[asterisk-users] not reaching at the destination number I dialed
Hi, From this morning, I am having a wired issue. If I dial a number from my asterisk exteion handset, it end up at different number than I dial and its happening everytime I dialed. I havn't change any thing in my asterisk box. And also, I tried my land phone handset which is not forked to a handset (another fork went to asterisk).. it was behaving same.. not sure..what is going on.. any idea? Thanks, -- You don't have any choice, you already made it before you came here. ___ --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] not reaching at the destination number I dialed
NVM, This is happening for a specific destination numbre only. I am considerting that destination number have a wired problem. Thanks, -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Mon, 31 Jul 2006 08:30:35 -0800 To: asterisk-users@lists.digium.com Subject: [asterisk-users] not reaching at the destination number I dialed Hi, From this morning, I am having a wired issue. If I dial a number from my asterisk exteion handset, it end up at different number than I dial and its happening everytime I dialed. I havn't change any thing in my asterisk box. And also, I tried my land phone handset which is not forked to a handset (another fork went to asterisk).. it was behaving same.. not sure..what is going on.. any idea? Thanks, -- You don't have any choice, you already made it before you came here. ___ --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] AGI and incoming call
Why don't you do something like this: exten = 12345678,1,Dial(10) exten = 45874521,1,Dial(11) exten = 32544884,1,Dial(12) replace Dial(10) and so on with apppriate extension. Thanks, -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Wed, 26 Apr 2006 08:47:03 +0200 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] AGI and incoming call Hello, I would like to intercept each incoming call and with an awk script, search the internal phone number ask. For example: I have a text database as this: External phone Internal Phone 12345678 10 45874521 11 32544884 12 When the client 45874521 call, Asterisk must routed the incoming call to the internal phone 11 I have an awk script able to find the good internal phone, but i don't know how to interface it with Asterisk. I thought that AGI is the best way. Is it? Best regards, -- Olivier Saulnier STEGANUX 35 Quai Louis Blanc 03100 Montluçon T: 04.70.02.80.55 F: 04.70.02.80.57 http://www.steganux.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
[Asterisk-Users] Detection of Answering Machine
Hello, To detect an answering machine I have found following two commands, BackgroundDetect (comes with asterisk) MachineDetect (asterisk add-ons) First question, does BackgroundDetect works well with g729? I havn't try MachineDetect yet, what is the benefit of MachineDetect over BackgroundDetect. If anybody used any of this command successfully, please help me. If possible, please let me know a good example of using these commands. What are the common mistakes users do to use these commands? Is there any documentation available arround other than voip-info.org Thanks -- You don't have any choice, you already made it before you came here.___ --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] AGI variables
When I read variables in AGI scripts, I see only the follwing 13 variables agi_request agi_channel agi_language agi_type agi_uniqueid agi_callerid agi_dnid agi_rdnis agi_context agi_extension agi_priority agi_enhanced agi_accountcode beside these, I found following variables documented on several sites. agi_calleridname agi_callingpres agi_callingani2 agi_callington agi_callingtns Where can I find list of agi variables? thanks, -- You don't have any choice, you already made it before you came here. ___ --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] wav to g729
hello, how can I convert my existing wav file to g729. Currently, i have all of them converted to gsm. Isn't it right, If I had all my sound files in g729 format, my server would use less resource and less channels. I have couple of g729 liscences from digium. Thanks, -- You don't have any choice, you already made it before you came here.___ --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] wav to g729
I prefer something 'sox' like program. -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Thu, 22 Dec 2005 19:44:36 +0100 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] wav to g729 Innocent Evil wrote: hello, how can I convert my existing wav file to g729. Currently, i have all of them converted to gsm. Isn't it right, If I had all my sound files in g729 format, my server would use less resource and less channels. I have couple of g729 liscences from digium. http://www.asteriskguru.com/tools/audio_conversion.php -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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] wav to g729
Where can I find it. Thanks, -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Thu, 22 Dec 2005 17:20:39 -0200 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] wav to g729 Try the new conversion module from redice li ..it is greate! Miklos IPFONE TELEFONIA IP Rua Caio Graco 735 São Paulo SP IPBX - +55 11 3488-3800 http://www.ipfone.com.br [EMAIL PROTECTED] Balbus balbum intellegit - Original Message - From: Innocent Evil [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, December 22, 2005 5:00 PM Subject: Re: [Asterisk-Users] wav to g729 I prefer something 'sox' like program. -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Thu, 22 Dec 2005 19:44:36 +0100 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] wav to g729 Innocent Evil wrote: hello, how can I convert my existing wav file to g729. Currently, i have all of them converted to gsm. Isn't it right, If I had all my sound files in g729 format, my server would use less resource and less channels. I have couple of g729 liscences from digium. http://www.asteriskguru.com/tools/audio_conversion.php -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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___ --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] wav to g729
how? where can I get more documentation. Thanks, -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Thu, 22 Dec 2005 13:52:03 -0600 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] wav to g729 Innocent Evil wrote: I prefer something 'sox' like program. -- You don't have any choice, you already made it before you came here. redice.krisk.org Do it from Asterisk! -- Kristian Kielhofner ___ --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] wav to g729
never mind ! I got the starting point. Looks like a good stuff !!! Thanks for helping -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Thu, 22 Dec 2005 13:52:03 -0600 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] wav to g729 Innocent Evil wrote: I prefer something 'sox' like program. -- You don't have any choice, you already made it before you came here. redice.krisk.org Do it from Asterisk! -- Kristian Kielhofner ___ --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] caller_id and law
I would like to send auto invitation to members of one of my community organization. I will use a trird party voip provider to make those call out. Question, what caller_id I can pass to that auto invitation message. What does law says? -- You don't have any choice, you already made it before you came here.___ --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] caller_id and law
Yes, I am talking about 'law' not ulaw. Sounds like, I can pass my home phone number, even I am not going to use my phone provider to make those call. -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Wed, 21 Dec 2005 13:47:18 -0800 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] caller_id and law On Wed, 2005-12-21 at 13:39 -0800, Innocent Evil wrote: I would like to send auto invitation to members of one of my community organization. I will use a trird party voip provider to make those call out. Question, what caller_id I can pass to that auto invitation message. What does law says? which law? The law of the republic of trixter states you can send anything you want, however you may be in a slightly different jurisdiction. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group___ --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] caller_id and law
I am located in United States. I am not going to send a caller_id that is bogus or not owned by me or someone of the commette of my community organization. Here is more about my auto invitation, I will pass caller_id of the General Secratery of my community organization. My concern is, as I am not using my GS's phone directly, can I send his phone number as caller_id even he permit to broadcast it. I hope I could explain my situation. Thanks -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Wed, 21 Dec 2005 17:16:11 -0500 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] caller_id and law trixter aka Bret McDanel wrote: On Wed, 2005-12-21 at 13:39 -0800, Innocent Evil wrote: I would like to send auto invitation to members of one of my community organization. I will use a trird party voip provider to make those call out. Question, what caller_id I can pass to that auto invitation message. What does law says? which law? The law of the republic of trixter states you can send anything you want, however you may be in a slightly different jurisdiction. Just do things honestly so there are no grounds for civil and criminal cases against you. Send a number that you own or one that you ahve been authorized to send by the owner. The number should identify the caller and work for callbacks. ___ --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
I was also following this thread. Would anybody please tell, what would be configuration file if mysql is a different machine than asterisk box? Thanks, --You don't have any choice, you already made it before you came here. -Original Message-From: [EMAIL PROTECTED]Sent: Mon, 12 Dec 2005 21:16:23 -0500To: asterisk-users@lists.digium.comSubject: Re: [Asterisk-Users] CDR MySQL I believe you are missing 2 variables in your conf file:table=cdr(the table your cdrs should be stored)sock=/var/lib/mysql/mysql.sock(the location to your mysql.sock)try something like this:[global]hostname=localhostdbname=dbasterisktable=cdrpassword=dbpassworduser=dbusersock=/var/lib/mysql/mysql.sockuserfield=1 On 12/12/05, Juanjo Portela [EMAIL PROTECTED] wrote: My cdr_mysql.conf is the same I was using for version.1.0.9 and it is as follow[global]hostname=localhostdbname=dbasteriskpassword=dbpassworduser=dbuseruserfield=1Any ideas?Thank you in advance, Juanjo___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Call screening script
I have an AGI script but goal was little different. But my script can do this by tweaking little bit. I have a two tables, caller_numbers and statuses every new number is logged in caller_numbers. If that caller call next time, based on the status call can be route to 1. Voicemail 2. Hangup 3. Busy 4. Telemarker Prompt 5. Reach a certain extension and play the recorded name For recorded name, I requested only at the first time caller called my number. I can prompt for re-record name using the status_id for a given caller. Thanks, -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Thu, 8 Dec 2005 15:20:16 -0500 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Call screening script Has anyone created a call screening script (AGI or other)? What I'm looking for is for Asterisk to: 1. Pickup line 2. Play Record your name to be connected 3. Call an extension 4. Play the recorded name 5. Prompt 1 to accept to 2 to refuse 6. Either connect the call if 1 or send to VM if 2. It's not that hard I just don't want to reinvent the wheel. Michelle Dupuis Technical Support Specialist Oxford Consulting Group Ltd. Making IT work for your business... T: (519) 672-8238 E: [EMAIL PROTECTED] W: www.ocg.ca ___ --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] Per Extension Password for Outgoing Routing
