RE: [asterisk-users] Using Asterisk with IVR connected with legacy pbxvia rs-232
HI. Depends the kind of PBX are you using. For example in some cases like Meridian is imposible to integrate the legacy PBX funcinalities like light on the phone for indicate the voicemail sign. So i don´t know other systems but i integrate the voicemail, IVR and ACD module with Meridian Option 11 and its works perfect. So the only problem that a got was the MWI on the existing meridian phone. We resolve the issue using the mail notification. But i got some ideas how to resolve the MWI issue but you need some develop depending of the Legacy PBX. Any. Let me know. Cristian. From: Paulo Garcia [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Using Asterisk with IVR connected with legacy pbxvia rs-232 Date: Thu, 21 Sep 2006 09:13:14 -0300 Hi, I have some cases that I need to use Asterisk as an IVR/VoiceMail only. It will be connected to legacy pbx using a serial port (R2-232) to exchange integrations and/or messages to allow pbx to send to terminal extensions 'message indications' (a led on in KS). I know Asterisk can do it alone and better, but in some cases isn't possible to change the pbx structure and this protocol via rs-232 is widely used for some big pbx systems. Any direction? Is there already a solution for this? Or I need to do a custom development? Thanks in advance! Paulo ___ --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 _ Find a local pizza place, music store, museum and more then map the best route! http://local.live.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
Re: [asterisk-users] voicemailmain
Ok you have two optionsthe iax extension is created under default
context???
The VoceMilMain could be configured with the options of wich context use
like this:
extensions.conf:
exten = 2999,1,Answer
exten = 2999,2,Wait,2
exten = 2999,3,Voicemailmain(@test)
Where test is the context where the iax client belong.
Let me know.
Chers.
Cris.
From: existx [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Subject: Re: [asterisk-users] voicemailmain
Date: Thu, 24 Aug 2006 16:39:35 -0400
Cristian,
The only other line in extensions.conf that references VoicemailMain is
this:
exten = a,1,VoicemailMain(${ARG1})
The error from the CLI is:
Aug 24 16:13:49 NOTICE[23174]: chan_iax2.c:7241 socket_read: Rejected
connect attempt from 192.168.0.23, request '[EMAIL PROTECTED]' does not
exist
Regards,
Jason
On 8/24/06, kritikus Araklidas [EMAIL PROTECTED] wrote:
Hi:
First it at all check if you have a different extension for
voicemailmain.?
Then use VoiceMailMain syntax.
And send me the CLI log when you try to connect to VoiceMailMain.
regards.
Cristian.
From: existx [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] voicemailmain
Date: Thu, 24 Aug 2006 16:08:01 -0400
Howdy,
I have a Debian box using Debian's Asterisk package. People can leave
voicemail for the extensions that are setup in the configuration, and
asterisk e-mail's the user a .wav file (voicemail.conf). This works
perfect.
However, I want to have VoicemailMain sit on an extension so people
can call in, change their greeting, listen too voicemail, etc.
extensions.conf:
exten = 2999,1,Answer
exten = 2999,2,Wait,2
exten = 2999,3,Voicemailmain()
My understand is, that this should allow any user to call up. Enter in
their mailbox number (currently the same as their extension) and
password. However, I cannot dial this extension after reloading
asterisk.
I'm thinking I should add something in another configuration file, or
perhaps my syntax is wrong. Any help would be much apperciated!
Thanks in advance.
