Re: [Asterisk-Users] Running asterisk within screen
On Thu, 2005-12-01 at 22:29 -0800, Luki wrote: Does anybody know, why it is not possible, to run asterisk within screen? Yes, it is possible but you can't scroll up so you only see the last ~40 lines. At least I didn't work for me but I didn't research this further. in screen you can enter copy/view scrollback mode by pressing Ctrl-a Esc then you can use page up/down. To leave scrollback mode press Esc again. =Stefan signature.asc Description: This is a digitally signed message part ___ --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] DTMF on Planet VIP153
Hi all. Does anybody use VIP 153 phone with asterisk and has DTMF works. Thank, Bob. ___ --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] IAX Configuration
See http://www.iaxtel.com/setup.html 2005/12/2, Ishanka Anuradha Ranasooriya [EMAIL PROTECTED]: Hi all, I have configured two asterisk Boxes.Then I need to communicate these asterisk boxes via the IAX.It is better if you can help me to configure two boxes to communicate via asterisk Thanks, Ishanka. - Original Message - From: Branko Samardzic [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, December 02, 2005 10:43 AM Subject: [Asterisk-Users] IAX trunking frequency parameter works only oninitiator side Hi, I am experimenting with trunkfreq parameter. When it is 20ms I can see both parties in IAX session sending IAX frames every 20ms. When I modify this parameter to 40ms then I can see that only server that initiated IAX connection works properly (i.e. sends IAX frames every 40ms while other side still sends IAX frames at 20ms per frame rate). I disabled jitter buffers on both sides and I use speex codec. Here is tcp dump of IAX traffic: 23:26:45.972072 IP SERVER_A.62142 SERVER_B.4569: UDP, length 58 23:26:45.976295 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:45.996264 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.006742 IP SERVER_A.62142 SERVER_B.4569: UDP, length 58 23:26:46.016270 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.036254 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.047891 IP SERVER_A.62142 SERVER_B.4569: UDP, length 58 23:26:46.056248 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.076286 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.091255 IP SERVER_A.62142 SERVER_B.4569: UDP, length 58 23:26:46.096262 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.116243 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.127494 IP SERVER_A.62142 SERVER_B.4569: UDP, length 58 23:26:46.136242 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 SERVER_A initiates connection while SERVER_B answers. SERVER_A iax.conf file === [SERVER_B] disallow=all allow=speex jitterbuffer=no dropcount=2 maxjitterbuffer=200 maxexcessbuffer=100 minexcessbuffer=60 jittershrinkrate=1 trunkfreq=40; How frequently to send trunk msgs (in ms) context = foo secret=zYX9VUt auth=md5 type=friend host=SERVER_B_IP_ADDRESS trunk=yes SERVER_B iax.conf file === [SERVER_B] disallow=all allow=speex jitterbuffer=no dropcount=2 maxjitterbuffer=200 maxexcessbuffer=100 minexcessbuffer=60 jittershrinkrate=1 trunkfreq=40; How frequently to send trunk msgs (in ms) context = default secret=zYX9V auth=md5 type=friend host=SERVER_B_IP_ADDRESS trunk=yes Any idea as to why trunking frequency is not symmetrical? Any help is appreciated ___ --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 -- This e-mail and any attachments are intended for the above named recipient(s) only and may be privileged. This message and any attachments has been scanned for viruses and dangerous content by ITABS Lanka Mail Scanner, and is believed to be clean. Although measures have been taken to ensure that this e-mail and attachments are free from any virus we advise that, in keeping with good computing practice, the recipient should ensure they are actually virus free. Please note that this message has been sent over public networks which may not be a 100% secure communications medium and ITABS Lanka cannot be held responsible for its integrity. -- This e-mail and any attachments are intended for the above named recipient(s) only and may be privileged. This message and any attachments has been scanned for viruses and dangerous content by ITABS Lanka Mail Scanner, and is believed to be clean. Although measures have been taken to ensure that this e-mail and attachments are free from any virus we advise that, in keeping with good computing practice, the recipient should ensure they are actually virus free. Please note that this message has been sent over public networks which may not be a 100% secure communications medium and ITABS Lanka cannot be held responsible for its integrity. ___ --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 -- This e-mail and any attachments are intended for the above named recipient(s) only and may be privileged. This message and any attachments has been scanned
Re: [Asterisk-Users] (no subject)
See http://www.iaxtel.com/setup.html 2005/12/2, P.G.C.K. Nirukshitha [EMAIL PROTECTED]: Dear Sir I have configured two asterisk Boxes.Then I need to communicate these asterisk boxes via the IAX.It is better if you can help me to configure two boxes to communicate via asterisk. Thanks Nirukshitha Gamage -- This e-mail and any attachments are intended for the above named recipient(s) only and may be privileged. This message and any attachments has been scanned for viruses and dangerous content by ITABS Lanka Mail Scanner, and is believed to be clean. Although measures have been taken to ensure that this e-mail and attachments are free from any virus we advise that, in keeping with good computing practice, the recipient should ensure they are actually virus free. Please note that this message has been sent over public networks which may not be a 100% secure communications medium and ITABS Lanka cannot be held responsible for its integrity. ___ --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 -- Giovanni Miano ___ --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] (no subject)
