Re: [asterisk-users] TE121 on Asterisk

Yeah thanks a lot, It worked that way... > Oguzhan Kayhan schreef: Oguzhan Kayhan wrote: > I want to change it to E1 instead of T1. > here comes the problem. > > If it's anything like the older cards, there is a jumper on the card that sets it to T1/E1 >>>

Re: [asterisk-users] codec_dahdi and Asterisk 1.6.0.6

2009/2/26 Brandon B. > I've got a question about codec_dahdi witrh a system running Asterisk > 1.6.0.6 and DAHDI 2.1.0.4 with a TE410P card. The system is used primary to > route calls between different PRI connections, so no transcoding between > codecs is happening as far as I am aware. > > 1)

Re: [asterisk-users] TE121 on Asterisk

On Thu, Feb 26, 2009 at 08:41:25AM +0200, Oguzhan Kayhan wrote: > >> Oguzhan Kayhan wrote: > >>> I want to change it to E1 instead of T1. > >>> here comes the problem. > >>> > >> > >> If it's anything like the older cards, there is a jumper on the card > >> that sets it to T1/E1 > >> > >> Doug > >>

[asterisk-users] Need US Dialing Account with Asterisk

Hello, We need a US Dialing Plan to use it with Asterisk. We need 3 - 4 Channels with it. Please suggest some good voice quality service. Regards, Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email: kas...@haditel

Re: [asterisk-users] TE121 on Asterisk

> On Thu, Feb 26, 2009 at 08:41:25AM +0200, Oguzhan Kayhan wrote: >> >> Oguzhan Kayhan wrote: >> >>> I want to change it to E1 instead of T1. >> >>> here comes the problem. >> >>> >> >> >> >> If it's anything like the older cards, there is a jumper on the card >> >> that sets it to T1/E1 >> >> >> >

Re: [asterisk-users] CDR - Asterisk-Stat and PHP5

Tiago Durante wrote: > Hi all, > > I don't know if its the right place to ask, but... Does any one have > the asterisk-stat-v2 running with PHP5? > > > Tks! > > # php --version PHP 5.2.0-8+etch13 (cli) (built: Oct 2 2008 08:26:18) Copyright (c) 1997-2006 The PHP Group Zend Engine v2.2.0, Co

Re: [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6

amit mehta wrote: > Hello Users, > > Is anyone aware about a solution to call incoming number and dictate the > files by using Dictate feature of Asterisk used for Medical > Transcription industry. I guess nobody will read your email as you: 1. hijacked a thread (http://www.internet-descriptio

[asterisk-users] Dictate

Hello Members, Sorry for hijacking the earlier thread and asking the question last time. Is anyone aware about a solution to call incoming number and dictate the files by using Dictate feature of Asterisk used for Medical Transcription industry. Thanks & Regards, Amit Mehta _

Re: [asterisk-users] Swichted digits on received number from fax on an fxs port

Does anyone have ideas on how to debug this? I connected a different and old fax machine and everything works fine. But for sure the other fax machine works fine too if I connect it to an analog line from the telco. Si it some incompatibility between my setup (asterisk, xorcom) and the fax ma

Re: [asterisk-users] Swichted digits on received number from fax on an fxs port

On Tue, Feb 24, 2009 at 10:15:27AM +0100, Loic Didelot wrote: > Hello, > I have connected a fax machine to a xorcom fxs port (signalling=fxo_ks). > But when I try to dial (send a fax) the number that I receive in > asterisk is wrong. Quite often a few digits are wrong but sometime is > correct. It

Re: [asterisk-users] HP DL380 G5 with TE420

On Thu, Feb 26, 2009 at 11:25 AM, Hans Konings wrote: > I've tried with a different server (HPdl320g5p) and the card is detected in > this but the cards generate NMI errors on many bootups. > Does anybody have this combination of hardware working? Or can anybody > think of something I've missed?

