Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
Well, look at it this way: if you get the working, you can buy one of those tiny form-factor 386 boards with the 2 pcmcia slots and get a pcmcia soundcard and a ethernet port. Run Linux off a CF card and have it setup to *only* interface with Skype and Asterisk. Basically, make a Skype ATA, but it would convert Skype to SIP. I think that could still be considered an ATA, right? Or a gateway at least. Since you can make a Skype account for free and can (for right now) make US and Canada LD calls for free, I think the cost and time to make them would be worth it. :) And if you figure out a good price for them, people might even buy them from you Undrhil --- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: How many channels have you guys been able to get with this? The only problem I have with this is that it takes skype and a soundcard (virtual or otherwise) and the API is really executing commands on a running skype process. In my opinion its not worth it for 1 concurrent call per account. I have written code that works with skype in linux that simulates a virtual sound device. I have used that and successfully done calls out with this. I havent played with the dbus stuff (how you control the skype app from within linux) but since I have a soundcard that I know the audio format of it wouldnt be difficult to integrate this into asterisk, I could tweak chan_oss and make it into chan_skype fairly easily since that takes care of the other half of the equation. The only thing missing would be the events via dbus, which there are plenty of examples on so its not like all new code would have to be written. But its just not worth it if you have to have skype running for each call. And then you would potentially have to have a new username for each running process, and skype really wants X on linux so you would have to at least have the X virtual frame buffer (it works and acts like X but never displays anything or uses any hardware). That seems like an aweful lot of wasted resources on a box to connect to skype. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461 DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! ___ --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 ___ --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] FXO for PSTN
1 FXO per PSTN, so you would need 16 FXO ports. That would be accomplished by 4 TDM100P with 4 FXO modules on each. Undrhil --- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: If I have 16 PSTN for my trunklines, how many FXO do I need? Thanks. Lito ___ --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 ___ --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] TDMoE question
I asked about a similar application a few weeks back. This is sometimes referred to as campusing since you are basically going to make the two systems sharing their resources appear to be one system. From what I understand, you have to have both boxes running Asterisk. I am pretty sure that it's the Asterisk software, not the Zapata hardware, that does the actual sharing. :) Undrhil --- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: On Jun 25, 2006, at 1:55 AM, Stelios Koroneos wrote: Greetings ! I am looking into the TDMoE functionality of the Zapata drivers and * and i am kind of confused. Lets say i have 2 linux boxes, one has * running but no fxs/fxo hardware the other has a card (for example an x100p) but does not have * installed. If i just want to use the card (no * reduduncy etc) from the machine that runs *, do i need to have * running on both boxes for this to work ? or loading the appropriate drivers to the second machine will be saficient ? The examples i have seen mention zapata.conf entries which make me think that * should be running on both machines, but i am not sure if this applies in my case. Any ideas, thoughts, links etc are more than welcome Sounds like you would definitely need asterisk on the box with the card. I don't think the driver can do anything all on it's own. I am really a newbie though, so don't take my word as gospel. Marty ___ --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 ___ --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] Asking for phone number to dial
I thought Background() only allowed you one digit dialing while it's playing. Is this not the case? I agree with the reply which said that you want to use DISA, the only problem with DISA is that you have no way to use the line for answering regular calls. Once you put the DISA command in the dialplan, you get the DISA dialtone for entering you code. I suppose if you know where you will be calling from, you could code in a specific dialplan based on your callerid info, but that just seems kind of tedious just for being about to dial out. Undrhil --- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: The number dialed after Background is stored in the EXTEN variable and can be used in the Dial application. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Don Sent: Friday, June 23, 2006 5:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asking for phone number to dial background just accepts input while other sounds...etc...are being played... instead of waiting for something to end and then accept input. It doesn't store the number...etc...then add it to dial command for a zap channel - Original Message - From: T. Shaw [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, June 23, 2006 5:19 PM Subject: RE: [Asterisk-Users] Asking for phone number to dial Isn't that what the Background() application does? [EMAIL PROTECTED] blah... From: Don [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asking for phone number to dial Date: Fri, 23 Jun 2006 15:51:00 -0400 Does anyone know where to find an example or able to provide an example of how to do the following: When asterisk answers a call... Ask for number to dial...then dial that number? I am basically dialing into the asterisk box and then wanting it to take the digits I enter and dial them on an outbound zap trunk... I basically am just not sure how to have asterisk accept the digits and then use them in the dial command... ___ --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 ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.2/373 - Release Date: 6/22/2006 ___ --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 ___ --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 ___ --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] Asking for phone number to dial