See http://www.iaxtel.com/setup.html 2005/12/2, Lakmal [EMAIL PROTECTED]: Hi all, I have configured two asterisk Boxes.Then I need to communicate these asterisk boxes via the IAX.It is better if you can help me to configure two boxes to communicate via asterisk Thanks, Ishanka. - Original Message - From: Branko Samardzic [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, December 02, 2005 10:43 AM Subject: [Asterisk-Users] IAX trunking frequency parameter works only oninitiator side Hi, I am experimenting with trunkfreq parameter. When it is 20ms I can see both parties in IAX session sending IAX frames every 20ms. When I modify this parameter to 40ms then I can see that only server that initiated IAX connection works properly (i.e. sends IAX frames every 40ms while other side still sends IAX frames at 20ms per frame rate). I disabled jitter buffers on both sides and I use speex codec. Here is tcp dump of IAX traffic: 23:26:45.972072 IP SERVER_A.62142 SERVER_B.4569: UDP, length 58 23:26:45.976295 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:45.996264 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.006742 IP SERVER_A.62142 SERVER_B.4569: UDP, length 58 23:26:46.016270 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.036254 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.047891 IP SERVER_A.62142 SERVER_B.4569: UDP, length 58 23:26:46.056248 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.076286 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.091255 IP SERVER_A.62142 SERVER_B.4569: UDP, length 58 23:26:46.096262 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.116243 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.127494 IP SERVER_A.62142 SERVER_B.4569: UDP, length 58 23:26:46.136242 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 SERVER_A initiates connection while SERVER_B answers. SERVER_A iax.conf file === [SERVER_B] disallow=all allow=speex jitterbuffer=no dropcount=2 maxjitterbuffer=200 maxexcessbuffer=100 minexcessbuffer=60 jittershrinkrate=1 trunkfreq=40; How frequently to send trunk msgs (in ms) context = foo secret=zYX9VUt auth=md5 type=friend host=SERVER_B_IP_ADDRESS trunk=yes SERVER_B iax.conf file === [SERVER_B] disallow=all allow=speex jitterbuffer=no dropcount=2 maxjitterbuffer=200 maxexcessbuffer=100 minexcessbuffer=60 jittershrinkrate=1 trunkfreq=40; How frequently to send trunk msgs (in ms) context = default secret=zYX9V auth=md5 type=friend host=SERVER_B_IP_ADDRESS trunk=yes Any idea as to why trunking frequency is not symmetrical? Any help is appreciated ___ --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 -- This e-mail and any attachments are intended for the above named recipient(s) only and may be privileged. This message and any attachments has been scanned for viruses and dangerous content by ITABS Lanka Mail Scanner, and is believed to be clean. Although measures have been taken to ensure that this e-mail and attachments are free from any virus we advise that, in keeping with good computing practice, the recipient should ensure they are actually virus free. Please note that this message has been sent over public networks which may not be a 100% secure communications medium and ITABS Lanka cannot be held responsible for its integrity. -- This e-mail and any attachments are intended for the above named recipient(s) only and may be privileged. This message and any attachments has been scanned for viruses and dangerous content by ITABS Lanka Mail Scanner, and is believed to be clean. Although measures have been taken to ensure that this e-mail and attachments are free from any virus we advise that, in keeping with good computing practice, the recipient should ensure they are actually virus free. Please note that this message has been sent over public networks which may not be a 100% secure communications medium and ITABS Lanka cannot be held responsible for its integrity. ___ --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 -- Giovanni Miano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:
[Asterisk-Users] WG: App_rxfax problem
so is there a solution in the next cvs udpate? Von: René Enskat [Teamware GmbH] [mailto:[EMAIL PROTECTED] Gesendet: Donnerstag, 1. Dezember 2005 14:47An: 'asterisk-users@lists.digium.com'Betreff: WG: App_rxfax problem I just sent the error in full log: Dec 1 15:01:08 VERBOSE[27950] logger.c: [app_rxfax.so]Dec 1 15:01:08 WARNING[27950] loader.c: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handler Dec 1 15:01:08 WARNING[27950] loader.c: Loading module app_rxfax.so failed! Von: René Enskat [Teamware GmbH] [mailto:[EMAIL PROTECTED] Gesendet: Donnerstag, 1. Dezember 2005 08:35An: 'asterisk-users@lists.digium.com'Betreff: App_rxfax problem When i load the fax modules into the asterisk i got this errors but compile was ok! I have the latest cvs head [res_musiconhold.so] = (Music On Hold Resource) == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' == Registered application 'StartMusicOnHold' == Registered application 'StopMusicOnHold'[app_rxfax.so]Warning, flexibel rate not heavily tested!Warning, flexibel rate not heavily tested!Warning, flexibel rate not heavily tested!Ouch ... error while writing audio data: : Broken pipeOuch ... error while writing audio data: : Broken pipeOuch ... error while writing audio data: : Broken pipe ___ --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] Hint: how to include dialplan files from remotesystems
