Re: [Asterisk-Users] Dial Plan Agents (1 of 2) agent-dialplan.conf

Oh i forgot to mention I have found a limitationcalls going through the queue system can NOT be parked properly. More precisely with my stdexten macro and/or the agent logic stuff the calls can NOT be rang-back to the original extension. They end up (in my example) in from-sip,s,1 whic

[Asterisk-Users] Out of 5 Grandstream BudgeTone 101 THREE are defect !!! (from Pulverstore)

I bought three plus two Grandstream BudgeTone 101 phones. The shipping cost more than the phone itself from Pulver store. The first shipping had one phone defect. Nothing on the display. (Can happen!) The second shipment had one phone with a defect display, but it still worked. The second phone'

[Asterisk-Users] Dial Plan Agents (2 of 2) extensions.com

Attached is the example extensions.conf extensions.conf Description: Binary data ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lis

[Asterisk-Users] Dial Plan Agents (1 of 2) agent-dialplan.conf

Well because I had sooo may problems with chan_agent.c I wrote this. I'm releasing it under LGPL but if you use it or anything please let me know. It'd be interesting if anyone finds this more useful than just a pile of junk. I've included a (working) example extensions file. SIP phones are as

[Asterisk-Users] Re: IAX2 doesn't respect bindaddr?

I wrote: > OK, I just [updated]. No change. :-( Here's what Asterisk says on startup: Asterisk CVS-HEAD-01/18/05-07:51:47, Copyright (C) 1999-2004 Digium. [...] == Parsing '/etc/asterisk/iax.conf': Found == Using TOS bits 0 == Binding IAX2 to '193.71.27.8:4569' == Registered channel ty

RE: [Asterisk-Users] Canadian Content: Telus and Shaw...

>I called Telus before Christmas requesting some sort of VOIP connection. >We are going with babytel. I'll advise how that works when it is up and >running, hopefully next week. [plug] www.thinktel.ca I know the guys they are competent they will sell IAX. Peered thru GT in Downtown Edmonton.

[Asterisk-Users] Auto Protocol (depending upon registration....

Hi folks, I'm sure I had this in a previous life :-) Basically the ability to dial with autoselection of either IAX2 or SIP depending upon the registration of the endpoint. Ok, I have probably missed it in the wiki as well. hints ? Gary . ___ A

[Asterisk-Users] Re: IAX2 doesn't respect bindaddr?

"Brian West" <[EMAIL PROTECTED]> writes: > update OK, I just did. No change. :-( -tih -- Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting www.eunet.no T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145 ___ Asterisk-Users m

[Asterisk-Users] Is anybody using an IAXy?

(second try !) Can anybody check my settings below. I doubt most settings in the extensions.conf. I get a dialtone if I pick up the phone! I have provisioned with iaxy.conf: ; ; IAXY Provisioning description ; dhcp codec: ulaw server: 61.220.xx.xx user: aaabbb pass: cccddd register iax.conf: ===

Re: [Asterisk-Users] Planning "hotel" phone system - Need input

--On Monday, January 17, 2005 22:20 -0800 [EMAIL PROTECTED] wrote: <...> The basic arrangement would be: Telco <-> T1 <-> Asterisk <-> T1 <-> Channel Bank <-> POTS <-> regular phone Here are my questions thus far: - Firstly, which channel bank would best suit me? I only need FXS, but I'll need C

[Asterisk-Users] Planning "hotel" phone system - Need input

Ok, I'm working on an implementation of Asterisk to service approximately 50 fractional (read: timeshare) residences. Basically what I'm starting with is a hotel phone system, but with additional functionality that Asterisk can provide. From the end user's perspective, I want the exact same fu

[Asterisk-Users] 2nd try Mediatrix 1204

Hi everybody, I have setup a Mediatrix 1204, the calls worked fine, both incoming and outgoing. The problem here is the delay. When I do a call to the PSTN or receive a call from the PSTN exists a delay of 4 seconds after answer or sending the call. For OUTGOING My Dialplan for the Mediatrix box