I just realized that I came up with a way to use DISA and still allow for voicemail-like activity. Set background() to play your greeting and then program in an exten with 9 which would then point you to your DISA dialtone. If the caller doesn't press 9, have them go to the general leaving a message stage or whatever you want to do with real callers. Undrhil --- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: The number dialed after Background is stored in the EXTEN variable and can be used in the Dial application. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Don Sent: Friday, June 23, 2006 5:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asking for phone number to dial background just accepts input while other sounds...etc...are being played... instead of waiting for something to end and then accept input. It doesn't store the number...etc...then add it to dial command for a zap channel - Original Message - From: T. Shaw [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, June 23, 2006 5:19 PM Subject: RE: [Asterisk-Users] Asking for phone number to dial Isn't that what the Background() application does? [EMAIL PROTECTED] blah... From: Don [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asking for phone number to dial Date: Fri, 23 Jun 2006 15:51:00 -0400 Does anyone know where to find an example or able to provide an example of how to do the following: When asterisk answers a call... Ask for number to dial...then dial that number? I am basically dialing into the asterisk box and then wanting it to take the digits I enter and dial them on an outbound zap trunk... I basically am just not sure how to have asterisk accept the digits and then use them in the dial command... ___ --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 ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.2/373 - Release Date: 6/22/2006 ___ --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 ___ --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 ___ --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] Act-Tel G11112DS Telephony Gateway
Hey everyone, I recently bought an Act-Tel G2DS telephony gateway (the web interface says it's model # is GS though.) Has anyone else on this list used one of these? It has one FXO and one FXS port. I have an account for it set up in sip.conf on my Asterisk box and it apparently logs in correctly because I can dial the extension I set up in extensions.conf and the FXS port rings and I can answer it. However, I cannot dial out through my Asterisk box on it. I need to get this part working before I even think of trying to put my dial tone on the FXO port. So, has anyone use one of these and might have some kind of documentation for it? Thanks Undrhil ___ --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] kiax - iax2 softphone
That was the first thing I checked for. I have since discovered that by checking off the comfort noise selection in the options for the SIP account and then restarted kiax, I don't get the noise anymore. I guess it's something that kiax uses as a way to make sure the caller knows I'm still on the line with them. Also, it sounds more like random computer noises (beeps and stuff, like in the old Inside Your PC screensaver from Windows 98 Plus! pack.) Thanks for the help! Undrhil --- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: On Wed, 2006-06-14 at 18:31 +, [EMAIL PROTECTED] wrote: Has anyone on here used kiax before? I am asking because I have it installed on several computers and have been able to get it to connect and register to my Asterisk box. I can even call between them and my SIP softphones. The problem I am having is this: when I use kiax to call someone else, they get some kind of background music playing while I am talking to them. I have called from kiax to kiax phone and get this on both ends but I cannot identify what it is or where it's playing from. I've watched the CLI when I make the call and there is no indication of playing music on hold or background music. The dialplan doesn't even reference music on hold, it just dials whichever phone I am trying to call. This may be dumb and/or obvious, but are you playing music on either machine making the call? Since kiax uses soundcard audio it's possible for music output from a music player (amarok, winamp etc) to get into the audio stream, especially if you're using something like Arts or ESD to mux audio. Incidentally I use kiax all the time with no strange music sources noted. Just a thought. /BAK/ -- Ben Klang Alkaloid Networks 404.475.4850 [EMAIL PROTECTED] http://projects.alkaloid.net ___ --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 ___ --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] One problem (MOH) and one question (incoming SIP calls)
OK, here's another problem I've run into with Asterisk. In the musiconhold.conf file, I can set the music on hold mode to files and the directory to the place where I have my MP3s stored and they play. If I set the music on hold mode to any other setting, instead of getting my MP3s, I get something that sounds like a motorcycle being cranked up and driving off. It sounds really weird, at any rate. I've even gone so far as to remove all the mp3 files from the /var/lib/asterisk/mohmp3 directory and this music is still the only thing that will play with any other setting. Is this normal or is there something in my Linux setup (I'm running Knoppix installed to a HD) which I can check for this? The question: if my Asterisk server is on a domain which is accessible from the public internet, is it possible for someone with a SIP client which is not on my Asterisk box to be able to dial something like ([EMAIL PROTECTED]) and get an extension on my box? If so, what would I need to do to enable it or does it come that way out of the box, as it were? Undrhil ___ --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] rollover simulation
Call Waiting will not allow you to simulate rollover because the second incoming call will simply beep on the current call on that POTS line. As far as I know, Asterisk has no way of picking that call up independently of the first call. Undrhil --- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: I am trying to perform a rollover when the primary number is busy. This is coming from a POTS line. Apparently I need call waiting on the POTS line as I get immediate busy from the FXS if I don't have it. So my question is this. I have an Aastra 480I CT. The call forward when busy here seems pretty straight forward. Choose the mode as busy enter the extension in the forward number (which points to another successfully registered line on the same phone) and number of rings 1 (although I have tried 2 and 3). This setting is on the line but I have tried global as well also. Any clues? Thanks Curt ___ --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 ___ --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] Problem trying to SayDigits when an invalid extension is dialed
Does SayDigits(${EXTEN}) not work in this case? I would imagine that it would
still maintain the dialled extension in that variable, would it not?