You can use EAGI 2005/12/2, Alexander Lopez [EMAIL PROTECTED]: Good Idea. I am doing a similar thing but for a different reason: I use the system call to bring in mp3 files for music on hold. We make custom Music on Hold messages and we store them at our DC. I am also using this to pull mp3 updates for holiday music. Try doing any of this with any OTHER PBX!!! Betcha can't -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: Thursday, December 01, 2005 9:52 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Hint: how to include dialplan files from remotesystems Every once in a while I find a nice, compact little project is good enough to share to the rest of the user community as a single post. Here's something that I was happy worked as planned. This is not particularly clever, but uses some infrequently-used tricks of running system commands from within Asterisk in two different ways: System and #exec. File this away under the heading of Not really clever, but impresses management. I've got a user community who doesn't want to log into their Asterisk servers much, but they have general housekeeping tasks they want to perform on their dialplan which is really just mapping usernames to extension (for SIP tasks, so that jwhorfin ends up calling Zap/g1/2939) This shouldn't involve them doing anything other than updating a file somewhere. They know how to put files on a webserver, so the trick was how to make them able to edit this file on their easy-to-use webserver and make it magically appear on the somewhat opaque Asterisk system. I gave them a template file for them to store on their well-understood and internally accessible webserver, that they can edit with WordPad or other text editors. This means that they don't have to learn any significant processes to update the list of user-to-number mappings if they know how to publish something on their webserver. Here's the template example: ; -- Start File -- ; Template for usernames-to-numbers ; ; Save in the privatefiles directory of the public ; webserver, accessible by anyone. ; ; Comments start with the semi-colon character ; ; After making changes on this list and saving it ; to the webserver, you must call ext. 2900 to ; cause the Asterisk system to update itself ; with the contents of this file. ; [username-to-numbers] exten = jwhorfin,1,Dial(Zap/g1/2939) exten = rnevada,1,Dial(Zap/g1/2988) ; ; -- end file -- ; So, in the dialplan, here's what I do to include this file (note that the echo user is just to illustrate that other names can be included in the chain of contexts manually): ; ...more extensions.conf above here. ; [from-internet] ; ; If any calls come in to user echo, play back an echo test ; exten = echo,1,Set(TIMEOUT(absolute)=500 exten = echo,n,Echo ; ; Now, include any users that have been configured by the client.. ; ; (watch out for accidental line wraps here! Next two lines start with #) #exec /usr/bin/curl -s http://webserver.domain.com/privatefiles/username-to-numbers /etc/asterisk/username-to-numbers #include username-to-numbers include = username-to-numbers ; ; ; more extensions.conf below here... ; ; Now, just as trivially clever is that in a different context (from their PBX) I can allow them to dial a special number that allows the administrator to re-load/re-parse this file at will: ; [from-pbx] ; exten = 2900,1,System(/usr/sbin/asterisk -rx extensions reload) ; Don't forget to add this set of 2 lines to asterisk.conf to allow the config files to execute commands with #exec: [options] execincludes=yes Good luck! JT ___ --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 -- Giovanni Miano ___ --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] IAX Configuration
Hi is there any way i can reduce Bandwidth ussage when iam making outbound?? right now each call taking 80k ram On 12/2/05, Giovanni Miano [EMAIL PROTECTED] wrote: Seehttp://www.iaxtel.com/setup.html2005/12/2, Ishanka Anuradha Ranasooriya [EMAIL PROTECTED]:Hi all,I have configured two asterisk Boxes.Then I need to communicate theseasterisk boxes via the IAX.It is better if you can help me to configure twoboxes to communicate via asteriskThanks,Ishanka. - Original Message - From: Branko Samardzic [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, December 02, 2005 10:43 AM Subject: [Asterisk-Users] IAX trunking frequency parameter works only oninitiator sideHi, I am experimenting with trunkfreq parameter. When it is 20ms I can see both parties in IAX session sending IAX frames every 20ms. When I modify this parameter to 40ms then I can see that only server that initiated IAX connection works properly (i.e. sends IAX frames every 40ms while other side still sends IAX frames at 20ms per frame rate). I disabled jitter buffers on both sides and I use speex codec. Here is tcp dump of IAX traffic: 23:26:45.972072 IP SERVER_A.62142 SERVER_B.4569: UDP, length 58 23:26:45.976295 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:45.996264 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.006742 IP SERVER_A.62142 SERVER_B.4569: UDP, length 58 23:26:46.016270 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.036254 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.047891 IP SERVER_A.62142 SERVER_B.4569: UDP, length 58 23:26:46.056248 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.076286 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.091255 IP SERVER_A.62142 SERVER_B.4569: UDP, length 58 23:26:46.096262 