[asterisk-users] HP DL380 G5 with TE420

Hi I'm having problems getting the TE420 working in HP DL380G5 servers. The cards don't seem to be detected 100% by the BIOS. With two cards in the server they are never detected. things I've tried: 1 Update firmware to latest (P56) for the server 2 change irq settings 3 disable all onboard dev

[asterisk-users] call-limit on a per destination basis

Hello, I use asterisk to to IAX2 trunking between London POP & Reunion Island pop. I would like to know if it's possible to do a kind of call-limit (i.e. restrict to XX) channels but on a per dialcode and / or destination basis. For example: [trunk] ; reunion proper, i want to send no more than

Re: [asterisk-users] HP DL380 G5 with TE420

Hans Konings schreef: Hi I'm having problems getting the TE420 working in HP DL380G5 servers. The cards don't seem to be detected 100% by the BIOS. With two cards in the server they are never detected. things I've tried: 1 Update firmware to latest (P56) for the server 2 change irq settings

Re: [asterisk-users] call-limit on a per destination basis

I have no clue about IAX, but if IAX does not support it you can program it yourself using the GROUP and GROUPCOUNT functions. regards klaus Jean-Michel Hiver wrote: > Hello, > > I use asterisk to to IAX2 trunking between London POP & Reunion Island > pop. I would like to know if it's possible

Re: [asterisk-users] Swichted digits on received number from fax on an fxs port

Yes, its the dialed number. I have asterisk 1.4.21.1 and zaptel-1.4.11. Reading your question I am pretty sure you advise me to update. Loic On Thu, 2009-02-26 at 12:44 +0200, Tzafrir Cohen wrote: > On Tue, Feb 24, 2009 at 10:15:27AM +0100, Loic Didelot wrote: > > Hello, > > I have connected a f

Re: [asterisk-users] DID's in a specific rate center

Look it up at www.localcallingguide.com. The name on record there may be the LECs name -- then you need to find a SIP provider or DID handler that has a business relationship with that LEC. For example, in the state of Iowa Vonage obtained (at least some of) it's numbering resources from McCleod.

Re: [asterisk-users] DID's in a specific rate center

As always, heed should be taken that number pooling and LNP have made the idea that someone "owns" a rate center increasingly meaningless. See the first or second question on our VoIP FAQ for more information @ www.evaristesys.com. It is related. I would paste direct URL but am not at comput

Re: [asterisk-users] HP DL380 G5 with TE420

On Thu, Feb 26, 2009 at 11:49 AM, stoffell wrote: > On Thu, Feb 26, 2009 at 11:25 AM, Hans Konings wrote: > >> >> > We have it working on a dl320, several ML350's, ml310, but never tried on a > dl380 yet. > > We had serious issues in the past when iLO was enabled on a 350. disabling > iLO on tha

Re: [asterisk-users] Multiple SIPGate accounts.

On Wed, 25 Feb 2009, Klaus Darilion wrote: > I supsect that the incoming request is tried to match against a peer - > based on IP:port. Thus, it will always match the same peer, regardsless > if the call is incoming from account1 or account2. > > Try using the same context in both peer definitions

Re: [asterisk-users] DID's in a specific rate center

Ok, then you can contact one of the myriad of providers to see if you can get a block of numbers. Keep in mind, there will be differences in quality and reliability depending on which provider you use and on whether or not you use PRI or pure VoIP. Since you already have the outbound capacity

Re: [asterisk-users] DID's in a specific rate center

According to Wikipedia no law was actually ever passed. I don't know if there is anything that falls under the FCC rules. In any event it would be unethical and evidence of fraudulent intent if one was trying to defraud someone in the process of doing so. Jason Aarons (US) wrote: > Any idea w

Re: [asterisk-users] Patton 5.3. How to get incoming calls ? [SOLVED]

Hi, Changing the line bellow helped to get incoming calls but I add to remove secret= option in sip.conf (otherwise, Patton wouldn't respond to 407 Auth required challenges). If someone could enable secret and still get incoming calls (in any SmartWare 5.X), please, do not hesitate to share here .