Undrhil
--- Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
wrote:
I am trying to modify a fairly complex digital receptionist dialplan
that has a number of included contexts. Right now the system is not
announcing the extension that the caller attempted to dial, so callers
get confused when they think they dialed a valid extension but
asterisk
didn't pick everything up. I would like to have the system
announce the
entension that they attempted to dial in addition to the
error message.
However, at the part where the error announcement is
made, the extension
is set to i, so I no longer know what digits the
caller dialed. I tried
inserting a wildcard extension before this
point that saves the dialed
digits in a variable, but since my
wildcard extension matches everything,
it no longer things that an
invalid extension was dialed, so it doesn't
go to the i extension.
Is there a way I can erase the fact that an extension
was matched? Or
is there some other way of accomplishing what I am trying
to do?
Thanks,
Carl
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Re: [Asterisk-Users] ztdummy
--- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: On Jun 13, 2006, at 7:52 PM, Josué ƒonti wrote: ?? Doug, If you it will not have hardware and if ztdummy will not have installed its moh will not function correctly I believe this is no longer be true with the new Native music on hold... Marty I was under the impression that you had to use ztdummy as long as you were using the 2.4 Linux kernel. It didn't matter whether it was the native MoH or not. If you are running the 2.6 Linux kernel, you can safely ignore using ztdummy. Undrhil ___ --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] kiax - iax2 softphone
Has anyone on here used kiax before? I am asking because I have it installed on several computers and have been able to get it to connect and register to my Asterisk box. I can even call between them and my SIP softphones. The problem I am having is this: when I use kiax to call someone else, they get some kind of background music playing while I am talking to them. I have called from kiax to kiax phone and get this on both ends but I cannot identify what it is or where it's playing from. I've watched the CLI when I make the call and there is no indication of playing music on hold or background music. The dialplan doesn't even reference music on hold, it just dials whichever phone I am trying to call. So, I am wondering if this is something which I need to set in kiax and if anyone who has used kiax might be able to give me a hint on where I can disable this at. I would prefer to use an IAX softphone since it only requires one port to be open into my firewall, but kiax is the only one I've found for free. Once I get Asterisk up and running properly, I'll probably buy a couple of licenses for a good quality softphone, but until then, free is the best for me. :) So, any ideas? Undrhil ___ --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] How to retrieve voicemail
Hey. Maybe you can give me a hand with configuring my Linux box to send out emails? I've installed sendmail as per *several websites* and it's installed and running. I've gone into the voicemail.conf file and specified to allow attachments, etc. And, yes, I restarted Asterisk. Technically, I rebooted the entire Linux box. :) Anyway, any help would be appreciated. Undrhil --- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: Hi, voicemail are working ok, I receive message as attach via email. My question is : how can the user call asterisk and listen to his voicemessages ? thanks Victor ___ --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 ___ --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] Question setting up a bat phone extension.
