Re: [asterisk-users] p2p t1 with sangoma hw

Awesome, I just figured this out myself but havent tested it yet, wasn't 100% sure I was right, now that I know, I will give it a shot! Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erik Anderson Sent: Sunday, November 18, 2007 1:01 AM To: Ast

Re: [asterisk-users] p2p t1 with sangoma hw

On Nov 17, 2007 11:49 PM, Michael J. Liberatore <[EMAIL PROTECTED]> wrote: > > I figured that one side would be pri net and the other would be pri cpe, > well I chose pri cpe and the next question was asking for a switch type, > national isdn 2, at&t, nortel, etc - that sounds really wrong. Pick

Re: [asterisk-users] p2p t1 with sangoma hw

Ok I am trying to setup this p2p t1 with the sangoma t1 cards, everything seemed ok till it asked me if I wanted fxs, fxo, or pri cpe or pri net. I figured that one side would be pri net and the other would be pri cpe, well I chose pri cpe and the next question was asking for a switch type, nation

Re: [asterisk-users] Trouble with asterisk-users mailman

I know this is a lame suggestion, but worth a shot if uninterrupted participation on this list is important to you, get a gmail address. gmail removes source IP info, effectively making their server address the src IP of the msg. Since @gmail is the most popular domain here, any problem resul

Re: [asterisk-users] problem with tdm2400p configuration

On Sun, Nov 18, 2007 at 10:46:57AM +0800, Mark Quitoriano wrote: > Hi i have a tdm2400p and installed asterisk 1.4.11 with zaptel 1.4.5 > im having an error message when in running asterisk with the tdm card > in. > > here's the error from the console of asterisk: > > [Nov 18 10:30:44] ERROR[5557

Re: [asterisk-users] Trouble with asterisk-users mailman

You came thru this time. On Nov 17, 2007 11:06 PM, Jesse Molina <[EMAIL PROTECTED]> wrote: > > I'm trying this again because the last attempt didn't go through (thus > more or less proving one of the below to be true.) > > > > Jesse Molina wrote: > > > > Test123 > > > > My messages to this maili

Re: [asterisk-users] Trouble with asterisk-users mailman

This message appears to have successfully gone through, but multiple others didn't. My messages that didn't get to the list were all sent within the first 48 hours of joining the list, but were after the first 30 minutes. I think there is something wrong. I joined the list both from my perso

Re: [asterisk-users] AsteriskNOW and TDM800P

rafael, it should work. both systems are auto configurable... daveC Rafael Canchola wrote: > > Hi all > > I sold new TDM800P card with 8 FXO ports, someone know if can be use > this card on AsteriskNOW or trixbox? > What can i do for use this card? > > Thanks. > > ---

Re: [asterisk-users] Trouble with asterisk-users mailman

I'm trying this again because the last attempt didn't go through (thus more or less proving one of the below to be true.) Jesse Molina wrote: > > Test123 > > My messages to this mailing list are disappearing. > > Is this list quietly being moderated? > > Have I been wrongly black-holed? >

Re: [asterisk-users] problem with tdm2400p configuration

ooops sorry i didn't cut and paste that correctly # Span 1: WCTDM/0 "Wildcard TDM2400P Board 1" fxsks=1 fxsks=2 fxsks=3 fxsks=4 fxsks=5 fxsks=6 fxsks=7 fxsks=8 fxsks=9 fxsks=10 fxsks=11 fxsks=12 fxsks=13 fxsks=14 fxsks=15 fxsks=16 fxsks=17 fxsks=18 fxsks=19 fxsks=20 fxsks=21 fxsks=22 fxsks=23 fxsk

Re: [asterisk-users] problem with tdm2400p configuration

You have 18 channels defined in zaptel.conf, but 24 channels configured in zapata.conf Mark Quitoriano wrote: > Hi i have a tdm2400p and installed asterisk 1.4.11 with zaptel 1.4.5 > im having an error message when in running asterisk with the tdm card > in. > > here's the error from the console

[asterisk-users] problem with tdm2400p configuration

Hi i have a tdm2400p and installed asterisk 1.4.11 with zaptel 1.4.5 im having an error message when in running asterisk with the tdm card in. here's the error from the console of asterisk: [Nov 18 10:30:44] ERROR[5557]: chan_zap.c:7489 mkintf: Unable to get span status: Inappropriate ioctl for d

Re: [asterisk-users] modifying a dialed exension before dialplan processing

On Nov 17, 2007 8:30 PM, Brian J. Murrell wrote: > Both for now. I am in transition from analog to a VSP (via IAX in > fact). > > I know where you are going with this, yes, it seems my VSP will > take a 1NXXNXX even for numbers which are local, and > we are in 10 digit dialling land here n

