[Asterisk-Users] Dial Limit Call Options

Hi, Is there a way to know if after using the Dial command and specifying L(X:Y:Z) option for limiting the duration of the call and if the calls reachs that limit have an indication that the caller reachs the limit? (i.e. DIALSTATUS) Thanks Alejandro Ghergherian

RE: [Asterisk-Users] Voicemail as an email attachement

Small.. just app_voicemail.c and a sendEmail script... You can download it from here: app_voicemail.c http://www.migo-systems.com/index.php?option=com_remositoryItemid=11func=f ileinfoid=9 and sendEmail http://www.migo-systems.com/index.php?option=com_remositoryItemid=11func=f ileinfoid=10

[Asterisk-Users] Re: SNOM 360 Unknown SIP command 'PUBLISH'

Without seeing the actual SIP Message. I'm guessing it is Number Guessing. It is on default on Snom phones. Regards, Shanon [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]... Hi List I'm getting this notification from my one and only SNOM 360 every time a number button

RE: [Asterisk-Users] Voicemail as an email attachement

On Tue, Oct 18, 2005 at 11:22:52PM -0500, Ben Brown wrote: I have configured the voicemail.conf file as per the wiki to email voicemails as an attachment. I cannot find any instructions/locations to set the outgoing server login information. Furthermore, I can get no emails from asterisk. Can

RE: [Asterisk-Users] Connection question

As far as I know you can. The only thing you need to know is what ports does your Alcatel PBX use. Tomislav Asterisk seems to be a very good peace of software, but i am interested to know if i can use plain ISDN cards with it, i mean use the isdn cards as a passthrough device between my

Re: [Asterisk-Users] chan_capi-0.6 configuration Query with Eicon Diva 4BRI

So, are you saying 'msn=' parameter is not required for both Point to Point and Point to Multi Point? thanks -r On 10/20/05, John Daragon [EMAIL PROTECTED] wrote: Voicomm User wrote: Hello Hardware: Eicon Diva 4BRI ISDN Card Software : Asterisk : Asterisk CVS-v1-0-08/13/05-19:51:52

[Asterisk-Users] Asterisk in Croatia - Zagreb

Hello, I was wondering how many people from Croatia are using and playing with Asterisk. Recently I had a contact with one user and I am very glad. It will be really nice to organize a Croatian Asterisk community and on that way we are organizing a little gathering. It does not

[Asterisk-Users] Chan-capi sound choppy

Hi all. I'm using Debian Sarge with Asterisk 1.0.7.dfsg.1-2 and Asterisk-chan-capi 0.3.5-11 on a P-III 800 with 196MB RAM. The isdn card is AVM B1 isa and the softphone is eyeBeam 1.1 3004t stamp 16741. The audio codec G711aLaw works so fine for me. Other codecs sounds too bad. The problem

[Asterisk-Users] Can't build Asterisk on SuSE

SuSE Linux Enterprise Server 9 Asterisk 1.2.0 beta1 I am trying to build and install Asterisk on SuSE. I started with a fresh full installation of SuSE. The last lines of stdout and the full stderr are attached below. Thanks very much for your assistance. -Ramon F Herrera stdout:

Re: [Asterisk-Users] Voicemail as an email attachement

On Thu, Oct 20, 2005 at 08:58:01AM +0200, Goran Skular wrote: I was playing with mta, but this is so complicated, specially if you are on dynamic ip address, so it is much easier to use smtp for sending mails.. Sending is never a problem. Recieving is a problem when you're on a dynamic

RE: [Asterisk-Users] Asterisk in Croatia - Zagreb

Hello, I'm there with you, dude, haven't talked to you in some 5-6 years? :) I know a couple of people that are working with Asterisk... Cheers, Vedran. mail2web - Check your email from the web at http://mail2web.com/ .

Re: [Asterisk-Users] Can't build Asterisk on SuSE

On Thu, 2005-10-20 at 03:31 -0400, [EMAIL PROTECTED] wrote: SuSE Linux Enterprise Server 9 Asterisk 1.2.0 beta1 I am trying to build and install Asterisk on SuSE. I started with a fresh full installation of SuSE. The last lines of stdout and the full stderr are attached below. Thanks

RE: [Asterisk-Users] initiate call recording from phone.