You can accomplish password per extention by using an AGI script. model would be, keep extension and password in a table Execute a simple script to authenticate before dial-out You can also accomplish dial-out time from an AGI script. Feel free to ask if you need further help. Thanks, --You don't have any choice, you already made it before you came here.-Original Message-From: [EMAIL PROTECTED]Sent: Tue, 6 Dec 2005 12:17:19 -0500To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Per Extension Password for Outgoing RoutingHi ! I'm planning to replacea legacyPanasonicPBX with Asterisk, but there are 2 issues to resolve before that. For default, the extensions only can dial to local numbers, but when they want to call to cell phones, long or international phones, there are authorized users, each one with their own password for dialing. I've checked the password for outgoing routing in Asterisk, but the password it's the same for everyone, or i am wrong? And the second issue, the extensions for default only can dial from 7.30am to 4.45pm (office hours); after that, nobody can dial out; but there are users which with a special sequence can dial out. is there a way to implement that functionality in Asterisk? Thanks for the help. ___ --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] h323 vs oh323
Hello, Would you please share your experience regarding h323 and oh323 in asterisk. I am confused to choose one. Thanks, -- You don't have any choice, you already made it before you came here.___ --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] h323 vs oh323
So, we have h323, oh323 and ooh323 I knew about h323 and oh323 but didn't know about ooh323. What is URL of ooh323, I want to know more about them. Thanks, -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Tue, 6 Dec 2005 09:16:05 +1100 To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] h323 vs oh323 I like the chan_ooh323. I like the idea of selfcontained H323 channel that doesn't rely external libraries, often with specific versions that conflict with something else. OOH323 works right out of box and since we started using it to interconnect Asterisk to Samsung OfficeServ 500 we had no problems whatsoever. regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, 6 December 2005 08:11 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] h323 vs oh323 Try chan_oh323 and if it is not ok, try chan_h323 Both work well in different situations/equipments. Isamar On Mon, 5 Dec 2005, Innocent Evil wrote: Hello, Would you please share your experience regarding h323 and oh323 in asterisk. I am confused to choose one. Thanks, -- You don't have any choice, you already made it before you came here.___ --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___ --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] prepaid application
Hi Scott, Yes, its possible pass 'm' option to Dial command for MusicOnHold If destination is unreachable, you need to get the return value of Dial and from that value you will know whether a call was connected or not. Based on that value you can execute Dial again or not. You can put everything in an AGI script. AGI is really fun ! Thanks, -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Thu, 01 Dec 2005 07:51:58 + To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] prepaid application Hi All I am using prepaid auth (callingcards), the idea is for a prepaid support line. It is up and running but I have a couple of questions with regards to modifications I would like to make. When a user calls and they go through the process of entering their card number. They are then asked for a destination. What I would like to be able to do is not have it ask for a destination and automatically dial a number? At present I ask them to enter a default number when it ask for a destination and this then takes them to a queue, if someone is available it rings and goes through, if no one is available rather than sit in the queue and listen to the lovely onhold music prepaid auth comes back and says that destination is unreachable, is there a way to get it to just wait in the queue? Many Thanks In Advance Scott Pinhorne ___ --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] Altering Incoming CallerID
Try,
Set(CALLERIDNAME=Innocent Evil)
Thanks,
--
You don't have any choice, you already made it before you came here.
-Original Message-
From: [EMAIL PROTECTED]
Sent: Thu, 01 Dec 2005 12:48:13 -0500
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Altering Incoming CallerID
What do I need to do to alter incoming CallerID? The below isn't
working...
Running Asterisk 1.2 CVS HEAD
exten = NXXNXX,1,Wait(1)
exten = NXXNXX,2,Set(CALLERID(name) = Fred)
exten = NXXNXX,3,NoOp(${CALLERID(name)})
-- Executing Wait(IAX2/A-9, 1) in new stack
-- Executing Set(IAX2/A-9, CALLERID(name) = Fred) in new stack
-- Executing NoOp(IAX2/A-9, Hugh Johnson) in new stack
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RE: [Asterisk-Users] Call Recording
What you wanna to do if there have more than 2 parties in the conversation ? !! -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Thu, 01 Dec 2005 19:27:45 +0100 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Call Recording Hi all, Perhaps a newby question, perhaps something impossible. While waiting for my HW to arrive, i've been studying the wiki's and TFOT to be preparred when it comes. Info is overwhelming. It seems that anything is possible... Is it possible to record allways from begin to end an entire conversation, without anybody having to press keys? I know it is possible as in an answering machine. But is it also possible for an answered call? And most ideal would be if it was ogg (or mp3), stereo, each call-party its own (L/R)channel. So if both parties speak at the same moment, eachr can still be heard, by turning either the left or right volume off. HtH, Hans -- pgp-id: 926EBB12 pgp-fingerprint: BE97 1CBF FAC4 236C 4A73 F76E EDFC D032 926E BB12 Registered linux user: 75761 (http://counter.li.org) ___ --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] Write to text file in dialplan
Yes, I have something like this in my extension.conf #include verizon.conf and in verizon.conf file have verizon related dialplan Thanks -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Thu, 01 Dec 2005 15:52:45 -0500 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Write to text file in dialplan Hi, Anyone has a way to write (append) to text file from the dial plan? Thanks, Andre ___ --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] Write to text file in dialplan
Sorry, to misinterpret.
I also never tried this.
Let me make a simple AGI script that will do this
#!/usr/bin/env ruby
message = ARGV.shift
$stderr.puts \n#{Time.now} #{message}
just put the above lines in a file in agi-bin direcotry say, 'Echo'
and call it like this
exten = s,7,AGI(Echo|Executing context, extension, priority)
this will put the message in standard error, you will see it in your CLI screen
of asterisk.
for any kind of further help, feel free to post a note.
Thanks,
--
You don't have any choice, you already made it before you came here.
-Original Message-
From: [EMAIL PROTECTED]
Sent: Thu, 01 Dec 2005 14:05:54 -0900
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Write to text file in dialplan
No I think what he means is more like a text file exists at
/tmp/something for example, and you might
exten = s,7,System(echo Call from ${CALLERIDNUM} /tmp/something)
but I haven't tested it yet. Can someone confirm this?
Innocent Evil wrote:
Yes,
I have something like this in my extension.conf
#include verizon.conf
and in verizon.conf file have verizon related dialplan
Thanks
--
You don't have any choice, you already made it before you came here.
-Original Message-
From: [EMAIL PROTECTED]
Sent: Thu, 01 Dec 2005 15:52:45 -0500
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Write to text file in dialplan
Hi,
Anyone has a way to write (append) to text file from the dial plan?
Thanks,
Andre
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[Asterisk-Users] ResetCDR with CDR
Hi,
I am trying to execute the following asterisk command from one of my AGI script.
By providing 'C' flag, I exected CDR would reset.
Problem is, CDR was reset but CDR didn't grab destination number (extension)
from the Dial command.
Well my AGI script was executed after answering a call on a channel.
EXEC DIAL IAX2/{context}/{Extention}|45|CH
What I am missing?
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Re: [Asterisk-Users] cdr_manager.conf
On Mon, 2005-11-28 at 12:24 -0800, Innocent Evil wrote: What is the purpose of cdr_manager.conf? cdr_manager.conf allows you to configure asterisk to send call detail records (cdr) via the Manager API. How I can configure it? to enable CDR via Manager API a cdr_manager.conf looks like this: ; ; Asterisk Call Management CDR ; [general] enabled = yes =Stefan Hi Stefan, I read this example in voip-info.org The main reason of my posting was to learn more about cdr_manager.conf Would anybody please tell me, If I keep enabled=yes, cdr_manager would be enable, I know but an 'enabled' cdr_manager would help me? How I can be benifited from this in terms of cdr management? What exactly it does if I keep enabled=yes? or, what are the next step(s)? Thanks, -- You don't have any choice, you already made it before you came here. ___ --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] AGI + CDR
Hi, I have an AGI script that is called after receving a call on a channel. And my script executel AGI cmd Dial to make another call. Is there any reason not to have CDR record for the call that was initiated in the AGI script? Or I am just missing something basics . Thanks, -- You don't have any choice, you already made it before you came here.___ --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] cdr_manager.conf
Hello, While I was trying to get right CDR record from AGI script, I came across cdr_manager.conf I am trying to learn about cdr_manager.conf What is the purpose of cdr_manager.conf? How I can configure it? I did google, really didn't have very good luck. Would anybody please write couple of sentenses regarding this. Any link on documentaion, tutorial would be great help. Thanks, -- You don't have any choice, you already made it before you came here.___ --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] Truncated CDR records
you can use 'w' option with 'Dial' on 1.2.x
I don't think w do anything like 'wait', If I am wrong, correct me someone
please
According to app_dial.c
w- Allow the called party to enable recording of the call by sending\n
the DTMF sequence defined for one-touch recording in
features.conf.\n
W- Allow the calling party to enable recording of the call by
sending\n
the DTMF sequence defined for one-touch recording in
features.conf.\n;
Thanks,
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-Original Message-
From: [EMAIL PROTECTED]
Sent: Fri, 25 Nov 2005 22:04:04 +0100
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Truncated CDR records
fixed by adding a w at the end of the dialstring:
w would normally indicate a wait, but putting it at the end of your
dialstring will cause asterisk to assume the number before is
complete.(sic!) ~markster
e.g. Dial(Zap/G4/${EXTEN}w)
regards
chris
On Fri, 25 Nov 2005 20:31:41 +0100
Christian B [EMAIL PROTECTED] wrote:
Hello Group!
