[asterisk-users] No Incoming Ring Tone (Even with "r" option)
Hello, I am using an incoming iax provider to bring calls to my server. When an incoming call comes in, the calling party does not hear a ring tone. I figured that this was no big deal, and that I just needed to enable the "r" flag to dial. This has not fixed the situation though. Just to try to make sure the line was being picked up properly, I tried the following: [inbound] exten => number,1,Answer exten => number,n,Background(vm-goodbye) exten => number,n,Dial(SIP/bdesktop,60,tr) exten => number,n,Hangup In this case, the caller hears the vm-goodbye message, and then silence (while the phone actually does ring). Does anyone know why the "r" option might not be working? The incoming channel is iax, and obviously the phone terminating the call is Sip. Thanks, --Brian ___ --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] installing asterisk for Ubuntu Synaptic
You need to make sure that you install the asterisk-config package as well. --Brian On 11/16/06, blackwater dev <[EMAIL PROTECTED]> wrote: I have an Ubuntu system and went into Synaptic and checked asterisk for installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc and got the following output with several errors and notices. Do I need to do more or are these ok? I expected to have some conf files in /etc/asterisk but there is nothing there. Thanks! Created by Mark Spencer <[EMAIL PROTECTED]> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. = Unable to open logger.conf: No such file or directory Asterisk Event Logger Started /var/log/asterisk/event_log Asterisk Dynamic Loader loading preload modules: == Manager registered action Ping == Manager registered action Events == Manager registered action Logoff == Manager registered action Hangup == Manager registered action Status == Manager registered action Setvar == Manager registered action Getvar == Manager registered action Redirect == Manager registered action Originate == Manager registered action Command == Manager registered action ExtensionState == Manager registered action AbsoluteTimeout == Manager registered action MailboxStatus == Manager registered action MailboxCount == Manager registered action ListCommands Nov 16 09:09:15 NOTICE[6622]: manager.c:1678 init_manager: Unable to open management configuration manager.conf. Call management disabled. Nov 16 09:09:15 NOTICE[6622]: cdr.c:1191 do_reload: CDR simple logging enabled. == RTP Allocating from port range 5000 -> 31000 Asterisk PBX Core Initializing Registering builtin applications: [AbsoluteTimeout] == Registered application 'AbsoluteTimeout' [Answer] == Registered application 'Answer' [BackGround] == Registered application 'BackGround' [Busy] == Registered application 'Busy' [Congestion] == Registered application 'Congestion' [DigitTimeout] == Registered application 'DigitTimeout' [Goto] == Registered application 'Goto' [GotoIf] == Registered application 'GotoIf' [GotoIfTime] == Registered application 'GotoIfTime' [ExecIfTime] == Registered application 'ExecIfTime' [Hangup] == Registered application 'Hangup' [NoOp] == Registered application 'NoOp' [Progress] == Registered application 'Progress' [ResetCDR] == Registered application 'ResetCDR' [ResponseTimeout] == Registered application 'ResponseTimeout' [Ringing] == Registered application 'Ringing' [SayNumber] == Registered application 'SayNumber' [SayDigits] == Registered application 'SayDigits' [SayAlpha] == Registered application 'SayAlpha' [SayPhonetic] == Registered application 'SayPhonetic' [SetAccount] == Registered application 'SetAccount' [SetAMAFlags] == Registered application 'SetAMAFlags' [SetGlobalVar] == Registered application 'SetGlobalVar' [SetLanguage] == Registered application 'SetLanguage' [Set] == Registered application 'Set' [SetVar] == Registered application 'SetVar' [ImportVar] == Registered application 'ImportVar' [Wait] == Registered application 'Wait' [WaitExten] == Registered application 'WaitExten' Asterisk Dynamic Loader Starting: [res_indications.so] => (Indications Configuration) == Registered application 'PlayTones' == Registered application 