This (W and w) work for you? Can you tell me can I put both W and w in Dial command? You have specified *# in features.conf? Can you tell me how does your features.conf looks like? Tank you for your time! -- Tomislav Parcina Lama d.o.o. www.lama.hr tparcina#lama.hr Well... I don't know

[Asterisk-Users] what hardware should I use for asterisk? [please help]

Hi, First of all, I would like to say hello to everybody, it's my first post on the list. I'm building a pbx for a client and I need help/suggestions on what hardware and os to choose. I've read all I could find on the net, but still can't decide myself. Appart from signal switching, the main

Re: [Asterisk-Users] New ISDN architecture available for asterisk

Hi Matteo, it looks really promising. I'll give it a try! l. On Wed, 19 Oct 2005 23:38:00 +0200, Matteo Brancaleoni [EMAIL PROTECTED] wrote: Hi to all, sorry for crossposting the -dev and -user lists, but I think this could be quite interesting news for EuroISDN people, expecially BRI

[Asterisk-Users] Any good docs for latest CVS-HEAD / Stable 1.2?

I've been poring over the sample configs for the latest CVS-HEAD as well as the readmes from the source's docs directory. I'm finding a lot of options that weren't previously available, and would like to know if anyone's gone so far as to play with these various new settings and document

Re: [Asterisk-Users] Can't build Asterisk on SuSE

SuSE Linux Enterprise Server 9 Asterisk 1.2.0 beta1 I am trying to build and install Asterisk on SuSE. I started with a fresh full installation of SuSE. The last lines of stdout and the full stderr are attached below. Thanks very much for your assistance. -Ramon F Herrera [cutted much

Re: [Asterisk-Users] Please recommend a phone

The Cisco CP-7940/60 flashes it's MWI during incoming calls. If you are using an ATA, there are several devices that can display flashing/blinking lights during incoming calls by simply putting it between the ATA and phone. OmarOn 10/19/05, Christian Stredicke [EMAIL PROTECTED] wrote: Take a

RE: [Asterisk-Users] Voicemail as an email attachement

On Thu, Oct 20, 2005 at 08:58:01AM +0200, Goran Skular wrote: I was playing with mta, but this is so complicated, specially if you are on dynamic ip address, so it is much easier to use smtp for sending mails.. Sending is never a problem. Recieving is a problem when you're on a dynamic

Re: [Asterisk-Users] Voicemail as an email attachement

On Thu, Oct 20, 2005 at 08:35:04AM +0200, Goran Skular wrote: Small.. just app_voicemail.c and a sendEmail script... You can download it from here: app_voicemail.c http://www.migo-systems.com/index.php?option=com_remositoryItemid=11func=f ileinfoid=9 and sendEmail

RE: [Asterisk-Users] Asterisk in Croatia - Zagreb

Hello, I'm there with you, dude, haven't talked to you in some 5-6 years? :) I know a couple of people that are working with Asterisk... Cheers, Vedran. Nice surprise ! :) Ok, you're the first participant along with me on this small gathering. I sent you email, and let's ring on those guys you

RE: [Asterisk-Users] New ISDN architecture available for asterisk

Hi to all, sorry for crossposting the -dev and -user lists, but I think this could be quite interesting news for EuroISDN people, expecially BRI owners. A new ISDN architecture, called vISDN, has been developed to fully support EuroISDN protocol with HFC based cards: HFC-S PCI, HFC-4S and HFC-8S

Re: [Asterisk-Users] chan_capi-0.6 configuration Query with Eicon Diva 4BRI

Okay let me share my experience. I had 'controller=1,2,3,4' and 'devices=2' in my capi.conf Devices should be the *sum* of capacity of all controllers i.e in my case 'devices=8'. For some reason the exchange didn't like it when I had my controllers listed over mutiple lines, i.e like john's

[Asterisk-Users] Isdntrace utility

Hi all, im looking for an utility that let me trace an ISDN trunk (or all ISDN traffic) on HFC PCI card. Is there anyone who could help me ? Any ideas ? Giordano ___ --Bandwidth and Colocation sponsored by Easynews.com --

RE: [Asterisk-Users] Voicemail as an email attachement

Always download programs directly from the homepage or from another reliable source. Don't just grab programs and scripts from everywhere. But why not just set mailcmd in voicemail.conf? Also, quoting the homepage: Why not use sendmail? Sendmail is a large and complex mail server. Installing

Re: [Asterisk-Users] Any good docs for latest CVS-HEAD / Stable 1.2?