Regards,
Jason
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RE: [asterisk-users] voicemailmain
Hi: First it at all check if you have a different extension for voicemailmain.? Then use VoiceMailMain syntax. And send me the CLI log when you try to connect to VoiceMailMain. regards. Cristian. From: existx [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] voicemailmain Date: Thu, 24 Aug 2006 16:08:01 -0400 Howdy, I have a Debian box using Debian's Asterisk package. People can leave voicemail for the extensions that are setup in the configuration, and asterisk e-mail's the user a .wav file (voicemail.conf). This works perfect. However, I want to have VoicemailMain sit on an extension so people can call in, change their greeting, listen too voicemail, etc. extensions.conf: exten = 2999,1,Answer exten = 2999,2,Wait,2 exten = 2999,3,Voicemailmain() My understand is, that this should allow any user to call up. Enter in their mailbox number (currently the same as their extension) and password. However, I cannot dial this extension after reloading asterisk. I'm thinking I should add something in another configuration file, or perhaps my syntax is wrong. Any help would be much apperciated! Thanks in advance. Regards, Jason ___ --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 _ Search from any web page with powerful protection. Get the FREE Windows Live Toolbar Today! http://get.live.com/toolbar/overview ___ --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] Idiot questions
So: The FXO car is for the Pots lines (I.E. bellsouth line) so if you need a analog phone cennected to asterisk you need a FXS card, so if you gonna use a SIP Soft Phone (or a regular SIP Phone) you only need a network connectivity between Asterisk and SIP Phone. Cris. From: joea, j4computers [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] Idiot questions Date: Thu, 24 Aug 2006 21:04:39 -0400 Thanks for all the replies. I take it that with one of these FXO boards, one would need an IP phone as there is no FXS ? BTW, cheapest I've seen is $19.95. Still, not bad. joea Nilesh Londhe[EMAIL PROTECTED] Boldly Declared: 8/24/2006 8:47 PM: I would suggest buying a very low price FXO to begin with which would probably be x100p PCI card at ebay for about $10 +shipping. On 8/24/06, Adam Collard [EMAIL PROTECTED] wrote: You will need a TDM400 with an FXO module for each line you want. A TDM400 supports up to four lines or analog stations. For two lines, you should get a TDM04B. -Original message- From: joea, j4computers [EMAIL PROTECTED] Date: Thu, 24 Aug 2006 14:58:21 -0700 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Idiot questions As a complete newcomer to Asterisk, Digium and PBX, I have several questions. But I'll start with this. To setup a simple system with only a couple of POTS lines, I gather I will need a TDM400 board with FXO and/or FXS modules. So, a TDM400 card will support up to two analog (POTS) lines? joea ___ --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 Adam Collard President Digital Telecom of Michigan, Inc. [EMAIL PROTECTED] (517) 233-1072 Direct Office (800) 420-3803 x4101 Office (517) 766-5902 Fax This email may be confidential. Any distribution, use or copying of this email or the information it contains by other than an intended recipient is unauthorized. If you received this email in error, please advise me (by return email or otherwise) immediately. ___ --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 _ Check the weather nationwide with MSN Search: Try it now! http://search.msn.com/results.aspx?q=weatherFORM=WLMTAG ___ --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] Asterisk Configuration
Hi: First at all: You SIP phones are right register on sip.conf file? Cris From: R.Linga Reddy [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk Configuration Date: Wed, 09 Aug 2006 19:40:50 +0530 Hi All I am new member to asterisk mailing list. I have complied the asterisk and it is running fine. I have configured two extensions in extensions.conf exten = 228,1,Dial exten = 234,1,Dial and configured the xlite soft phone. when I am calling from 234 to 228 it is unable to establish the call. I am able to here all automated playback IVR. ex.500, 600 can any one help to configure the inbound / outbound calls and how to add sip users. -Linga Reddy ___ --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 _ Dont just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ --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] MWI from Asterisk to Meridian
Hi. Tks. What it is TAPI license and how much we have to pay for that? Cris. From: (AstATN) [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] MWI from Asterisk to Meridian Date: Fri, 4 Aug 2006 09:34:55 +0800 If it's control by serial port, you need TAPI license, and need some investigation period to integrate with Nortel system. ( you have to pay for the license, if your system can do so. ) Cheers. Andrew: The key here is try to create a way to integrate Asterisk Voicemail with existing Meridian PBX and send the MWI to M2616. I'm investigating the propietary protocol used by Nortel in the integration between Octel-Dialogic and Meridian for MWI. I believe is good idea to create an appl (and execute that appl in voicemail.conf like external application) for connect to the Meridian console port (Serial interface) and send the command to the end M2616. So, if you have some other idea is very welcome. Cheers. ___ --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 _ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --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] MWI from Asterisk to Meridian