'StopPlayTones' [res_monitor.so] => (Call Monitoring Resource) == Registered application 'Monitor' == Registered application 'StopMonitor' == Registered application 'ChangeMonitor' == Manager registered action Monitor == Manager registered action StopMonitor == Manager registered action ChangeMonitor [res_adsi.so] => (ADSI Resource) [res_agi.so] => (Asterisk Gateway Interface (AGI)) == Registered application 'DeadAGI' == Registered application 'EAGI' == Registered application 'AGI' [res_features.so] => (Call Features Resource) -- Registered extension context 'parkedcalls' -- Added extension '700' priority 1 to parkedcalls == Registered application 'ParkedCall' == Registered application 'Park' == Manager registered action ParkedCalls [res_config_odbc.so] => (ODBC Configuration) Nov 16 09:09:15 NOTICE[6622]: config.c :863 ast_config_engine_register: Registered Config Engine odbc res_config_odbc loaded. [res_odbc.so] => (ODBC Resource) Nov 16 09:09:15 NOTICE[6622]: res_odbc.c:599 load_module: res_odbc loaded. [res_crypto.so] => (Cryptographic Digital Signatures) -- Loaded PUBLIC key 'freeworlddialup' -- Loaded PUBLIC key 'iaxtel' [res_musiconhold.so] => (Music On Hold Resource) == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Reg
Re: [asterisk-users] installing asterisk for Ubuntu Synaptic
You need to make sure that you install the asterisk-config package as well. --Brian On 11/16/06, blackwater dev <[EMAIL PROTECTED]> wrote: I have an Ubuntu system and went into Synaptic and checked asterisk for installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc and got the following output with several errors and notices. Do I need to do more or are these ok? I expected to have some conf files in /etc/asterisk but there is nothing there. Thanks! Created by Mark Spencer <[EMAIL PROTECTED]> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. = Unable to open logger.conf: No such file or directory Asterisk Event Logger Started /var/log/asterisk/event_log Asterisk Dynamic Loader loading preload modules: == Manager registered action Ping == Manager registered action Events == Manager registered action Logoff == Manager registered action Hangup == Manager registered action Status == Manager registered action Setvar == Manager registered action Getvar == Manager registered action Redirect == Manager registered action Originate == Manager registered action Command == Manager registered action ExtensionState == Manager registered action AbsoluteTimeout == Manager registered action MailboxStatus == Manager registered action MailboxCount == Manager registered action ListCommands Nov 16 09:09:15 NOTICE[6622]: manager.c:1678 init_manager: Unable to open management configuration manager.conf. Call management disabled. Nov 16 09:09:15 NOTICE[6622]: cdr.c:1191 do_reload: CDR simple logging enabled. == RTP Allocating from port range 5000 -> 31000 Asterisk PBX Core Initializing Registering builtin applications: [AbsoluteTimeout] == Registered application 'AbsoluteTimeout' [Answer] == Registered application 'Answer' [BackGround] == Registered application 'BackGround' [Busy] == Registered application 'Busy' [Congestion] == Registered application 'Congestion' [DigitTimeout] == Registered application 'DigitTimeout' [Goto] == Registered application 'Goto' [GotoIf] == Registered application 'GotoIf' [GotoIfTime] == Registered application 'GotoIfTime' [ExecIfTime] == Registered application 'ExecIfTime' [Hangup] == Registered application 'Hangup' [NoOp] == Registered application 'NoOp' [Progress] == Registered application 'Progress' [ResetCDR] == Registered application 'ResetCDR' [ResponseTimeout] == Registered application 'ResponseTimeout' [Ringing] == Registered application 'Ringing' [SayNumber] == Registered application 'SayNumber' [SayDigits] == Registered application 'SayDigits' [SayAlpha] == Registered application 'SayAlpha' [SayPhonetic] == Registered application 'SayPhonetic' [SetAccount] == Registered application 'SetAccount' [SetAMAFlags] == Registered application 'SetAMAFlags' [SetGlobalVar] == Registered application 'SetGlobalVar' [SetLanguage] == Registered application 'SetLanguage' [Set] == Registered