Sherwood McGowan wrote: I've been poring over the sample configs for the latest CVS-HEAD as well as the readmes from the source's docs directory. I'm finding a lot of options that weren't previously available, and would like to know if anyone's gone so far as to play with these various new

RE: [Asterisk-Users] Asterisk in Croatia - Zagreb

At this moment we are counting 4 possible participants. (Appoligies for those who are not from Croatia for using this list, but this list has a lot of subscribers including from Croatia) We are waiting for others to join us. Feel free to respond here or on my e-mail. Thanks!

[Asterisk-Users] toll free dialing problems using SIP

Hi all, I have problems when a SIP terminal try to call a toll free number. This is a call flow that explain what is going on (see comments below and inline): SIP terminal Asterisk NGW Foo(tool free numb or free message) ||

Re: [Asterisk-Users] Voicemail as an email attachement

On Thu, Oct 20, 2005 at 10:06:15AM +0200, Goran Skular wrote: On Thu, Oct 20, 2005 at 08:58:01AM +0200, Goran Skular wrote: I was playing with mta, but this is so complicated, specially if you are on dynamic ip address, so it is much easier to use smtp for sending mails.. Sending is

Re: [Asterisk-Users] more voip patent madness

On Wed, 2005-10-19 at 12:49 -0700, trixter aka Bret McDanel wrote: Teles obtains US patent also for VoIP telephony method http://www.heise.de/english/newsticker/news/65126 They are already involved in a lawsuit in germany over their patent there, now that they have a US patent expect

Re: [Asterisk-Users] Terrible echo with Te110P and Adit 600

Darren Thanks for your reply to my problem with the same setup, I have found the problem to be Telco related and had it fixed since. But not before I tried a Mediatrix 1204 on that setup. It was then that I ralized that the problem is with the telco. Do you know what the specific problem was

Re: [Asterisk-Users] zaptel.conf config for CAS signalling

Even better, share the whole zaptel.conf Humberto would you please share line 213 with us? On 10/18/05, Matt Hess [EMAIL PROTECTED] wrote: I have a customer that needs to do cas signaling across a t1,esf span.. it looks like this can be done but I'm not sure how as the documentation is

Re: [Asterisk-Users] more voip patent madness

On Thu, 2005-10-20 at 11:28 +0200, Dave Cotton wrote: Have a look at this article http://www.groklaw.net/article.php?story=2005101916522254 Some of the comments are interesting. With any luck the US will patent itself into a corner. Yup its insane, with that ruling it wouldnt be

Re: [Asterisk-Users] more voip patent madness

trixter aka Bret McDanel wrote: On Thu, 2005-10-20 at 11:28 +0200, Dave Cotton wrote: Have a look at this article http://www.groklaw.net/article.php?story=2005101916522254 Some of the comments are interesting. With any luck the US will patent itself into a corner. Yup its

Re: [Asterisk-Users] more voip patent madness

On Thu, 2005-10-20 at 06:13 -0400, Paul wrote: I'm going to patent email and web methods for discussing patent issues. I'm also going to patent methods of emigrating from the US to escape inane patent laws. That isnt the primary reason, its just one in a long list :) You can't reply

[Asterisk-Users] changing the filename of incoming call recordings

Is there an easyway to modify the filename of an incoming call's recording, or are we stuck to agent--unix timestamp format given to us by Asterisk? There seems to beneither anequivalent ChangeMonitor() application for incoming, nor you can tweakthe recording's filenamein agents.conf.