Re: [asterisk-users] HP DL380 G5 with TE420

Hans Konings escribió: > Hi > > I'm having problems getting the TE420 working in HP DL380G5 servers. > > The cards don't seem to be detected 100% by the BIOS. With two cards > in the server they are never detected. > Did you test the TE420 card on another server? It may be a defective card... --

[asterisk-users] Getting SIP field P-Asserted-Identity from EAGI

Hi, using EAGI variables like agi_request agi_channel agi_language agi_type agi_uniqueid agi_callerid agi_dnid agi_rdnis agi_context agi_extension agi_priority agi_enhanced agi_accountcode I get a lot of data about a call, but I need to obtain P-Asserted-Identity value from a SIP call. Are tehe

[asterisk-users] Congestion Tone

Hello! I?ve connected an avaya PABX with an asterisk box through h323, all calls from Avaya are sended to the asterisk. What I need is send to the AVAYA PABX a congestion tone when Zap channels are full. How I do it?Thanks for any idea! Gustavo A. González Dto. de Infraestructura Despegar

[asterisk-users] ATA recommendation (wih FTP provisioning)

Hi, I am looking for a good ATA recommendation, ideally something: 1) with one FXS and one LAN port (so it's as inexpensive as possible) 2) That can be provisioning using FTP (configuration and firmware upon reboot, ideally remote reboot from a sip notify) 3) Supports T.38 Nice

Re: [asterisk-users] Problems with Outbound Calls

hi you should first solve this Warning [2630]: config.c:768 process_text_line: Unknown Directive at line 231 of /etc/asterisk/../zaptel.conf check what do you have in the line 231 of your zaptel.conf file. David 2009/2/26 Wye-khe Kwok > Hi everyone! > > I'm quite a newbie at this Asterisk stu

Re: [asterisk-users] ATA recommendation (wih FTP provisioning)

I used the Patton M-ATA a year or so ago and it was a piece of junk. T.38 didn't work and there was no way to troubleshoot why it didn't work. Also, the web interface was horrible. Mike wrote: > Hi, > > > > I am looking for a good ATA recommendation, ideally something: > > > > 1) with o

Re: [asterisk-users] Getting SIP field P-Asserted-Identity from EAGI

Based on this page - http://blog.herbertm.ca/2007/09/03/extracting-dids-from-the-sip-header You could put this in your dialplan [ext-did-custom] exten => s,1,Set(ASSERT="${SIP_HEADER(Call-ID)}") exten => s,2,AGI(xxx.pl,${ASSERT}) _ From: asterisk-users-boun...@lists.digium.co

Re: [asterisk-users] ATA recommendation (wih FTP provisioning)

Thanks for the comment, it's good to know. Mike > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of pe...@networkoblivion.com > Sent: Thursday, February 26, 2009 9:16 > To: Asterisk Users Mailing List - Non-

[asterisk-users] Odd Read App Issues

So I'm using the READ() application within an IVR, and having a strange issue, and wondering if anyone else has had this problem. When calling from an outside line, and entering the digits during the read() part of my dialplan, it's accepting some of the digits twice, though it's only keyed in

Re: [asterisk-users] Gosub behavior change <=1.6.0.5 to 1.6.0.6

Klaus Darilion writes: > What about a config option >gototriggersinvalid=yes (default=no) > in extensions.conf for users which are using this feature? Please, no more options. There are way too many options already. Since I personally believe the use of the special extensions should be limi

Re: [asterisk-users] Odd Read App Issues

Btw, I'm using Asterisk SVN-branch-1.4-r178640 Robert Broyles wrote: > So I'm using the READ() application within an IVR, and having a strange > issue, and wondering if anyone else has had this problem. > > When calling from an outside line, and entering the digits during the > read() part of m

Re: [asterisk-users] Getting SIP field P-Asserted-Identity from EAGI

Can you access channel functions from EAGI? SIP_HEADER(P-Asserted-Identity) contains what you need. It would be nice if Asterisk supported it natively and forgot about the deprecated Remote-Party-ID header. /Benny ___ -- Bandwidth and Colocation Pro

Re: [asterisk-users] ATA recommendation (wih FTP provisioning)

