RE: [Asterisk-Users] Telephone Tree

> I'd like to use Asterisk to build a phonetree (www.phonetree.com) type > of application, like this: > > 1. Read a text-based name/phonenumber file. > 2. Call every number and play a recorded message. > 3. If a beep is detected, replay the message from scratch (to leave > messages on an answering

[Asterisk-Users] Thank you very much

To James, Robert, Woody, and last but not least, Leo. Thank you very much for your suggestions on Zaurus mic/headphone configurations and the link for the softphone apps. Your help is much appreciated. Daniel ___ Asterisk-Users mailing list [EMAIL P

Re: [Asterisk-Users] Opportunistic VoIP

At 12:58 11-6-2003 -0700, you wrote: I see large benefits in using TRIP versus ENUM. I'll list some below, with #1 and #2 being the most important, and the others in no particular order. 1) The ENUM architecture is controlled by national or international governing bodies. Ultimately, they can

Re: [Asterisk-Users] Testing two E400P with E1 cross-cable

I'm using a self-made cable with just 4-wires to hook up 2 E100P. Has been working for few months without trouble. Just connect: Pin 1 to Pin 4 Pin 2 to Pin 5 Pin 4 to Pin 1 Pin 5 to Pin 2 It's the same as a T1 crossover cable. IIRC, the E1/T1 sends signal out on pair 4/5 and receives on 1/2. So

Re: [Asterisk-Users] How do i make best use of Macro?

Since there was some interest in this, here is the diff against current cvs. Someone that is better at C should look into my use of strsep because there is a couple of warnings. Also there is a warning on my use of pbx_builtin_setvar_helper, but I can't see whats wrong here. BTW, SayNumber doesn't

Re: [Asterisk-Users] AGI and SET VARIABLE

Notice that you should refer to PHONE_NUM variable this way: ${PHONE_NUM} Martin On Wed, 11 Jun 2003, Mark Street wrote: > I am having a problem understanding/visualizing the environment of AGI and how > variables defined there can be used in my dial plan. I am so close I can > taste it. I jus

RE: [Asterisk-Users] Using Linux traffic shaping to prioritise SIP/IAX traffic?

Hi Alberto, Being a QOS newbie, your example was invaluable! I'm testing your example, and so far so good. Once I have something I'm happy with, I'll post it on my Asterisk website: http://www.wwworks-inc.com/asterisk Nice work Alberto, and thanks. -wade > On Tue, J

Re: [Asterisk-Users] Voicemail notification

On Wednesday 11 June 2003 19:10, Andy Powell wrote: > I'd like to use either the message waiting light or stutter tone but > on searching the archives I found conflicting answers. > > Everyone seems to agree that you should add > > mainbox= > > but some people are saying that it should be added to

Re: [Asterisk-Users] Telephone Tree

On Wed, Jun 11, 2003 at 07:44:47PM -0600, Dylan VanHerpen wrote: > > > > > Well, I guess you'd have to include a disclaimer not to use it for > marketing or political purposes ;) Perhaps an 'abuse' clause is needed. -Andrew ___ Asterisk-Users mailing

Re: [Asterisk-Users] Telephone Tree

Steve wrote: On Wednesday 11 June 2003 08:08 pm, Dylan VanHerpen wrote: Hi everyone, I'd like to use Asterisk to build a phonetree (www.phonetree.com) type of application, like this: 1. Read a text-based name/phonenumber file. 2. Call every number and play a recorded message. 3. If a beep is d

Re: [Asterisk-Users] AGI and SET VARIABLE

On Wednesday 11 June 2003 17:10, Steven Critchfield wrote: > Why bother returning the value when you can just dial directly from AGI. Because my feeble mind is being streched a bit by AGI. Throw me a bone man. I downloaded and installed the asterisk-perl modules and changed my script

Re: [Asterisk-Users] Voicemail notification

it should be added to zapata.conf, and you can specify multiple mailboxes separated by , On Wed, 2003-06-11 at 20:10, Andy Powell wrote: > I'd like to use either the message waiting light or stutter tone but on searching > the archives I found conflicting answers. > > Everyone seems to agree tha