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.116243 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 23:26:46.127494 IP SERVER_A.62142 SERVER_B.4569: UDP, length 58 23:26:46.136242 IP SERVER_B.4569 SERVER_A.62142: UDP, length 25 SERVER_A initiates connection while SERVER_B answers. SERVER_A iax.conf file === [SERVER_B] disallow=all allow=speex jitterbuffer=no dropcount=2 maxjitterbuffer=200 maxexcessbuffer=100 minexcessbuffer=60 jittershrinkrate=1 trunkfreq=40; How frequently to send trunk msgs (in ms) context = foo secret=zYX9VUt auth=md5 type=friend host=SERVER_B_IP_ADDRESS trunk=yesSERVER_B iax.conf file === [SERVER_B] disallow=all allow=speex jitterbuffer=no dropcount=2 maxjitterbuffer=200 maxexcessbuffer=100 minexcessbuffer=60 jittershrinkrate=1 trunkfreq=40; How frequently to send trunk msgs (in ms) context = default secret=zYX9V auth=md5 type=friend host=SERVER_B_IP_ADDRESS trunk=yes Any idea as to why trunking frequency is not symmetrical? Any help is appreciated ___ --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 -- This e-mail and any attachments are intended for the above named recipient(s) only and may be privileged. This message and any attachments has been scanned for viruses and dangerous content by ITABS Lanka Mail Scanner, and is believed to be clean. Although measures have been taken to ensure that this e-mail and attachments are free from any virus we advise that, in keeping with good computing practice, the recipient should ensure they are actually virus free. Please note that this message has been sent over public networks which may not be a 100% secure communications medium and ITABS Lanka cannot be held responsible for its integrity. -- This e-mail and any attachments are intended for the above named recipient(s) only and may be privileged. This message and any attachments has been scanned for viruses and dangerous content by ITABS Lanka Mail Scanner, and is believed to be clean. Although measures have been taken to ensure that this e-mail and attachments are free from any virus we advise that, in keeping with good computing practice, the recipient should ensure they are actually virus free. Please note that this message has been sent over public networks which may not be a 100% secure communications medium and ITABS Lanka cannot be held responsible for its integrity. ___ --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 -- This e-mail and any attachments are intended for the above named recipient(s) only and may be privileged. This message and any attachments has been scanned for viruses and
Re: [Asterisk-Users] voipbuster
2005/12/1, Tony Hoyle [EMAIL PROTECTED]: The following works in iax.conf for me: [voipbuster] host=iax.voipbuster.com type=peer username=username secret=password qualify=yes context=inbound Context=inbound? I'm using from-pstn. The problem is this: I configure asterisk (through amp) with the username and password of one just created account (without credit) and I'm able to make calls of one minute. All OK. Then I change the username and password for the one of one account that has credit and is working ok with the propietary software. The only change I make is username and password. Then, the registration is refused. I double checked and the username and password works fine with the propietary software. Also the asterisk configuration works with another username/password without credit. I sent a mail to voipbuster support with the user that has credit but they never answered. -- Alejandro Vargas ___ --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] voipbuster
2005/12/1, Francesco Peeters [EMAIL PROTECTED]: Mine works just fine. It's a pain though if you have to get a new account, as minimum amount is now EUR 5... OTOH, for free calls, it might be worth it... The account that I'm testing is one of 5 EUR. I'm trying to test it with asterisk because if it works, I will place credit on my account. But if I can't make it work with an account with credit, I must suspect they are blocking the accounts with credit for avoiding people to use it with other software than their propietary client. -- Alejandro Vargas ___ --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] IAX Configuration
2005/12/2, ram [EMAIL PROTECTED]: is there any way i can reduce Bandwidth ussage when iam making outbound?? right now each call taking 80k Choose a different codec. The problem is that the codec must be supported by the other end. I think the better sould be aspeex, or iblc. But the more commonly supported is the propietary g729. To use it you should pay for each communication. If your phone does not support it, asterisk must do transcoding in both directions and you must pay twice. There is a royalty-free version for private use provided by intel. -- Alejandro Vargas ___ --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] IAX Configuration
2005/12/2, Giovanni Miano [EMAIL PROTECTED]: See http://www.iaxtel.com/setup.html I'm also interested on doing this. I already set up iax connections to providers like free world dialup, voipbuster and voipjet, but I don't know how to configure asterisk to receive the registration from the other side. I'm using amp (from [EMAIL PROTECTED]). -- Alejandro Vargas ___ --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
On Thu, Dec 01, 2005 at 04:43:25PM -0800, Innocent Evil wrote:
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
$stderr.puts \n#{Time.now} #{message}
just put the above lines in a file in agi-bin direcotry say, 'Echo'
and call it like this
exten = s,7,AGI(Echo|Executing context, extension, priority)
this will put the message in standard error, you will see it in your CLI
screen of asterisk.
