannot migrate to testing. 1: http://bjorn.haxx.se/debian/testing.pl?package=asterisk 2: http://bjorn.haxx.se/debian/testing.pl?package=yate 1.4.12 -- Digium released 1.4.12 the day before yesterday. I have committed all the changes needed and we are now up to date. Fortunately, many of our fixes that I reported upstream have been merged. I have manually ported bristuff 0.4.0-test4 to 1.4.12; it needed many changes compared to the previous upstream updates. I will forward my changes to kapejod so that he can hopefully release a new version. supplementary packages -- * asterisk-addons (-mp3, -mysql, -ooh323c) are finally present in Debian and should be ready to migrate to lenny after Asterisk does. Digium released a new version along with 1.4.12 and I will update this ASAP. * asterisk-chan-capi, asterisk-spandsp-plugins, asterisk-oh323 had recents uploads and all are in a good shape. * I am going to drop rate-engine from the archive (#444712) since it has no users, it wasn't released with etch, has open bugs for a really long time and is unmaintained by upstream. * I tried compiling chan_misdn together with the mISDN maintainer (Simon Richter) and failed because of an mISDN API mismatch. Need to take another look. * asterisk-gui needs to be uploaded; Tzafrir? * are we going to upload ARI? If not, we should drop it from our SVN. * zaptel is in a good status and it's the only package from the suite that is migrating to testing. Things TODO that come to mind are: a) fixing a bug which results in /lib/modules/2.6.foo/modules.* files in amd64 and b) evaluate a switch to OSLEC as the default echo cancellator. Tzafrir is doing an excellent job on maintaining this package by himself :) * Right now, we are shipping asterisk-sounds-main which is the main asterisk sounds in English in GSM format -- exactly as shipped in the original tarball by Digium. Kilian, Tzafrir and me were pondering on the idea of shipping separately all sounds as shipped by Digium in all formats (besides WAV), each in a separate package. This should serve our users better but has an obvious problem of size. This is not decided yet. ABI issues -- Most -if not all- of these plugins build-depend on asterisk-dev i.e. use Asterisk's development headers. These headers are tied to the ABI and this can only be expressed in dependencies manually. asterisk-chan-capi was compiled with 1.2 asterisk-dev, had a = 1.2 dependency but segfaults on 1.4 (#441237). There are currently no similar problems that I know of. However, we should expect more of these when we transition to 1.6 which will most probably have a different ABI. I'm leaning towards a solution: * Add a Provides: asterisk-1.4 to asterisk. * Replace Depends: asterisk (= 1.4.0) (or similar) with Depends: asterisk-1.4 on all external modules. This should help in *breaking*, dpkg-wise, the modules when a new version is uploaded which in turn will prevent a new version from entering testing until all plugins are recompiled. pushing our work upstream - On the 1.4.11-1.4.12 cycle, I tried pushing all of our patches to Digium's BTS (mantis). This has worked well since they're quite responsive (contrary to our secondary upstream, Klaus-Peter
Re: [asterisk-users] Where to download Junghanns ISDNguard software?
Le vendredi 05 octobre 2007 à 12:13 +1000, Nick Richardson a écrit : What if you don't use or want to use bristuff? We use Digium PRI cards and don't need any of the BRIstuff yep i know ... but res_watchdog is part of bristuff. My tarball have just the res_watchdog module with a makefile. It's not necessary to using bristuff :-) Sylvain On 10/4/07, Sylvain Boily [EMAIL PROTECTED] wrote: Hi Nick, For using ISDNGuard, you can using res_watchdog from the bristuff patch. I attach for you a version who have patched and work for asterisk 1.2.24. It's an archive with a makefile with just res_watchdog. We have some clients who using this version from many month ago. Sylvain Le jeudi 04 octobre 2007 à 08:51 +1000, Nick Richardson a écrit : Hi list, I recently purchased an ISDNguard from Junghanns. It came with no software and there is no sign on their website or in any of their documentation where to download it. I have looked in http://www.junghanns.net/downloads/ and there is no sign of it there either. The only thing remotly close ther is isdnguard-asterisk-1.2.13.patch. Their documentation refers to /usr/sbin/ISDNguard. Where does one get this mysterious binary from? I have emailed their support a few times and get no response, needless to say I am NOT a happy customer. Can anyone help me with a download link? Thanks in advance.. - Nick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sylvain BOILY Proformatique - 67 rue Voltaire - 92800 Puteaux Tel. : 01 41 38 99 64 - Fax. : 01 41 38 99 70 Email : [EMAIL PROTECTED] - http://proformatique.com/ Vers un monde plus libre Proformatique est membre de l'ASS2L http://www.ass2L.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sylvain BOILY Proformatique - 67 rue Voltaire - 92800 Puteaux Tel. : 01 41 38 99 64 - Fax. : 01 41 38 99 70 Email : [EMAIL PROTECTED] - http://proformatique.com/ Vers un monde plus libre Proformatique est membre de l'ASS2L http://www.ass2L.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