[Asterisk-Users] Sound quality - commercial vs. Asterisk

So far in my playing with Asterisk I've messed with soft phones (x-ten, sjphone), hard phones (Grandstream 102), and ATA adapters (Grandstream 286, Digium IAXy).   I've also got a Vonage line, using a Linksys ATA.   None of the devices I've connected to my Asterisk server have been able to m

[Asterisk-Users] VoIP Routes and Terminations

Hello All, I have been trying to do a lot of reading and researching while still being very new to the VoIP arena so I will appreciata any and all help that you may wish to provide. In my research I have come actross the idea of VoIP routes and terminations but am not exactly clear on them. Could

Re: [Asterisk-Users] I Don't Want Asterisk in the Media Path

Thank you all for replies and comments. I finally figured out what was happening. I was performing these tests using Xten Pro. There was an option enabled under Codec Order on Xten preferences. For some reason when "Use Remote Preferred Coded as Local Preferred Codec" option is enabled, Asterisk

Re: [Asterisk-Users] internal dial tone on password from outside

> > exten => s,1,Authenticate(X) > > exten => s,2,DISA,no-password|local > > > > Can someone explain to me what passcode is used for? > > > > If I enter "no-password" I can make a call but if I enter any number > > instead of word "passcode" it will not let me IN. > > Is passcode a second lev

Re: [Asterisk-Users] Wait(n) -v- Background(silence/n) ?

On Tue, 2005-01-18 at 10:44 +1100, Howard Lowndes wrote: > Will Wait(n) still listen for DTMF input from the caller after there has > been a Background(some-message) prompt, or do I need to use > Background(silence/n) to still listen for DTMF? You don't need anything but a proper gap. You need to

Re: [Asterisk-Users] On Hold music

Computer Onsite Support wrote: You right the sound card is NOT necessary like you just said. I mean when I dial extension 216 I should be able to hear the on-hold music but is not working on a new machine I want to put in production but it works on a PIII 500Mh that I want to retire. And you have t

Re: [Asterisk-Users] internal dial tone on password from outside

Joseph wrote: On Mon, 2005-01-17 at 21:43 -0500, Brian Dingman wrote: http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20DISA Thank you! DISA (Direct Inward System Access) - that is what I need. DISA,passcode|context exten => s,1,Authenticate(X) exten => s,2,DISA,no-password|local

Re: [Asterisk-Users] On Hold music

Computer Onsite Support wrote: Please be more specific regarding symbolic link of mpg321 so I can troubleshoot it myself. The strength thing is that I tried this in three other different computers and can't get it to work using same installation guide was able to get it to run on a PIII 500 which I

Re: [Asterisk-Users] transfers with zap channel

Ah, suddenly everything becomes clear.   I've never looked in features.conf before.  I now understand that 700 is supposed to intitiate the call park, and it's taking precidence over the extension I was trying to dial of 7007.  I've changed the call parking extension and now I can do regular

[Asterisk-Users] Granstream Phone "Login incorrect message"

Can anybody send me a web screen shot of their Granstream Phone configuration so I can figure out why I'm NOT able to check voice mail throw my granstream Phone. I Appreciate in advance. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Matt Riddell Sent: Mond

Re: [Asterisk-Users] internal dial tone on password from outside

On Mon, 2005-01-17 at 21:43 -0500, Brian Dingman wrote: > http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20DISA Thank you! DISA (Direct Inward System Access) - that is what I need. DISA,passcode|context exten => s,1,Authenticate(X) exten => s,2,DISA,no-password|local Can someo

RE: [Asterisk-Users] On Hold music

You right the sound card is NOT necessary like you just said. I mean when I dial extension 216 I should be able to hear the on-hold music but is not working on a new machine I want to put in production but it works on a PIII 500Mh that I want to retire. -Original Message- From: [EMAIL PROT

RE: [Asterisk-Users] On Hold music

This may sound kind of crazy and I maybe missing something. But are you placing the call on hold so you can hear the hold music. This may not be the case but you may have to place the call on hold to here the music. Also you mentioned sound, you do not need a sound card in the asterisk box to us

