Re: [asterisk-users] What is the best softphone work with Asterisk
I tried several and had very poor luck with each I tried. These included IaxComm, IaxComm Pro, Diax and Firefly II. Also, One other one from I think Germany that had just changed it's name. All of these had issues. I could not get Firefly configured at all to talk to Asterisk. Diax, when the user places a call, just keeps ringing even when the person answered. Both IaxComms would crash. I'm sure there is one out there but I have not found it, although I have not yet tried the SIP soft phones. --On Tuesday, July 24, 2007 2:09 PM -0700 bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; I need to configure a softphone to be client and use it with Asterisk, which is the recommended one? Is it iax2? Regards Bilal _ ___ Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Debian etch and web voice mail - how to configure it?
Hi Everyone... I am running Asterisk 1.2.13 on Debian Etch. I installed it from the package. I also installed the web voice mail package, which installed Apache2 and a bunch of other stuff. When I point my browser at my PBX machine, the web page says It Works! but of course it does not. It does not seem that Apache is configured to run the vmail.cgi script. In the docs directory there is just the change log and googling it has not helped. Can someone give me a hint as to how to configure this or else point me at some docs? Thanks very much. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Debian etch and web voice mail - how to configure it?
--On Sunday, July 22, 2007 1:17 PM -0400 dave cantera [EMAIL PROTECTED] wrote: the asterisk gui doesn't interact with apache or apache2... it has it's own httpd... perhaps you can move the vmail.cgi script to the apache2 directory structure cgi-bin. I haven't tried that as of yet so I don't know how that would work. Hi Dave, thanks very much. Well I have no burning desire to use Apache at all. The Debian package for web voice mail installed it. I assumed it was required since the package manager included it. If I don't need it, great. One less thing to maintain. But, how do I activate the http server in asterisk then? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Debian etch and web voice mail - how to configure it?
--On Sunday, July 22, 2007 8:35 PM +0300 Tzafrir Cohen [EMAIL PROTECTED] wrote: We're talking about http://packages.debian.org/stable/comm/asterisk-web-vmail ( http://packages.debian.org/asterisk-web-vmail ) It actually requires httpd-cgi. Apache happens to be one of the packages that provide that... Ah, okay. When I point my browser at my PBX machine, the web page says It Works! What web page, exactly? It seems like the sample page bundled with the package, but it's not the typical Apache was just installed page I am used to seeing. It really is nothing more than a header that says It Works! which I thought odd. That package doesn't have many files: Right, but it has many dependencies, like Apache 2. So lots of stuff got dragged in along the way. The only relevant one is: /usr/lib/cgi-bin/asterisk/vmail.cgi That's there, but I was not sure if it came with the asterisk package or the asterisk-web-vmail package. So what do you get when you try: http://yourhost/cgi-bin/asteriskvmail.cgi I got NOT FOUND: The requested URL /cgi-bin/asteriskvmail.cgi was not found on this server ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Debian etch and web voice mail - how to configure it?
--On Sunday, July 22, 2007 9:03 PM +0300 Tzafrir Cohen [EMAIL PROTECTED] wrote: Oops: http://yourhost/cgi-bin/asterisk/vmail.cgi Thanks Tzafrir! That got the script to work. When I try to log in though, I get an odd error: Bleh, no /etc/asterisk/voicemail.conf at /usr/lib/cgi-bin/asterisk/vmail.cgi line 152. Line 152 seems straightforward, so I double checked to make sure the file is present. It is. I flagged it world readable just as a test. Actually, Apache runs as user nobody on Debian, so that may be needed anyhow. Regardless, it didn't fix the problem. Any ideas? Thanks again! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Debian etch and web voice mail - how to configure it?
--On Monday, July 23, 2007 7:40 AM +0300 Tzafrir Cohen [EMAIL PROTECTED] wrote: That got the script to work. When I try to log in though, I get an odd error: Bleh, no /etc/asterisk/voicemail.conf at /usr/lib/cgi-bin/asterisk/vmail.cgi line 152. It cannot read that file, or it cannot read /etc/asterisk . I can't think of a solution for this I really like. Basically add the web server to the group asterisk or any other permissions games. I had to set the /etc/asterisk directory to a+x. There are a variety of other permission issues, which I'll work through. Thanks again for your help. Best, Jim ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Slow to get dialtone when going off hook - big problem for me :(
Hi All... My Asterisk system uses VoIP and also 2 POTS lines from Cox Communications. Recently, the dial tone presentation from Cox seems to have slowed, so it can take as long as 3 seconds to get a dial tone. The problem I am having is that Asterisk does not seem to wait for the dial tone when dialing out. I'm using zaptel T400 cards. Is there any way to configure it such that I can insert a delay between the time the card goes off hook and the time it starts dialing? Alternatively, can I make it wait until there is a dial tone? Incoming calls are just fine, so I am almost certain this is what's happening. Thanks! Jim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slow to get dialtone when going off hook - big problem for me :(
--On Sunday, November 12, 2006 10:06 PM -0500 Steve Totaro [EMAIL PROTECTED] wrote: add a couple or few w's before you dial. Okay, but where? I didn't see a w option for the dial command, and if I add a wait before the dial won;t that just delay going off hook? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slow to get dialtone when going off hook - bigproblem for me :(
--On Sunday, November 12, 2006 11:53 PM -0500 John Novack [EMAIL PROTECTED] wrote: Dovid B wrote: snip How hard would it be to have asterisk detect a dial tone ? I really can't say. I am not a C programmer, so I wouldn't even know where to start. Given that cheap dial up modems have, for the past ??20?? years, have been able to do just that, I would think it should have been an early consideration For those 1% of users, the last time I tried, the insertion of a w had no effect for pulse dialing either. Well thanks to everyone who responded, and thanks to multiple w's I am back in operation. I went off hook a bunch of times and the worst case seemed to be 3 seconds to get a dial tone (which is pretty bad). It's hard to google one letter, but I eventually found that each w is .5 seconds, so 7 w's were inserted to be safe. I also called Cox and griped but I doubt that will do me any good. I am a C programmer, but I don't know anything about the inards of Asterisk. However, I would expect that dial tone detection would be a function of the hardware, not the Asterisk software. The cheap modems do this on board and export a simple command set. But I also don't know anything about Digium's hardware either. Thanks again! I really appreciate the help! Jim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voice mailbox on the fly?