Andrew: The key here is try to create a way to integrate Asterisk Voicemail with existing Meridian PBX and send the MWI to M2616. I'm investigating the propietary protocol used by Nortel in the integration between Octel-Dialogic and Meridian for MWI. I believe is good idea to create an appl (and execute that appl in voicemail.conf like external application) for connect to the Meridian console port (Serial interface) and send the command to the end M2616. So, if you have some other idea is very welcome. Cheers. Chris. From: Leo Ann Boon [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] MWI from Asterisk to Meridian Date: Thu, 03 Aug 2006 18:00:56 +0800 Andrew Kohlsmith wrote: Aside from using a Norstar ATA connected to an FXS port on Asterisk and executing a hookflash *1, no. There isn't a really good way to do it, as that is part of what keeps you hooked into their proprietary crap. It's the same as trying to tie in SIP phones through Asterisk to a Norstar system. I have it done through a PRI but even then Norstar sees the phones as external destinations, so I can't pick up Norstar parked calls and can't transfer calls over. Andrew, Is the Norstar VM integration the same as that the Meridian-1? I remember VM boxes like Octel use a Dialogic card that emulates a bunch of Nortel 2616 phones to send the MWI. Cisco Unity also does it in a similar manner except they need to use external handset emulators like PBXLink. Cheers. ___ --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 _ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --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] RAM memory high Comsumption
Hi everyone: I specting high volume of RAM memory consumption in my Asterisk server, the version of Asterisk is 1.2.4, It could be a bug in my Asterisk version? When i reboot the server, the asterisk day per day increase the use of the RAM memory. Any help is appreciated. Chris. _ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --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] Asterisk suddenly die
Hi everyone: My asterisk application (Version 1.2.4) with no reason sunddenly die. Can i have the way for debug the reason why the asterisk appl die? Regards. _ On the road to retirement? Check out MSN Life Events for advice on how to get there! http://lifeevents.msn.com/category.aspx?cid=Retirement ___ --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] MWI from Asterisk to Meridian
Yeah is true.but we have to sincronize this console command with Asterisk SIP MWI Regards. Cris. From: Johann Steinwendtner [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] MWI from Asterisk to Meridian Date: Tue, 01 Aug 2006 17:06:41 +0200 May be you can build an application which controls the background terminal of the Meridian. (This would be a serial connection to the M1) This application sends background commands like: se mw 3000. This could be a try. Best regards Hans Andrew Kohlsmith schrieb: Please keep responses to the list, so this can help everyone. On Tuesday 01 August 2006 09:26, you wrote: Thak you for you response. My interconection between Asterisk (Voicemail) and my meridian is througth PRI T1, so the only stuff that i can't activate is the light in the meridian digital phones, i understand the asterisk see those phones like a external devices, but i don't know is somebody create o modify the SIP MWI and generate TDM messages to meridian. This isn't about modifying Asterisk to work with the Meridian. This is about the Meridian simply having no way to accept that information from an external trunk. There are VM message centers but they are extraordinarily limited and you can't give a unique one to every user, or even to a group of users. They're line-based. Similarly, you can buy an expensive NAPN or MCDN license which will allow the Norstar to see a PRI as an internal trunk line, but now you are running an undocumented and proprietary PRI signaling protocol called SL-1. It's what Norstar systems use to communicate with each other (imagine two Norstar systems connected together over a leased T1). We have no documentation on it, and Nortel is very likely unwilling to give us the information. So, as I said, you are stuck using a Nortel ATA and an FXS port on Asterisk and using a hookflash *1 sequence to toggle it. Unfortunately the VM callback # will be the ATA's DN, so only one person at a time can access voicemail. I spent some time digging into this last year, but came up without an acceptable solution. I may be forgetting or misremembering some of the details but the end result is the same: you can hack something into it but it's a shitty solution. -A. ___ --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 _ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --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] MWI from Asterisk to Meridian