While parsing my cdrs of the last week, i realized that approx. 1 in
100 _successful_ outbound zap-calls are recorded with a truncated
destination number in the verbose logs and in the cdrs.
Several digits are simply missing. eg 0049 is recorded as
0049. I have received cdrs from the telco, and have verified that the
long version was definitely dialed, and answered.
Most of my traffic (99%) is via DISA, and so far this behaviour has
only surfaced with outbound DISA calls. I can't be certain that its a
DISA only problem, though.
I've added the dialplan that applies to these calls(dialplan_disa.txt),
the agis
that are beeing executed(disa-usergroup1.sh, disacut.sh), samples of
verbose
logs for wrong numbers, cdr's and corresponding cdr's from my
telco(examples_cdr_verboselog.txt). I've upgraded to 1.2 final on
25.11.2000
03:00am in the hope this would be resolved but also today i have faulty
records.
Hope someone can give me information on this issue.
regards
christian
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[Asterisk-Users] cdr enhancement with 'rate' column
Hello, I am trying to enhance my cdr records. What I am trying to are: 1. Add an option in Dial, Say R. to pass rate (price per minute) for the call to do that I will have to modify app_dial.c 2. Dial would use option R to set cdr 3. I will also need to add one more function in cdr.c, say something like ast_cdr_setrate(...) 4. I will also have to edit cdr_addon_mysql.c to have correct record in cdr table, ofcourse I need to have 'rate' column in cdr table too. I am sure, someone of you already did it. Would anybody please mind to share your patch if you already done/have it Or any help to accomplish my above goal would be very much appreciated. Thanks, -- You don't have any choice, you already made it before you came here.___ --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 enhancement with 'rate' column
1. Add an option in Dial, Say R. to pass rate (price per minute) for the call to do that I will have to modify app_dial.c 2. Dial would use option R to set cdr 3. I will also need to add one more function in cdr.c, say something like ast_cdr_setrate(...) None of this is necessary. CDRs can have variables (just like channels), and the CDR posting module can place those variables into storage. Right now only the cdr_csv_custom module knows how to do this, but adding that functionality to cdr_addon_mysql would be relatively easy. Would you please give me little more clue. Well, I want this functionality to cdr_addon_mysql 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
RE: [Asterisk-Users] AGI and AUTOHANGUP
Comeo'n AGI guys.. Please say something. Hi, Using AUTOHANGUP, I can force a call duration within a time limit. I would like to playback a message before 1 minute of autohangup. How can I accomplish it? Would anybody please give me right direction. Thanks, You don't have any choice, you already made it before you came here.___ --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___ --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] International Dialing Code
eh 200lb !! as setup 50lb /month !! You must like like pound starling notation !! I am looking for something free! -Original Message- From: [EMAIL PROTECTED] Sent: Tue, 22 Nov 2005 08:55:05 -0800 To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] International Dialing Code May I recommend www.numberingplans.com as a resource for checking international dial codes and indeed doing a reverse lookup to find out about a number. We have used this as a resource in the past. Regards Neil -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter aka Bret McDanel Sent: Tuesday, November 22, 2005 7:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] International Dialing Code Those came from astbill. I will make the changes and reupload, I have gotten a few more changes as well.. Thanks :) On Mon, 2005-11-21 at 22:03 -0800, Innocent Evil wrote: Lots of country have wrong prefix. Andorra,376 should be 1376 Angola,244should be 1244 Antarctica,6721 should be 1672 http://en.wikipedia.org/wiki/List_of_country_calling_codes have good calling codes, but they are not complete and not downloadable :-( You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Sun, 20 Nov 2005 09:27:09 -0800 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] International Dialing Code On Sun, 2005-11-20 at 15:20 -0200, Hermann Wecke wrote: Innocent Evil wrote: I am trying to download a list of international dialing codes. Would anybody please post a link to get it Google IS your friend. Did you try? http://www.0xdecafbad.com has one in the first few links of the main page/articles page. In asterisk dialplan format as well as csv for use in LCR scripts or whatever. :) There are 2 known issues though, I just havent bothered yet. vienna austria is a bit off, and someone is sending me updates to israel sometime (to make it more complete and verify for errors). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group___ --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 -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group ___ --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___ --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] AGI and AUTOHANGUP
thanks a lot, Darren. I opened up app_dial.c and learned more !! -Original Message- From: [EMAIL PROTECTED] Sent: Tue, 22 Nov 2005 22:18:14 -0700 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] AGI and AUTOHANGUP This is easy. :-) How are you creating the call? From an AGI script? Here's how I do it. $maxtime = $timelimit * 1000; #$maxtime is maxlength in seconds $timelimit = |30|HL($maxtime:6:3); #This will provide a warning @ 60 and 30 seconds. return $timelimit; Append $timelimit onto the end of your dialcommand. You can look at the code for ASTCC in the asterisk cvs or look at astpp-callingcard.agi in the cvs code available @ www.aleph-com.net/astpp Darren Wiebe [EMAIL PROTECTED] Innocent Evil wrote: Comeo'n AGI guys.. Please say something. Hi, Using AUTOHANGUP, I can force a call duration within a time limit. I would like to playback a message before 1 minute of autohangup. How can I accomplish it? Would anybody please give me right direction. Thanks, You don't have any choice, you already made it before you came here.___ --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___ --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 ___ --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 ___ --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] AGI and AUTOHANGUP
Hi, Using AUTOHANGUP, I can force a call duration within a time limit. I would like to playback a message before 1 minute of autohangup. How can I accomplish it? Would anybody please give me right direction. Thanks, You don't have any choice, you already made it before you came here.___ --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] International Dialing Code
Lots of country have wrong prefix. Andorra,376 should be 1376 Angola,244should be 1244 Antarctica,6721 should be 1672 http://en.wikipedia.org/wiki/List_of_country_calling_codes have good calling codes, but they are not complete and not downloadable :-( You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Sun, 20 Nov 2005 09:27:09 -0800 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] International Dialing Code On Sun, 2005-11-20 at 15:20 -0200, Hermann Wecke wrote: Innocent Evil wrote: I am trying to download a list of international dialing codes. Would anybody please post a link to get it Google IS your friend. Did you try? http://www.0xdecafbad.com has one in the first few links of the main page/articles page. In asterisk dialplan format as well as csv for use in LCR scripts or whatever. :) There are 2 known issues though, I just havent bothered yet. vienna austria is a bit off, and someone is sending me updates to israel sometime (to make it more complete and verify for errors). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group___ --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] International Dialing Code
Never mind, I was wrong in my previous posting. Sorry about it. -Original Message- From: [EMAIL PROTECTED] Sent: Mon, 21 Nov 2005 22:03:45 -0800 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] International Dialing Code Lots of country have wrong prefix. Andorra,376 should be 1376 Angola,244should be 1244 Antarctica,6721 should be 1672 http://en.wikipedia.org/wiki/List_of_country_calling_codes have good calling codes, but they are not complete and not downloadable :-( You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Sun, 20 Nov 2005 09:27:09 -0800 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] International Dialing Code On Sun, 2005-11-20 at 15:20 -0200, Hermann Wecke wrote: Innocent Evil wrote: I am trying to download a list of international dialing codes. Would anybody please post a link to get it Google IS your friend. Did you try? http://www.0xdecafbad.com has one in the first few links of the main page/articles page. In asterisk dialplan format as well as csv for use in LCR scripts or whatever. :) There are 2 known issues though, I just havent bothered yet. vienna austria is a bit off, and someone is sending me updates to israel sometime (to make it more complete and verify for errors). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group___ --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___ --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] International Dialing Code