Hi All... I'm trying to figure out how to get Asterisk to answer a number, prompt the caller for a code 6 digit code and then prompt the caller to leave a message. I then want to email that message out. I realize this is not likelt t be readily available, but could someone offer a suggestion about how I might implement this? Could I do it with the existing Asterisk apps or do I have to write a new one? Thanks... ___ 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/asterisk-users
Re: [Asterisk-Users] Voice mailbox on the fly?
Thanks for the reply... Well I need the voice mail WAV file mailed to a different email address, depending upon what the code is. But this looks interesting: http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime+Voicemail The only problem I see with this is that the mailbox ID is only int(5) and I need 6 or 7... Maybe I could modify the voicemail app to read the data directly from my own database structure Thinking... --On Wednesday, July 27, 2005 2:18 PM +0200 Altus Snyman [EMAIL PROTECTED] wrote: Why not exten = 123,1,BackGround(whatIsthe6Digets) exten = 123456,1,Voicemail(u123456) Jim Archer wrote: Hi All... I'm trying to figure out how to get Asterisk to answer a number, prompt the caller for a code 6 digit code and then prompt the caller to leave a message. I then want to email that message out. I realize this is not likelt t be readily available, but could someone offer a suggestion about how I might implement this? Could I do it with the existing Asterisk apps or do I have to write a new one? Thanks... ___ 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/asterisk-users ___ 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/asterisk-users
Re: [Asterisk-Users] DID + 800 Providers
http://www.junctionnetworks.com --On Sunday, July 24, 2005 9:20 PM +0200 Marc Storck [EMAIL PROTECTED] wrote: Hello, I'm looking for US DID and US50/CA 800# Providers. I found voiceconduits.com 8 month ago, there interface looks good, but there are still not live, I believe they won't be any time soon. I found sixtel, but order take eternities, they probably won't get my orders right any soon. So i'm looking for a good provider for DIDs and 800# from the US and CA, who offer online signup and ordering. The provisioning should be less than 12 hours, preferably instantly. If anybody knows or even uses such a provider, please leave me a note. Many thanks, Marc ___ 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/asterisk-users ___ 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/asterisk-users
Re: [Asterisk-Users] Best VoIP provider
I have bee using Junction Networks. Quality and reliability has been fine and they are very responsive. http://www.junctionnetworks.com --On Tuesday, July 19, 2005 2:02 AM -0400 Bernie Courtney [EMAIL PROTECTED] wrote: looking at setting up an asterisk box at my home-- what VOIP providers are you all using with the best results (and low costs! lol) thanks Bernie ___ 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/asterisk-users ___ 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/asterisk-users
[Asterisk-Users] Free Music for MOH from Digium?