RE: [Asterisk-Users] On Hold music

Please be more specific regarding symbolic link of mpg321 so I can troubleshoot it myself. The strength thing is that I tried this in three other different computers and can't get it to work using same installation guide was able to get it to run on a PIII 500 which I want to get rid of it now. --

Re: [Asterisk-Users] On Hold music

Computer Onsite Support wrote: Thanks you but that didn't work. Any other solutions? Make sure you are using the correct version of mpg123... I think from memory the correct version is 0.59r Also, I think Redhat has simply made a symlink to mpg321 (which is *not* the same). -- Cheers, Matt Riddel

Re: [Asterisk-Users] transfers with zap channel

Have you looked at features.conf?   Lyle - Original Message - From: Paul Fielding To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, January 17, 2005 8:53 PM Subject: Re: [Asterisk-Users] transfers with zap channel The outside line

RE: [Asterisk-Users] On Hold music

Thanks you but that didn't work. Any other solutions? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Joseph Sent: Monday, January 17, 2005 9:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] On Hold music On Mo

Re: [Asterisk-Users] transfers with zap channel

The outside line isn't actually being dropped - the outside line hanging up is me hanging up the outside line after finding that my transfer failed.   I must be not understanding how the flash-hook works then.  My understanding was that when I flash-hook and get a second dialtone I should be

Re: [Asterisk-Users] Wait(n) -v- Background(silence/n) ?

On Tue, 2005-01-18 at 13:18, Eric Wieling wrote: > Howard Lowndes wrote: > > > Will Wait(n) still listen for DTMF input from the caller after there has > > been a Background(some-message) prompt, or do I need to use > > Background(silence/n) to still listen for DTMF? > > > > The WaitExten and Re

Re: [Asterisk-Users] internal dial tone on password from outside

When I experimented with DISA, I found it to be very unreliable - sometimes it would ignore my key presses and just keep giving dialtone, sometimes it would work. I couldn't find a rhyme or reason to it. I ended up just giving up and going with the silence Paul - Original Message ---

Re: [Asterisk-Users] On Hold music

On Mon, 2005-01-17 at 20:53 -0500, Computer Onsite Support wrote: > Can anyone of you help me out with this issue. My Asterisk is working > fine except my music-on-hold will NOT work even though I just retry > three different other machines with different board and sound. You don't need any sound

Re: [Asterisk-Users] callers who don't press any keys

> Warren Burstein wrote: > > I've noticed that some callers listen to our main menu and don't > > press any keys. > Remember Rotary Phones? They are still in use in some homes/areas John M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.c

Re: [Asterisk-Users] internal dial tone on password from outside

http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20DISA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/m

RE: [Asterisk-Users] On Hold music

I'm confuse because I was able to install on a shit machine and everything works fine even though It was not necessary installing such "timer" what you just mentioned. NOW that I just installed it on a better machine "Celeron 850Mhz 394M of Ram instead of PIII 500Mhz 128M and this thing will no pla

[Asterisk-Users] internal dial tone on password from outside

Is it possible to get an internal dial tone when I call to my asterisk and enter password? I would like to call my line enter extension - password - and get internal dial tone. once I'm in I would like to dial based on what context permits, mostly long distance calls VOIP. I can not preset the ex

Re: [Asterisk-Users] transfers with zap channel

How long between getting parked is the orginal call dropping?    Depending on your dialplan, yes dialing 700x will almost immediately send the call to call parking. (IMHO, poor extension planning can also cause this.)   I don't use the t or T options.  IMHO, you just lose the ability to use

Re: [Asterisk-Users] Wait(n) -v- Background(silence/n) ?

Howard Lowndes wrote: Will Wait(n) still listen for DTMF input from the caller after there has been a Background(some-message) prompt, or do I need to use Background(silence/n) to still listen for DTMF? The WaitExten and Read applications won't work for you?