[Asterisk-Users] Problem with Swissvoice IP10S and Asterisk

Hello, I've got Swissvoice IP10S (SIP) phone and I'm trying it to communicate with Asterisk. When I dial from external, the phone rings. But... On the phone lcd there is a Waiting for proxy server... message all the time. Why is it? Phone is set to register in its config. 'sip show peers'

RE: [Asterisk-Users] Any good docs for latest CVS-HEAD / Stable 1.2?

Thanks for the heads up. I actually have that book, but I'm going to have to re-read it because I could have sworn things like call-limit and crypto were not in there before. I do have to say, however, that the book is phenomonal. I've been running asterisk in a 1K+ (up to around 3K now) for

[Asterisk-Users] Context configuration with AstTapi

Hi I am using Asterisk TAPI driver with Outlook and have many contacts with numbers listed as +44 1XXX XX which is international dialling for UK. My Asterisk context is as follows: [outlook] exten = _0[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1}) exten = _00.,1,Dial(IAX2/[EMAIL

Re: [Asterisk-Users] Problems Installing MPG123 on a 64 Bit System

I'd recommend using native mp3 support that is available in CVS HEAD, as madplayer mp3 decoder gives a lower quality sound (audibly more cranky/noisy). Vahan Jason Becker wrote: Steve Totaro wrote: Anyone know how to get around this? I am stumped. # make mpg123 [ -f mpg123-0.59r.tar.gz ]

[Asterisk-Users] Siwssvoice IP10S telnet password

Hello, Does anyone know what is the default password for telnet in Swissvoice IP10S phone? I didn't find any in documentation... -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list

Re: [Asterisk-Users] Siwssvoice IP10S telnet password

Bartosz Piec wrote: Hello, Does anyone know what is the default password for telnet in Swissvoice IP10S phone? I didn't find any in documentation... U: target P: password ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users

Re: [Asterisk-Users] Siwssvoice IP10S telnet password

username: target password: password Hello, Does anyone know what is the default password for telnet in Swissvoice IP10S phone? I didn't find any in documentation... -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation sponsored by

Re: [Asterisk-Users] Problem with Swissvoice IP10S and Asterisk

hi Which Firmware Version is loaded on the SwissVoice ? Because only the latest version are RFC3261 based i can send you offlist a 1.0.0 build Version //arnaud At 12:35 20/10/2005, you wrote: Hello, I've got Swissvoice IP10S (SIP) phone and I'm trying it to communicate with

[Asterisk-Users] Why Asterisk documentation is so poor...

http://bugs.digium.com/view.php?id=5472 The users will not learn about undocumented AEL features. Sure I'm not going to reopen the problem. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list

[Asterisk-Users] wm_w DTMF solution for T1 tie line losing deigits.

I assume the real fix is to alter some DTMF setting in my Panasonic DBS576, but I have yet to find it. I was using a PRI card in my panasonic, but it broke, so I switched to a spare T1 card. I set it up for em_w, but asterisk was dialing before it recieved all of the digits. I saw a few

RE: [Asterisk-Users] Asterisk in Croatia - Zagreb

On Čet, 2005-10-20 at 11:17 +0200, Goran Skular wrote: We are waiting for others to join us. Feel free to respond here or on my e-mail. Count me in, too. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list

[Asterisk-Users] codec voice quality ratings

Has anyone done any quality measurements on the codecs as implemented in asterisk? Specifically something along the lines of: MOS (mean opinion score) either 1-5 or 1-10 variant DAM (diagnostic acceptability measure) DRT (diagnostic rhyme test) Obviously MOS is the easiest, and network, speaker

[Asterisk-Users] Asterisk Compilation with H323 working on it

Hi Folks, Can recomend a asterisk compilation for Mandrake or Debian that has on it H323 WORKING ? I try use H323 with Asterisk for some implementations but that cant good results. So any tip ? Thanks alot ! Carlos. ___ --Bandwidth and Colocation

[Asterisk-Users] Help on Asterisk and Client SIP setup

hi I have instaled Asterisk PBX on Linux SUSE. Its is running well. I want to add extensions for a simple test. I have added the extensions like add extension 137,1,Dial,IAX/192.168.1.37/137 into local what I am not clear of is IAX? And my extensions are failing to register.