Hi All... I installed the Debian Sarge Asterisk package and in the docs it had the licensing terms for the MOH, explaing that Digium (or someone) had licensed the mucic for distribution as MOH only. That's fine, but I can't find the music! Does anyone know where it can be found? Is there another source of free MOH that sounds good with Asterisk? Thanks... ___ 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/asterisk-users
Re: [Asterisk-Users] Festival questions
I'm working on this now. I don't expect it to be too useful though. --On Wednesday, July 13, 2005 3:47 PM -0400 [EMAIL PROTECTED] wrote: Hi, Is it possible to setup an Asterisk system that can allow someone to dial in using a DID and listen to their e-mail? Has anyone done this? Thanks, Mike C. ___ 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/asterisk-users ___ 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/asterisk-users
[Asterisk-Users] Strange softphone issue - audio open before answer
Hi All... I'm not sure if this is a bug or a feature. When I use a soft phone such as iaxcomm and firefly, I find that when the extension is rung from any channel (zap, IAX, SIP) that while the phone is ringing, before it is answered, audio is passed between the caller and called phone. This even happens when people call from the outside world. Is there a way to stop this? Thanks very much... ___ 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/asterisk-users
[Asterisk-Users] MeetMe problem - some parameters ignored
Hi All... I set up a conference bridge using MeetMe. It works nicely, except that it seems that certain parameters I give it are ignored or else don't work. Here is the line from my dial plan: exten = 6500,1,absolutetimeout,0 exten = 6500,2,MeetMe,100|ciMpPs|1234 The MOH and * work, but users are not announced when they join or leave and the pin is not requested. Maybe I am misunderstanding what these are supposed to do? ___ 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/asterisk-users
[Asterisk-Users] Cepstral
I have been reading about Cepstral, their voices and the Digium partner agreement with them. I see where they sell the voices and the licenses for them, but what I can't find is how to buy or get Swift? If I understand correctly, swift is the actual program that makes the speech? Strangely, the Cepstral web site does not explain this... Can someone shed some light? Thanks... ___ 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/asterisk-users
Re: [Asterisk-Users] Cepstral
Thanks William and John, I'll look again for that download. Comments below... --On Sunday, July 10, 2005 1:50 PM +0200 Wilson Pickett [EMAIL PROTECTED] wrote: FWIW? I bought that voice and I find it amusing, but not ready for prime time. I had it read articles from a publication and it was ludicrous. I can understand the people talking about ATT, I think I heard a demo that was very convincing. What is ATT? Is it another text to speech engine? I installed Festival a few days ago and have been playing with it. It sounds okay, but I decided to look to see if I could find something better. Some searching on this list and elsewhere revealed that people were raving about Cepstral, so I figured I would try it. I found their demo page and, honestly, didn't think it sounded much better than Festival. But I like that it had different voice options and Festival seems to have an Irish accent. Not that I mind an Irish accent, but in the US it would not be expected. Is there another product I should be looking at? I don't even know for sure what I am going to do with it yet, but I am certain I'll think of something. This is too cool not to use, but only if it is useful. So much depends on what you are trying to do. I just wanted to have a way to allow asterisk to talk in a demo, to show the concept. Unfortunately, showing a talking server with Cepestral's David is little like showing a prototype website: people don't always have the imagination (like we all do here :) to see what this would be like when actually done (or using a better voice in this case). People can be turned off very quickly. That's exactly why, whatever I end up doing with this, it needs to sound clear and be understandable. No one gives anything a second chance :( Jim ___ 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/asterisk-users
[Asterisk-Users] Flash and zap and # key
Hi All... I'm not sure if this is supposed to happen, but when I press the # key it seems to have the effect of flashing the hook, or at least letting me transfer. I am using Zap hardware. The problem with this is that I need to be able to dial a # key to access other systems, like my cell phone voice mail... Can I stop this somehow? Thanks... ___ 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/asterisk-users
[Asterisk-Users] Upgrading to 1.x from 0.7 on Linux
Hi All... I have an Asterisk 0.7x server running and have forever now. I would like to upgrade it to 1.0 (or whatever the current version is). It's running on Linux. I have been told there is now a Debian package for Asterisk on Sarge! I was looking at the Asterisk web site and I noticed that the Wildcard X100P cards are deprecated. I am using two of these cards to interface to POTS lines. If I upgrade to Asterisk 1.x, will I still be able to use these cards? Are there better cards I should look at that will improve quality or offer more features? Thank you very much! Jim ___ 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/asterisk-users
Re: [Asterisk-Users] Upgrading to 1.x from 0.7 on Linux
Thanks Walt, that's great! You just remindedme about something, although I don't know why. When I first set this up, I wanted Asterisk to detect distinctive ring patterns and only answer a particular pattern, so that I could share a fax line. At the time, it was not possible. Has this changed? Will new hardware do this? Thanks! Walt Reed said: On Fri, May 06, 2005 at 07:43:15PM -0400, Jim Archer said: Hi All... I have an Asterisk 0.7x server running and have forever now. I would like to upgrade it to 1.0 (or whatever the current version is). It's running on Linux. I have been told there is now a Debian package for Asterisk on Sarge! I was looking at the Asterisk web site and I noticed that the Wildcard X100P cards are deprecated. I am using two of these cards to interface to POTS lines. If I upgrade to Asterisk 1.x, will I still be able to use these cards? Are there better cards I should look at that will improve quality or offer more features? Yes. The old X100P cards still work fine (in the US, in most cases) with both 1.x and cvs HEAD (the dev branch.) That said, I'm migrating a similar setup to one X100P and one SPA3000 to cut the number of interrupts in half and free up a slot. ___ 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/asterisk-users
[Asterisk-Users] Asterisk as an outbound call machine?
Hi All... I have a need to phone a large number of people and collect information from them. I know Asterisk has a nice IVR system, but can it be used to initiate a call to people listed in a database or text file? Don't worry, this is not an annoying marketing thing. Thanks... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as an outbound call machine?
Hi Ken and thanks! That's great! I had never heard of the superdialer app
for Asterisk. I'm not sure what you mean by predictive dialing. But we
can write php code to go through the database.
I need to get this thing to run unatended. Is it possible for Asterisk to
recognize busy signals and answering machines? I realize answering
machines are tricky, but can it at least detect silence?
Thank you!