application 'Set' [SetVar] == Registered application 'SetVar' [ImportVar] == Registered application 'ImportVar' [Wait] == Registered application 'Wait' [WaitExten] == Registered application 'WaitExten' Asterisk Dynamic Loader Starting: [res_indications.so] => (Indications Configuration) == Registered application 'PlayTones' == Registered application 'StopPlayTones' [res_monitor.so] => (Call Monitoring Resource) == Registered application 'Monitor' == Registered application 'StopMonitor' == Registered application 'ChangeMonitor' == Manager registered action Monitor == Manager registered action StopMonitor == Manager registered action ChangeMonitor [res_adsi.so] => (ADSI Resource) [res_agi.so] => (Asterisk Gateway Interface (AGI)) == Registered application 'DeadAGI' == Registered application 'EAGI' == Registered application 'AGI' [res_features.so] => (Call Features Resource) -- Registered extension context 'parkedcalls' -- Added extension '700' priority 1 to parkedcalls == Registered application 'ParkedCall' == Registered application 'Park' == Manager registered action ParkedCalls [res_config_odbc.so] => (ODBC Configuration) Nov 16 09:09:15 NOTICE[6622]: config.c :863 ast_config_engine_register: Registered Config Engine odbc res_config_odbc loaded. [res_odbc.so] => (ODBC Resource) Nov 16 09:09:15 NOTICE[6622]: res_odbc.c:599 load_module: res_odbc loaded. [res_crypto.so] => (Cryptographic Digital Signatures) -- Loaded PUBLIC key 'freeworlddialup' -- Loaded PUBLIC key 'iaxtel' [res_musiconhold.so] => (Music On Hold Resource) == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Regi
Re: [asterisk-users] Asterisk IVR functionality
On Mon, Nov 13, 2006 at 12:46:14PM +0100, nik600 wrote: > Hi > > i have an application developed with bayonne. > > Recentely i'm experiencing some problems and i am planning to migrate > to asterisk. > > I would like to know if i can do these things whit asterisk: > > - IVR integration with database (mysql, insert,delete,update,select) Asterisk uses a system called AGI to provide IVR. For more information see http://www.voip-info.org/wiki-Asterisk+AGI. As such, IVR applications are just scripts. You can use whatever the underlying platform supports. > - TTS Text to Speech in Asterisk is supported by Festival (http://www.voip-info.org/wiki/view/Festival). > - record exploration (for example, check if some resources are > available in the database, and list them to the user (via TTS)) Again, this just done by your script. If you can do it in the underlying langugae, you can do it in your IVR app. > > Are these things possible? > ___ > --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] simultaneous ring - call groups or queues or something else?
You can just seperate multiple phones with "&" in the Dial command, as the voip-info wiki page shows: http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial On Tue, Oct 31, 2006 at 10:28:32AM -0700, Stephen Bosch wrote: > Hi, folks: > > I need to be able to have a single DID ring multiple remote (IP and > PSTN) extensions, and then pass the call to whichever picks up first. > I'm sure this is old hat -- lots of providers offer it. > > I see that Trixbox will do it, but it's not clear how it's doing it. > They use different terminology -- a "ring group" and "hunt strategy" > > How can it be done with a straight Asterisk server? > > Thanks for the help! > > -Stephen- > ___ > --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] 1.4 and slow sound playback
I have seen this if you do not include -c1 for stereo audio files. --Brian On Tue, Oct 10, 2006 at 02:31:59PM -0400, Bill Merriam wrote: > I am testing 1.4 and am having trouble with the sound files. The gsm > files are much larger than they used to be. Sox (12.18.2) plays them > back really sllo. Apparently it thinks the > sampling rate is 8000. When I specify -r 48000 it play back properly. > > I mention the sox behavior because Asterisk plays them back the same way > sox does, very slowly. > > I am using the ulaw codec and I installed the ulaw sound files. > Asterisk still plays the sound very slowly. I don't know if it is using > the ulaw files or gsm files. > > How do I tell Asterisk to use the right sampling rate? > > Bill > ___ > --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] single conference, multiple numbers