Re: [Asterisk-Users] Telephone Tree

On Wednesday 11 June 2003 08:08 pm, Dylan VanHerpen wrote: > Hi everyone, > > I'd like to use Asterisk to build a phonetree (www.phonetree.com) type > of application, like this: > > 1. Read a text-based name/phonenumber file. > 2. Call every number and play a recorded message. > 3. If a beep is det

Re: [Asterisk-Users] Voicemail notification

I'd like to use either the message waiting light or stutter tone but on searching the archives I found conflicting answers. Everyone seems to agree that you should add mainbox= but some people are saying that it should be added to zapata.conf and others are saying zaptel.conf Can someone who h

Re: [Asterisk-Users] AGI and SET VARIABLE

Why bother returning the value when you can just dial directly from AGI. On Wed, 2003-06-11 at 18:35, Mark Street wrote: > I am having a problem understanding/visualizing the environment of AGI and how > variables defined there can be used in my dial plan. I am so close I can > taste it. I jus

[Asterisk-Users] Telephone Tree

Hi everyone, I'd like to use Asterisk to build a phonetree (www.phonetree.com) type of application, like this: 1. Read a text-based name/phonenumber file. 2. Call every number and play a recorded message. 3. If a beep is detected, replay the message from scratch (to leave messages on an answeri

Re: [Asterisk-Users] PDA's over SIP channels on Asterisk

> "flickds" == flickds <[EMAIL PROTECTED]> writes: flickds> Is it possible for two PDA's to communicate like flickds> telephones via SIP channels on a PC running Asterisk? Certainly. There are several softphone apps available for the various platforms. And you can of course talk to any othe

[Asterisk-Users] AGI and SET VARIABLE

I am having a problem understanding/visualizing the environment of AGI and how variables defined there can be used in my dial plan. I am so close I can taste it. I just want to return a number to dial from a list of numbers in a file. from extensions.conf [talk2doc] ; Please Hold While I Tran

Re: [Asterisk-Users] Voicemail notification

On 11 Jun 2003 16:53:24 -0500, Steven Critchfield wrote: >On Wed, 2003-06-11 at 15:16, Derek Beaumont wrote: >> Besides email notification, is there another way to get asterisk notify >> the user that they have a message? >> >> Example: Some analog phones have a blinking light that lets the user

Re: [Asterisk-Users] Asterisk Hardware - Channelbank vs SIP etc

On Wed, Jun 11, 2003 at 12:42:57AM -0500, denon wrote: > We're doing a new * installation at a remote office soon, and I was just > curious what people's opinions were on hardware these days .. I've had > decent luck with T100Ps and Adtran, but I know times change .. > > I'm looking to do roughl

Re: [Asterisk-Users] Only noise in zap channel

On Wed, Jun 11, 2003 at 11:36:26AM -0300, Eduardo Goncalves wrote: > On Tue, 10 Jun 2003 14:36:25 -0400 > Scott Lambert <[EMAIL PROTECTED]> wrote: > > > Is the noise loud and sounds like you have picked up the phone in the > > middle of a modem call? > > > > If so, I had a similar problem with

Re: [Asterisk-Users] NewbieQ: SOHO setup with x100p

On Wed, Jun 11, 2003 at 10:12:22AM +0200, Tielman Koekemoer wrote: > > > On Tue, 10 Jun 2003, Tielman Koekemoer wrote: > > > > welcome to * > > Good to be here. > > > c) get a T400P + channel bank (expensive, but it does give you 24 > ports) > > I'm also considering a PRI from our local Telco

Re: [Asterisk-Users] Voicemail notification

On Wed, 2003-06-11 at 15:16, Derek Beaumont wrote: > Besides email notification, is there another way to get asterisk notify > the user that they have a message? > > Example: Some analog phones have a blinking light that lets the user > know that they have a voicemail message. > Is asterisk capab

Re: [Asterisk-Users] Opportunistic VoIP

At 12:58 11-6-2003 -0700, you wrote: I see large benefits in using TRIP versus ENUM. I'll list some below, with #1 and #2 being the most important, and the others in no particular order. 1) The ENUM architecture is controlled by national or international governing bodies. Ultimately, they can

[Asterisk-Users] Problems configuring Asterisk with SIP

Hi everybody Could someone give a tip  on how can I configure asterisk to use 2 ATA's 186 to communicate each other using SIP with asterisk. I know this most be a very simple task, however this is the very first aproach I have to asterisk. I set the following config but I don't get dial-tone w

[Asterisk-Users] Voicemail notification

Besides email notification, is there another way to get asterisk notify the user that they have a message? Example: Some analog phones have a blinking light that lets the user know that they have a voicemail message. Is asterisk capable of doing this?