Jim
--On Saturday, September 18, 2004 11:25 AM -0700 Kenneth Shaw
[EMAIL PROTECTED] wrote:
Jim,
What you are probably looking for is a superdialer mechanism, as it is
tricky to get Asterisk to do predictive dialing.
A superdialer (if you don't know what it is) is basically a forward call
succession plan. What happens is that you connect to one phone number
after another in successive order, with your agent sitting and listening
on one end.
Accomplishing this in Asterisk is fairly easy, and I approached a
similar problem with an AGI script (written in PHP to interface with our
database).
All you would need to do is have an agent connect to a specific
extension and then launch the AGI script. Here's something off the top
of my head:
extensions.conf
exten = 5000,1,Wait(1)
exten = 5000,2,Answer
exten = 5000,3,AGI(superdialer.php,${INSERT_SOME_PARAM_HERE})
exten = 5000,4,Hangup
superdialer.php (using PHP AGI)
?php
require_once('phpagi/phpagi.php');
require_once('AwesomePhoneNumberSelectCode.inc.php');
$agi = new AGI();
$numbers = AwesomePhoneNumberSelectCode_Execute();
// assume it returns an array of phone numbers
foreach ($numbers as $number) {
$agi-conlog(SuperDialing: $number);
// dial with a 30 second timeout (approx. 5 or 6 rings)
$result = $agi-agi_exec(EXEC Dial
IAX2/[EMAIL PROTECTED]/$number|30);
if ($result['code'] != 200) {
// error here
}
$result = $agi-agi_exec('channel status');
if (!is_array($result) || $result['code'] != 200) {
// asterisk terminated on us, so exit out
break;
}
}
?
Something like the above should allow you to accomplish what you're
looking to accomplish.
-Ken Shaw...
On Sat, 2004-09-18 at 10:56, Jim Archer wrote:
Hi All...
I have a need to phone a large number of people and collect information
from them. I know Asterisk has a nice IVR system, but can it be used to
initiate a call to people listed in a database or text file?
Don't worry, this is not an annoying marketing thing.
Thanks...
___
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[Asterisk-Users] ACD autologoff problem
Hi All, I am wondering if anyone has seen this particular problem. We're running Asterisk 0.7.2 on a Debian Sid system (using the Debian package). I have a queue set up with 4 Zap channels using a TDM400P card. I have also tried this with soft phones using iaxComm, and the problem appears to be the same (or at least the symptom is the same). I have set this up so that agents can log in using the AgentCallbackLogin application like this: exten = 5300,1,AgentCallbackLogin(@default); exten = 5300,2,Hangup() When we start Asterisk, 4 agents log in. If one walks away, they are logged out because I have the autologoff parameter set to 12 seconds. This allows the call to go the the next agent in sequence. (if I don't use this parameter, the agent's phone rings forever, and other calls remain stacked up in the queue). So far, so good. However, when the logged off human agent returns from the bathroom of lunch or whatever, they log back in. Asterisk will send a call to them, and it will say it is ringing the channel on the console. Problem is, the phone never actually rings, and so they get logged back off again. The only way to clear this is by restarting Asterisk. Has anyone else experienced this and if so, is there an error on my part? Thanks... Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hardware requirements
Hi All... I just bought an IBM xSeries 205 (2.8GHZ P4 processor, 768MB RAM, EIDE HD, 5 PCI slots). My plan is to put two TDM400P cards in it and connect to an IAX2 provider to run a call center. Is this hardware adequate? I run another Asterisk box on much less hardware, so I thought this would be adequate, but someone has expressed concerns. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Are IAX2 providers ready for prime time?
Hi All... I have been putting together a call center and I was hoping to use a VoIP provider who can terminate IAX2. A participant here has been kind enough to help me off list, for which I am grateful. I have called two VoIP providers, Coloco and NuFone. Coloco has yet to return my calls and although I spoke with someone at NuFone yesterday, today when I tried to call to get service I got several all circuits are busy messages. Calling on the non-toll free number I left voice mail, but got no call back, or response to my email. I looked at Voice Pulse, but have read here that there are issues with reliability. They also seem set up for end user consumers, as do a few others. So I am starting to get a bad feeling about VoIP providers. They seem like really, really small outfits. Realistically, is the service provided by VoIP providers reliable and stable? Can you bet your business on it? How often does it go down? I would appreciate any feedback. Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Are IAX2 providers ready for prime time?
--On Friday, February 20, 2004 8:55 AM + WipeOut [EMAIL PROTECTED] wrote: The way I see it VoIP services across the internet (in the short term anyway) will never be as reliable as the standard telco provided PSTN lines.. The main reason being the nature of the Internet.. So I think the question Can you bet you business on it? actually needs to be asked in a differnt way, Can your business afford to be out of communication for the an unknown period of time for the sake of possibly cheaper phone calls?.. You are really the only one who can answer that... Well, even the most established telcos can go down once in a great while. Heck, a backhoe can dig up the wires. These things happen. What I am asking is is VoIP service reasonably reliable? Does it go down for extended periods daily? Weekly? Does it sort of stop working for a few minutes here and there during the day? Does the sound quality remain reasonable consistent? Has it progressed from the experimental stage, or is it still to be considered a toy? And, do VoIP providers support their customers? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need to interface to BRIs
Hi All... I would like to interface 4 BRI lines to Asterisk. I looked at Digium's hardware list and, although they have solutions for PRI and T1, I didn't see anything for BRI. I would like to avoid ISDN4Linux if possible. Does anyone know of any hardware suppoted by Asterisk I can use for this? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need to interface to BRIs
I forgot to mention, I am in North America. --On Monday, February 16, 2004 4:10 AM -0500 Jim Archer [EMAIL PROTECTED] wrote: Hi All... I would like to interface 4 BRI lines to Asterisk. I looked at Digium's hardware list and, although they have solutions for PRI and T1, I didn't see anything for BRI. I would like to avoid ISDN4Linux if possible. Does anyone know of any hardware suppoted by Asterisk I can use for this? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk for a call center?