But will run ruby, rather than simply /bin/sh (for every call) as in the
System() case.
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il | | a Mutt's
[EMAIL PROTECTED] | | best
ICQ# 16849755 | | friend
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[Asterisk-Users] Zap Channels
Hi guys,
on my Asterisk box I've got 3 PRI, and I need to use 2 of those, with
one guest, but I see that only the 3rd is used.
This is what I've put into my extensions.conf:
---
[trunk]
exten = _7653.,1,SetCallerID(${CALLERID(number)})
exten = _7653.,2,Dial(${PRITRUNK2}/${EXTEN})
exten = _7653.,3,Dial(${PRITRUNK3}/${EXTEN})
exten = _7653.,4,Congestion
--
What's wrong?
Thanks!
--
.:FaberK:.
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Re: [Asterisk-Users] IAX Configuration
Hi if you are using AMP go to trunk and start regitering your account to check the accounts registered and Go to Console asterisk sip show registry ram On 12/2/05, Alejandro Vargas [EMAIL PROTECTED] wrote: 2005/12/2, Giovanni Miano [EMAIL PROTECTED]: See http://www.iaxtel.com/setup.htmlI'm also interested on doing this. I already set up iax connections toproviders like free world dialup, voipbuster and voipjet, but I don't know how to configure asterisk to receive the registration from theother side. I'm using amp (from [EMAIL PROTECTED]).--Alejandro Vargas___--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] Hint: how to include dialplan files from remote systems
On Thu, Dec 01, 2005 at 06:51:54PM -0800, John Todd wrote: #exec /usr/bin/curl -s http://webserver.domain.com/privatefiles/username-to-numbers /etc/asterisk/username-to-numbers #include username-to-numbers Nice. However, what happens if curl takes longer than expected? your reload waits for it. And what if you get a broken copy? I figure you should generally fetch to a temporary file and only replace the working copy if the download was successful. Something like (untested): filename=`mktemp` destination=/etc/asterisk/username-to-numbers wget -q -O$filename cp $filename $destination rm $filename This still won't report errors up, and won't do any single sanity check, but you get my point. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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] voipbuster
On Fri, December 2, 2005 9:26, Alejandro Vargas said: 2005/12/1, Tony Hoyle [EMAIL PROTECTED]: The following works in iax.conf for me: [voipbuster] host=iax.voipbuster.com type=peer username=username secret=password qualify=yes context=inbound Context=inbound? I'm using from-pstn. The problem is this: I configure asterisk (through amp) with the username and password of one just created account (without credit) and I'm able to make calls of one minute. All OK. Then I change the username and password for the one of one account that has credit and is working ok with the propietary software. The only change I make is username and password. Then, the registration is refused. I double checked and the username and password works fine with the propietary software. Also the asterisk configuration works with another username/password without credit. I sent a mail to voipbuster support with the user that has credit but they never answered. My (Working!) VB AMP settings are: Trunk Name: voipbuster Peer Details: host=iax.voipbuster.com secret= type=peer username=VBUSERname USER Context: VBUSERname USER Details: context=from-pstn secret=picard type=user REGISTRATION String: VBUSERname:[EMAIL PROTECTED] HTH -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Users Newsgroup
Are there any Asterisk Users Newsgroup? For me it's much easier to follow newsgroup then to read all e-mails. Especially with news readers with so many features. So, if anybody knows for any newsgroup that has big discussions about Asterisk, please let me know. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)393447 e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Zap Channels
Please, paste your zapata and zaptel files.
have you created groups in those files?