Re: [Asterisk-Users] How do i make best use of Macro?

On Wed, 2003-06-11 at 14:43, Tilghman Lesher wrote: > On Wednesday 11 June 2003 02:09 pm, Steven Critchfield wrote: > > Okay, while reading over this thread it occured to me one more > > feature that should be real simple to add to app_meetme.c that > > would solve quite a bit of what is trying to

[Asterisk-Users] filling suppressed silence with chan_oh323

After some more analysis of my "dropped fragment" problem, things look like this: Cisco 7940 phone -- RTP --> chan_oh323 --> Asterisk (running, eg., VoiceMailMain) That RTP connection was negotiated via H.323 on a third machine running Cisco CallManager

Re: [Asterisk-Users] Opportunistic VoIP

This is slightly off-topic I suppose, but: I'd say it's on-topic, since it's something (if implemented) could radically change the way Asterisk moves calls between servers. If understood correctly, I believe it could be the single biggest change that the VOIP industry ("movement"?) could use to

Re: [Asterisk-Users] How do i make best use of Macro?

On Wednesday 11 June 2003 02:09 pm, Steven Critchfield wrote: > Okay, while reading over this thread it occured to me one more > feature that should be real simple to add to app_meetme.c that > would solve quite a bit of what is trying to be done here. The > feature that needs to be added is a func

[Asterisk-Users] New Asterisk System

Hello! I'm new to Asterisk, although I've had my eye on it for about a year now. I just recently installed in on RedHat 8 on a 2 GHZ system, but the sound was choppy - presumably from the onboard sound card (I read about that in the archives). So I stuck Asterisk on an old ISA system -

Re: [Asterisk-Users] How do i make best use of Macro?

Okay, while reading over this thread it occured to me one more feature that should be real simple to add to app_meetme.c that would solve quite a bit of what is trying to be done here. The feature that needs to be added is a function to pass in a variable and let meetme populate it with the current

Re: [Asterisk-Users] lost variables

Why do you think so? Local variables get lost only when the call gets hanged up. Martin On Wed, 11 Jun 2003, Paulo Mannheimer wrote: > Hi, > > Seems that my local variable content get lost when I call an AGI > program. Is this the correct functionality? > > Thanks, > > Paulo H. Mannheimer > > _

Re: [Asterisk-Users] How do i make best use of Macro?

On Wednesday 11 June 2003 01:01 pm, Christopher Arnold wrote: > On Wed, 11 Jun 2003, Tilghman Lesher wrote: > > On Wednesday 11 June 2003 10:43 am, Christopher Arnold wrote: > > > a) Is there state building up if my macro calls itself > > > recusivly? > > > > A macro is NOT a function. It simply i

[Asterisk-Users] Busy message with call waiting?

Is it possible to have both a busy and an away message when the call waiting feature is enabled? extensions.conf ... exten=>403,1,Dial,Zap/3|10 exten=>403,2,Voicemail2,u403 exten=>403,103,Voicemail2,b403 ... Because I have enabled call waiting, I can't see how it will be possible to get the busy

Re: [Asterisk-Users] How do i make best use of Macro?

On Wed, 11 Jun 2003, Tilghman Lesher wrote: > On Wednesday 11 June 2003 10:43 am, Christopher Arnold wrote: > > a) Is there state building up if my macro calls itself recusivly? > A macro is NOT a function. It simply is a shortcut to doing a > longer series of commands. A macro cannot itself b

Re: [Asterisk-Users] Testing two E400P with E1 cross-cable

Something for you to think about, your machines should be more than powerful enough to move that much data. If the system load is high, maybe you might want to look into what file type you where playing on the line to simulate the call. If it is playing a mp3, or a GSM file there is decompression a

[Asterisk-Users] segmentation asterisk oh323

B. Katisi Electrical Engineer when i call the error is give *CLI> WrapH323Connection::WrapH323Connection: WrapH323Connection created. WrapH323Connection::OnReceivedSignalSetup: Received SETUP message... 0:18.185 H225 RAS:80efe50 RAS admissionRequest rejected: callerNotRegistered 0:

RE: [Asterisk-Users] Bandwidth measurement tool: bmtools

Title: RE: [Asterisk-Users] Bandwidth measurement tool: bmtools http://s-tech.elsat.net.pl/bmtools/ -Original Message- From: Steve Bourg [mailto:[EMAIL PROTECTED]] Sent: Wednesday, June 11, 2003 11:44 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Bandwidth measurement tool:

Re: [Asterisk-Users] Testing two E400P with E1 cross-cable

Martin Pycko escribió: It should be good enough. The problem is propably in software configuration Martin On Wed, 11 Jun 2003, [UTF-8] Carlos Carús wrote: So if cable is ok, the problem must be one of these three: 1.- Config error, as Martin points (most probably) 2.- System can't hold

RE: [Asterisk-Users] Bandwidth measurement tool: bmtools

30 seconds with google gave me http://s-tech.elsat.net.pl/bmtools/ looks like the same thing at a glance. lemme see, at £60 an hour, that's 50p u owe me.. :-) Mike Pellatt > -Original Message- > From: ml.asteriskusers [mailto:[EMAIL PROTECTED] > Sent: 11 June 2003 17:44 > To: aste

RE: [Asterisk-Users] Bandwidth measurement tool: bmtools

Looks like they changed their site to http://s-tech.elsat.net.pl/bmtools/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Bourg Sent: Wednesday, June 11, 2003 12:44 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Bandwidth measurement tool: b

[Asterisk-Users] lost variables

Hi,   Seems that my local variable content get lost when I call an AGI program. Is this the correct functionality?   Thanks,   Paulo H. Mannheimer  

RE: [Asterisk-Users] Re:Some SIP questions AGAIN

Nat=1 is so that mgcp functions properly behind a NAT gateway. What kind of problems are you having with your SIP? What type of SIP phone do you have? Can you elaborate a little more or even post you SIP.conf? Here's what ours looks like so you can do a comparison: Sip.conf --- ; ; SIP

Re: [Asterisk-Users] Testing two E400P with E1 cross-cable

It should be good enough. The problem is propably in software configuration Martin On Wed, 11 Jun 2003, [UTF-8] Carlos Carús wrote: > Jared Smith escribió: > > >I have a funny feeling your crossover cable might be wrong... I'm not > >sure about an E1 crossover, but I know that a T1 crosso

Re: [Asterisk-Users] Bandwidth measurement tool: bmtools

I can't resolve this host from anywhere. Is there a mirror somewhere? Thanks, Steve Bourg On Sat, 7 Jun 2003, John Todd wrote: > > This is not specifically on-topic for Asterisk, but I have found on > many occasions while working with Asterisk that it would have been > very handy to be able to

Re: [Asterisk-Users] Testing two E400P with E1 cross-cable

Jared Smith escribió: I have a funny feeling your crossover cable might be wrong... I'm not sure about an E1 crossover, but I know that a T1 crossover is different than a standard ethernet crossover. (See http://www.jaredsmith.net/misc/cables/) If you do find the pinout for an E1 crossover, let

Re: [Asterisk-Users] How do i make best use of Macro?

On Wednesday 11 June 2003 10:43 am, Christopher Arnold wrote: > Hi, > > im trying to setup a chat system. And i belive the best way is > using an macro. But a couple of questions regarding using macros > pops up. > > a) Is there state building up if my macro calls itself recusivly? A macro is NOT

[Asterisk-Users] Re:Some SIP questions AGAIN

Hi Edwin I have my mgcp.conf almost the same as yours, except from "nat=1" , why do you put it? Anyway, DL102s now works more or less acceptably so now I'm having a battle with sip.conf Thank you for your help Michelle - Tu cuenta de correo gratuita Mixmail http://mixmail.ya.com Ya.com

Re: [Asterisk-Users] Testing two E400P with E1 cross-cable

I have a funny feeling your crossover cable might be wrong... I'm not sure about an E1 crossover, but I know that a T1 crossover is different than a standard ethernet crossover. (See http://www.jaredsmith.net/misc/cables/) If you do find the pinout for an E1 crossover, let me know and I'll add it

Re: [Asterisk-Users] Configuring zhone zplex to 24 fxs ports

On Wednesday 11 June 2003 09:04 am, Brancaleoni Matteo wrote: > I've already bought it (2 of them) ;) > So was wondering if anyone has a hint or a restore > file to put it into all fxs mode... There are two different models of the Zhone Zplex 10B. One is a combination 8/16 and the other is 24 FXS

[Asterisk-Users] How do i make best use of Macro?