Hi All... I am using Asterisk successfully in my small office as a fairly ordinary PBX. I am quite happy with it. I have a friend who needs to build a call center. The call center will be used to take orders. I have two big questions. First, can Asterisk be configured accept calls on a bunch of incoming lines, answering with a greeting and telling the person that they will be transferred to the next available operator. Then, can it watch all the extensions, and route the calls to these extensions on a first in, first out basis? Can operators somehow tell Asterisk they are ready for another call or are on break? Second, could I use a VoIP service instead of BRI lines? I experimented with iConnect quite a while ago, and the biggest problem I had with it was that I could only have one line of iConnect. I expect the software has improved since then. Thanks! Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BIG problem with multiple rings before pickup
Hi All... I have a maddening problem... I have Asterisk configured to pick up a line after 4 rings. I do this to allow my fax machine to pick up a particular distinctive ring pattern, so I don't have to pay for a dedicated fax line. If someone calls the line, lets it ring 3 times and then hangs up, Asterisk answers the line, and holds it off hook forever, constantly playing the prompts. My hardware is 2 X100P cards. Any ideas? Thanks... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BIG problem with multiple rings before pickup
Thanks, but this is not a great solution. It still leaves the line off hook for the length of the timeout and limits real calls to the timeout as well. --On Wednesday, July 02, 2003 2:11 PM -0500 Tilghman Lesher [EMAIL PROTECTED] wrote: On Wednesday 02 July 2003 01:34 pm, Jim Archer wrote: Hi All... I have a maddening problem... I have Asterisk configured to pick up a line after 4 rings. I do this to allow my fax machine to pick up a particular distinctive ring pattern, so I don't have to pay for a dedicated fax line. If someone calls the line, lets it ring 3 times and then hangs up, Asterisk answers the line, and holds it off hook forever, constantly playing the prompts. show application AbsoluteTimeout -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BIG problem with multiple rings before pickup
Ah! I see, thanks! And thanks to everyone else. I have plenty to ways to proceed now. I'm going to try the cheap solutions first. Jim --On Wednesday, July 02, 2003 3:34 PM -0500 Tilghman Lesher [EMAIL PROTECTED] wrote: On Wednesday 02 July 2003 02:42 pm, Jim Archer wrote: Thanks, but this is not a great solution. It still leaves the line off hook for the length of the timeout and limits real calls to the timeout as well. Not so. Once the user presses any DTMF, the first thing you can do is to set AbsoluteTimeout(0), which turns off the AbsoluteTimeout. If you originally set it to 60, the line will remain off hook for a maximum of 60 seconds before it will hang up. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BIG problem with multiple rings before pickup
Ok, how do I detect the pressing of any touch tone so I can set the timeout back to 0? --On Wednesday, July 02, 2003 2:51 PM -0500 Tilghman Lesher [EMAIL PROTECTED] wrote: On Wednesday 02 July 2003 02:12 pm, Joe Antkowiak wrote: How do you tell asterisk to detect for fax tones? Zaptel devices will detect fax tones automatically. If it finds them, Asterisk will attempt to go to extension fax, priority 1, if it exists. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BIG problem with multiple rings before pickup
Mark, I took a look at this app. How would I do this? Put wait for ring at the top of the loop? How do I detect and act on the return value? --On Wednesday, July 02, 2003 4:38 PM -0500 Mark Spencer [EMAIL PROTECTED] wrote: Actually there is a WaitForRing app that would probably solve this more easily. Mark On 2 Jul 2003, Steven Critchfield wrote: On Wed, 2003-07-02 at 13:34, Jim Archer wrote: Hi All... I have a maddening problem... I have Asterisk configured to pick up a line after 4 rings. I do this to allow my fax machine to pick up a particular distinctive ring pattern, so I don't have to pay for a dedicated fax line. If someone calls the line, lets it ring 3 times and then hangs up, Asterisk answers the line, and holds it off hook forever, constantly playing the prompts. My hardware is 2 X100P cards. Any ideas? Get a TDM10B, cancel your distinctive ring, and let asterisk answer immediately and detect fax tones and forward it to your fax machine. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BIG problem with multiple rings before pickup
Ok, thanks. I was hoping I would not have to set timeout back to 0 for each extension... --On Wednesday, July 02, 2003 4:54 PM -0500 Tilghman Lesher [EMAIL PROTECTED] wrote: On Wednesday 02 July 2003 04:32 pm, Jim Archer wrote: Ok, how do I detect the pressing of any touch tone so I can set the timeout back to 0? Pressing DTMF while waiting for a timeout (or while playing a file with Background) will redirect you to a new extension. For example, if you call us (700-382-4758), you'll hear a looping greeting. If you type in my extension (103), it'll jump to extension 103, priority 1, where I run AbsoluteTimeout(0), then it runs Dial on my extension. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiple X100P cards
Hi All... If I have more than 1 X100P card, how do I configure them so I know which one is which? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to transfer a call??