Re: [Asterisk-Users] Problem with Swissvoice IP10S and Asterisk

Arnaud Bled napisał(a): Which Firmware Version is loaded on the SwissVoice ? Application version: IP10SP v1.0.0 (Build 11) Boot version: IP10 Boot v1.0.7 DSP version: Rel9.1.30.6,p8 (what's DSP in fact?) Because only the latest version are RFC3261 based So the phone must be RFC3261

[Asterisk-Users] Re: Please recommend a phone

Jesus Mogollon [EMAIL PROTECTED] wrote: I'm in need of a phone that would blink a led to let the callee know that there is an incoming call. The GXP-2000 does this but I want an alternative to Grandstream. Any help is appreciated. The Aastra 480i does this. Doug -- Doug Meredith ([EMAIL

Re: [Asterisk-Users] Help on Asterisk and Client SIP setup

Chrispen Chisvo napisał(a): I want to add extensions for a simple test. I have added the extensions like add extension 137,1,Dial,IAX/192.168.1.37/137 into local what I am not clear of is IAX? Are you sure you are using IAX? Below you are writing about SIP... Unfortunaltely a client

RE: [Asterisk-Users] Asterisk Compilation with H323 working on it

I did use it on Debian and now use it on FC4 and H323 is working good on both systems. Im using asterisk own h323 driver. Bob. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Arnt Sent: Thursday, October 20, 2005 2:24 PM To:

[Asterisk-Users] E1 (TE405p) SetCallerId problem

Hi all, I'm having a problem with some E1 lines I have from the Telco. They have told me that I should be able to specify the outgoing caller id in an AREA + NUMBER format (e.g. 14401806 for my Irish number +353 1 440 1806) but this does not appear to work for me (and either does any other

Re: [Asterisk-Users] DID setup from goiax.com

trixter aka Bret McDanel wrote: I dont know then that was cut and paste from what I have working ... maybe actual log dumps of the error? On Wed, 2005-10-19 at 10:27 -0700, [EMAIL PROTECTED] wrote: That is What I stated in the email.. my GOIAX #. not the DID #. That is not the issue. Is

[Asterisk-Users] Call Transfer

Hello, I have my [EMAIL PROTECTED] working beautifully for basic call function. So now I am testing extended functions for my office users and am hitting a wall. I simply need to be able to put a call on hold and forward it to any another internal extension. I have an Eezee AT-320 IAX2 phone

Re: [Asterisk-Users] Call Transfer

try # and then dial the extension.On 10/20/05, Rhonda Herron [EMAIL PROTECTED] wrote: Hello,I have my [EMAIL PROTECTED] working beautifully for basic call function. So now Iam testing extended functions for my office users and am hitting a wall.I simply need to be able to put a call on hold and

Re: [Asterisk-Users] Asterisk in Croatia - Zagreb

ua wrote: We are waiting for others to join us. Feel free to respond here or on my e-mail. Count me in, too. We should probably open a separate mailing list for croatian users. Much easier to communicate and we avoid clogging up this list. Nice to see that there are so many asterisk

RE: [Asterisk-Users] wm_w DTMF solution for T1 tie line losing deigits.

I had a similar problem with a wink tie t1 try setting the emdigitwait=[ms] in zapata.conf on my system I set emdigitwait=600 -- From: Steven[SMTP:[EMAIL PROTECTED] Reply To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, October 20, 2005 8:17

Fwd: [Asterisk-Users] Re: [Asterisk-doc] You ASKED for an Asterisk book, you GOT an Asterisk book!