Mike writes: > 2) That can be provisioning using FTP (configuration and firmware upon > reboot, ideally remote reboot from a sip notify) What's wrong with HTTP? > I have been recommended the PAP2T in the past, and although I have used it > and sort of liked it, it wasn't possible to provision i

Re: [asterisk-users] Odd Read App Issues

Robert Broyles wrote: > So I'm using the READ() application within an IVR, and having a strange > issue, and wondering if anyone else has had this problem. > > When calling from an outside line, and entering the digits during the > read() part of my dialplan, it's accepting some of the digits tw

Re: [asterisk-users] Problems with Outbound Calls

On Thu, Feb 26, 2009 at 11:50:57AM -0200, David fire wrote: > hi > you should first solve this > > Warning [2630]: config.c:768 process_text_line: Unknown Directive at > line 231 of /etc/asterisk/../zaptel.conf In zaptel.conf it is perfectly legal to have lines beginning with a '#'. With Asteris

Re: [asterisk-users] Odd Read App Issues

Okay. I'm using this all over SIP Trunking with Vitelity. Any other suggestions? -- Regards, Robert Broyles Eric Wieling, Asteria Solutions Group wrote: Robert Broyles wrote: So I'm using the READ() application within an IVR, and having a strange issue, and wondering if anyone else has h

[asterisk-users] [cdr_odbc] error: Cannot insert the value NULL into column 'calldate'

Hi, I am trying to get * log to mssql server. I have odbc and freetds configured, but my insert query is missing calldate which is a NOT NULL field in database schema. cdr_adaptive_odbc: Insert failed on 'sqlserver:cdr'. CDR failed: INSERT INTO cdr (clid,src,dst,dcontext,channel,lastapp,lastdata

Re: [asterisk-users] ATA recommendation (wih FTP provisioning)

On Thu, 26 Feb 2009, Mike wrote: > Hi, > > > > I am looking for a good ATA recommendation, ideally something: > > > > 1) with one FXS and one LAN port (so it's as inexpensive as possible) > > > > 2) That can be provisioning using FTP (configuration and firmware upon > reboot, ideally remote rebo

Re: [asterisk-users] Odd Read App Issues

Robert Broyles wrote: > Okay. I'm using this all over SIP Trunking with Vitelity. > Any other suggestions? Sorry for wasting your time. -- Eric Wieling * Asteria Solutions Group * Huntsville, AL Call centers * IVRs * Enterprise PBXs * Conferencing applications 256-705-0277 * http://www.asterias

Re: [asterisk-users] Getting SIP field P-Asserted-Identity from EAGI

On Thursday 26 February 2009 08:39:29 Benny Amorsen wrote: > Can you access channel functions from EAGI? > > SIP_HEADER(P-Asserted-Identity) contains what you need. > > It would be nice if Asterisk supported it natively and forgot about > the deprecated Remote-Party-ID header. Of course you can.

Re: [asterisk-users] Dictate

amit mehta wrote: > Hello Members, > > Sorry for hijacking the earlier thread and asking the question last time. > > Is anyone aware about a solution to call incoming number and dictate > the files by using Dictate feature of Asterisk used for Medical > Transcription industry. > > Thanks & Rega

Re: [asterisk-users] DID's in a specific rate center

>> What is illegal is to set caller-id to a fraudulent value such that the >> person on the other end will not be able to correctly identify the >> originator of the call. >I don't know if there is anything that falls under the FCC rules. In any >event it >would be unethical and evidence of

Re: [asterisk-users] Odd Read App Issues

Not at all. In fact, I found that relaxdtmf=yes is now available for sip.conf as of 1.4 as well. However, that didn't resolve the problem. -- Regards, Robert Broyles Eric Wieling, Asteria Solutions Group wrote: Robert Broyles wrote: Okay. I'm using this all over SIP Trunking with Vitel

Re: [asterisk-users] Getting SIP field P-Asserted-Identity from EAGI

Thanks to all, I'll try it! On Thu, Feb 26, 2009 at 12:25 PM, Danny Nicholas wrote: > Based on this page - > http://blog.herbertm.ca/2007/09/03/extracting-dids-from-the-sip-header > > > > You could put this in your dialplan > > > > [ext-did-custom] > > exten => s,1,Set(ASSERT="${SIP_HEADER(Cal