Re: [Asterisk-Users] iaxtel - -- Format for call is ADPCM

There was a bug with codecs for a very long time with Asterisk. In [general] remove the bandwidth= line (all it does is allow specific codecs) and disallow=all and allow= for eac codec you want. Joseph wrote: When I try to call iaxtel it goes to codec ADPCM even though I have define in iax.conf

RE: [Asterisk-Users] On Hold music

Do you have Zaptel cards installed? You need to have a timer installed (whatever that means). If you don’t have a zaptel card, then use ztdummy to fake one. You need to download and compile the zaptel drivers (from asterisk website). Edit the makefile and find the line: TZOBJS=zonedata.lo tonezon

[Asterisk-Users] On Hold music

Can anyone of you help me out with this issue. My Asterisk is working fine except my music-on-hold will NOT work even though I just retry three different other machines with different board and sound. [Manny Teixeira]   al Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Beha

[Asterisk-Users] TDM13B - FXO ports not seeing incoming calls

I seem to have tripped over a problem that just seems plain weird to me... I have the TDM card with 1 FXS and 3 FXO interfaces. The FXS connects to a modem and EFTPOS terminal, which is working fine. The 3 FXO ports are connecting to another PBX as internal extensions. Usually, everything works

[Asterisk-Users] SIP URL for incoming

I want to set up my asterisk to receive SIP calls using the URL [EMAIL PROTECTED] . I have my own DNS server but would like know what entry goes into it as I have never set up SRV records before. (if it matter it is a BIND dns server).   thanx ___

Re: [Asterisk-Users] callers who don't press any keys

Warren Burstein wrote: I've noticed that some callers listen to our main menu and don't press any keys. I have it set up to restart the menu a few times and eventually hang up. I'm wondering if these are wrong numbers (in that case, why don't they hang up) or they really want to speak to someo

RE: [Asterisk-Users] pattern matching problem

> > How do I solve the problem with between patterns: > > _1800 > > _1NXX > > > > I would like all numbers 1800, 1877 etc to go through iaxtel > > but all other numbers 1xxx via voipjet > > > In your default context (i.e. the one specified in sip.conf/iax.conf) > include the iaxtel context befo

[Asterisk-Users] here's my IAX callthrough app and some questions about problems I have.

Hello all, What my app does is accepts a call in on a Dial-In Number (DID) via IAX, and then prompts the caller for the top secret password (123) and then authenticates the user and prompts them to dial in the number they'd like to call. Once they press pound after dialing in the number it will re

Re: [Asterisk-Users] callers who don't press any keys

On Mon, 2005-01-17 at 19:32 -0500, Warren Burstein wrote: > I've noticed that some callers listen to our main menu and don't press > any keys. I have it set up to restart the menu a few times and > eventually hang up. I'm wondering if these are wrong numbers (in that > case, why don't they han

[Asterisk-Users] Transferring calls on Asterisk with X-Lite

I am having trouble transferring calls using asterisk. I think it is my * installation, because this worked fine with the same system when it was hosted at our VoIP providers.   I receive a call on my IAX Trunk, to my extensions. I speak to the incoming call and tell them I’ll just tran

[Asterisk-Users] Re: Media Path Optimization & NAT

Rich Adamson wrote: Now, I would very much like to remove the "canreinvite=no" from the provider's definition on sip.conf, but doing so causes Asterisk to send a re-invite to the provider pointing to a private IP. I thought that correct localnet entries would solve this... By changing to canrei

Re: [Asterisk-Users] callers who don't press any keys

Warren Burstein wrote: I've noticed that some callers listen to our main menu and don't press any keys. I have it set up to restart the menu a few times and eventually hang up. I'm wondering if these are wrong numbers (in that case, why don't they hang up) or they really want to speak to someo

[Asterisk-Users] spandsp and app_txfax

Hi all,     Ok, I've been bashing my head for a few hours now on this, trying to figure out if I've done something wrong, but everything seems to me hunky-dory. So here's the deal:   1. I've compiled the spandsp 0.0.2pre10 source code successfully and also the asterisk     application a

Re: [Asterisk-Users] Offtopic: improving softphone latency on Linux?