Re: [asterisk-users] p2p t1 with sangoma hw

Jesse, thanks a lot! Its funny I just checked my email cause I just finished running the cable and was about to terminate it, perfect timing! Thanks. Now that just leaves my sangoma questions, if anyone can help me with that, I would be grateful. Thanks! Mike -Original Message- From:

Re: [asterisk-users] modifying a dialed exension before dialplan processing

On Sat, 2007-11-17 at 20:14 -0500, Baji Panchumarti wrote: > > just out of curiosity are you dialing out of * on an analog > line or are you terminating thru a sip provider, Both for now. I am in transition from analog to a VSP (via IAX in fact). I know where you are going with this, yes, it

Re: [asterisk-users] p2p t1 with sangoma hw

Michael J. Liberatore wrote: > > Also for the wiring, i have a verizon smartjack for the p2p t1 and i am > running cat5e from the smartjack to the asterisk box, do i wire this > like a standard ethernet cable t568b? Yes. > or does it need to be wired > differently? No. > Verizon was g

Re: [asterisk-users] modifying a dialed exension before dialplan processing

On Nov 17, 2007 Brian J. Murrell wrote: > [...] which means having to buy something like what > I have and spoofing a little to make it do what I want. just out of curiosity are you dialing out of * on an analog line or are you terminating thru a sip provider, or some other way ? -- __

[asterisk-users] p2p t1 with sangoma hw

Hi all, i am trying to setup my first t1 in asterisk, i have been using asterisk for several years but ahve never needed a t1 line before. I have a sangoma card already in the server with 4fxo ports. Now i ordered two single port t1 line cards from sangoma for the two servers i am connecting with

Re: [asterisk-users] modifying a dialed exension before dialplan processing

On Sat, 2007-11-17 at 12:50 -0600, Eric "ManxPower" Wieling wrote: > The real solution is for the phone to stop screwing with the digits you > are dialing. No disrespect intended, but don't you think if I could change the behaviour of the phone, I would? But like most "black box" consumer device

Re: [asterisk-users] Multiple B410P's in one machine

In an effort to better understand the interaction between multiple B410P's, mISDN, chan_misdn and Asterisk, I hope someone can add a bit more details to the clear and welcome answers presented so far. > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED

[asterisk-users] Connecting Ericsson 4422 or similar set to Asterisk ?

Hi, Has anyone managed to connect an Ericsson 4422 or similar set to Asterisk ? Thanks in advance. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lis

Re: [asterisk-users] Multiple B410P's in one machine

[EMAIL PROTECTED] wrote: > 1) Is it possible/supported to install two or more B410P Digium cards in one > computer (single Asterisk installation)? Yes, both possible and supported. > 2) Do they need to be hard-wired together with a PCM cable like I've seen > explained in some beronet manuals (al

[asterisk-users] Polycom Provisioning Tool Source Code Released

I have had so many requests for it, I have released the source. http://www.wintrisk.com/ppt.html Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-d

[asterisk-users] Building and running mISDN for B410P on Ubuntu 7.04

Hi. Using Asterisk 1.4.13 running on Ubuntu 7.04 with Intel CPU: 1) Not being able to build mISDN on Ubuntu using "make b410p" I have used mISDN-1_1_7 which seems to work ok. QUESTION: Should I expect this version of mISDN to work ok with these cards? Or is there a way to build using "make b410P"

[asterisk-users] Multiple B410P's in one machine

Hi. Using Asterisk 1.4.13 running on Ubuntu 7.04 with Intel CPU: 1) Is it possible/supported to install two or more B410P Digium cards in one computer (single Asterisk installation)? 2) Do they need to be hard-wired together with a PCM cable like I've seen explained in some beronet manuals (alth

Re: [asterisk-users] Pass CallerID when call forwards to PSTN?

On Thu, Nov 15, 2007 at 06:18:33PM +, Russell Horn wrote: > Hi, > > Incoming calls to one of my lines are set to ring two internal lines > and simultaneously start ringing my cell phone. Something like this: > > exten => s,1,Dial(SIP/2201&SIP/2202&IAX2/[EMAIL PROTECTED]),90) > > The internal

Re: [asterisk-users] Changing audio message to text message

On Fri, Nov 16, 2007 at 02:28:45PM +0100, Anthony Chapellier wrote: > Hi all, > > I know Asterisk is able to send a waiting message (audio) to people > trying to call a busy user agent using a queue. However, I'd like to > change this audio message to a text message to be able to print it on >

[asterisk-users] Page Command

Hello List, I'm looking at the page command. I was wondering if there was a way to set a wild card to dial all registered sip devices. For example page all 1XXX extensions. Thanks in advance Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Man

Re: [asterisk-users] modifying a dialed exension before dialplan processing

The real solution is for the phone to stop screwing with the digits you are dialing. It's obvious that if you plugged that phone into the PSTN it would not work at all. Find out why and fix it. Tilghman Lesher wrote: > On Saturday 17 November 2007 09:31:35 Brian J. Murrell wrote: >> On Fri, 20