Absolutely, the MeetMe command just takes a conference number. You could have as many extensions invoke it as you would like. --Brian On Tue, Oct 10, 2006 at 11:37:49AM +0100, Mike Williams wrote: > Hi, > > Is it within the realms of possibility to have a single conference with > multiple numbers? > > I'm thinking of getting PSTN numbers in a number of different countries so > that people in those countries only pay for a local call. > At this stage doing it with VoIP is out of the question. > > Thanks > > -- > Mike Williams > ___ > --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] CDR - mysql with asterisk 1.2.12 not working
I know that this is a silly suggestion but you should check to make sure that you actually have the cdr_mysql module, because at some point (I believe at the 1.2 release or shortly thereafter), it was moved into asterisk-addons. --Brian On Tue, Oct 10, 2006 at 08:31:43AM +0200, Garth van Sittert wrote: > Joseph wrote: > >On Mon, 2006-10-09 at 01:45 -0300, Hermann Wecke wrote: > > > >>On Sun, Oct 08, 2006 at 10:39:26PM -0600, Joseph wrote: > >> > >>>I have bind-address = 127.0.0.1 in my.cnf > >>>the cdr was working find with asterisk 1.0.1 just after upgrade > >>>something is not connecting. > >>> > >>I don't know if asterisk will use the localhost or the "network" IP to > >>connect. Just try to comment your line and see what happens. This is > >>really > >>a guess... > >> > > > >Make no difference if I use IP or "localhost" it is still not > >connecting; it could be something with the cdr_addon_mysql.so > > > >Anybody has any other ideas / suggestions? > > > > > Have you tried turning on debug in logger.conf. You should be able to > see what is wrong from there. > > Kind Regards > Garth > > ___ > --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] how to play pre-recorded file in meetme conference
I don't know if there is a better way to do this with meetme itself, but you could use the manager interface (or even the file method described in http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out). You can pass a Data argument with the filename, to an extension that simply plays a file into the conference. You may also be able to do something with the 'b' argument to MeetMe. --Brian On Mon, Oct 09, 2006 at 04:42:02PM -0400, Barry D. Hassler wrote: > Hey folks, Is it possible to play a pre-recorded file in a meetme > conference? That is, I'd like to get everyone into a conference, then > somehow play a previously recorded file (in this case, a podcast). This > isn't for individuals to call into to listen to the cast, but for it to > be played simultaneously for all in the conference. > > This would be handy for me! > > ___ > --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] astcc help-pleasssssseeee
As I recall, you nee to make sure you run this script with the DeadAGI command, not just AGI. This will make sure that the dial command will return to your script only after it is done. --Brian On Sat, Oct 07, 2006 at 10:45:10PM -0700, Ali wrote: > So what should I do? > > > > On 10/7/06, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > > > >On Fri, Oct 06, 2006 at 10:39:18PM -0700, Ali wrote: > >> Hi, > >> > >> I am wondering if astcc has ever worked for someone because it always > >return > >> 0 for answeredtime! I tracked every bit of informaion on google and wiki > >and > >> finally found out that its because of asterisk returning to dial plan > >after > >> executing Dial, so astcc.agi runs through the end without wating for > >call > >> completion. > >> > >> Am I missing something crazy? please someone give me a hint. > >> > >> > > > >astcc doesn't use strict. This is perl code I wouldn't like to touch. > > > >-- > >Tzafrir Cohen sip:[EMAIL PROTECTED] > >icq#16849755 iax:[EMAIL PROTECTED] > >+972-50-7952406 jabber:[EMAIL PROTECTED] > >[EMAIL PROTECTED] http://www.xorcom.com > >___ > >--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] Help in MySQL + Asterisk.