I am trying to download a list of international dialing codes. Would anybody please post a link to get it Thanks in advance. ___ --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] SDT Message Signal
Hello, My phone's VMWI (Visual Message Waiting Indicator) is able to detect SDT message signal. But how I would configure asterisk to send SDT message signal to a certain extension? Thanks, ___ --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] wait before accepting the call
Why don't you write a couple of lines AGI scripts that will call asterisk command WAIT(5) Thankx -Original Message- From: [EMAIL PROTECTED] Sent: Tue, 27 Sep 2005 13:42:31 +0200 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] wait before accepting the call hello! i'm looking for a way to prolonge a pstn-call for 5 seconds before it enters the extensions.conf. this is for testing purposes, all numbers of a ddi should be received by asterisk before the call is walking through the extensions. how can i achive this? i've not seen a feature like this for zapata or zaptel, does anyone have an idea how this could be done? thx christian ___ --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___ --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] Fedora Core 4 not recognizing X100P cards
como'n folks.. ... Well, as I told earlier.. my asterisk was running great with one fxo and one fxs module of a TDM400P All i tried last night to run asterisk with non-root I must did something wrong while I was trying to do that FXO module on channel # 1 FXS module on channel # 4 /etc/zaptel.conf - loadzone = us defaultzone=us fxoks=1 fxsks=4 /etc/asterisk/zapata.conf signalling=fxo_ks channel=1 signalling=fxs_ks channel=4 Did I made any mistake above? /etc/modprobe.conf - install wcfxo /sbin/modprobe --ignore-install wcfxo /sbin/ztcfg install wctdm /sbin/modprobe --ignore-install wctdm /sbin/ztcfg # there have more line in it.. i guess they are not important here alias wcfxs wctdm /etc/rc.d/init.d/asterisk - #important lines are below start) /sbin/modprobe wctdm daemon /usr/sbin/asterisk stop) killproc asterisk /sbin/modprobe -r wctdm Here is the output of asterisk -vvvc Sep 13 22:18:11 WARNING[3982]: chan_zap.c:887 zt_open: Unable to specify channel 1: No such device Sep 13 22:18:11 ERROR[3982]: chan_zap.c:6612 mkintf: Unable to open channel 1: No such device here = 0, tmp-channel = 1, channel = 1 Sep 13 22:18:11 ERROR[3982]: chan_zap.c:9990 setup_zap: Unable to register channel '1' Sep 13 22:18:11 WARNING[3982]: loader.c:403 __load_resource: chan_zap.so: load_module failed, returning -1 Sep 13 22:18:11 WARNING[3982]: loader.c:543 load_modules: Loading module chan_zap.so failed! see I have channel # 1 [EMAIL PROTECTED] ~]# ls -l /dev/zap/ total 0 crw-rw 1 root asterisk 196, 1 Sep 13 22:20 1 crw-rw 1 root asterisk 196, 2 Sep 13 22:20 2 crw-rw 1 root asterisk 196, 3 Sep 13 22:20 3 crw-rw 1 root asterisk 196, 4 Sep 13 22:20 4 crw-rw 1 root asterisk 196, 254 Sep 13 22:20 channel crw-rw 1 root asterisk 196, 0 Sep 13 22:20 ctl crw-rw 1 root asterisk 196, 255 Sep 13 22:20 pseudo crw-rw 1 root asterisk 196, 253 Sep 13 22:20 timer After all of this, if I comment out this from /etc/asterisk/zapata.conf /etc/asterisk/zapata.conf ;signalling=fxo_ks ;channel=1 signalling=fxs_ks channel=4 My asterisk run fine .. I just dont able to use my phone set attaced to my fxs module. And all I had to do is .. forward all incoming call to my voicemail box !! Should I consider my fxs card has burnt out !! Thanks for reading.. hope someone will reply me to help. Thanks again, ___ --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] Fedora Core 4 not recognizing X100P cards
just an update for all curious folks I just replaced my FXS card and everything is working great .. I am running asterisk as non-root user. Moreover, I couldn't figure it out how/why my FXS card got damaged. Thanks, -Original Message- From: [EMAIL PROTECTED] Sent: Wed, 14 Sep 2005 06:44:45 -0800 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Fedora Core 4 not recognizing X100P cards como'n folks.. ... Well, as I told earlier.. my asterisk was running great with one fxo and one fxs module of a TDM400P All i tried last night to run asterisk with non-root I must did something wrong while I was trying to do that FXO module on channel # 1 FXS module on channel # 4 /etc/zaptel.conf - loadzone = us defaultzone=us fxoks=1 fxsks=4 /etc/asterisk/zapata.conf signalling=fxo_ks channel=1 signalling=fxs_ks channel=4 Did I made any mistake above? /etc/modprobe.conf - install wcfxo /sbin/modprobe --ignore-install wcfxo /sbin/ztcfg install wctdm /sbin/modprobe --ignore-install wctdm /sbin/ztcfg # there have more line in it.. i guess they are not important here alias wcfxs wctdm /etc/rc.d/init.d/asterisk - #important lines are below start) /sbin/modprobe wctdm daemon /usr/sbin/asterisk stop) killproc asterisk /sbin/modprobe -r wctdm Here is the output of asterisk -vvvc Sep 13 22:18:11 WARNING[3982]: chan_zap.c:887 zt_open: Unable to specify channel 1: No such device Sep 13 22:18:11 ERROR[3982]: chan_zap.c:6612 mkintf: Unable to open channel 1: No such device here = 0, tmp-channel = 1, channel = 1 Sep 13 22:18:11 ERROR[3982]: chan_zap.c:9990 setup_zap: Unable to register channel '1' Sep 13 22:18:11 WARNING[3982]: loader.c:403 __load_resource: chan_zap.so: load_module failed, returning -1 Sep 13 22:18:11 WARNING[3982]: loader.c:543 load_modules: Loading module chan_zap.so failed! see I have channel # 1 [EMAIL PROTECTED] ~]# ls -l /dev/zap/ total 0 crw-rw 1 root asterisk 196, 1 Sep 13 22:20 1 crw-rw 1 root asterisk 196, 2 Sep 13 22:20 2 crw-rw 1 root asterisk 196, 3 Sep 13 22:20 3 crw-rw 1 root asterisk 196, 4 Sep 13 22:20 4 crw-rw 1 root asterisk 196, 254 Sep 13 22:20 channel crw-rw 1 root asterisk 196, 0 Sep 13 22:20 ctl crw-rw 1 root asterisk 196, 255 Sep 13 22:20 pseudo crw-rw 1 root asterisk 196, 253 Sep 13 22:20 timer After all of this, if I comment out this from /etc/asterisk/zapata.conf /etc/asterisk/zapata.conf ;signalling=fxo_ks ;channel=1 signalling=fxs_ks channel=4 My asterisk run fine .. I just dont able to use my phone set attaced to my fxs module. And all I had to do is .. forward all incoming call to my voicemail box !! Should I consider my fxs card has burnt out !! Thanks for reading.. hope someone will reply me to help. Thanks again, ___ --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___ --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] problem with FXS module
All of sudden my FXS module is not working. I have a TDM card with one FXS and one FXO, FXO module seems working fine. I also noticed the LED is not on for my FXS module while it is on for my FXO module. Sep 13 12:11:44 WARNING[9870]: chan_zap.c:887 zt_open: Unable to specify channel 1: No such device Sep 13 12:11:44 ERROR[9870]: chan_zap.c:6612 mkintf: Unable to open channel 1: No such device here = 0, tmp-channel = 1, channel = 1 Sep 13 12:11:44 ERROR[9870]: chan_zap.c:9990 setup_zap: Unable to register channel '1' Sep 13 12:11:44 WARNING[9870]: loader.c:403 __load_resource: chan_zap.so: load_module failed, returning -1 Sep 13 12:11:44 WARNING[9870]: loader.c:543 load_modules: Loading module chan_zap.so failed! [EMAIL PROTECTED] zaptel]# ls -l /dev/zap/ total 0 crw-rw 1 root root 196, 1 Sep 13 01:51 1 crw-rw 1 root root 196, 2 Sep 13 01:51 2 crw-rw 1 root root 196, 3 Sep 13 01:51 3 crw-rw 1 root root 196, 4 Sep 13 01:51 4 crw-rw 1 root root 196, 254 Sep 13 01:51 channel crw-rw 1 root root 196, 0 Sep 13 01:51 ctl crw-rw 1 root root 196, 255 Sep 13 01:51 pseudo crw-rw 1 root root 196, 253 Sep 13 01:51 timer Please help to solve this issue. Thanks,___ --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] Fedora Core 4 not recognizing X100P cards
My TDM400 on fc4 was working great.. all of sudden ..i am having the same issue ..you guys are having all i tried to run asterisk as non-root user.. and I was able to run it as non-root and was able to receive and send call using asterisk.. I am not sure.. what thing I did wrong and coz all the trouble.. Thanks -Original Message- From: [EMAIL PROTECTED] Sent: Tue, 13 Sep 2005 15:09:52 -0500 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Fedora Core 4 not recognizing X100P cards I have the same problem. I've been having a bit of trouble getting the cards to work with asterisk, and I thought perhaps you might know what I might be doing wrong. I installed them in a linux box, and when I check to see if the OS has recognized them it looks fine: They show up as HSP56 MicroModem (rev 04) [EMAIL PROTECTED] lspci 00:00.0 Host bridge: Intel Corporation 82850 850 (Tehama) Chipset Host Bridge (MCH) (rev 04) 00:01.0 PCI bridge: Intel Corporation 82850 850 (Tehama) Chipset AGP Bridge (rev 04) 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev 04) 00:1f.0 ISA bridge: Intel Corporation 82801BA ISA Bridge (LPC) (rev 04) 00:1f.1 IDE interface: Intel Corporation 82801BA IDE U100 (rev 04) 00:1f.2 USB Controller: Intel Corporation 82801BA/BAM USB (Hub #1) (rev 04) 00:1f.3 SMBus: Intel Corporation 82801BA/BAM SMBus (rev 04) 00:1f.5 Multimedia audio controller: Intel Corporation 82801BA/BAM AC'97 Audio (rev 04) 01:00.0 VGA compatible controller: nVidia Corporation NV18 [GeForce4 MX 4000 AGP 8x] (rev c1) 02:01.0 USB Controller: NEC Corporation USB (rev 41) 02:01.1 