Basically, I am looking to set up an extension which will be used as a help-line. I want it to function kind of like the bat phone from the old Batman series, where Commissioner Gordon would pick up the extension in his office and it would ring the phone back at Wayne's mansion. Is there a way to duplicate this functionality with Asterisk? I just need asterisk to auto-dial an extension when I go off-hook on another one. For instance, I have two SIP phones, Brian and Susan. When Susan goes off-hook, it should automatically ring Brian, without any other activity taking place. If Brian goes off-hook, it acts like a normal extension. Any ideas? Undrhil ___ --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] Question setting up a
Easy to do on the Linksys PAP2, if that helps. The functionality probably depends on the make and model of the phone... maybe if you gave those details as well? James Well, there's the rub. I don't have any of the hardware yet. I am looking at the various options before buying anything. I know that sounds like an odd way to do things: research it before buying it, but what can I say? :) Besides, this is all still theorhetical for me. I'm still working on getting a FXO card for my home phone line so I can get this all configured for testing. :) Undrhil ___ --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] sip
As silly as it sounds, I found the demo setup which was shown in the video by Systm (www.systm.org - episode 5) works really well for testing these. :) Once you get the SIP phone configured in sip.conf and you get the phone connected, you can make the following context in extensions.conf to test. Make sure that whatever context you use in extensions.conf is the same that you assinged your sip phone to. [sip-ext] exten = _x.,1,Answer exten = _x.,n,Wait(2) exten = _x.,n,Playback(tt-monkeys) exten = _x.,n,Hangup The file tt-monkeys.gsm is included in the standard asterisk install, so it should play. If you want to demo the system for a client, you might want to go through the list of sounds and pick a more appropriate one. :) Also, feel free to replace _x. with 555 or 1337 or something. That way you can tell that the system is getting the proper signalling from your softphone. :) I downloaded, installed and configured X-Lite last night on my Windows XP machine. The only problem I ran into was trying to figure out how to tell Asterisk the login name I would use (that's the context you define in sip.conf) and how to tell X-Lite the IP address of the Asterisk machine (domain address is where to put this.) Good luck. :) Undrhil --- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: http://www.voip-info.org/wiki/view/Asterisk+phones Scroll down and you will see a list of Softphones that you can choose from. best way to test it, imho, use: 1. Echo test - http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Echo 2. configure another sip phone on another PC and call! you may also asterisk -r on the asterisk server to get to asterisk CLI and see call progress as you make calls. rajeev On 6/8/06, issam [EMAIL PROTECTED] wrote: hello how can i configure asterisk to use soft sip phone and when asterisk is running how can I know he work correctly thanks ___ --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 ___ --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 ___ --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] Phone recommendations?
If the end-user PCs are setup with speaker/microphones or are using headsets with boom mikes, you could look around at some of the free softphones, like X-Lite or their pay-for cousins. Unfortunately, phones of any quality higher than 1-line, caller id, speaker phone are going to cost some money, especially VoIP phones, since this is a new market and the companies are trying to find the business models for selling these phones. Since people buy them at the prices they are, the companies aren't going to go lower unless forced to by (a) competition in the market or (b) newer models being released making older models less desirable. I'm afraid that's the telecom industry in a nutshell. Undrhil --- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: Because some people want a great phone for US$25. PaulH -- Paul Hales Technical Manager AsteriskIT www.asteriskit.com.au bus: 03 8320 8100 mob: 0434 673 529 Tom Vile wrote: Search the list. This question gets ask almost every week. I prefer: Snom Polycom Cisco What not too pricey? On 6/8/06, Derek [EMAIL PROTECTED] wrote: Hi All, I'm looking for a good voip hardphone that has a decent set of the regular features (conference, 2 lines, etc) thats reliable, has decent quality, and isn't too pricey. Does anyone have any suggestions? Thanks in advance. Derek -- Derek Fedel ___ --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 ___ --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 ___ --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] hangup lag causing the answering of already answered calls
So, your dialplan for that incoming call is just the one line? exten = s,1,Dial(IAX2/carey) Nothing else? Try adding a Hangup command on the next priority and see if that helps any. exten = s,2,Hangup If you already have a Hangup command in there, then I apologize for wasting your time. :) Undrhil --- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: I have a TDM-400P with one FXO module. On an incoming call, I have set Asterisk to dial my phone (exten = s,1,Dial(IAX2/carey)), which is basically the only thing in my dialplan. When the call is answered by the PSTN phone first, or when the ringing call is hung up, Asterisk keeps ringing for 5+ seconds, which causes trouble (the answering of already answered calls). I noticed in the Asterisk console that my phone is called twice every time there is an incoming call. Is this normal, and could it be causing this behaviour? If not, any ideas as to what could be causing this? I can provide full debug logs and my relevant configuration if needed. Console log: -- Starting simple switch on 'Zap/4-1' -- Executing Dial(Zap/4-1, IAX2/carey) in new stack -- Called carey -- Starting simple switch on 'Zap/4-1' -- Executing Dial(Zap/4-1, IAX2/carey) in new stack -- Called carey -- Call accepted by 10.0.12.102 (format ulaw) -- Format for call is ulaw -- Call accepted by 10.0.12.102 (format ulaw) -- Format for call is ulaw -- IAX2/carey-1 is ringing -- IAX2/carey-1 is ringing -- Hungup 'IAX2/carey-1' == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Hungup 'IAX2/carey-1' == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' ___ --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 ___ --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] This is what I want to do...