Check out the MySQL realtime module. It is in asterisk-addons. You can read more about this at: http://www.voip-info.org/wiki/view/Asterisk+RealTime You will need to compile the add-ons yourself though (unless your distribution includes a package for them). --Brian On Wed, Oct 04, 2006 at 04:41:51PM +0530, raviprakash sunkara wrote: > Hello Users... > Can any one help on Asterisk with MySqL > I don't want to use ODBC+MySqL. for RealTime... > Just need the MySql and Asterisk integration.. > On That i need extension.conf ,sip.conf,and voicemail.conf,meetme.conf, > musiconhold.conf are in MySql Databases accesing > > In Flaf files its working fine... with OpenSER > > Help me.. > > -- > Thanks and Regards > Ravi Prakash Sunkara > [EMAIL PROTECTED] > M:+91 9985077535 > O:+91 40 23114549 > F:+91 40 40208727 > [EMAIL PROTECTED] > www.hyperion-tech.com > ___ > --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] Running Multiple Instances of Asterisk
On Mon, Sep 25, 2006 at 12:52:43PM -0600, Douglas Garstang wrote: > > -Original Message- > > From: Brian Rogan [mailto:[EMAIL PROTECTED] > > Sent: Monday, September 25, 2006 12:40 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [asterisk-users] Running Multiple Instances of Asterisk > > > > > > to use the realtime stuff, and build your own management tools, which > > would allow you to do this (you could drastically cut the complexity > > with the right tools). Even if you could run them together, how > > would you put everything on the appropriate ports? How would you deal > > with multiple instances accessing hardware? > > Realtime is resource intensive, requiring many queries to perform simple > lookups. Check out the static config option, which just loads everything to memory at startup (just like the config file method). http://www.voip-info.org/wiki-Asterisk+RealTime (Extconfig-Static Configs section) > > > > I'm not convinced that maintaining the config files, binaries > > and other > > components of multiple asterisk's is easier than just building better > > tools to configure one. > > I am. I look at our configuration which is currently for one customer, and > there's already several dozen contexts in order to cover a lot of complexity. > Multiply that by a couple of hundred, and I won't want to be administering it! That's one way to look at it. The flip side, is you just need to maintain the same complexity just a bunch of times. Either way, I wouldn't want to be administering it ;-), but with good configuration utilities, you shouldn't have to deal with this complexity at all: you should have utilities that maintain configuration for you, and if you're going to do this, realtime is by far the best way to go. I don't pretend to know what you want in your application, but It seems clear that YOU NEED GOOD TOOLS to manage it. If you build these though, I still don't see what you could do with multiple instances that you can't do with one. If you abstract away the dial plan with your tools, what does it matter that the underlying plan is a complicated mess. In any case, take that for what its worth. --Brian ___ --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] Running Multiple Instances of Asterisk
Doug, Why do you want to do this to begin with? I think the best solution is to use the realtime stuff, and build your own management tools, which would allow you to do this (you could drastically cut the complexity with the right tools). Even if you could run them together, how would you put everything on the appropriate ports? How would you deal with multiple instances accessing hardware? I'm not convinced that maintaining the config files, binaries and other components of multiple asterisk's is easier than just building better tools to configure one. You could also try User-Mode-Linux or something like that. --Brian On Mon, Sep 25, 2006 at 12:28:30PM -0600, Douglas Garstang wrote: > > -Original Message- > > From: Eric "ManxPower" Wieling [mailto:[EMAIL PROTECTED] > > Sent: Monday, September 25, 2006 11:24 AM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [asterisk-users] Running Multiple Instances of Asterisk > > > > > > Asterisk does not support this, as it already has features for > > multi-client configuration within a single Asterisk > > installation/process. > > > > Douglas Garstang wrote: > > > I'd like to know if anyone has sucessfully managed to run > > multiple instances of Asterisk on the same system. > > > > > > - Did you run each instance as a separate user? > > > - Did you have any install or config problems? > > > - It looks like the G729 codec registration utility doesn't > > work when files aren't installed in standard places. Did you > > have this problem? > > > - How many instances could be run on a single Asterisk box? > > What do you mean 'does not support'? > > How easy do you think the management of the configuration files is going to > be if your trying to host several dozen companies on the one Asterisk > instance? Sure, you can split things into contexts, but just try and imagine > how complex the management is going to become when several companies comprise > the same file space. > > Doug > ___ > --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] OT: Opinions on Aastra 480i CT?