USB Controller: NEC Corporation USB (rev 41) 02:01.2 USB Controller: NEC Corporation USB 2.0 (rev 02) 02:09.0 Modem: PCTel Inc HSP56 MicroModem (rev 04) 02:0a.0 Modem: PCTel Inc: Unknown device 2181 (rev 04) 02:0b.0 Modem: PCTel Inc HSP56 MicroModem (rev 04) 02:0c.0 Ethernet controller: 3Com Corporation 3c905C-TX/TX-M [Tornado] (rev 78) 02:0d.0 Modem: PCTel Inc HSP56 MicroModem (rev 04) But when I try to start the wcfxo module it doesn't work: [EMAIL PROTECTED] modprobe wcfxo Notice: Configuration file is /etc/zaptel.conf line 146: Unable to open master device '/dev/zap/ctl' And the linux kernel doesn't quite recognize them: Sep 6 18:58:45 asterisk2 kernel: zaptel: no version for struct_module found: kernel tainted. Sep 6 18:58:45 asterisk2 kernel: Zapata Telephony Interface Registered on major 196 When I try to configure the ztcfg it doesn't find anything on channel 1: [EMAIL PROTECTED] /sbin/ztcfg -vvv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) And Asterisk can't get them working: [chan_zap.so] = (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found Sep 6 19:01:34 WARNING[2549]: chan_zap.c:778 zt_open: Unable to specify channel 1: No such device or address Sep 6 19:01:34 ERROR[2549]: chan_zap.c:6239 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Sep 6 19:01:34 ERROR[2549]: chan_zap.c:9191 setup_zap: Unable to register channel '1-4' Sep 6 19:01:34 WARNING[2549]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' Sep 6 19:01:34 WARNING[2549]: loader.c:440 load_modules: Loading module chan_zap.so failed! I've tried most of the pci cards and they all give the same result. When using a different type of card in that same PC i don't get those errors. I thought perhaps there is some software upgrade necessary for them to work, or something along those lines? Any help you could mention would be very appreciated. Thanks -- Ing CIP Alejandro Celi Mariátegui [EMAIL PROTECTED] El mar, 13-09-2005 a las 15:01, Mojo with Horan Company, LLC escribió: hisax seems to be a loadable module for an ISDN card. if: # lsmod | grep hisax prints any output, try # rmmod hisax; modprobe zaptel ? hth Mojo Shawn Porter wrote: I am getting quite frustrated today, so please bear with me. I just installed Fedora Core 4 (was running RedHat 9 with a working Asterisk) now my Fedora does not appear to be recognizing my X100P (clone) at all. Hardware browser just shows them as unknown device. driver: hisax So, of course, my zaptel drivers do not work and therefore my asterisk does not work. any help would be greatly appreciated….. Shawn ___
Re: [Asterisk-Users] Fedora Core 4 not recognizing X100P cards
Well, as I told earlier.. my asterisk was running great with one fxo and one fxs module of a TDM400P All i tried last night to run asterisk with non-root I must did something wrong while I was trying to do that FXO module on channel # 1 FXS module on channel # 4 /etc/zaptel.conf - loadzone = us defaultzone=us fxoks=1 fxsks=4 /etc/asterisk/zapata.conf signalling=fxo_ks channel=1 signalling=fxs_ks channel=4 Did I made any mistake above? /etc/modprobe.conf - install wcfxo /sbin/modprobe --ignore-install wcfxo /sbin/ztcfg install wctdm /sbin/modprobe --ignore-install wctdm /sbin/ztcfg # there have more line in it.. i guess they are not important here alias wcfxs wctdm /etc/rc.d/init.d/asterisk - #important lines are below start) /sbin/modprobe wctdm daemon /usr/sbin/asterisk stop) killproc asterisk /sbin/modprobe -r wctdm Here is the output of asterisk -vvvc Sep 13 22:18:11 WARNING[3982]: chan_zap.c:887 zt_open: Unable to specify channel 1: No such device Sep 13 22:18:11 ERROR[3982]: chan_zap.c:6612 mkintf: Unable to open channel 1: No such device here = 0, tmp-channel = 1, channel = 1 Sep 13 22:18:11 ERROR[3982]: chan_zap.c:9990 setup_zap: Unable to register channel '1' Sep 13 22:18:11 WARNING[3982]: loader.c:403 __load_resource: chan_zap.so: load_module failed, returning -1 Sep 13 22:18:11 WARNING[3982]: loader.c:543 load_modules: Loading module chan_zap.so failed! see I have channel # 1 [EMAIL PROTECTED] ~]# ls -l /dev/zap/ total 0 crw-rw 1 root asterisk 196, 1 Sep 13 22:20 1 crw-rw 1 root asterisk 196, 2 Sep 13 22:20 2 crw-rw 1 root asterisk 196, 3 Sep 13 22:20 3 crw-rw 1 root asterisk 196, 4 Sep 13 22:20 4 crw-rw 1 root asterisk 196, 254 Sep 13 22:20 channel crw-rw 1 root asterisk 196, 0 Sep 13 22:20 ctl crw-rw 1 root asterisk 196, 255 Sep 13 22:20 pseudo crw-rw 1 root asterisk 196, 253 Sep 13 22:20 timer After all of this, if I comment out this from /etc/asterisk/zapata.conf /etc/asterisk/zapata.conf ;signalling=fxo_ks ;channel=1 signalling=fxs_ks channel=4 My asterisk run fine .. I just dont able to use my phone set attaced to my fxs module. And all I had to do is .. forward all incoming call to my voicemail box !! Should I consider my fxs card has burnt out !! Thanks for reading.. hope someone will reply me to help. Thanks again, -Original Message- From: [EMAIL PROTECTED] Sent: Wed, 14 Sep 2005 09:37:00 +1000 (EST) To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Fedora Core 4 not recognizing X100P cards We had our share of problems with the Fritz Card on FC4. Everything was OK except the speech was one-way in the outgoing direction. We spent days on it and couldn't find a solution to it. We discovered that the device nodes were not created correctly by udev. Enough was enough, we switched to SuSE 9.2 and within a day everything worked beautifully without much effort. I think FC4 has some serious problems with udev (some even claimed that it's not stable release) and I have never seen anyone successfully ran Asterisk on it. If you have a choice, switch to SuSE or other Linux distribution such as Debian. /Y.T. --- Innocent Evil [EMAIL PROTECTED] wrote: My TDM400 on fc4 was working great.. all of sudden ..i am having the same issue ..you guys are having all i tried to run asterisk as non-root user.. and I was able to run it as non-root and was able to receive and send call using asterisk.. I am not sure.. what thing I did wrong and coz all the trouble.. Thanks -Original Message- From: [EMAIL PROTECTED] Sent: Tue, 13 Sep 2005 15:09:52 -0500 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Fedora Core 4 not recognizing X100P cards I have the same problem. I've been having a bit of trouble getting the cards to work with asterisk, and I thought perhaps you might know what I might be doing wrong. I installed them in a linux box, and when I check to see if the OS has recognized them it looks fine: They show up as HSP56 MicroModem (rev 04) [EMAIL PROTECTED] lspci 00:00.0 Host bridge: Intel Corporation 82850 850 (Tehama) Chipset Host Bridge (MCH) (rev 04) 00:01.0 PCI bridge: Intel Corporation 82850 850 (Tehama) Chipset AGP Bridge (rev 04) 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev 04) 00:1f.0 ISA bridge: Intel Corporation 82801BA ISA Bridge (LPC) (rev 04) 00:1f.1 IDE interface: Intel Corporation 82801BA IDE U100 (rev 04) 00:1f.2 USB Controller: Intel Corporation 82801BA/BAM USB (Hub #1) (rev 04) 00:1f.3 SMBus: Intel Corporation 82801BA/BAM SMBus (rev 04) 00:1f.5 Multimedia audio controller: Intel Corporation 82801BA/BAM AC'97 Audio
[Asterisk-Users] AGI problem with library path
Hi List, My AGI seems work well in asterisk -vvvc mode, other than that it doesn't work. Its seems to me, when I run asterisk as daemon (service asterisk start .. on fc4), it doesn't know about my library path. How can pass libray path to my AGI script or asterisk? Thanks___ --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] DTMF not working
Everywhere it is RFC2833 including in SIP phone, Asterisk's sip.conf. DTMF work only from the phone that is hooked with asterisk box. Thanks, -Original Message- From: [EMAIL PROTECTED] Sent: Wed, 24 Aug 2005 12:04:04 -0400 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] DTMF not working Innocent Evil wrote: I am having same problem .. DTMF is not working from a SIP phone while sending to Asterisk cmd VoiceMailMain. Have you set DTMF to out of band RFC2833? In band won't work. At least in my version of HEAD John Novack Would you please explain this line !941+1336/100,!0/100, /* 0 */ what value is what and how it affect on DTMF tone generation. Thanks, I had a similar problem that seems to be caused by the DTMF tone lengths being to short. Try this: Asterisk generates DTMF tones in do_senddigit() in the file channel.c. The tones are defined in a const char array called dtmf_tones[]. Each DTMF tone is a string that looks something like: !941+1336/100,!0/100, /* 0 */ The part that reads !941+1336/100 is the part that you want. Change the 100 to something bigger and recompile. You will have to do that for every tone. I'm using 400 right now, and it seems to be working. I hope that helps. Rob Peter Osborne wrote: Hi all, I just upgraded from Asterisk 1.0RC1 to Asterisk 1.0.7 and our dtmf no longer works with external phone systems. I have a Wildcard TDM400P with 4 FXO's? (it connects to analog lines). No changes were made to the config files. Here's my config: /etc/zaptel.conf fxsks=1-4 loadzone = us defaultzone=us /etc/asterisk/zapata.conf [channels] usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echotraining=yes rxgain=2.0 txgain=2.0 callgroup=1 pickupgroup=1 musiconhold=default context=incoming group=1 signalling=fxs_ks echocancel=64 echocancelwhenbridged=yes relaxdtmf=yes channel = 1-3 [pete_desk] ;Pete's Desk phone (Polycom IP 300) type=friend username=pete_desk secret=pass context=longdistance callerid=Pete 601 host=dynamic mailbox=601 dtmfmode=inband disallow=all allow=ulaw allow=alaw Thanks, Pete ___ 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 -- Robert Tarte Pacific CodeWorks P.O. Box 29050 San Francisco, CA 94129 (p) 831-426-7582 (f) 831-426-7584 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] where can I get low cost g723.1 liscence