Hi everyone: Anyone know some idea if the Asterisk voicemail (WMI) can send the messages to meridian for activate the light on meridian digital phones for voicemail notification Thank Cris. _ Dont just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ --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] TDMoE
Hello everyone: Somebody knows what i have to do to configure TDMoE between two asterisk and use PRI signalling in between??? Regards. Cristian. _ Dont just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ --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] Noise on IAX or SIP trunk between 2 Asterisk
Hi everyone: My escenario is: Meridian PBX (Connected to the PSTN) is connected to Asterisk-1 via PRI T1, the Asterisk 1 is connected to Asterisk 2 via IAX trunk or SIP trunk, the Asterisk 2 is used for predictive dialer with Answer Machine Detector, so for some reason, the AMD is starting at the moment the first ring back is detected, wich is not a normal behavior, because the AMD detect words. So finally i found the source of the problem: When the call is start to ring back generate some noise afecting to AMD. Any idea why the trunk or my configuration generate noise when the call is connected.? Regards. Cristian. _ Dont just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ --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] res_config_mysql.so: undefined symbol: __stack_chk_fail
Hi everyone: I installed the lates version of Asterisk with Asterisk Add-Ons. A month ago i upgraded my database form mysql 4.1 to mysql 5.0. So after to start Asterisk i have the following error: [res_config_mysql.so]Apr 11 17:25:51 WARNING[31300]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: __stack_chk_fail Apr 11 17:25:51 WARNING[31300]: loader.c:554 load_modules: Loading module res_config_mysql.so failed! Any idea.?? Regards. Cristian. _ FREE pop-up blocking with the new MSN Toolbar get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/ ___ --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] res_config_mysql.so: undefinedsymbol: __stack_chk_fail
I followed the upgrade procedures from 4.1 to 5.0 using the sources. Regards. Cristian. From: Tim Connolly [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] res_config_mysql.so: undefinedsymbol: __stack_chk_fail Date: Tue, 11 Apr 2006 21:32:37 -0500 Did you upgrade all the mysql packages, or just the server? I would bet you missed the -dev or -lib package. kritikus Araklidas wrote: Hi everyone: I installed the lates version of Asterisk with Asterisk Add-Ons. A month ago i upgraded my database form mysql 4.1 to mysql 5.0. So after to start Asterisk i have the following error: [res_config_mysql.so]Apr 11 17:25:51 WARNING[31300]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: __stack_chk_fail Apr 11 17:25:51 WARNING[31300]: loader.c:554 load_modules: Loading module res_config_mysql.so failed! Any idea.?? Regards. Cristian. ___ --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 _ Dont just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ --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] Agent supervisor configuration
Hi everyone. I have the follow problem: I need to configure an Agent (Supervisor) for monitoring and intercept calls regarding to different Queue, Any help is appreciated. Regards. Cristian. _ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --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] MWI integration between Asterisk and Meridian
I try to integrate my old PBX Meridian and Asterisk througth a PRI T1 (I'm gonna use only the asterisk voicemail system) but i don't know how to integrate the MWI protocol between Asterisk Voicemail and my Nortel meridian. Anyone know what i have to do for that.? Any idea is appreciated. Regards. Cristian. _ Dont just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] MWI integration between Asterisk and Meridian
TKS buddy if i find o develop myself something regarding that i told you. Cristian. From: Gary Reuter [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] MWI integration between Asterisk and Meridian Date: Wed, 12 Oct 2005 17:12:29 -0400 I've been wanting to do exactly the same thing, but I believe it's beyond my coding skills. I think we need a function similar to the SirrixMWI. Some initial code for MWI exists in libpri, but nothing in the rest of asterisk calls those functions yet. On 10/12/05, kritikus Araklidas [EMAIL PROTECTED] wrote: I try to integrate my old PBX Meridian and Asterisk througth a PRI T1 (I'm gonna use only the asterisk voicemail system) but i don't know how to integrate the MWI protocol between Asterisk Voicemail and my Nortel meridian. Anyone know what i have to do for that.? Any idea is appreciated. Regards. Cristian. _ Don't just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com-- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 _ Dont just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] astGUIclient installation problem
Hi everyone: I try to install astGUIclient for my call center. I'm interesting to put in work the monitoring client, i follow step by step the installation from scratch but when i try to run the application from my Windows XP astGUIclient i got the follow error: Client does not support authentication protocol requested by server; consider up grading MySQL client at astGUIclient_1.1.0.pl line 4704 Any idea will be appreciated. Regards. Kritikus. _ Dont just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ 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] Re: CDR for PSTN