Re: [asterisk-users] CDR - Asterisk-Stat and PHP5

Hi Tiago. I've it working on PHP 5.2.6 but only after having modified the php.ini default configuration keys: zend.ze1_compatibility_mode = Off short_open_tag = Off setting together to On and restarting apache forces PHP5 to behave like PHP 4.x version. regards, Marco Signorini ===

Re: [asterisk-users] Odd Read App Issues

Robert Broyles wrote: Okay. I'm using this all over SIP Trunking with Vitelity. Any other suggestions? -- Regards, Robert Broyles Eric Wieling, Asteria Solutions Group wrote: Robert Broyles wrote: So I'm using the READ() application within an IVR, and having a strange issue, and wonde

Re: [asterisk-users] DTMF tones mid conversation

Hi list, I wonder if anyone is able to offer any [polite ;-)] words of wisdom?? I'm having the same problem too, with random DTMF heard during ISDN calls. Am running Asterisk 1.4.22 and mISDN 1.1.8. (kernel 2.6.18 CentOS) Symptoms are on incoming and out outgoing ISDN calls, particular

Re: [asterisk-users] Odd Read App Issues

Yea, I tried that too. I have it: dtmfmode=rfc2833 -- Regards, Robert Broyles Brent Davidson wrote: Robert Broyles wrote: Okay. I'm using this all over SIP Trunking with Vitelity. Any other suggestions? -- Regards, Robert Broyles Eric Wieling, Asteria Solutions Group wrote: Robert Bro

Re: [asterisk-users] Dictate

Hi Brent, Thanks for the help. I am working to achieve the middle solution. The scenario that i am trying to achieve : Customer calls in the tollfree number: Enter the ID and press"#" Pwd: # Press * to begin dictate,4 to pause and 2 to start again. PRess * to end and start new Upto this has bee

Re: [asterisk-users] ATA recommendation (wih FTP provisioning)

Just beware that most PAP2s on places like E-Bay are from Vonage. They lock the things up quite seriously. There are procedures out there to unlock them but it requires stuff like setting up an isolated LAN with a DNS server and FTP server and a special file. If someone would make a live boot

Re: [asterisk-users] ATA recommendation (wih FTP provisioning)

On Thu, 26 Feb 2009, Wilton Helm wrote: > Just beware that most PAP2s on places like E-Bay are from Vonage. They > lock the things up quite seriously. There are procedures out there to > unlock them but it requires stuff like setting up an isolated LAN with a > DNS server and FTP server and

[asterisk-users] Residential portals and real world scalability

Hey all, I have a couple of questions. 1) What is the maximum amount of registrations and ongoing calls you've been able to achieve on your Asterisk systems. Please do not respond with marketing hyperbole. I'm looking for real world implementation in the thousands range. For instance, max I recal

Re: [asterisk-users] CDR - Asterisk-Stat and PHP5

Marco Signorini wrote: > Hi Tiago. > > I've it working on PHP 5.2.6 but only after having modified the php.ini > default configuration keys: > > zend.ze1_compatibility_mode = Off > short_open_tag = Off Though my zend.ze1_compatibility_mode is set to Off, short_open_tag is set to On and it is wo

[asterisk-users] Can't build today's AGX Asterisk Addon with spandsp0.0.6pre3 or 4

Hi, With 0.0.6pre3: # ./build.sh CMake Warning (dev) in CMakeLists.txt: No cmake_minimum_required command is present. A line of code such as cmake_minimum_required(VERSION 2.6) should be added at the top of the file. The version specified may be lower if you wish to support older CMa

Re: [asterisk-users] Can't build today's AGX Asterisk Addon with spandsp0.0.6pre3 or 4

I must add I tried spandsp0.0.6xxx as a warning message advised me to do so (using 0.0.4 would be ok for me but current trunk doesn't allow this anymore, it seems). 2009/2/26 Olivier > Hi, > > With 0.0.6pre3: > # ./build.sh > CMake Warning (dev) in CMakeLists.txt: > No cmake_minimum_required