Hello, Would you please suggest me, where can I buy g723.1 liscence in cheap. I might need a liscence for 10-50 channels. Thanks,___ --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] DTMF not working
I am having same problem .. DTMF is not working from a SIP phone while sending to Asterisk cmd VoiceMailMain. Would you please explain this line !941+1336/100,!0/100, /* 0 */ what value is what and how it affect on DTMF tone generation. Thanks, I had a similar problem that seems to be caused by the DTMF tone lengths being to short. Try this: Asterisk generates DTMF tones in do_senddigit() in the file channel.c. The tones are defined in a const char array called dtmf_tones[]. Each DTMF tone is a string that looks something like: !941+1336/100,!0/100, /* 0 */ The part that reads !941+1336/100 is the part that you want. Change the 100 to something bigger and recompile. You will have to do that for every tone. I'm using 400 right now, and it seems to be working. I hope that helps. Rob Peter Osborne wrote: Hi all, I just upgraded from Asterisk 1.0RC1 to Asterisk 1.0.7 and our dtmf no longer works with external phone systems. I have a Wildcard TDM400P with 4 FXO's? (it connects to analog lines). No changes were made to the config files. Here's my config: /etc/zaptel.conf fxsks=1-4 loadzone = us defaultzone=us /etc/asterisk/zapata.conf [channels] usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echotraining=yes rxgain=2.0 txgain=2.0 callgroup=1 pickupgroup=1 musiconhold=default context=incoming group=1 signalling=fxs_ks echocancel=64 echocancelwhenbridged=yes relaxdtmf=yes channel = 1-3 [pete_desk] ;Pete's Desk phone (Polycom IP 300) type=friend username=pete_desk secret=pass context=longdistance callerid=Pete 601 host=dynamic mailbox=601 dtmfmode=inband disallow=all allow=ulaw allow=alaw Thanks, Pete ___ 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 -- Robert Tarte Pacific CodeWorks P.O. Box 29050 San Francisco, CA 94129 (p) 831-426-7582 (f) 831-426-7584 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI + Ruby
I would like to write AGI script in Ruby Would anybody please show me right direction.. Thanks___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI + Ruby
What IDE are you talking about? Any URL would be helpful. Thanks, -Original Message- From: [EMAIL PROTECTED] Sent: Wed, 24 Aug 2005 15:16:18 -0400 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] AGI + Ruby I think you might find amethyst much simpler and possibly cheaper too. I believe the current IDE is 12.4K Innocent Evil wrote: I would like to write AGI script in Ruby Would anybody please show me right direction.. Thanks___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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 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] DTMF not working
Hi Rob, I am using RFC2833 everywhere including SIP phone, asterisk's sip.conf Do you think, to raise the value from 100 to 400, would solve my issue? Thanks, -Original Message- From: [EMAIL PROTECTED] Sent: Wed, 24 Aug 2005 08:46:43 -0700 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] DTMF not working Hi Mr. Evil, I'm not sure if the problem that I am describing relates to the problem that you are having. It seems that when you press a key on a SIP phone that is set for inband DTMF, asterisk absorbs the tones until you release the key. This way if you are using DTMF to do things like transfer calls, the user won't get tone blasts in their ear until asterisk has had a chance to interpret the tones. After asterisk has figured out what to do with the tone, it generates and transmits it's own tones in the routine do_senddigit() (assuming that the DTMF tone should be passed on). The duration of the DTMF tones that asterisk generates is fixed and independent of how long you pressed the key on your phone. In the line !941+1336/100,!0/100, the 941 is one tone of the DTMF (dual tone multi-frequency), and 1336 is the other tone. The 100 is the duration of those tones. The tones are in Hz. I'm not sure what units the duration is in, but I bumped mine from 100 to 400 and that seems to do the trick. The part of the string that reads !0/100 just shuts the tone generator off. Rob Innocent Evil wrote: I am having same problem .. DTMF is not working from a SIP phone while sending to Asterisk cmd VoiceMailMain. Would you please explain this line !941+1336/100,!0/100, /* 0 */ what value is what and how it affect on DTMF tone generation. Thanks, I had a similar problem that seems to be caused by the DTMF tone lengths being to short. Try this: Asterisk generates DTMF tones in do_senddigit() in the file channel.c. The tones are defined in a const char array called dtmf_tones[]. Each DTMF tone is a string that looks something like: !941+1336/100,!0/100, /* 0 */ The part that reads !941+1336/100 is the part that you want. Change the 100 to something bigger and recompile. You will have to do that for every tone. I'm using 400 right now, and it seems to be working. I hope that helps. Rob Peter Osborne wrote: Hi all, I just upgraded from Asterisk 1.0RC1 to Asterisk 1.0.7 and our dtmf no longer works with external phone systems. I have a Wildcard TDM400P with 4 FXO's? (it connects to analog lines). No changes were made to the config files. Here's my config: /etc/zaptel.conf fxsks=1-4 loadzone = us defaultzone=us /etc/asterisk/zapata.conf [channels] usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echotraining=yes rxgain=2.0 txgain=2.0 callgroup=1 pickupgroup=1 musiconhold=default context=incoming group=1 signalling=fxs_ks echocancel=64 echocancelwhenbridged=yes relaxdtmf=yes channel = 1-3 [pete_desk] ;Pete's Desk phone (Polycom IP 300) type=friend username=pete_desk secret=pass context=longdistance callerid=Pete 601 host=dynamic mailbox=601 dtmfmode=inband disallow=all allow=ulaw allow=alaw Thanks, Pete -- Robert Tarte Pacific CodeWorks P.O. Box 29050 San Francisco, CA 94129 (p) 831-426-7582 (f) 831-426-7584 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AGI + Ruby
Well,, I never heard about 'amethyst' -Original Message- From: [EMAIL PROTECTED] Sent: Wed, 24 Aug 2005 15:59:12 -0400 To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] AGI + Ruby U joke - duh! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Innocent Evil Sent: Wednesday, August 24, 2005 3:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] AGI + Ruby What IDE are you talking about? Any URL would be helpful. Thanks, -Original Message- From: [EMAIL PROTECTED] Sent: Wed, 24 Aug 2005 15:16:18 -0400 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] AGI + Ruby I think you might find amethyst much simpler and possibly cheaper too. I believe the current IDE is 12.4K Innocent Evil wrote: I would like to write AGI script in Ruby Would anybody please show me right direction.. Thanks___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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 mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users 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] AGI + Ruby
Thanks guys, for your great humors, joke and fun writes .. Well this link helped to start writing AGI in Ruby (An object oriented scripting language, for the people who dont knows Ruby is only a expensive stone !!) http://home.cogeco.ca/~camstuff/agi.html Thanks again, -Original Message- From: [EMAIL PROTECTED] Sent: Wed, 24 Aug 2005 17:48:05 -0400 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] AGI + Ruby YEah but the problem with pearls are that they come in different colours and are often of varying quality. Black one (which are actually green) are the best. Huddleston, Robert wrote: Actually Perl is even better -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: Wednesday, August 24, 2005 4:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] AGI + Ruby Y'see it? There it goes! Right over his head. Huddleston, Robert wrote: U joke - duh! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Innocent Evil Sent: Wednesday, August 24, 2005 3:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] AGI + Ruby What IDE are you talking about? Any URL would be helpful. Thanks, -Original Message- From: [EMAIL PROTECTED] Sent: Wed, 24 Aug 2005 15:16:18 -0400 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] AGI + Ruby I think you might find amethyst much simpler and possibly cheaper too. I believe the current IDE is 12.4K Innocent Evil wrote: I would like to write AGI script in Ruby Would anybody please show me right direction.. Thanks___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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 mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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 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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --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___ --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] Out of G.729 Decoder Licenses!