On Tue, 2005-01-18 at 07:43 +0800, Steve Underwood wrote: > Latencies that big should not be due to the softphone. They are often > due to the sound card driver. Yeah, it's what I thought, but then, as said, I tried the planetccrma kernel and drivers, which are supposed to support professional a

[Asterisk-Users] callers who don't press any keys

I've noticed that some callers listen to our main menu and don't press any keys. I have it set up to restart the menu a few times and eventually hang up. I'm wondering if these are wrong numbers (in that case, why don't they hang up) or they really want to speak to someone here but don't unde

Re: [Asterisk-Users] Wait(n) -v- Background(silence/n) ?

Howard Lowndes wrote: Will Wait(n) still listen for DTMF input from the caller after there has been a Background(some-message) prompt, or do I need to use Background(silence/n) to still listen for DTMF? WaitExten(n) will ___ Asterisk-Users mailing list

[Asterisk-Users] transfers with zap channel

Ok, I've seen discussion before on doing transfers (attended and unattended), there seems to be much confusion over it.   As things sit, I've been trying (unsuccessfully) to do transfers with a zap channel (analog phone attached to TDM400).  I have no idea what I'm missing.  My current under

RE: [Asterisk-Users] Re: Any interest in a Canadian Asterisk

[EMAIL PROTECTED] wrote: > Quoting [EMAIL PROTECTED]: > >>> Would it be considered trolling to start a thread on Cleaning Maple >>> Syrup off of Dial Pads, or Wiring your Moose for Wi-Fi? >> >> Let's not forget the weekly "tooques and telephony" segment, and a >> review of the best block heaters

[Asterisk-Users] RE: Canadian Content: Telus and Shaw...

Kim Lux wrote: > I called Telus before Christmas requesting some sort of VOIP > connection. Here is what I learned: > > a) the guy I was talking to never heard of * That'll change. > b) they didn't think there was any way that a PC could > perform the duties of a PBX He was probably thinking of

Re: [Asterisk-Users] iaxtel - -- Format for call is ADPCM

> > Why is it switching me to Codec: ADPCM? > > PS. It seems to me iaxtel has a problem with connection today, can > anybody confirm it? I just tried to place a call via iaxtel and watched the packets with ethereal. The iaxtel server is very very slow to respond to _any_ packet, indicating its n

Re: [Asterisk-Users] Passing PIN Numbers

Title: Passing PIN Numbers This is a long shot, I am not sure if it will solve your problem:   Did you try to change dtmfmode in sip.conf?   Rene Kluwen Chimit   - Original Message - From: Michael Di Martino To: asterisk-users@lists.digium.com Sent: Friday, January 14

Re: [Asterisk-Users] Voice Mail Notification

Alternatively, What I (we) do personally: In stead of having * call my cellphone, it sends an MMS message with the message audio as content.   Rene Kluwen Chimit - Original Message - From: Mike Boger Jr To: Asterisk Users Mailing List - Non-Commercial Discussion Se

Re: [Asterisk-Users] SMS Gateway

There's lots. www.clickatell.com is one of them. Google for "sms gateway" and you will find a bunch - especially in the paid-add section. Rene Kluwen Chimit - Original Message - From: "Brian C. Fertig" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent

Re: [Asterisk-Users] China direct route

Direct VoIP in and out of China is illegal there, and is generally blocked by the great fire wall of China. Try an IAX connection. I don't think they have blocked those yet. :-) Steve mohammad wrote: HI; We need China direct route over H323. plz contcat me offline MSN: [EMAIL PROTECTED]

Re: [Asterisk-Users] Wait(n) -v- Background(silence/n) ?

In my AGI script I made the next trick: $digit = $AGI->get_data("vm-enter-num-to-call-then-pound", 15000, 1); while ( $digit eq 0 or $digit ) { $phoneNum .= $digit; $digit = $AGI->get_data("empty", 7000, 1); } where file empty.gsm have 0 byte length. It works like a charm

Re: [Asterisk-Users] Offtopic: improving softphone latency on Linux?