Re: [asterisk-users] ATA recommendation (wih FTP provisioning)

2009/2/26 pe...@networkoblivion.com > I used the Patton M-ATA a year or so ago and it was a piece of junk. > T.38 didn't work and there was no way to troubleshoot why it didn't > work. Have you tried again recently ? I've just tried its T.38 capabilities and I'm not successful yet. > Also, t

Re: [asterisk-users] ATA recommendation (wih FTP provisioning)

2009/2/26 Mike > a) PoE powered > Unfortunately, I don't think any PoE powered exists, AFAIK. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://list

Re: [asterisk-users] AGI problem using mono (.Net)

Steve: = Thanks for the info on the agi debug command. We'll see what information we can garner with that. Thanks also for the advanced logging info. Unfortunately, we are pretty aware of how AGI works (at least at the level that you explained it). Thanks for the illustrati

[asterisk-users] incoming call problem

Dear All, I have created an inbound context in SIP .conf that forward incoming call to opensips server...The problem appears as soon as I enable t38pt_udptl = yes...The Asterisk negotiate the SIP session with OpenSIPS without adding voice codec to INVITE packet...It just contains T.38 protocol...Wh

Re: [asterisk-users] AGI problem using mono (.Net)

An update here. Yesterday's problem has been solved (partly). After looking closer at the results of my perl script, as well as looking at the documentation, I realized that we were leaving the escape character argument off of the STREAM FILE command in the mono application. After appending "" to t

Re: [asterisk-users] DTMF tones mid conversation

On Thu, Feb 26, 2009 at 6:08 PM, Simon Dixey wrote: > I wonder if anyone is able to offer any [polite ;-)] words of wisdom?? > I can be polite, I'm not sure about the wisdom .. :-) > DTMF threshold in misdn-init look high doesn't it... I'm not entirely sure > what it "should" be set to, to be h

Re: [asterisk-users] CDR - Asterisk-Stat and PHP5

On Thu, Feb 26, 2009 at 12:52 PM, Paulo Santos wrote: > Marco Signorini wrote: >> Hi Tiago. >> >> I've it working on PHP 5.2.6 but only after having modified the php.ini >> default configuration keys: >> >> zend.ze1_compatibility_mode = Off >> short_open_tag = Off > > Though my zend.ze1_compatibil

[asterisk-users] asterisk 1.4.23.1 and mISDN 1.1.8 segfaults

hi all, I'm have a bit of a hard time with some segfaults on running 1.4.23.1 and mISDN 1.1.8. I already enabled " DONT_OPTIMIZE" and " DEBUG_THREADS" in asterisk so I can now generate a bt. I did that (following the instructions on voip-info) but I'm not sure how to "read' the output now. By loo

Re: [asterisk-users] Odd Read App Issues

I turned on DTMF debugging. It looks like the extra digits coming in are less than the minimum duration of 100ms Anyone know how to force that minimum duration? [Feb 26 12:15:07] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '1' received on SIP/carrier-c4022740 [Feb 26 12:15:07] DTMF[3564]

Re: [asterisk-users] Problems with Outbound Calls

sorry but how do you know the warning is from an # ? he only post this from zaptel.conf span = 1,1,0,esf,b8zs bchan = 1-23 dchan = 24 loadzone = jp defaultzone = jp David 2009/2/26 Tzafrir Cohen > On Thu, Feb 26, 2009 at 11:50:57AM -0200, David fire wrote: > > hi > > you should first solve th

[asterisk-users] Friday Feb 27th at 12 Noon EST: Polycom Applications

Hi, We're moving ahead on this, there are some interesting files to check out thanks to davevg's perl skills. If you find a way to write an interesting application to run on a Polycom, you can win a Polycom IP 450. Even if you don't care about that, you may be interested in knowing more about prog

Re: [asterisk-users] Odd Read App Issues

Robert Broyles wrote: I turned on DTMF debugging. It looks like the extra digits coming in are less than the minimum duration of 100ms Anyone know how to force that minimum duration? [Feb 26 12:15:07] DTMF[3564]: channel.c:2280 __ast_read: DTMF begin '1' received on SIP/carrier-c4022740 [Feb

[asterisk-users] Current state of Asterisk and Virtualization?