Matthew, thanks for answering me. I think, I have found the problem. Yes, the 2 liscenses was intalled. If I make a phone call from SIP phone, asterisk use 1/1 encoder/decoder.. If I make a call and asterisk forward to voice mailbox.. just before it starts recording voice mail, it use 1/1 encode/decode.. but right after recording voice mail, i start getting that liscence violation error. May be I need more channel. And I need to understand 'channel' properly.. Would anybody please explain on channel.. when channel number increase based on uses, link, interportaion. Thanks, -Original Message- From: [EMAIL PROTECTED] Sent: Fri, 19 Aug 2005 09:02:21 -0500 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Out of G.729 Decoder Licenses! Innocent Evil wrote: Hello, I have SIP and Asterisk. On Asterisk I have 2 liscenses of g729 (from digium website) SIP user (100) is calling another SIP user (101). As 101 is not online, my SIP server is redirecting that call to Asterisk. Asterisk forward it to 101's voice mail box. SIP user 100's phone have g729 codec. I havn't buy any codec for SIP server itself. But when 100 reach at 101's voice mail, I get this: Aug 18 18:31:36 WARNING[14511]: codec_g729.c:180 g729tolin_framein: Out of G.729 Decoder Licenses! I didn't get it. Would anybody please explain it. Are the licenses installed? Do show g729 from CLI. You will need a g729 license to access asterisk voicemail from a g729 phone. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sending digits from SIP to Asterisk's VoiceMailMain
Hi, I am using Asterisk cmd VoiceMailMain to manage voice mail. Problem is, voice mail box can't read password sent from SIP phone, but I don't have any problem to read password from the handset attached to my asterisk box. Your help will be greatly appreciated. Thanks,___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sending digits from SIP to Asterisk's VoiceMailMain
my sip phone have dtmf relay: rfc2833 asterisk sip.conf have dtmf relay: rfc2833 in associated context. I tried with Inband.. but g729 doesn't support it. I have g729 liscence from digium I havn't try with INFO yet. I prefer to have rfc2833 as dtmf relay. Is there any other thing that can cause this issue? Thanks, -Original Message- From: [EMAIL PROTECTED] Sent: Fri, 19 Aug 2005 14:21:27 -0400 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Sending digits from SIP to Asterisk's VoiceMailMain It sounds to me like an issue of transmitting DTMF tones from the SIP phones. There are several methods that can be used to accomplish DTMF from SIP phones. Of course, you may ask why it isn't just sent as audio (like a regular POTS phone would.) What happens if you are using a SIP phone, hold down the number 4 button for two seconds (so it sends 2 seconds worth of DTMF on the audio stream) and there is some packet loss during that time? You'll have an audio dropout (thus, tone followed by brief silence and tone again.) The remote end will see this as two tones, not one, which obviously can cause undesired results (and is why it's not a good idea to send DTMF in the audio stream.) That being said, look in your sip.conf for a dtmfmode parameter. You can use inband (in the audio stream, not recommended), RFC2833, or SIP INFO. Your SIP phone should also allow you to set how DTMF is sent (although it may not support all of these formats.) Preferably, use RFC2833 or SIP INFO. Find a setting that is available on your phone and on *, and make sure they're set to match. Once you do that, it should work. Jeremy Innocent Evil wrote: Hi, I am using Asterisk cmd VoiceMailMain to manage voice mail. Problem is, voice mail box can't read password sent from SIP phone, but I don't have any problem to read password from the handset attached to my asterisk box. Your help will be greatly appreciated. Thanks,___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jeremy Gault, KD4NED[EMAIL PROTECTED] Network Administrator, WinWorld Corporation Member: Bradley County ACS/RACES/SkyWarn voice: +1.423.473.8084 fax: +1.423.472.9465 fwd: 461771 msn msgr: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sending digits from SIP to Asterisk's VoiceMailMain
Other than below: Got RTP packet from x.y.z.sip_phone:10006 (type 18, seq 25407, ts 191360, len 40) Sent RTP packet to x.y.z.asterisk:10006 (type 18, seq 63928, ts 193744, len 20) I dont see any message while sending digits. -Original Message- From: [EMAIL PROTECTED] Sent: Fri, 19 Aug 2005 14:33:14 -0500 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Sending digits from SIP to Asterisk's VoiceMailMain If you can get an rtp debug while your pressing digits I can see if maybe your device is sending the digits incorrectly. /b On Aug 19, 2005, at 1:46 PM, Innocent Evil wrote: my sip phone have dtmf relay: rfc2833 asterisk sip.conf have dtmf relay: rfc2833 in associated context. I tried with Inband.. but g729 doesn't support it. I have g729 liscence from digium I havn't try with INFO yet. I prefer to have rfc2833 as dtmf relay. Is there any other thing that can cause this issue? Thanks, -Original Message- From: [EMAIL PROTECTED] Sent: Fri, 19 Aug 2005 14:21:27 -0400 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Sending digits from SIP to Asterisk's VoiceMailMain It sounds to me like an issue of transmitting DTMF tones from the SIP phones. There are several methods that can be used to accomplish DTMF from SIP phones. Of course, you may ask why it isn't just sent as audio (like a regular POTS phone would.) What happens if you are using a SIP phone, hold down the number 4 button for two seconds (so it sends 2 seconds worth of DTMF on the audio stream) and there is some packet loss during that time? You'll have an audio dropout (thus, tone followed by brief silence and tone again.) The remote end will see this as two tones, not one, which obviously can cause undesired results (and is why it's not a good idea to send DTMF in the audio stream.) That being said, look in your sip.conf for a dtmfmode parameter. You can use inband (in the audio stream, not recommended), RFC2833, or SIP INFO. Your SIP phone should also allow you to set how DTMF is sent (although it may not support all of these formats.) Preferably, use RFC2833 or SIP INFO. Find a setting that is available on your phone and on *, and make sure they're set to match. Once you do that, it should work. Jeremy Innocent Evil wrote: Hi, I am using Asterisk cmd VoiceMailMain to manage voice mail. Problem is, voice mail box can't read password sent from SIP phone, but I don't have any problem to read password from the handset attached to my asterisk box. Your help will be greatly appreciated. Thanks,___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jeremy Gault, KD4NED[EMAIL PROTECTED] Network Administrator, WinWorld Corporation Member: Bradley County ACS/RACES/SkyWarn voice: +1.423.473.8084 fax: +1.423.472.9465 fwd: 461771 msn msgr: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk- users___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Festival error
Hello, I have installed festival (the rpm package came with fc4). But getting this: client(1) : rejected from myserver not in access list whenever I try to access it from asterisk . I found in documentation: If you see a message such as: client(1) : rejected from myserver.mydomain.com not in access list then edit the festival/bin/festival_server startup script to include that FQDM in the line with localhost.*. but I dont see any file named 'festival_server' in my fc4 box. How can I get arround this issue? Thanks,___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Festival error
Sorry, the file I am talking about is in right place !! but not sure what to add in festival_server script. Thanks, Hello, I have installed festival (the rpm package came with fc4). But getting this: client(1) : rejected from myserver not in access list whenever I try to access it from asterisk . I found in documentation: If you see a message such as: client(1) : rejected from myserver.mydomain.com not in access list then edit the festival/bin/festival_server startup script to include that FQDM in the line with localhost.*. but I dont see any file named 'festival_server' in my fc4 box. How can I get arround this issue? Thanks___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Festival error
From: [EMAIL PROTECTED] Sent: Thu, 18 Aug 2005 08:23:45 -0600 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Festival error festival_server.scm -- should be in /usr/share/festival grep for '(localhost) and replace with nil (don't use double quotes). Isn't this change going to make my festival server accept connect from anybody? If it is, I dont want to do that. I just want to add my asterisk sever to festival's client list. Thanks, Innocent Evil wrote: Hello, I have installed festival (the rpm package came with fc4). But getting this: client(1) : rejected from myserver not in access list whenever I try to access it from asterisk . I found in documentation: If you see a message such as: client(1) : rejected from myserver.mydomain.com not in access list then edit the festival/bin/festival_server startup script to include that FQDM in the line with localhost.*. but I dont see any file named 'festival_server' in my fc4 box. How can I get arround this issue? Thanks,___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ 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] Festival sounds too wired !!
Hello, I was just able to connect to my festival server.. but the voice generated by festival sounds too wired ..really. I installed only festival, i didn't install speech_tools and couple progams as was documented in voip-info.org How can I tune up festival to have better voice (not as good as like human speech!) Thanks,___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Searching For a Voip Provider
Please change the subject to 'Advertisement of a VoIP Provider' -Original Message- From: [EMAIL PROTECTED] Sent: Thu, 18 Aug 2005 11:55:50 -0700 (PDT) To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Searching For a Voip Provider Hi: Please advice me of a voip provider with reasonable reseller program. I was using voipjet and it has a lot of problems. Did anyone experienced asteriskout.com service? They have good prices. Regards; Chawki Hammoud Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoipJet Problems - anyone?
I noticed their mysql server is down or can't connect to mysql server. I tried to download there cvs format price list. -Original Message- From: [EMAIL PROTECTED] Sent: Thu, 18 Aug 2005 16:04:30 -0400 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] VoipJet Problems - anyone? Hi, Does anyone know what is going on with voipjet? This morning/afternoon they just seem to have gone down no word on their website. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Out of G.729 Decoder Licenses!