Hi Bruno, Latencies that big should not be due to the softphone. They are often due to the sound card driver. The driver for the Yamaha sound chip in my Vaio, for example, has massive latency like you are seeing. There seems to be no way to configure that driver to stop it. With the good drivers

[Asterisk-Users] Wait(n) -v- Background(silence/n) ?

Will Wait(n) still listen for DTMF input from the caller after there has been a Background(some-message) prompt, or do I need to use Background(silence/n) to still listen for DTMF? -- Howard. LANNet Computing Associates; Your Linux people -

[Asterisk-Users] iaxtel - -- Format for call is ADPCM

When I try to call iaxtel it goes to codec ADPCM even though I have define in iax.conf gsm Call accepted by 69.73.19.178 (format ADPCM) -- Format for call is ADPCM My settings: [general] port=4569 register => :[EMAIL PROTECTED] bandwidth=high jitterbuffer=no tos=lowdelay [voipjet] type=

RE: [Asterisk-Users] DIDs anywhere but here?

We have DID's in 5 Australian cities for $5 per month. Mike Sander Operations Manager Suite 4 / 38-48 Waterloo St Surry Hills N.S.W 2010 Phone:(02) 8307 8877 Fax:(02)93182254 Mobile:0401 010 289 Email: [EMAIL PROTECTED] Website: www.corporatebankinginternational.com -Original Message- Fro

Re: [Asterisk-Users] Canadian Content: Telus and Shaw...

So what stops this from happening a year from now, leaving our VOIP system high and dry for a decent broadband connection ? Matt: do you do SIP on broadband with "Telcom" Has their latency addition wrecked your connection ? Thanks. On Tue, 2005-01-18 at 12:03 +1300, Matt Riddell wrote: > K

Re: [Asterisk-Users] Canadian Content: Telus and Shaw...

Kim Lux wrote: That is a good point. I never thought of doing that. They could kill SIP connections by subtly delaying any upload traffic streams, ie introducing latency or jitter. LOL! As our New Zealand Telecom's Provider/Internet Provider has done. A little background - in New Zealand we

[Asterisk-Users] Looking for Asterisk termination in Russia

I would like to make inlimited call to russia in exchange to USA. Any idea are welcome. Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options vis

RE: [Asterisk-Users] Any interest in a Canadian Asterisk mailing list?

[EMAIL PROTECTED] wrote: > On January 17, 2005 04:47 pm, Jim Van Meggelen wrote: >> LOL. I hadn't thought of it that way. Little vignettes amidst the >> commercials? > > Exactly -- It's precisely why I hang around on linux-elitists > and a couple > other oddball lists... a good 90% of what's ther

Re: [Asterisk-Users] Offtopic: improving softphone latency on Linux?

On Mon, 2005-01-17 at 16:51 -0500, Steve Kann wrote: > What softphone are you using on Linux? > > > iaxcomm, linphone and sjphone, and they all give > > If you use an iaxclient-based softphone on linux as root, it runs with > RT priority, and pretty low latency Hmmm, on my side I can't say it

Re: [Asterisk-Users] Euro ISDN and Caller ID (Sweden)

On Mon, 17 Jan 2005, Daniel Nyström wrote: > Do anyone have experiences with Euro ISDN in Sweden? Yes, it works. We have not connected with Telia though, only other operators. > Does CallerID work properly? Both in and out. Yes. > Do anyone know of a reseller for Digium cards and/or CarrierAc

Re: [Asterisk-Users] Re: Media Path Optimization & NAT

> >>I have a bunch of setups where an Asterisk system with a public IP > >>doubles as a router/gateway/firewall for a set of phones on a private > >>network. > >> > >>We're using external SIP providers. > >> > >>Everything works quite nicely. > >> > >>Now, I would very much like to remove the "ca

Re: [Asterisk-Users] X100P Unstable.