Hi all, In a pure VoIP env, what is the current state of do's and don't s of virtualizing * in order to provide multiple separate instances, say for hosting lots of Asterisk-gui/FreePBX/a-n-other gui? I've read lots of threads going back to 2007 and I'm in the general option that kvm is the way t

Re: [asterisk-users] ATA recommendation (wih FTP provisioning)

Nope, I gave up and haven't gone back. Not worth the hassle. We use the SPA2100/SPA2102 and they work great for providing analog/fax lines as they support T.38. Olivier wrote: > > 2009/2/26 pe...@networkoblivion.com > mailto:pe...@networkoblivion.com>> > >

Re: [asterisk-users] AGI problem using mono (.Net)

On Thu, 26 Feb 2009, Douglas Mortensen wrote: > New problem: > == > Although we see the following in our logs / asterisk console: [Feb 26 > 11:11:05] VERBOSE[9824] logger.c: -- Playing 'filename' > (escape_digits=) (sample_offset 0) > > All that the caller hears is a very brief clic

[asterisk-users] changing /etc/dahdi/system.conf

At the top of my /etc/dahdi/system.conf file is this line: # Autogenerated by /usr/sbin/dahdi_genconf on Wed Feb 25 18:25:10 2009 -- do not hand edit OK, so how do I adjust the timing source and LBO numbers, and echo cancellers if I'm not supposed to edit this file? Brandon.

Re: [asterisk-users] dahdi wcb4xxp and fax

2009/2/25 stoffell > Hi all, > > I wanted to switch from my current setup (mISDN) to the native dahdi with > b410p support (wcb4xp). All works fine for normal phone calls but not for > faxing. Faxes are distorted, if arriving at all, and hylafax logs the usual > bad stuff (HDLC frame not byte-ori

Re: [asterisk-users] changing /etc/dahdi/system.conf

>At the top of my /etc/dahdi/system.conf file is this line: > >    # Autogenerated by /usr/sbin/dahdi_genconf on Wed Feb 25 18:25:10 2009 -- >do not hand edit > >OK, so how do I adjust the timing source and LBO numbers, and echo cancellers >if I'm not supposed to edit this file? Well, if you han

Re: [asterisk-users] changing /etc/dahdi/system.conf

On Thu, Feb 26, 2009 at 01:44:19PM -0700, Brandon B. wrote: > At the top of my /etc/dahdi/system.conf file is this line: > > # Autogenerated by /usr/sbin/dahdi_genconf on Wed Feb 25 18:25:10 2009 > -- do not hand edit > > OK, so how do I adjust the timing source and LBO numbers, and echo > ca

Re: [asterisk-users] changing /etc/dahdi/system.conf

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Brandon B. wrote: > At the top of my /etc/dahdi/system.conf file is this line: > > # Autogenerated by /usr/sbin/dahdi_genconf on Wed Feb 25 18:25:10 > 2009 -- do not hand edit > > OK, so how do I adjust the timing source and LBO numbers, and echo

Re: [asterisk-users] Dropping RTP packets

You need "canreinvite=no" in the config for your sip phone and the veracity connection, otherwise Asterisk will just mediate the call setup then try to allow the sip phone and veracity to talk directly to one another. Jim Dickenson wrote: > I have a SIP phone at home behind a NAT router registe

Re: [asterisk-users] ATA recommendation (wih FTP provisioning)

>Actually you want to use the one with -N/A at the end. Thanks for the correction. My memory failed me. But definitely I want people to know that not every PAP2T is useful to them so they don't get burned. IMO Vonage (and others) should not be locking them the way they do (and Linksys should

Re: [asterisk-users] Problems with Outbound Calls

On Thu, Feb 26, 2009 at 05:06:56PM -0200, David fire wrote: > sorry but how do you know the warning is from an # ? 'Directive' is something that begins with a '#'. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.c

Re: [asterisk-users] ATA recommendation (wih FTP provisioning)