I was petty sure that t and T worked from calls from one extension to another. I did notice that the caller can not transfer a call that goes to an outside line. I can double check tomorrow. --On Friday, March 14, 2003 11:45 AM -0800 TC [EMAIL PROTECTED] wrote: I have T working here. Is your definition of T, that the caller can transfer ?? ie I P/U Zap 1 call Zap 2, Now I press #701 I can Park Zap2 If so is this a patch ??? pls post ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Commercial release of SIP-based IVR system
--On Saturday, March 15, 2003 12:39 PM -0600 Eric Wieling [EMAIL PROTECTED] wrote: That said, I just don't have enough experience with running Java apps in a server enviroment and not enough experience dealing with Java on a server and so I won't use Java unless there's a VERY, VERY good reason (and there seldom is). Actually, after 15 years of C, C++ and Java development I can say that for many, many server applications java is an ideal environment. The Enterprise Java Bean system (EJB) is particularly good for deploying middleware systems where shared database access is required. My company uses java to arbitrate access to a number if resources that are accessed via TCP and this works quite well. Development under java really is quicker than under C/C++ for several reasons. That said, I don't think I would choose java to build something like Asterisk. Asterisk does have realtime or at least very near real time constraints. And you can't do drivers in Asterisk. Yeah, it's tacky for this guy to have announced the product on the mailing lists, but do you really think many asterisk users will switch to it? Even if no one will, its not an appropriate use of this list, IMHO. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ATA186 MGCP or SIP?
--On Saturday, March 15, 2003 10:43 AM +0100 Michiel Betel [EMAIL PROTECTED] wrote: 266Mhz is too slow, voiceprompts get choppy, and codec conversion is not fast enough Ok, the odd part is that talking from one extension to another sounds great. The only time I have audio quality problems is when the X100P FXO is involved. Is this symptomatic of a slow processor, or should I look elsewhere? Regardless, the new machine is due in next week. Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Suggestion regarding 3 way calling
Hello... I noticed that during a three way call, if one of the three parties parks the call the other two get treated to the hold music. I was wondering if its possible for Asterisk to not play hold music if there is still more than one party on the parked call? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Suggestion regarding Mute
Hi All... I was thinking it would be really nice if there was a code users could dial to mute their microphones. Just a wish list suggestion. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to transfer a call??
I have T working here. --On Friday, March 14, 2003 9:40 AM -0600 Martin Pycko [EMAIL PROTECTED] wrote: Of courese: exten = 9998,1,Dial,SIP/9998|30|tTm Notice when you don't use the timeout you do have to use the options separator | like this: exten = 9998,1,Dial,SIP/9998||tTm but I think that T is not yet implemented regards Martin On Fri, 14 Mar 2003, WipeOut . wrote: Thanks the 'show application dial' was useful.. Can multiple options be specified? eg. exten = 9998,1,Dial,SIP/9998|30|t|T - Original Message - From: Pertti Pikkarainen [EMAIL PROTECTED] Date: Fri, 14 Mar 2003 15:15:14 +0200 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] How to transfer a call?? I have it like this exten = 9998,1,Dial,SIP/9998|30|t 30 is a timeout value Check 'show application dial' WipeOut ? wrote: What is the correct syntax to use the 't' option?? This is the current line in my extensions.conf exten = 9998,1,Dial,SIP/9998 So would I change it to exten = 9998,1,Dial,SIP/9998,t Thanks. - Original Message - From: Pertti Pikkarainen [EMAIL PROTECTED] Date: Fri, 14 Mar 2003 13:50:21 +0200 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] How to transfer a call?? Negative side effect with 't' option: all the local SIP-to-SIP media streams travel trough Asterisk instead of going direct. Right now I'm using SNOM's transfer option instead. But now I can't use * call parking because of that. Using # is probably better if there are no scaling problems. Regards Pertti Steven Critchfield wrote: If you search the archives you would find that for IP phone you need to add a 't' option to the end of your dial command. The 't' option will let the user dial '#' to get the systems attention, then dial an extention for the transfer. On Fri, 2003-03-14 at 03:32, =?iso-8859-1?B?V2lwZU91dCCZ ?= wrote: Hi, Firstly let me start off by saying that asterisk is one of the most amazing pieces of open source I have seen, it rates right up there with Apache, OpenOffice, MySQL and even Linux itself.. Nice work!! I have just installed my first server, thanks to the astinstall script.. and I have read the Handbook (ver 1) and the white paper PDF's.. and I have managed to setup 2 extentions and make calls between them using MSN Messenger, nothing fantastic but its a start.. One answer is still missing.. How do I transfer a call to another ext?? I am looking at only using IP phones and so for the test system I am using MSN Messenger.. The final solution will probably use a linux softphone line gnophone or linphone.. All I have been able to find in the docs about call transfer is using a normal phone handset and hook-flash (not quite sure what that it, I am new to telephony).. So I guess what I am asking is what do I need to configure or do to be able to transfer a call from one IP ext to another?? Thanks.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- ** Nordic LANWAN Communication Oy Pertti Pikkarainen vp of engineering E-Mail: [EMAIL PROTECTED] tel: +358-9-5024100 fax: +358-9-5023840 gsm: +358-500-511467 Sinikalliontie 16 02630 Espoo FINLAND ** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Gain settings