Regards
FaberK escribió:
Hi guys,
on my Asterisk box I've got 3 PRI, and I need to use 2 of those, with
one guest, but I see that only the 3rd is used.
This is what I've put into my extensions.conf:
---
[trunk]
exten = _7653.,1,SetCallerID(${CALLERID(number)})
exten = _7653.,2,Dial(${PRITRUNK2}/${EXTEN})
exten = _7653.,3,Dial(${PRITRUNK3}/${EXTEN})
exten = _7653.,4,Congestion
--
What's wrong?
Thanks!
--
.:FaberK:.
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[Asterisk-Users] voice problems under 8 concurent calles
hi guys: we suffer strange voice shakings after only8 concurrent PSTN calls, any one knows why? we use g729, which can be cpu intensive, is this the coz? Matt ___ --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] voipbuster
On Fri, December 2, 2005 9:29, Alejandro Vargas said: 2005/12/1, Francesco Peeters [EMAIL PROTECTED]: Mine works just fine. It's a pain though if you have to get a new account, as minimum amount is now EUR 5... OTOH, for free calls, it might be worth it... The account that I'm testing is one of 5 EUR. I'm trying to test it with asterisk because if it works, I will place credit on my account. But if I can't make it work with an account with credit, I must suspect they are blocking the accounts with credit for avoiding people to use it with other software than their propietary client. Mine is a EUR 5 account, and is working fine... I set up below outgoing routes for VB free calls: 0030. 00311. 00312. 00313. 00314. 00315. 00317. 0034. 00352. 00353. 00358. 0041. 0043. 0045. 0046. 0047. 0049. (The 0031x are set up in this manner to avoid Cellphone (0031-6.) and Premium (0031-8. 0031-9.) numbers.) HTH -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --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] Re: Very Weird problem with MeetMe, SIP, Zap and the combo of the three
In article [EMAIL PROTECTED], Nir Simionovich - CTO [EMAIL PROTECTED] wrote: Hi All, I have a really funky problem, which I can't seem to pin point.I have a setup that looks something like this: SS7 Networks --SS7-- Veraz IGate4000 --SIP-- Asterisk Now, Asterisk has a second connection, that looks like this: Asterisk --PRI-- Avaya CTI Now, I'll describe several sceanrios that I'm testing, with some really Weird results: My first guess would be something to do with silence suppression. Try applying the async patch from http://bugs.digium.com/view.php?id=5374 and see if it helps. The patch is not very involved, and can be applied by hand if necessary (I did so successfully for Asterisk 1.0.x). Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.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
Re: [Asterisk-Users] Error on using queue.
You'll have to have agents join the queues by issuing the commands AgentLogin() or AgentCallBackLogin() from an extension in your dialplan. l. On Thu, 01 Dec 2005 17:53:19 +0100, gc [EMAIL PROTECTED] wrote: Thanks. I made change to joinempty=yes. And now I can hear the music on hold. But it would not ring the agent even if I login agent in. When I run show queue command under CLI, I got these messages: queue1 has 1 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:2, SL:0.0% within 0s Members: Agent/555997 (Unavailable) has taken no calls yet Agent/555998 (Unavailable) has taken no calls yet It seems that something wrong with my config file, it did not login any agent. - Original Message - From: Dov Bigio To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, December 01, 2005 8:33 AM Subject: Re: [Asterisk-Users] Error on using queue. If you are using 1.2, it might be the joinempty and leavewhenempty parameters. Their default are different than the 1.0.x releases - Original Message - From: gc To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, December 01, 2005 11:27 AM Subject: [Asterisk-Users] Error on using queue. I am trying to use * as ACD server for our sip proxy. I first dial 55 to login 98 as ACD agent it worked fine and then when I dialed 98, I got these messages from * CLI: -- Executing Answer(SIP/98-f718, ) in new stack -- Executing Ringing(SIP/98-f718, ) in new stack -- Executing Wait(SIP/98-f718, 2) in new stack -- Executing Queue(SIP/98-f718, queue1) in new stack Nov 30 16:54:12 WARNING[7579]: app_queue.c:3078 queue_exec: Unable to join queue 'queue1' -- Executing Hangup(SIP/98-f718, ) in new stack == Spawn extension (default, 99, 5) exited non-zero on 'SIP/5025155598-f718' Can anybody tell me what cause this problem? The followings are my configuration files: extensions.conf: [default] ;For incoming call to ring into the queue. exten= 99,1,Answer exten= 99,2,Ringing exten= 99,3,Wait(2) exten= 99,4,Queue(queue1) exten= 99,5,Hangup ;Agent login exten = 55,1,AgentCallBackLogin(|[EMAIL PROTECTED]) ;Agent logout exten = 55,1,AgentCallBackLogin(|1) exten = 97,1,Dial(SIP/97) exten = 98,1,Dial(SIP/98) agents.conf: [Agent1] agent = 97,,Gary1 agent = 98,,Gary2 queues.conf: [queue1] musiconhold = default strategy=ringall timeout=15 retry=5 wrapuptime=0 maxlen = 0 announce-frequency = 0 announce-holdtime = no member = Agent1/555997 member = Agent1/555998 sip.conf: port=5060 bindaddr=192.168.111.11 context=default allow=ulaw [97] type=friend username=97 insecure=very canreinvite=no context=default host=192.168.111.2 [98] type=friend username=98 insecure=very canreinvite=no context=default host=192.168.111.2 ___ --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 -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --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] Limiting DID calls
I have 3 DID numbers and one E1. How to limit incoming calls so first DID can accept 10, second 15 and the third 5 councurent calls. 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] Zap Channels
HI, here they are:
--
zapata.conf
[channels]
language=it
;context=incoming
context=default
switchtype=national
pridialplan=unknown
signalling=pri_cpe
echocancel=yes
group = 1
channel = 1-15,17-31
group = 2
channel = 32-46,48-62
group = 3
channel = 63-77,79-93
transfer=yes
threewaycalling=yes
callwaitingcallerid=yes
callwaiting=yes
cancallforward=yes
usecallerid=yes
hidecallerid=no
echocancel=yes
echotraining=yes
zaptel.conf
defaultzone=it
loadzone=it
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
span=2,1,0,ccs,hdb3,crc4
bchan=32-46,48-62
dchan=47
span=3,1,0,ccs,hdb3,crc4
bchan=63-77,79-93
dchan=78
span=4,1,0,ccs,hdb3,crc4
bchan=94-109,111-124
dchan=110
--
2005/12/2, Xisco Mateu [EMAIL PROTECTED]:
Please, paste your zapata and zaptel files.
have you created groups in those files?
Regards
FaberK escribió:
Hi guys,
on my Asterisk box I've got 3 PRI, and I need to use 2 of those, with
one guest, but I see that only the 3rd is used.
This is what I've put into my extensions.conf:
---
[trunk]
exten = _7653.,1,SetCallerID(${CALLERID(number)})
exten = _7653.,2,Dial(${PRITRUNK2}/${EXTEN})
exten = _7653.,3,Dial(${PRITRUNK3}/${EXTEN})
exten = _7653.,4,Congestion
--
What's wrong?
Thanks!
--
.:FaberK:.
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--
.:FaberK:.
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[Asterisk-Users] AGI Problem
Hi,
I'm running the last CVS asterisk version (I was running an older version
before with the same problem) and I've a problem with agi scripts. Commands
results are not always correct.
I've made a small agi test script that execute ChanIsAvail on an inexistent
extension:
#!/usr/bin/perl
$|=1;
while(STDIN) {
chomp;
last unless length($_);
if (/^agi_(\w+)\:\s+(.*)$/) {
$AGI{$1} = $2;
}
}
# Check
print EXEC ChanIsAvail IAX/24\n;
$result = STDIN;
print VERBOSE \$result\ 0\n;
# Check
print EXEC ChanIsAvail IAX/24\n;
$result = STDIN;
print VERBOSE \$result\ 0\n;
# Check
print EXEC ChanIsAvail IAX/24\n;
$result = STDIN;
print VERBOSE \$result\ 0\n;
Result is :
-- Executing DeadAGI(SIP/200-60d2, b) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/b
-- AGI Script Executing Application: (ChanIsAvail) Options: (IAX/24)
Dec 2 10:29:37 WARNING[15776]: channel.c:2520 ast_request: No channel type
registered for 'IAX'
b: 200 result=-1
-- AGI Script Executing Application: (ChanIsAvail) Options: (IAX/24)
Dec 2 10:29:37 WARNING[15776]: channel.c:2520 ast_request: No channel type
registered for 'IAX'
b: 200 result=1
-- AGI Script Executing Application: (ChanIsAvail) Options: (IAX/24)
Dec 2 10:29:37 WARNING[15776]: channel.c:2520 ast_request: No channel type
registered for 'IAX'
b: 510 Invalid or unknown command
-- AGI Script b completed, returning 0
The first result is ok (-1) but not the second and the third.