Hi , How did you go with this problem? We have had our * box working for over 6 months now with out a problem. then all of a sudden we now have the n same problem that you have/had? Any solutions? Thanks Alex Broad Jefferson Carvalho <[EMAIL PROTECTED]> Sent by: [EMAIL PROTECTED] 29/09

Re: [Asterisk-Users] Canadian Content: Telus and Shaw...

That is a good point. I never thought of doing that. They could kill SIP connections by subtly delaying any upload traffic streams, ie introducing latency or jitter. On Mon, 2005-01-17 at 17:11 -0500, Sergey Kuznetsov wrote: > If they will do it, you are welcome to write the letter to CRTC

[Asterisk-Users] Re: Any interest in a Canadian Asterisk

Quoting [EMAIL PROTECTED]: > > Would it be considered trolling to start a thread on Cleaning Maple > > Syrup off of Dial Pads, or Wiring your Moose for Wi-Fi? > > Let's not forget the weekly "tooques and telephony" segment, and a > review of > the best block heaters for your wi-fi fones. > Oh, we

Re: [Asterisk-Users] Canadian Content: Telus and Shaw...

The CRTC is the biggest joke in the world. Ten people sit on their ass making decisions for 30+ million and no one has ever done anything to remove them from power. Heck even HBO is illegal in Canada. Why? Because they few that run the country want the many to remain their captive audience. We can

Re: [Asterisk-Users] Agent Status on FOP

On Monday 10 January 2005 04:10 pm, Richard Lyman wrote: > Joe Dennick wrote: > > The hype and documentation for the last couple of releases of the Flash > > Operator Panel claim that the Panel can be configured to either change > > the LED for a phone, or the name of a phone to indicate when that

RE: [Asterisk-Users] ntp Server and Zultys 4X4

That was the trick. Thanks for the assistance Have you had success getting the park button to work with Asterisk? Ron -Original Message- From: Bruce Komito [mailto:[EMAIL PROTECTED] Sent: Monday, January 17, 2005 11:05 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non

[Asterisk-Users] China direct route

HI;     We need China direct route over H323. plz contcat me offline   MSN: [EMAIL PROTECTED]     Regards Mohammad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or

[Asterisk-Users] Re: Media Path Optimization & NAT

Rich Adamson wrote: I have a bunch of setups where an Asterisk system with a public IP doubles as a router/gateway/firewall for a set of phones on a private network. We're using external SIP providers. Everything works quite nicely. Now, I would very much like to remove the "canreinvite=no" from

[Asterisk-Users] Multiple Line Caller Id With Polycom IP500

Greetings, I'm wondering if it's possible to display line breaks with caller ID display. I have the Polycom ip500 phone, and what I am trying to accomplish is instead of the phone saying 'Incoming call from: name/number' i want it to appear on the phone like this Incoming Call from: Menu Context

[Asterisk-Users] VOIP CONNECTION, NO AUDIO AT THE OTHER END, NEWBIE

I INSTALLED FEDORA CORE AND ASTERISK YESTERDAY. THEN I MADE A THIRD PARTY PC-PHONE CALL THROUGH VOIPJET AND IT WENT FINE. TODAY I TRIED TO MAKE A CALL AGAIN, PEOPLE AT THE OTHER END CAN'T HEAR ANY THING. I TESTED MY SOUND CARD AND IT'S WORKING PROPERLY. IT SEEMS MY CALL IS GETTING LOST SOMEWHERE IN

Re: [Asterisk-Users] Canadian Content: Telus and Shaw...

If they will do it, you are welcome to write the letter to CRTC and other governmental agencies for uncompetitive behavior. I think it should work. All the Best! Sergey. Kim Lux wrote: I called Telus before Christmas requesting some sort of VOIP connection. Here is what I learned: a) the guy I wa

Re: [Asterisk-Users] Any interest in a Canadian Asterisk mailing list?

I would be interested in this list as well. I have an positive experience how to get License Class A from CRTC. As well as I am interested to talk about LNP portability. All the Best! Sergey. Andrew Kohlsmith wrote: On January 17, 2005 04:47 pm, Jim Van Meggelen wrote: LOL. I ha

[Asterisk-Users] Queue and Normal Transfer.