Hello, I have SIP and Asterisk. On Asterisk I have 2 liscenses of g729 (from digium website) SIP user (100) is calling another SIP user (101). As 101 is not online, my SIP server is redirecting that call to Asterisk. Asterisk forward it to 101's voice mail box. SIP user 100's phone have g729 codec. I havn't buy any codec for SIP server itself. But when 100 reach at 101's voice mail, I get this: Aug 18 18:31:36 WARNING[14511]: codec_g729.c:180 g729tolin_framein: Out of G.729 Decoder Licenses! I didn't get it. Would anybody please explain it. Thanks,___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Port
[EMAIL PROTECTED] ~]# netstat -naptu | grep asterisk tcp0 0 0.0.0.0:20000.0.0.0:* LISTEN 9231/asterisk udp0 0 0.0.0.0:27270.0.0.0:* 9231/asterisk udp0 0 0.0.0.0:45200.0.0.0:* 9231/asterisk udp0 0 xx.yy.zz.ww:50600.0.0.0:* 9231/asterisk udp0 0 0.0.0.0:45690.0.0.0:* 9231/asterisk Hi, My asterisk server is listening to the above ports. would somebody explain, what ports are for what. Is there any security issue with these ports? what firewall messure you do regarding these open ports? Thanks___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problem with sound device
On Monday 15 August 2005 21:08, Innocent Evil wrote: I am getting this whenever I start asterisk. Aug 15 15:03:58 WARNING[11754] chan_oss.c: Read error on sound device: Resource temporarily unavailable sounds like your soundbard is blocked by another program. Sometimes applications like KDE or XMMS block the sound card, even after these are turned off. It then takes a while for the soundbard to become available again. Christoph No, I don't have KDE installed. Today, I have chaged sound card. Now I have this one: Multimedia audio controller: Ensoniq 5880 AudioPCI (rev 02) Still I get the same WARNING message. BTW, where can I know more information about /etc/asterisk/alsa.conf Thanks___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and LCR
Hello, How do you guys implement LCR in Asterisk? Thanks,___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and LCR
How about this: 1. Put all the routes of all the providers in a MySQL table 2. Write a script with a 'clever' algorithm to find out cheapest route of each prefix. 3. Based on #2.. make a lcr_cheapest_route.conf 4. include lcr_cheapest_route.conf in extension.conf But I don't know, how much resource asterisk will take after loading lcr_cheapest_route.conf Also, I don't have any idea about the performance would be. What do you think? Thanks -Original Message- From: [EMAIL PROTECTED] Sent: Tue, 16 Aug 2005 12:57:14 -0400 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Asterisk and LCR On Tue, Aug 16, 2005 at 10:22:01AM -0600, Damon Estep wrote: Any input from others that have already done what I am doing would be helpful, what works best? For 100k routes+, you will have trouble holding them in a SQL database, particularly if your route selection query is complex. With a modern PC running PostgreSQL, you'll run into trouble at around 250k BHCA even with a much smaller number of routes. (This is quite apart from Asterisk itself, try writing a simple program that runs sample queries in a loop, perhaps with several threads. To a certain extent it depends on how you write the query and how judiciously you place indexes on the tables) When you want NPANXX granularity from several carriers (commonly 75-100k routes each) you'll get hit even worse. In my experience the safe limits of this approach are about a 2x DS3 worth of traffic with 10,000 routes in the table... After that you've got to pull everything into RAM and write a clever route selection algorithm... -w -- William Waites ww [EMAIL PROTECTED] magicphone.ca ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and LCR
Are you saying realtime mysql is not clever? That is exactly what it is supposed to do. BTW, how do you integrate mysql with asterisk? any link, documention, tutorials would be greatly helpful. Thanks,___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Switch between FXS ports
Hello, I have two FXS port on my TDM card. channel 4 is attached with a telco line that I use frequently. And channel 3 have another telco line. but I dont publish that number to my friends. If I receive a call through channel 4, how can I handover that call to channel 3 ..so that I can keep channel 4 open for incoming call. Thanks,___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Switch between FXS ports
Sorry for the typo. Do I need to ask my telco, if I want to use Asterisk as a PBX in a home/small biz/large biz and I want one hunting number. Thanks, -Original Message- From: [EMAIL PROTECTED] Sent: Mon, 15 Aug 2005 13:20:17 -0500 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Switch between FXS ports Innocent Evil wrote: Hello, I have two FXS port on my TDM card. channel 4 is attached with a telco line that I use frequently. And channel 3 have another telco line. but I dont publish that number to my friends. If I receive a call through channel 4, how can I handover that call to channel 3 ..so that I can keep channel 4 open for incoming call. First, FXS = handset / FXO = telco line. Second, you don't. Does the telephone company let you do this now, if so, how - otherwise, no you can't. Chris -- Christopher L. Wade ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Switch between FXS ports
As far as I remember, you can't really do that (because the telco isn't switching the call), what you'll want to do is have a hunt group set up Yesss... this is exactly I am looking for. How can I do that? Thanks, --Original Message- -From: [EMAIL PROTECTED] -[mailto:[EMAIL PROTECTED] On Behalf Of -Innocent Evil -Sent: Monday, August 15, 2005 2:17 PM -To: asterisk-users@lists.digium.com -Subject: [Asterisk-Users] Switch between FXS ports - -Hello, - -I have two FXS port on my TDM card. -channel 4 is attached with a telco line that I use -frequently. And channel 3 have another telco line. but I dont -publish that number to my friends. -If I receive a call through channel 4, how can I handover -that call to channel 3 ..so that I can keep channel 4 open -for incoming call. - http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problem with sound device
I am getting this whenever I start asterisk. Aug 15 15:03:58 WARNING[11754] chan_oss.c: Read error on sound device: Resource temporarily unavailable This is my sound card: Multimedia audio controller: Fortemedia, Inc Xwave QS3000A I am not sure... what I am doing wrong. Please help. Thanks,___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Switch between FXS ports
I am clear with this issue. Thanks everybody for answering me. -Original Message- From: [EMAIL PROTECTED] Sent: Mon, 15 Aug 2005 10:16:34 -0800 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Switch between FXS ports Hello, I have two FXS port on my TDM card. channel 4 is attached with a telco line that I use frequently. And channel 3 have another telco line. but I dont publish that number to my friends. If I receive a call through channel 4, how can I handover that call to channel 3 ..so that I can keep channel 4 open for incoming call. Thanks,___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] how may channels
keep approx. 32kb per channel.. -Original Message- From: [EMAIL PROTECTED] Sent: Fri, 5 Aug 2005 11:16:32 -0700 (PDT) To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] how may channels how many channels using codec g729 can be done by an internet bandwidth to 512kb dedicated. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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 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] number 'register = ' in sip.conf
how many 'register =' I can have in sip.conf___ 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] call outside from FXS through FXO
Hi, I am trying to make an outbound call from phone attached to FXS port. My telephone (VoIP) line is connected to FXO port (Zap/4) Default context for channel # 4 is 'directdial' here is part of my extension.conf [directdial] ignorepat = 9 exten = 9,1,Dial,Zap/4/ exten = 9,2,Congestion include = international [international] ignorepat = 9 exten = _9011.,1,Dial,Zap/4/BYEXTENSION exten = _9011.,2,Congestion include = longdistance [longdistance] ignorepat = 9 exten = _91NXXNXX,1,Dial,Zap/4/BYEXTENSION exten = _91NXXNXX,2,Congestion I am new in asterisk, having very hardtime to find my mistake.. :p Please help .. I could successfully redirect any incoming from outside to phone attached on FXS port (Zap/1).___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk as PSTN gateway, voice mail server with SIP
Hello, I am sure this has been answered so many times as it is one of the most fundamental features of Asterisk. Here is my scenario, I have setup my asterisk server with a TDM400p which have one FXO and FXS card. My SIP server is up and its working fine only in SIP network ( I used ser) For my daily use telephone, I have a VoIP telephone from a major service provider. What I want is to hook my telephone line to Asterisk server FXO port. So Asterisk is going to work as PSTN gateway for my SIP server. How do I would do these: 1. If I receive a call to my telephone line, I would like to forward it from my asterisk server to my SIP phone. 2. Using my SIP phone, I would like to make PSTN call using my asterisk server. How I know how to pass call to asterisk. But I dont know how to receive that call from SIP server and initiate call using my telephone line on asterisk server. 3. Just in case, I am not able to answer from my sip phone, I would like forward that call to asterisk so that caller can leave message. 4. I would like to retrive stored message from asterisk server. Thanks,___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] g729 liscence question
I have a TDM400P with one FXS and one FXO.. how many liscence(2) I will have to buy? Thanks___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g729 liscence question
Thanks everybody for answering me. Yes, I plan to connect SIP phone outside of my network. Infact, I am going to use Asterisk as my PSTN gateway and voice mailbox. Also, I have plan to add two more FXO card when I will have bigger network. Sounds like, I should get two liscences at this moment. Thanks again. -Original Message- From: [EMAIL PROTECTED] Sent: Mon, 01 Aug 2005 18:19:03 -0500 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] g729 liscence question On Mon, 2005-08-01 at 10:53 -0800, Innocent Evil wrote: I have a TDM400P with one FXS and one FXO.. how many liscence(s) I will have to buy? Short answer: None. Long answer: Zap interfaces use G711 and do not need G729 to work. Only if you plan to connect SIP or IAX phones from outside your local network do you really need voice compression. -- Telecomunicaciones Abiertas de Mexico Carlos Chavez Director de Tecnologia +52-55-91169161 Ext. 2001___ 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