Tried ForkCDR. Kritikus. From: Kamran Ahmad [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: CDR for PSTN Date: Fri, 6 May 2005 03:26:30 -0700 (PDT) hello Thanks for replying. i know duration and billsec. but i am getting wrong billsec. for example in one call billsecduration 48 55 and actually in this call phone rings 10 seconds. and accual duration on my cell phone is 35 Hi, Look at http://www.voip-info.org/wiki-Asterisk+billing duration: Total time in system, in seconds (integer), from dial to hangup What are you looking for (from my point of view) is billsec: Total time call is up, in seconds (integer), from answer to hangup -b - Original Message - From: Kamran Ahmad [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, May 05, 2005 7:39 AM Subject: [Asterisk-Users] Re: CDR for PSTN hello Any help. CDR duration starts from 183 Session Progress. cdr duration should start from 200 OK when both parties are inside session. i am using Quintum gw for PSTN Calls. here is the call flow between Asterisk and QuintumGateway. ASTERISK GW 1 |-INTITE--| 2 |183 Session Progress-| cdr starts 3 |180 Ringing--| 4 |200 Ok---| 5 |ACK--| should here 6 |AUDIO Session---| any idea why call duration is starting from step 2. actually session starts from step 5. Kamran Discover Yahoo! Get on-the-go sports scores, stock quotes, news and more. Check it out! http://discover.yahoo.com/mobile.html ___ 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 _ FREE pop-up blocking with the new MSN Toolbar get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/ ___ 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] Account Code in all cases?
Could you use ForkCDR. Regards. From: Matthew Boehm [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Account Code in all cases? Date: Thu, 5 May 2005 13:52:29 -0500 Scenario #1: SIP UA 1 - Asterisk - PSTN CDR shows account code of SIP UA 1; as expected, works great. Scenario #2: PSTN - Asterisk - SIP UA 1 CDR shows no account code. How can I get the account code of SIP UA 1 to appear in the CDR of Scenario #2? Unless someone else has a better method of finding out who called SIP UA 1. SIPUA1 can be reached by internal office 4 digit dialing and by several outside lines (PSTN). Any ideas? -Matthew -- Matthew Boehm, IT DirectorCypress Telecommunications [EMAIL PROTECTED] 3838 N. Sam Houston Parkway E #400 T: 832-200-8640 x3044 Houston, TX 77032 My girlfriend was recently diagnosed with multiple personality disorder; When she called yesterday, my CallerID box exploded. ___ 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 _ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ 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] On live extension monitoring
Hi Team: Somebody knows how to configure some extension for monitoring on live a group of other extensions. Regards. Kritikus. _ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ 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] Music Onhold Problem
Hi every one: I have * RT latest CVS version, the problem is when i make calls from soft SIP phone to other soft phone (Between extensions) if the receiver put on hold i cannot hear the music in the other end, if the caller put the call on hold, happen the same thing. I'm using SJphone version 2.48 and the error messages is the following: Apr 21 21:13:17 WARNING[10144]: res_musiconhold.c:865 local_ast_moh_start: No class: default Regards. Kritikus. _ Dont just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ 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] Extensions unavailable after to sucessfull call (Registration lose)
Hi everyone: I have asterisk latest version (Downloaded and installed today) with two softphone (X-Lite and SJPhone) and one Cisco ATA 186, The problem is when i make call between extensions, after to hang up if i try to do inmediatley a new call the extensions looks die, i have to wait for new registration for make a new cal (Looks afetr make a call the phone lose the registration). Sometimes i see some warning messages like: WARNING[4979]: channel.c:528 ast_channel_walk_locked: Avoided deadlock for 'SIP/268-b551', 10 retries! WARNING[4979]: chan_sip.c:862 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 1 (Non-critical Response) After this the phone recover all privileges for make a calls. Any idea. Thank. Kritikus. _ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ 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] Authentication with DB Support
Hi: Somebody know how to configure the Authentication cmd with DB (Mysql) suport. its work with single password and password file, but i cannot find information for use database in conjunction with DB. Any help will be appreciated. Regards. Kritikus. _ FREE pop-up blocking with the new MSN Toolbar get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/ ___ 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] Macro Extension with Realtime and Mysql DB
Thank Matthew:
I do that, i create the database with tables for support RT Asterisk, then i
create the context deafult in the database, but the macro that i use is
steel in the etension.conf and its works.