Hi, im trying to setup a chat system. And i belive the best way is using an macro. But a couple of questions regarding using macros pops up. a) Is there state building up if my macro calls itself recusivly? Pseudo example: [macro-chat] to_many? Macro(chat, next_room) increase # of users in chat

Re: [Asterisk-Users] Testing two E400P with E1 cross-cable

Martin Pycko escribió: Do you see in /proc/interrupts that tor2 receives IRQs on both CPUs ? Martin On Wed, 11 Jun 2003, Carlos Carús wrote: Yes, it does too: $>cat /proc/interrupts CPU0 CPU1 0: 410297 0 local-APIC-edge timer 1: 2 2IO-APIC

Re: [Asterisk-Users] SIP phone behind NAT

Hi Olaf, I've just started working on a SIP and RTP proxy to handle exactly this. I'm really just in proof of concept at the moment but just one hour ago I got a completely successful connection out over NAT in which both endpoints thought they were talking to the proxy. I should have the code

Re: [Asterisk-Users] Testing two E400P with E1 cross-cable

Do you see in /proc/interrupts that tor2 receives IRQs on both CPUs ? Martin On Wed, 11 Jun 2003, Carlos Carús wrote: > Martin Pycko escribió: > > >Did you recompile zaptel for -D__SMP__ ? > >Check the zaptel/Makefile > > > >Martin > > > > Yes, I did :-( > > -- > Carlos Carús > Ingeniero de Si

Re: [Asterisk-Users] Testing two E400P with E1 cross-cable

Martin Pycko escribió: Did you recompile zaptel for -D__SMP__ ? Check the zaptel/Makefile Martin Yes, I did :-( -- Carlos Carús Ingeniero de Sistemas [EMAIL PROTECTED] Alisys Software Alisys Sof

Re: [Asterisk-Users] Only noise in zap channel

On Tue, 10 Jun 2003 14:36:25 -0400 Scott Lambert <[EMAIL PROTECTED]> wrote: > Is the noise loud and sounds like you have picked up the phone in the > middle of a modem call? > > If so, I had a similar problem with my TDM20 while it was sharing an IRQ > with the unused AC97 chip. I shuffled the

Re: [Asterisk-Users] Testing two E400P with E1 cross-cable

Did you recompile zaptel for -D__SMP__ ? Check the zaptel/Makefile Martin On Wed, 11 Jun 2003, Carlos Carús wrote: > Hi! > > I have the chance to play with a couple of E400P cards, each installed > in a IBM e330 XSeries servers (2 x 1GHz P-III CPU 2 Gb RAM, 36Gb SCSI > HDD with RH8.0 2.4.18-sm

Re: [Asterisk-Users] E100P Setup

Did you configure the circuit in /etc/asterisk/zapata.conf ? What do you see when you do "pri intense debug span 1" ? Do you see SABME being sent out by asterisk and no response ? Martin On Wed, 11 Jun 2003, Mark McKibbin wrote: > Can anyone give us a clue on setting up a E100P we just get Busy

Re: [Asterisk-Users] Configuring zhone zplex to 24 fxs ports

I've already bought it (2 of them) ;) So was wondering if anyone has a hint or a restore file to put it into all fxs mode... Matteo. Il mer, 2003-06-11 alle 15:27, Jeremy McNamara ha scritto: > Sure, contact the sales department at Digium. > > Jeremy McNamara > > > > Brancaleoni Matteo wrote

Re: [Asterisk-Users] Testing two E400P with E1 cross-cable

See archives on the general uselessness of RH. Your fist try should be to recompile the kernels if for nothing else than the fact that once again there is 2 bug reports against the RH shipping kernels this week, and recently announced rootable bug in the shipping kernels. RH does not have a good tr