I don't have experience using the 480i CT, only using the 9112i, so you should take what I say with a grain of salt. I have been nothing but impressed with this phone. In terms of being friendly with *, they dedicate a section of their manual to asterisk configuration, which makes things go quite smoothly (not that the configuration is particularly difficult: its a totally standard SIP setup). As for the "No hold, no one-touch voicemail," this isn't strictly true. It has programmable soft-keys, and though I haven't experimented that extensively with them, they can be configured to dial a line (i.e. your voicemail), or "park" a call (i.e. hold). There's also other cool features, like the ability to write custom menus for the phone, that get called over HTTP. All in all, my 9112i has been pretty good (I had a few lockups, but none since I upgraded the firmware), and I would say its definitely worth buying one to see if it will work for your needs. --Brian On Mon, Sep 25, 2006 at 10:48:00AM -0400, Richard wrote: > > It's excellent home phone. I wouldn't use it in a business environment. No > hold, no one-touch voicemail. However, it works great! > > /R > > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson > Sent: Monday, September 25, 2006 10:25 AM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: [asterisk-users] OT: Opinions on Aastra 480i CT? > > Looks good, great price: > > http://www.aastratelecom.com/ipphones/pro_243.asp > > Anybody using these? How's the cordless? Does it play nice with * ? > ___ > --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] Issues with AGI+Dial command
Hi, I am still working on trying to figure out why I cannot use the Dial command from my AGI script. Can anyone tell me what I can do to get more information about what's going on. I've tried asterisk -v with as many v's as I can put on one line (like 40), and I was wondering if there is anything that I can do to debug this problem. Thanks a lot, --Brian On Fri, Sep 15, 2006 at 09:51:01AM -0400, Brian Rogan wrote: > Hello, > > I am trying to write an AGI application that will transfer the caller to > a phone number on certain conditions. From what I understand (from the > astcc application and voip-info wiki), I should just be able to EXEC the > dial command. I'm having problems with this though. I send asterisk > the following: > > EXEC Dial "Zap/g1/8475881188|30" > > I get back: > > 200 result=-1 > > On the asterisk console I see: > > -- AGI Script Executing Application: (Dial) Options: > (Zap/g1/8475881188|30) > -- Requested transfer capability: 0x00 - SPEECH > -- Called g1/8475881188 > -- Hungup 'Zap/1-1' > > > The HANGUPCAUSE variable is set to 0. When I put this dial in my > dialplan as Dial(Zap/g1/8475881188|30), the call goes through fine, so I > don't think that its the T card or any configuration. > > Has anyone ever seen this before? > > Thanks, > > --Brian > ___ > --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] Starting out
On Sun, Sep 17, 2006 at 01:25:10PM +0200, Timothy Parez wrote: > We'll have about 10 internal phones. > One of the phones should be like a central station, where all other > calls can be monitored (if possible) > and from that phone the user should be able to press a button to take > over a call which is rining on another phone. > > Then we need less advanced phones for the rest of us, but we should > still be able to pick up calls that are rining > on a phone in the same room. (if possible) > > I live in Belgium and we are using ISDN lines. > If I had to select phones from this page: > http://www.voipsolutions.be/index.php/cPath/54_24 > What whould you sugest and why ? I have used the Aastra 9112i and it seems to be an a very solid, reliable (not to mention inexpensive) phone. It also has instructions on how to configure it with Asterisk right out of the box, which is nice. I would recommend it without reservation. You might also want to look at the Aastra480i for your central station. One of the neat things that the phone offers is a whole XML based markup language that allows you to design custom menus which call HTTP pages and stuff (much like HTML). This might be just the thing for being able to take over any current calls from the central phone. --Brian ___ --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] Issues with AGI+Dial command
Hello, I am trying to write an AGI application that will transfer the caller to a phone number on certain conditions. From what I understand (from the astcc application and voip-info wiki), I should just be able to EXEC the dial command. I'm having problems with this though. I send asterisk the following: EXEC Dial "Zap/g1/8475881188|30" I get back: 200 result=-1 On the asterisk console I see: -- AGI Script Executing Application: (Dial) Options: (Zap/g1/8475881188|30) -- Requested transfer capability: 0x00 - SPEECH -- Called g1/8475881188 -- Hungup 'Zap/1-1' The HANGUPCAUSE variable is set to 0. When I put this dial in my dialplan as Dial(Zap/g1/8475881188|30), the call goes through fine, so I don't think that its the T card or any configuration. Has anyone ever seen this before? Thanks, --Brian ___ --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