I cranked them up around 15 and now the voice levels appear to match the levels for the automatic voice (like the voice mail and the directory). Doing that seems to distort the caller id so Asterisk can't decode it. I'm still experimenting. Thanks! --On Wednesday, March 12, 2003 8:55 PM + T Aksoy [EMAIL PROTECTED] wrote: rxgain and txgain are in db. We have a similar problem which is even more noticeable since we divert calls by receiving on one fxo card #1 and sending out on fxo card #2. I can't seem to find a properly working solution for the attentuation which is taking place. For your issue, try setting txgain to around 6.0 and see if it's any better. I think that you will need to restart asterisk for settings to take effect. Tan - Original Message - From: Jim Archer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, March 12, 2003 2:06 AM Subject: [Asterisk-Users] Gain settings Hi All... I am using Asterisk on Debian with a single FXO card. I find that when I dial into it it sounds very soft. I also noticed that when I record VM greetings (I use the USB device for FXS) they are very soft. I saw the rxgain and txgain. Can some one tell me how these are used? I have seen examples with 0.0 and others with 100%. I have played with integers and found they can make it very loud and distorted. Whats the proper system? Thanks... Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect quality?
Hi Greg and thanks very much... A few questions... First, regarding the prefix, it seemed that this acts as a toggle, switching from the one codec to the other. But how do I set which me system uses by default? Or does iconnect always use the high bandwidth one by default (such that the always switches to the low bandwidth one)? Next, I am still struggling to understand the SIP options and what goes where. Could you please tell me where the format command goes? Is this an option on the channel? I thing the allow goes in sip.conf. Finally, does this have any impact on the problem where the person called can not be heard? Thanks!!! Jim --On Tuesday, March 11, 2003 1:35 PM -0500 Gregg Lebovitz [EMAIL PROTECTED] wrote: Jim, I changed my extensions entry for iconnect to: exten = _1XX,1,Dial,SIP/[EMAIL PROTECTED] and my calls work fine with ulaw. I am calling from a linejack card with format=ulaw and SIP with allow=ulaw. Gregg On Mon, 2003-03-10 at 23:01, Jim Archer wrote: --On Monday, March 10, 2003 4:47 PM -0300 Dan Fernandez [EMAIL PROTECTED] wrote: Iconnect uses codecs g723 and g711 that can be configured for each account (you can change them by the prefix) I tried adding the in front of a number and it reliably generates error 488 invalid media. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnect quality?
--On Tuesday, March 11, 2003 5:31 PM -0800 John Todd [EMAIL PROTECTED] wrote: Because Asterisk doesn't implement RTCP. That should have nothing to do with it, right? If a SIP BYE message gets sent to the remote end by Asterisk, the RTP connection should get shut down.Or am I missing something obvious here? this was my thought as well. There are a number of conditions under which Asterisk does not tell iconnect BYE. I often see temporarialy unavailable and other errors from iconnect, and the Dail app exits, but iconnect keeps on sending the error messages. I think if we hang up on our end we should be able to end the call. I checked the iconnect account page, and I have been billed for lengthy calls that I hung up on right at the beginning. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Gain settings
Hi All... I am using Asterisk on Debian with a single FXO card. I find that when I dial into it it sounds very soft. I also noticed that when I record VM greetings (I use the USB device for FXS) they are very soft. I saw the rxgain and txgain. Can some one tell me how these are used? I have seen examples with 0.0 and others with 100%. I have played with integers and found they can make it very loud and distorted. Whats the proper system? Thanks... Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cheap sourc of music on hold music?
Hi all... I have been shopping around and noticed that licensed music on hold music can be a bit expensive if you want to assemble a variety of types. Does anyone know of an inexpensive source? Thanks... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iconnect quality?
Does anyone here use iconnect regularly with Asterisk? If so, what do you think of its reliability and quality? I used up my 10 free minutes just getting it to work. By the end it was working, but I found that (1) many calls did not connect, due to a variety of errors reported by them (service unavailable and such) and (2) when I did connect, I could be heard but I could not hear the person I was talking to. I would appreciate hearing any experiences. Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Iconnect protocol errors? (was iconnect quality?)