Database Extension:
IDCONTEX EXTENPRIORITYAPP APPDATA
1 default _2XX 1 Macro
test1|SIP/${EXTEN:0}
2 default _3XX 1 Macro
test1|SIP/${EXTEN:0}
3 default _4XX 1 Macro
test1|SIP/${EXTEN:0}
Extension.conf:
[default]
switch = Realtime/default@
[macro-test1]
exten = s,1,Dial(${ARG1},20,tTr)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(b${MACRO_EXTEN})
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain([EMAIL PROTECTED])
Its works, but if i change the macro test to database, its doesn't works, my
database and extensions.conf loooks like:
Database Extension:
IDCONTEX EXTEN PRIORITYAPPAPPDATA
1 default _2XX 1 Macro
test1|SIP/${EXTEN:0}
2 default _3XX 1 Macro
test1|SIP/${EXTEN:0}
3 default _4XX 1 Macro
test1|SIP/${EXTEN:0}
4 test1 s 1 Dial
${ARG1}|20|tTr
5 test1 s 2 Goto
s-${DIALSTATUS}|1
6 test1 s-NOANSWER 1Voicemail
u${MACRO_EXTEN}
7 test1 s-NOANSWER 2Goto
default|s|1
8 test1 s-BUSY1Voicemail
b${MACRO_EXTEN}
9 test1 s-BUSY2Goto
default|s|1
10test1 _s-. 1Goto
s-NOANSWER|1
11test1 a1 VoicemailMain
[EMAIL PROTECTED]
And the extensions.conf looks:
[default]
switch = Realtime/default@
[macro-test1]
switch = Realtime/test1@
The error on CLI Asterisk is the context macro-test1 no exist for macro
test1
But, this configuration don't work.
Any idea.
Thank.
Kritikus.
ehm [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Macro Extension with Realtime and Mysql DB
Date: Sat, 02 Apr 2005 23:04:01 -0600
AFAIK, you would configure a macro extension in RealTime just like you
configure a regular extension/context in RealTime.
-Matthew
From: kritikus Araklidas [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Sun, 03 Apr 2005 04:11:38 +
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Macro Extension with Realtime and Mysql DB
Hi Everyone:
I need to know if somebody know how to configure macro extension
(extension.conf) in the database for Asterisk Realtime support if is
suported.
Regards,
Kritikus
_
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[Asterisk-Users] Long Distance Acces Code
Hi everyone: I need some help for configuring access code for long distance call for each extension (Sip friend), i cannot find any clear information for asigned acces code for each user. Any idea will be appreciated. Kritikus. _ Dont just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ 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] Macro Extension with Realtime and Mysql DB
Hi Everyone: I need to know if somebody know how to configure macro extension (extension.conf) in the database for Asterisk Realtime support if is suported. Regards, Kritikus _ Dont just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ 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] Parked Call Issue with realtime Asterisk version
Hi: I need hel for resolve the follow issue: I have the Realtime Asterisk version (v1.1) and the issue is at the momento to receive call from internal extension and that call is put in parking, then when the parking call exceed the expiration time the call is returned back, after that when i try to park the call again, is not possible, i can see the follow error in the asterisk console: Mar 30 17:41:29 WARNING[11367]: chan_sip.c:2141 sip_indicate: Don't know how to indicate condition 16 Mar 30 17:41:32 WARNING[11367]: chan_sip.c:2141 sip_indicate: Don't know how to indicate condition 17 Mar 30 17:41:32 WARNING[11367]: chan_sip.c:2141 sip_indicate: Don't know how to indicate condition 16 Any idea, will be apreciated. Regards. Kritikus. _ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ 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