I want to set up a test system at my house using a FXO card (one of the X100P cards) and a FXS device. I am going to have the one line running into the machine and the house phones running off of the FXS port. I will be installing a softphone on my laptop and will be using Hamachi to tunnel back to my Asterisk machine while I am out and about. This way, I can leave my firewall intact and still use SIP if I need to. I know the basic differences between SIP and IAX (many ports vs. one port) but are there any bandwidth issues or other differences which might come into play? Also, I am going to set up my Asterisk box so that when I call in through the FXO number, I'll be able to dial a code and have Asterisk read me back my Caller ID / ANI info. Is this feasible? There's no need to post me example code right now unless you really really want to. I want to try to figure it out on my own. But I do want to know whether or not it's even worth trying to do. Eventually, I'm going to be getting a VoIP line to complement my one FXO line at which time I will configure Asterisk for call transfers (For John, dial 1. For Sally, dial 2. etc.) Can Asterisk trigger a true three-way user transfer if the FXO line supports it? It would require triggering a switch hook and dialing the number to transfer to and then hanging up, obviously. If it can, then I won't even need the VoIP line. :) Finally, I want to end up with a voice-mail system which will replace our answering machine (which usually blanks out when we lose power) and have the ability to call in on my FXO and have Asterisk transfer me to a number I dial on the dial-pad through the VoIP line. Is Asterisk capable of doing on-the-fly transfers like this? I'm sure I'll think of more scenarios eventually. I work in the telecom industry and I like that Asterisk can do what our top-of-the-line phone system can do at a fraction of the cost. :) Undrhil ___ --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] Campusing two Asterisk boxes?
I have been looking around some and I can't seem to find anything which will answer my question. If I have two Asterisk boxes in different locations which are linked to each other over the internet, can I configure the boxes to use each other's lines as local? In other words, let's say Site A has Phone1 for a 1FB line going into it on an FXO port. Site B has Phone2 for a 1FB line going into it on an FXO port. Is there a way to configure Site A to use Phone2 from Site B and vice versa? Undrhil ___ --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] New Member, saying Hi. :)
Hello everyone. I had heard about this open-source PBX once a while back. I wasn't too interested in it at the time but I kept the info filed away for possible future use. A couple of days ago, I was walking around Barnes and Nobles and I found this book, called Asterisk: The Future of Telephony. I paged through it a little and I was really excited by what I read. Then I remembered the open-source PBX I had read about before: it was Asterisk! This book was about that open-source PBX. It was very enlightening and I decided to buy the book so I could learn more. When I got home, I read through a few chapters and I also started looking online to find a download. I somehow managed to find a ready-made appliance called PoundKey which I downloaded and installed on my spare PC. Now I got confused because I wasn't sure where to go from the command-line prompt. So, I'm starting over at square one and I am going to download plain-jane Asterisk and get it running on a Knoppix HD installation... I hope. :) Anyway, this has been a brief (trust me, brief is good!) introduction of myself to the group. I'm sure I'll be asking lots of questions. :) Undrhil ___ --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] fine-tuning asterisk questions
For Problem #1:
exten = _X.,1,SetGroup(${EXTEN})
exten = _X.,2,GotoIf($[${GROUPCOUNT}
= 1]?104:3)
exten = _X.,3,Dial,SIP/username
exten = _X.,104,voicemail(u${EXTEN})
exten = _X.,105,hangup
This will limit the amount of incoming calls
to 1 and send everything else
to the VM.
Hey. I was under the impression
that Asterisk would, by default, send calls to priority n + 101 if the called
station was busy. Is this not the case? Why would you have to set up something
special for this to work?
Undrhil
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Re: [Asterisk-Users] fine-tuning asterisk questions
Yes you are correct... by default asterisk will send the call to priority N+101... what is your point? You asked about turning off call waiting. In the example that I provided, if the amount of active calls is 1 then it will forward to VM without dialing the exten. That is what you asked for... right? bp Nope. I am a different poster just wanting to clarify (for myself) that Asterisk would do exactly what the original poster wanted without any special programming. I wasn't aware that there would be any kind of notification to the station being called that there was a second call incoming. Everything I've read so far just says that if the station is in use, the call is routed to priority n + 101 as a busy call. Undrhil ___ --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