Re: [asterisk-users] Two B-Channel Transfer (2BCT/TBCT) Trobule on DMS100 PRI

Jacob Lefkowitz wrote: > I have not been able to get two B-channel transfer to work on DMS100 PRI. I > consistently get the following errors: > > [Nov 6 11:12:49] ERROR[2774]: chan_zap.c:8178 zt_pri_error: ROSE RETURN > ERROR: > [Nov 6 11:12:49] ERROR[2774]: chan_zap.c:8178 zt_pri_error:OPE

Re: [asterisk-users] modifying a dialed exension before dialplan processing

On Sat, 2007-11-17 at 09:57 -0600, Tilghman Lesher wrote: > > func_odbc.conf > > [ISLOCAL] > dsn=foo > read=SELECT COUNT(*) FROM localexchanges WHERE prefix='${ARG1:0:6}' > > extensions.conf > > exten => _011-1-NXX-NXX-,1,GotoIf(${ODBC_ISLOCAL(${EXTEN:4})}? > ${EXTEN:4},1:${EXTEN:3},1) Ver

Re: [asterisk-users] modifying a dialed exension befor e dialplan processing

On Saturday 17 November 2007 09:31:35 Brian J. Murrell wrote: > On Fri, 2007-11-16 at 22:58 -0600, Eric "ManxPower" Wieling wrote: > > exten => _0111NXXNXX,1,Goto(${EXTEN:4},1) > > > > exten => _NXXNXX,1,Dial( > > Of course! Couldn't be any simpler. Almost! > > User dials 6135551212,

Re: [asterisk-users] modifying a dialed exension before dialplan processing

On Fri, 2007-11-16 at 22:58 -0600, Eric "ManxPower" Wieling wrote: > exten => _0111NXXNXX,1,Goto(${EXTEN:4},1) > > exten => _NXXNXX,1,Dial( Of course! Couldn't be any simpler. Almost! User dials 6135551212, phone sends 01116135551212, above rules processes as: Goto(6135551212,1)

Re: [asterisk-users] r2 multiframe error - continue

Ok here is some more info. Currently Elastix 0.9.0 installed and nothing more changed because i dont want to create confusion I'm in Poland, my teleco is Telekomunikacja Polska (TP) and they are using Siemens EWSD on my link. Cas, hdb3, crc4 mfcr2 are in use on link. My card is from http://www.p

[asterisk-users] chan_ss7 0.10

hi, i made tarball with some ss7 patches from www.voip-info.org and other places and put this at http://www.freevoice.cz/chan_ss7-0.10.tgz Sifira is not in active development anymore :( (but they made good work! thanks) from Changelog New in version 0.10 (community version) - port to asterisk

Re: [asterisk-users] Change the Voice promps in asterisk 1.4

http://www.voip-info.org/wiki/view/Asterisk+sound+files+international http://voicevector.com/ http://www.voip-info.org/wiki/view/Asterisk+sound+files I recall running into really great sounding French recordings somewhere, hope you find the info and sounds in the above. -baji. -- On

[asterisk-users] Astmanproxy Yahoo Group

Hi All, Is the astmanproxy yahoo group (http://tech.groups.yahoo.com/group/asterisk-astmanproxy/) still working? It seems to me the most recent posts are 2006's. I have sent a message but didn't receive any feedback and the post was not listed. Is the project still being maintained? Is there an

[asterisk-users] Blackberry MVS and Asterisk.

Hi,just spotted this on the RIM site. http://na.blackberry.com/eng/services/blackberry_mvs/ Just wondered if there is anybody working on somehow linking MVS and Asterisk, or if it is even possible? thanks Mike ___ --Bandwidth and Colocation Provided by h

[asterisk-users] California based PSTN connections

Hi, Can anyone recommend any company that can provide PSTN termination for SIP calls, at least USA based, preferably California area. One of my A*k servers is US based and I don't want my traffic flowing back and forth via my current UK PSTN provider for US<>US calls. Thanks, Adrian _

Re: [asterisk-users] Change the Voice promps in asterisk 1.4

voip crazy wrote: > Hello all, > > Which is the best way to change the default Voice promps in asteriosk > 1.4from english to french? You obtain/record new voice recordings, and add those under /var/lib/asterisk/sounds/fr/. Maybe someone has done this already and you can borrow their recordings

[asterisk-users] The call does not disconnect at the softphone when caller hangup the mobile

Hi List; I am using Firefly, and when I am calling from my mobile to the Zaptel (via pstn analoge lines), and select the firefly extension, it starts ring at the firefly, if I decided to hangup the mobile before answering the call at the firefly, then it stays ringing till being transferred for th