--On Monday, March 10, 2003 5:13 AM -0800 William X Walsh [EMAIL PROTECTED] wrote: I have had no problems with call quality, but right now inbound calling from them is having an issue with completing the call through to asterisk. I have noticed a few little problems. From watching the udp packets fly by, it seems that if iconnect returns an error to the dial application, the dial application returns without telling the SIP server never mind. For example, if I dial a number that is invalid (this often happens because my USB device is very noisy and tones are taken wrong) iconnect returns with 404 not found. Dial then exits, but iconnect keeps sending the not found error. Asterisk receives it and dumps a note to DEBUG saying That's odd ... Other errors I have seen are temporarily unavailable, and invalid trunk. If everything works correctly, asterisk seems to issue a terminate command to iconnect. I confess, I did not read the RFC, but given this appears to be a stateful protocol based on udp should the dial application advise iconnect that it will be exiting? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP LD carriers
Thanks!
Now, I am a bit confused...
The example has (in sip.conf):
[iconnect]
type=friend
secret=stillnotmypassword
username=12691220
host=sipauth.deltathree.com
and in extensions.conf:
[macro-dialiconnect]
exten = s,1,SetCallerID(${ICONNECT1})
exten = s,2,SetCIDName(${MYNAME})
exten = s,3,Dial(SIP/[EMAIL PROTECTED],${ARG2})
exten = s,4,Playback(invalid)
exten = s,5,Hangup
exten = s,104,Playtones(busy)
exten = s,105,Wait,30
exten = s,106,Hangup
In my sip.conf I have (mostly from another example):
[iconnect]
type=friend
username=12345678
password=1234
host=sipauth.deltathree.com
callerid=My COmpany 1 222 333
In extensions.conf I have:
exten = _91NXXNXX,1,StripMSD,1
exten = _1NxxNXX,2,Dial,SIP/iconnect
When I dial, it seemd to connect to icinnect, but I never hear any riunging
and it never actually goes thorugh. If I call my cell phone, or my friends
VM, it never answers.
What did I do wrong?
Thanks...
--On Sunday, March 09, 2003 7:57 PM -0800 John Todd [EMAIL PROTECTED]
wrote:
Is there a way to configure a channel to ethernet, such that we
could use one of the VoIP long distance carriers, like Vonage?
Yes.
See http://www.loligo.com/asterisk/
for some sample configs using iconnect.
vonage does not (supposedly) allow people to use the username/password
they give you. If you can figure out some way to weasel it out of them
or the ATA-186, I'm sure the list would love to know.
JT
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Re: [Asterisk-Users] # Ouch ... error while writing audio data: :Broken pipe
My buuld is just a week old. We see that error all the time, but its never a problem... --On Monday, March 10, 2003 12:03 AM -0600 James Sizemore [EMAIL PROTECTED] wrote: Hell rm -rf asterisk cvs checkout asterisk make samples Same thing! Mark Spencer wrote: make clean ; make install? Mark On Sun, 9 Mar 2003, James Sizemore wrote: Just check-out asterisk from cvs, It compile but crashes right off with? # Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe Any ideal how far back I need to go to get a working build? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Windows XP client?
Can anyone recommend a client / phone that runs on Windows XP, with either a sound card or some other hardware? Ideally free, but does not have to be. Thanks... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Very slow welcome message...
--On Friday, March 07, 2003 10:41 PM -0600 Steven Critchfield [EMAIL PROTECTED] wrote: limp mode? Sometimes if a Linux server was not shut down correctly it will come up with the CPU speed reduced. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATA186 (was Windows XP client?)
I would prefer to be able to use actual phones. I read the earlier thread on the ATA186 device but was confused... Cisco's web site says: The Cisco ATA 186 is installed at the subscriber`s premises and supports two voice ports, each with its own independent phone number. But there seemed to be something about the licensing that prohibited the use of the second port? Also, is this device completely compatible with Asterisk? Could I use it to have two separate extensions? Will it display caller ID on an analog phones display? There seemed to be some drawbacks. --On Saturday, March 08, 2003 1:02 PM -0800 William X Walsh [EMAIL PROTECTED] wrote: But for the pricenot sure if they are worth it, when you can pick up a hardware device like an ATA186 for $150 and use a regular phone. Search for ATA186 on ecost.com. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dev kit and poor audio quality
Hi All... I spent many hours playing with Asterisk after my dev kit arrived today. One thing I noticed right off is that the audio quality of the voice mail menu items is quite poor. I thought this was a little odd, since the quality of the recorded messages was quite good. I am running on quite a low end PC, a P2 266MHz, just for testing. Is this an issue confined to the dev kit or perhaps the low end PC? Thanks... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Inexpensive VoIP phones
Hi All... I was wondering if someone could recomend an inexpensive VoIP phone. It need not be fancy. I need a few extensions for places where I need no computers. Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P question about odd behavior
Hi All... I have installed a single X100P card in my PC and am playing with Asterisk. The wire I plugged into the X100P has two POTS lines on it, wired on the RJ45 in the normal way. I am getting odd behavior. It seems when I dial out that the X100P dials both lines at the same time. I have two questions. First, I see that the X100P is only a single channel. Does this mean that I can only use one POTS line with it? When I installed it I thought that it would support two POTS lines. I guess I thought this because it has an ordinary phone jack that had 4 little metal fingers in it. Is it possible that the X100P is really dialing both lines at the same time and if so is there a way to stop this? Thanks... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distinctive ringing
Hi All... Can Asterick detect distinctive ringing on a POTS line and answer with different configurations? Thanks... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