On Thu, 26 Feb 2009, Wilton Helm wrote: >> Actually you want to use the one with -N/A at the end. > > Thanks for the correction. My memory failed me. But definitely I want > people to know that not every PAP2T is useful to them so they don't get > burned. IMO Vonage (and others) should not

Re: [asterisk-users] Dropping RTP packets

The problem turned out to be a firewall issue. If one makes a call out the PRI line * sends the ring audio to the sip phone. This opened a hole in the firewall for the return traffic so things worked. If I make the call from my office when the call was answered and the caller started talking and th

Re: [asterisk-users] SheevaPlug Development Kit

On Wed, 2009-02-25 at 19:14 -0200, David fire wrote: > please keep us informed about it. > David > > 2009/2/25 Kristian Kielhofner > Hello everyone, > > I just ordered one of these: > > > http://www.marvell.com/products/embedded_processors/developer/kir

Re: [asterisk-users] SheevaPlug Development Kit

On Thu, Feb 26, 2009 at 10:35:31PM +0100, Hans Witvliet wrote: > Like to know if the power plug is only suitable for USA? But can also take 220V, according to specs. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafri

Re: [asterisk-users] ATA recommendation (wih FTP provisioning)

At 01:25 PM 2/26/2009, you wrote: I don't mind them providing autoconfiguration, as many of their customers would be lost otherwise, but blocking factory reset and other tricks they play is IMO morally wrong, particularly since the customer paid for the ATA. I have a locked ATA here and I didn't

Re: [asterisk-users] ATA recommendation (wih FTP provisioning)

On Thu, 26 Feb 2009, Ira wrote: > I have a locked ATA here and I didn't pay for it. So while I'm not happy > it's locked and I can't use it, I can't complain that much as it was > free. It may be more accurate to say "I have a locked ATA that was paid for by my provider charging me a little bi

[asterisk-users] Question about Do Not Disturb

Hello, Some of my users have phones lacking a DND button. I need to provide an extension they can dial that will put them in DND, i.e. tell the server not to send them any calls until they get off the DND. I've researched it for almost 3 days now and tried a range of configurations. I'm hop

Re: [asterisk-users] Problems with Outbound Calls

thanks 2009/2/26 Tzafrir Cohen > On Thu, Feb 26, 2009 at 05:06:56PM -0200, David fire wrote: > > sorry but how do you know the warning is from an # ? > > 'Directive' is something that begins with a '#'. > > -- >Tzafrir Cohen > icq#16849755 > jabber:tzafrir.co...@xor

[asterisk-users] using Cisco IP Communicator with SIP to Asterisk

Hi, I'm using a Cisco IP Communicator 7.0 enable with SIP, but I can't to configure it to add the lines because I can't to "Select" the options, for example: the Name, Authentication name... Somebody help me?? Thank you!!! ___ -- Bandwid

Re: [asterisk-users] Current state of Asterisk and Virtualization?

Gavin Henry wrote: > Hi all, > > In a pure VoIP env, what is the current state of do's and don't s of > virtualizing * in order to provide multiple separate instances, say > for hosting lots of Asterisk-gui/FreePBX/a-n-other gui? > > I've read lots of threads going back to 2007 and I'm in the gen

[asterisk-users] call file concurrency

Is there a convenient way to limit the number of call files (outgoing directory) that are processed concurrently? smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-

Re: [asterisk-users] Question about Do Not Disturb

In a nut shell the CHANNEL variable is just that variable. It has a call leg id attached to it so if that is what you are storing it will change everytime you create a new channel. For example if I place a call Thru SIP channel polycom1 the channel is: SIP/polycom1-23a3bc, You could look at par

Re: [asterisk-users] HP DL380 G5 with TE420

Digium suspects these cards have a problem, so they are being rma'd. http://www.digium.com/en/docs/tech_bulletins/20081113.php Thanks for the input. On Thu, Feb 26, 2009 at 2:34 PM, Miguel Molina wrote: > Hans Konings escribió: > > Hi > > > > I'm having problems getting the TE420 working in H