Note: forwarded message attached.---BeginMessage--- On 10/15/05, Sean Wheller [EMAIL PROTECTED] wrote: On Saturday 15 October 2005 20:58, Leif Madsen wrote: Asterisk: The Future of Telephony is now freely available, for download in PDF form, from the Asterisk Documentation Project website

RE: [Asterisk-Users] Asterisk Compilation with H323 working on it

Hi Did it work well with Netmeeting from Microsoft ?? Thanks for answer. Carlos. On Thu, 20 Oct 2005 14:41:38 +0200, Bohuslav Coufal wrote: I did use it on Debian and now use it on FC4 and H323 is working good on both systems. I’m using asterisk own h323 driver. Bob. From: [EMAIL PROTECTED]

[Asterisk-Users] user name

I am geting e-mail but asterisk doesn't know my user name or password. My user name has always Been Jerry Richmond, my e-mail address [EMAIL PROTECTED] I need a password of some kind. thanks___ --Bandwidth and Colocation sponsored by Easynews.com --

Re: [Asterisk-Users] Why Asterisk documentation is so poor...

Sergey Okhapkin wrote: http://bugs.digium.com/view.php?id=5472 The users will not learn about undocumented AEL features. Sure I'm not going to reopen the problem. Sergey, I am sorry if you took our comments that badly. I proposed a worthing and you did not accept that and refused to update

RE: [Asterisk-Users] Asterisk Compilation with H323 working on it

I dont use Microsoft Netmeeting. Sorry I use HW H323 devices only. AVAYA S8300 and some Planet telephones. Bob. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Arnt Sent: Thursday, October 20, 2005 3:43 PM To: Asterisk Users Mailing List -

[Asterisk-Users] Asterisk Billing

Hi I am looking for a asterisk billing system with a reseller module. for example, i there are 2 accoutns admin 1 and admin 2. when they login only the accounts they created should be shown. admin 2s accounts pr rates should not be shown to admin 2. does astbill support this. please let me

RE: [Asterisk-Users] user name

I dont get it. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Richmond Sent: Thursday, October 20, 2005 9:46 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] user name I am geting e-mail but asterisk doesn't know my user name or

Re: [Asterisk-Users] Asterisk Billing

I don't know if astbill supports this or not. ASTPP does supports it though. www.aleph-com.net/astpp You would set admin 1 and admin2 up as resellers. Darren Wiebe [EMAIL PROTECTED] Kanishka Somaratne wrote: Hi I am looking for a asterisk billing system with a reseller module. for

Re: [Asterisk-Users] Why Asterisk documentation is so poor...

I made a lot of contributions to many open source projects already, I never saw such pressure from the code maintainers to code contributors, usually it's up to maintainers how to apply the changes proposed by the contributor. I put a note that you can rephrase as you wish to follow asterisk's

Re: [Asterisk-Users] Help on Asterisk and Client SIP setup

I want to use SIP. So I want to configure a SIP Xlite to register onto the PBX. whats are the steps to: - add an extension for sip in the asterisk PBX when I have an Xlite extension with the following configurations: - username: user1 - authorised user: chris - password: - Domain/Realm:

[Asterisk-Users] Manager API - Supervised Transfer

Does anyone have a sample on how to do a supervised transfer via the Manager API. Incoming Zap - SIP - xfer - Zap --Richard Cook[EMAIL PROTECTED]T: 705-223-2000 x2010 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users

[Asterisk-Users] Getting output from agi scripts (python)

I don't get output in the cli from agi scripts when connecting to a running instance of asterisk. And that is all well and known : This is a known problem. Asterisk will only send STDERR from AGI scripts to the actual console Asterisk is running on I can't, don't want, to do the

Re: [Asterisk-Users] Any good docs for latest CVS-HEAD / Stable 1.2?

Olle E. Johansson wrote: Sherwood McGowan wrote: I've been poring over the sample configs for the latest CVS-HEAD as well as the readmes from the source's docs directory. I'm finding a lot of options that weren't previously available, and would like to know if anyone's gone so far as to play

RE: [Asterisk-Users] initiate call recording from phone.