Re: [Asterisk-Users] E100P Setup

On Wed, 2003-06-11 at 02:47, Mark McKibbin wrote: > Can anyone give us a clue on setting up a E100P we just get Busy tone > all the time. The LED on the back of the card shows green which I assume > is good. You left out a lot of information. First, is your E100P connecting to the telco? Green i

Re: [Asterisk-Users] Testing two E400P with E1 cross-cable

Michael Bielicki escribió: on your links one side has to be pri_cpe and the other one pri_net Yes, they are..One side is pri_net and the other side is obviously pri_cpe ;-) -- Carlos Carús Ingeniero de Sistemas [EMAIL PROTECTED] Alisys Software ---

Re: [Asterisk-Users] Configuring zhone zplex to 24 fxs ports

Sure, contact the sales department at Digium. Jeremy McNamara Brancaleoni Matteo wrote: Hi. I was wondering if the zplex in the dev kit could be configured to have all fxs ports, instead of the standard 8 fxo + 16 fxs. If so, anyone managed to do that? Matteo.

Re: [Asterisk-Users] Testing two E400P with E1 cross-cable

on your links one side has to be pri_cpe and the other one pri_net On Wednesday 11 Jun 2003 1:03 pm, Carlos Carús wrote: > Hi! > > I have the chance to play with a couple of E400P cards, each installed > in a IBM e330 XSeries servers (2 x 1GHz P-III CPU 2 Gb RAM, 36Gb SCSI > HDD with RH8.0 2.4.1

[Asterisk-Users] Configuring zhone zplex to 24 fxs ports

Hi. I was wondering if the zplex in the dev kit could be configured to have all fxs ports, instead of the standard 8 fxo + 16 fxs. If so, anyone managed to do that? Matteo. -- Matteo Brancaleoni Powered by RedHat Linux 8.0 Linux User #153521 -BEGIN GEEK CODE BLOCK- Version: 3.12 GS d?

[Asterisk-Users] Testing two E400P with E1 cross-cable

Hi! I have the chance to play with a couple of E400P cards, each installed in a IBM e330 XSeries servers (2 x 1GHz P-III CPU 2 Gb RAM, 36Gb SCSI HDD with RH8.0 2.4.18-smp kernel), and I'm trying to test/benchmark this e330/E400P combo generating calls thru /var/spool/asterisk/outgoing One e40

RE: [Asterisk-Users] some sip questions AGAIN

Hi Michelle, For the d-link VoIP gateways you need to configure your mgcp.conf the following is my configuration for a dg104s. Mgcp.conf -- ; ; MGCP Configuration for Asterisk ; [general] port = 2427 bindaddr = 0.0.0.0 [mgcp01] threewaycalling=yes transfer=yes callwaiting=yes callwai

RE: [Asterisk-Users] Correction regarding price of Grandstream Budgetone 100 series phone

in general, i think it is most appropriate to discuss technical issues and avoid specific price discussions for fear of having another "here is all the stuff i have to sell" discussions. however, i think bill was totally fine to correct the erroneous price quoted on the list. i think that many pe

[Asterisk-Users] SIP phone behind NAT

Hi all, I have a Asterisk at a public Network (official IP address). In the local network I have isntalled a Snom 200 IP phone and in my home network (behind NAT) a Snom 100 device. I can dial the Snom200 device from my home location without any problems but the Snom200 can not dial m

Re: [Asterisk-Users] Dialing out through a Hardware PBX

On Wed, 11 Jun 2003 [EMAIL PROTECTED] wrote: > is there any possiblity that asterisk can make calls like that, ie, first dialing 9, > and then > wait for the dial tone and then dialing the number? > how do i pause between 9 and the telephone number, will comma ( , ) do the job? > for ex. will Dia

Re: [Asterisk-Users] Underwater in 10 - 20 seconds

Sounds like a shared IRQ problem. Having had this problem myself I suggest you make sure you are not sharing any IRQ's and you may have to try a different NIC card. (I had a broadcom built on the mommy broad thing that would NOT play nice with X100P's Switching to a PCI card solved the problem) J

[Asterisk-Users] Dialing out through a Hardware PBX

hello All,   our Asterisk pbx is sitting behind a normal analog hardware pbx, we have to dial 9 to take an outside call through the hardware pbx, our fxo interface is also connected to one of the extensions of it. we can make calls to internal hardware pbx extensions by dialing through the fxo int