Hi, Does anyone know how to get the normal transfer button to work when transferring a queue call. There seems to be a bug in app_queue that prevents calls from reaching agents. If a call is directed to an agent, and that agent transfers the call using the transfer facility on Cisco phones the cal

Re: [Asterisk-Users] Having trouble with T405P and PPP: ZT_SPANCONFIG failed

Eric Wieling aka ManxPower wrote: Adam Goryachev wrote: On Fri, 2005-01-14 at 14:38 -0800, Ben Greear wrote: Hello! I am trying to set up multi-link PPP using two T100P cards in one machine, and 1 T405P card (the 4-port one) in another machine. I have previously been able to get PPP working betwee

[Asterisk-Users] Canadian Content: Telus and Shaw...

I called Telus before Christmas requesting some sort of VOIP connection. Here is what I learned: a) the guy I was talking to never heard of * b) they didn't think there was any way that a PC could perform the duties of a PBX c) they told me they didn't have any VOIP connections, but then told me t

Re: [Asterisk-Users] Any interest in a Canadian Asterisk mailing list?

On January 17, 2005 04:47 pm, Jim Van Meggelen wrote: > LOL. I hadn't thought of it that way. Little vignettes amidst the > commercials? Exactly -- It's precisely why I hang around on linux-elitists and a couple other oddball lists... a good 90% of what's there is crap but man when something go

Re: [Asterisk-Users] Media Path Optimization & NAT

> (This message is not a dumb NAT question!) > > I have a bunch of setups where an Asterisk system with a public IP > doubles as a router/gateway/firewall for a set of phones on a private > network. > > We're using external SIP providers. > > Everything works quite nicely. > > Now, I would ve

Re: [Asterisk-Users] Offtopic: improving softphone latency on Linux?

Bruno Hertz wrote: Hi folks last weekend, I tried Windows Messenger first time and was stunned by the little latency it gives. Until now, I've been using softphones on Linux exclusively, like iaxcomm, linphone and sjphone, and they all give me about 1, at times even 2 secs delay. Whereas Messenger

Re: FW: [Asterisk-Users] Radius on *

http://voipbill.sf.net/asterisk_b2bua_v0.1.tgz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/aste

RE: [Asterisk-Users] Any interest in a Canadian Asterisk mailing list?

[EMAIL PROTECTED] wrote: > On January 17, 2005 01:57 pm, Jim Van Meggelen wrote: >> I'd have a hard time appreciating regulatory challenges in regions >> that I don't have involvement with, so I assume that the reciprocal >> is generally true as well. Granted, I shouldn't presume that everyone >> t

Re: [Asterisk-Users] Snom hint for ZAP channels?

no we have a tdm400 at this site does this still apply? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/lis

[Asterisk-Users] Echo on SIP -- not on analog.

Okay, I'm stumped. When I call the PSTN (through POTS lines), my analog phone phone works fine. My SIP phones -- a Grandstream and a Polycom -- have major echo; roughly a .25 second delay. Eventually, it goes away, which I guess is echo cancellation in action. But, dammit, why does my analo

[Asterisk-Users] X-Ten lite troubles.

Hi guys, I do have some weird situation. I do have an * box, and I want to connect to that box from my Windows box by SIP via X-Ten Lite. I made configuration of that soft phone as it was suggested by lots of tutorials I found by Google. But... it doesn't work! I don't know what is wrong there, b

[Asterisk-Users] Media Path Optimization & NAT

(This message is not a dumb NAT question!) I have a bunch of setups where an Asterisk system with a public IP doubles as a router/gateway/firewall for a set of phones on a private network. We're using external SIP providers. Everything works quite nicely. Now, I would very much like to remove th

[Asterisk-Users] How to call an extension number from ohphone to astersisk

Hi friends Can you please say me "How to send an extension number from ohphone to astersisk". For eg I have an extension 5454 at the asterisk. How can I make a call to that extension from ohphone. I tried with the command ohphone [EMAIL PROTECTED] But I could n't call that number. I want to

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