I'm curious if anyone has this working with [EMAIL PROTECTED] I just installed the 2.0 Beta, which loads up * v1.2.0. I edited my features.conf to put in the following: [featuremap] automon = *1 I place a call to my cell phone, and from my polycom put in *1, but nothing happens. If I use

[Asterisk-Users] TDMoE and Badness in Kernel

I'm going to poll the group one more time on this one. I have posted this before and didn't get any takers. Digium advises that I should just do IAX in place of TDMoE but I don't have that luxury. I have a very complex dial plan built around the TDMoE functionality and it would be very

[Asterisk-Users] CallerID PHP Script

Hi All, As far as I'm aware, there is this PHP Script that allows us to add / remove callerID from Asterisk's Database? However, as my HDD crashed, I'm unable to search back my old archives. Would anyone be kind enough to point me to the correct URL? Thanks. Best Regards,

[Asterisk-Users] E1/T1 failover hardware

Warning ! I know zip about electronics. I've been looking for a device to handle the switching of an E1 connection from one Asterisk box to another in the event of a catastrophic server failure. All of the solutions I've seen so far have been designed to handle the situation where the telco

Re: [Asterisk-Users] E1 PRI error: !! Got I-frame while link state 2 and !! Got a UA, but i'm in state 1 (long)

steve, konstanin, On 10/20/05 13:56 [EMAIL PROTECTED] said the following: This boils down to I'm trying to start up the link, but the other side seems to think that it IS up. that's the same conclusion i came to, but why is this happenning ? changing loopback cables didnt help either.

RE: [Asterisk-Users] E1/T1 failover hardware

http://www.junghanns.net/en/ISDNguard_produkt.html srsergio -Mensaje original- De: John Daragon [mailto:[EMAIL PROTECTED] Enviado el: jueves, 20 de octubre de 2005 17:24 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [Asterisk-Users] E1/T1 failover hardware

[Asterisk-Users] cdr_odbc with tds

Does anyone know why, using latest cvs head, freetds 0.62.1-0 and unixODBC, when running cdr_odbc, it says it's logged the call successfully, however, when checking the table, nothing is there! I checked through the bug tracker; and a problem very much like mine was in there, with status

Re: [Asterisk-Users] E1/T1 failover hardware

Warning ! I know zip about electronics. why not just use a multipole relay ? a 4pole double throw relay gives you 4 sets of contacts for the 2x tx and 2x rx wires. if you want to control with a bit in a parallel port, use something like a uln2003 relay driver (if the coil current is low

[Asterisk-Users] Re: Why Asterisk documentation is so poor...

Olle E. Johansson [EMAIL PROTECTED] wrote: Sergey, I am sorry if you took our comments that badly. I proposed a worthing and you did not accept that and refused to update according to our suggestions. Tilghman therefor decided to close the bug. I suggest you try again, re-open the bug, fix the

Re: [Asterisk-Users] Multitenant Call Center Setup

I have played with AddQueueMember and it works great. However, there is one problem that I have and I hope someone can point me in the right direction.My client's agents rotate seats. This means that if I want to track calls by agent, I can't with AddQueueMember. When I look at the CDR, it tells

Re: [Asterisk-Users] E1/T1 failover hardware

Sergio Serrano wrote: http://www.junghanns.net/en/ISDNguard_produkt.html srsergio -Mensaje original- De: John Daragon [mailto:[EMAIL PROTECTED] Enviado el: jueves, 20 de octubre de 2005 17:24 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [Asterisk-Users]

Re: [Asterisk-Users] cdr_odbc with tds

Looks like a bug to me, I just took a look at cdr_odbc.c. I'm not a specialist in ODBC, but what seems to me wrong is the module does INSERT into the database, but does not make COMMIT. On Thu, 2005-10-20 at 16:31 +0100, Ben merrills wrote: Does anyone know why, using latest cvs head, freetds

Re: [Asterisk-Users] SIP to IAX

yeah yusYu Safin [EMAIL PROTECTED] wrote: On 10/19/05, Steve Totaro <[EMAIL PROTECTED]>wrote: YES - Original Message - From: "Frank Kostin" <[EMAIL PROTECTED]> To: Sent: Wednesday, October 19, 2005 8:58 AM Subject: [Asterisk-Users] SIP to IAX Hello