[Asterisk-Users] Underwater in 10 - 20 seconds

Title: Underwater in 10 - 20 seconds I'm running a X100P connected to a POTS line and a TDMP400P w/ two FXS daughter cards. Both calling out from one of the FXS phones (internally) or calling my home number (externally) the FXO card starts to freak out. By freak out I mean I can still hear bu

Re: [Asterisk-Users] Newbie : i try and test to use asterisk

Hi, You need to change your settings in X-lite: >Display name : roseau >user name : 1000 <--- this is wrong! >authorization user : >Password : >Domain/Realme : 192.168.0.2 >SIP Proxy : 192.168.0.2:5060 ; i can't have this field empty to: user name : roseau (That should match the definition

Re: [Asterisk-Users] Newbie : i try and test to use asterisk

> > i use X-Lite on windows > in setup ; > > Display name : roseau > user name : 1000 > authorization user : > Password : > Domain/Realme : 192.168.0.2 > SIP Proxy : 192.168.0.2:5060 ; i can't have this field empty > You're using a username that is different from [username] in sip.conf. You can

[Asterisk-Users] All extensions busy

Hi Firstly could I thnk everyone who has helped me so far, I just have a couple of queries I have not had chance to debug this much yet but When using the tdm40p all extesions busy themselves out, and * cannot rint the extensions for incoming calls is this because I don't have a hangup statem

[Asterisk-Users] how to receive call on iaxclient

Hello, I have successfully tested the new IAXCLIENT release (even with GUI) to initiate calls. I wonder now how I could receive call on this client (using dynamic IP address) as I didn't see any kind of registration. Thanks and regards. Francois. ___

[Asterisk-Users] Newbie : i try and test to use asterisk

I try to use X-lite with asterisk on intranet In sip.conf i have [general] port = 5060 bindaddr = 0.0.0.0 context = default [roseau] type=friend host=dynamic dtmfmode=inband context=sip [bambou] type=friend host=dynamic dtmfmode=inband context=sip and in extensions.conf [sip] exten => 1000,1

RE: [Asterisk-Users] NewbieQ: SOHO setup with x100p

> On Tue, 10 Jun 2003, Tielman Koekemoer wrote: > > welcome to * Good to be here. > c) get a T400P + channel bank (expensive, but it does give you 24 ports) I'm also considering a PRI from our local Telco (thanks to Mr Davies)connected to an E100P but am waiting for a quote from said Telco to

[Asterisk-Users] E100P Setup

Can anyone give us a clue on setting up a E100P we just get Busy tone all the time. The LED on the back of the card shows green which I assume is good. Regards Mark McKibbin DCS Internet 64 Queen St Warragul Victoria3820 Australia www.dcsi.net.au [EMAIL PROTECTED] Ph. 1300 665575 Fx. 1300 55

Re: [Asterisk-Users] Opportunistic VoIP

This is slightly off-topic I suppose, but: At 20:37 10-6-2003 -0700, you wrote: You should investigate TRIP (RFC 3129): http://www.zvon.org/tmRFC/RFC3219/Output/ Find BSD-licensed proof-of-concept code at http://www.vovida.org/downloads/trip/trip-1.0.0.tar.gz If someone could incorporate this

[Asterisk-Users] some sip questions AGAIN

I write the email again, the third time!!, cause the other two ones, I have had problems while sending them. I hope this time it works. Here is the email again: Hi (and sorry) everybody I'm starting with SIP and I wanted to ask some questions, perhaps silly ones, but I hope people can answer

[Asterisk-Users] some sip questions

I write the email again, cause the first one I have had problems while sending it. Here is the email again: Hi everybody, I'm starting with SIP and I wanted to ask some questions, perhaps silly ones, but I hope people can answer me! 1) Which codecs may I use? I want the SIP phones to call to the

[Asterisk-Users] (no subject)

Hi everybody I'm starting with SIP and I wanted to ask some questions, perhaps silly ones, but I hope people can answer me! 1) Which codecs may I use? I want the SIP phones to call to the PSTN above all, but I have two dlink dg102s (MGCP) and I'd like to can call them too. The problem is that whe