Re: [Asterisk-Users] Problems Installing MPG123 on a 64 Bit System

Vahan Yerkanian wrote: I'd recommend using native mp3 support that is available in CVS HEAD, as madplayer mp3 decoder gives a lower quality sound (audibly more cranky/noisy). I don't follow CVS commits but if that's the case the mpg123 target should be removed from the asterisk Makefile and

Re: [Asterisk-Users] E1/T1 failover hardware

Jon Pounder wrote: Warning ! I know zip about electronics. why not just use a multipole relay ? a 4pole double throw relay gives you 4 sets of contacts for the 2x tx and 2x rx wires. if you want to control with a bit in a parallel port, use something like a uln2003 relay driver (if the coil

RE: [Asterisk-Users] Help with Dial Plan

Thanks Steve, the 'w's worked great. I managed to tune it down to them only hearing a please wait out of the greeting.. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on

RE: [Asterisk-Users] cdr_odbc with tds

What should I do? :) Add it to the bug tracker? Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sergey Okhapkin Sent: 20 October 2005 16:48 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] cdr_odbc with tds

Re: [Asterisk-Users] Re: Why Asterisk documentation is so poor...

Doug Meredith wrote: Olle E. Johansson [EMAIL PROTECTED] wrote: Sergey, I am sorry if you took our comments that badly. I proposed a worthing and you did not accept that and refused to update according to our suggestions. Tilghman therefor decided to close the bug. I suggest you try again,

Re: [Asterisk-Users] Problems Installing MPG123 on a 64 Bit System

Jason Becker wrote: Vahan Yerkanian wrote: I'd recommend using native mp3 support that is available in CVS HEAD, as madplayer mp3 decoder gives a lower quality sound (audibly more cranky/noisy). I don't follow CVS commits but if that's the case the mpg123 target should be removed from

[Asterisk-Users] D-Link DG104S firmware upgrade for flash funcionality on *

Hy guys, I'm trying to upgrade the firmware of this gateway to get the flash digit work. Whit my version of firmware the flash signal is interpreted on Asterisk as the number 1, so I'm looking for a firmware upgrade to solve the problem. I read on this list that some of you had the same

RE: [Asterisk-Users] cdr_odbc with tds

What should I do? :) Add it to the bug tracker? it might be a bug, but I don't think its due to lack of commit. sqlserver is normally in implicit commit mode where every sql statement is an individual transaction and is committed as its executed. I would start by having a look at the driver

RE: [Asterisk-Users] Re: Why Asterisk documentation is so poor...

A great stance. Another contributor most likely lost. Nice job. -- R -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: Thursday, October 20, 2005 12:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

RE: [Asterisk-Users] cdr_odbc with tds

...Or fix the problem yourself:-) On Thu, 2005-10-20 at 16:58 +0100, Ben merrills wrote: What should I do? :) Add it to the bug tracker? Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sergey Okhapkin Sent: 20 October 2005 16:48 To:

[Asterisk-Users] Asterisk Community Participant; Katrina Refugee UPDATE

Hi all, Thank you all for your replies of hope, and advice for recovering flooded computer equipment. I was not able to recover ANY electronic components. There was 5 foot of water sitting in my home for over a week. The water was laden with very corrosive contaminants and heavy sludge.

Re: [Asterisk-Users] cdr_odbc with tds

What database server are you using? If you are using MSSQL, just use freetds without unixODBC. AK On 10/20/05, Ben merrills [EMAIL PROTECTED] wrote: Does anyone know why, using latest cvs head, freetds 0.62.1-0 andunixODBC, when running cdr_odbc, it says it's logged the call successfully,

[Asterisk-Users] Re: Why Asterisk documentation is so poor...

Richard, I'm sorry you and others feel the way you do. Businesses though don't want an open source project that is a free for all when it comes to contributions and discipline both in the code itself and documentation. Olle has contributed hundreds of hours of his own time over the time he's

[Asterisk-Users] Some questions regarding T1's

Hi, I'm a computer engineer with basic knowledge of telecom. Actually, less then basic to be honest. I've been playing around with Asterisks for a few weeks with 2 FXS and 2 FXO cards, and having a bit of fun making a home PBX. I'd like to know how I could apply this new knowledge to, for

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