Why do I get different results for the same command?
Thank you,
Regards,
Cyrille
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Re: [Asterisk-Users] Zap Channels
On Fri, Dec 02, 2005 at 11:00:34AM +0100, FaberK wrote:
HI, here they are:
--
zapata.conf
[channels]
language=it
;context=incoming
context=default
switchtype=national
pridialplan=unknown
signalling=pri_cpe
echocancel=yes
group = 1
channel = 1-15,17-31
group = 2
channel = 32-46,48-62
group = 3
channel = 63-77,79-93
transfer=yes
threewaycalling=yes
callwaitingcallerid=yes
callwaiting=yes
cancallforward=yes
usecallerid=yes
hidecallerid=no
echocancel=yes
echotraining=yes
zaptel.conf
defaultzone=it
loadzone=it
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
span=2,1,0,ccs,hdb3,crc4
bchan=32-46,48-62
dchan=47
span=3,1,0,ccs,hdb3,crc4
bchan=63-77,79-93
dchan=78
span=4,1,0,ccs,hdb3,crc4
bchan=94-109,111-124
dchan=110
--
2005/12/2, Xisco Mateu [EMAIL PROTECTED]:
Please, paste your zapata and zaptel files.
have you created groups in those files?
Regards
FaberK escribió:
Hi guys,
on my Asterisk box I've got 3 PRI, and I need to use 2 of those, with
one guest, but I see that only the 3rd is used.
This is what I've put into my extensions.conf:
---
[trunk]
exten = _7653.,1,SetCallerID(${CALLERID(number)})
exten = _7653.,2,Dial(${PRITRUNK2}/${EXTEN})
exten = _7653.,3,Dial(${PRITRUNK3}/${EXTEN})
exten = _7653.,4,Congestion
--
What's wrong?
What is PRITRUNK1? where is it defined?
How do you know something is wrong? Could you please paste the trace
from the logs/cli when verbosity is set to a high enough value? (e.g: 3)
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il | | a Mutt's
[EMAIL PROTECTED] | | best
ICQ# 16849755 | | friend
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Re: [Asterisk-Users] Running asterisk within screen
On Fri, Dec 02, 2005 at 09:32:35AM +0800, Marcus Deluigi (intern) wrote: Hi! I downloaded asterisk 1.2.0 and compiled it myself. The default behaviour is that calling 'asterisk' will return the prompt and calling 'asterisk -v' is returning the CLI. I want to run asterisk within screen, however '/usr/bin/screen -L /usr/sbin/asterisk -v' outputs: [screen is terminating] Maybe you can get more details using strace: /usr/bin/screen -L strace -f -o /tmp/trace /usr/sbin/asterisk -v and I have no screen session running and I also have no asterisk CLI to connect to. I can't explain the behaviour and the screenlog is empty. permissions? If that is what you suspect, strace the whole screen session. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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] Zap Channels
PRITRUNK1 is defined into the extensions.conf globals:
--
[globals]
PRITRUNK1=Zap/g1
PRITRUNK2=Zap/g2
PRITRUNK3=Zap/g3
--
Well I know what's happening, from my asterisk CDRs, and also from the PRI-CDRs.
We use a Teles.
2005/12/2, Tzafrir Cohen [EMAIL PROTECTED]:
On Fri, Dec 02, 2005 at 11:00:34AM +0100, FaberK wrote:
HI, here they are:
--
zapata.conf
[channels]
language=it
;context=incoming
context=default
switchtype=national
pridialplan=unknown
signalling=pri_cpe
echocancel=yes
group = 1
channel = 1-15,17-31
group = 2
channel = 32-46,48-62
group = 3
channel = 63-77,79-93
transfer=yes
threewaycalling=yes
callwaitingcallerid=yes
callwaiting=yes
cancallforward=yes
usecallerid=yes
hidecallerid=no
echocancel=yes
echotraining=yes
zaptel.conf
defaultzone=it
loadzone=it
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
span=2,1,0,ccs,hdb3,crc4
bchan=32-46,48-62
dchan=47
span=3,1,0,ccs,hdb3,crc4
bchan=63-77,79-93
dchan=78
span=4,1,0,ccs,hdb3,crc4
bchan=94-109,111-124
dchan=110
--
2005/12/2, Xisco Mateu [EMAIL PROTECTED]:
Please, paste your zapata and zaptel files.
have you created groups in those files?
Regards
FaberK escribió:
Hi guys,
on my Asterisk box I've got 3 PRI, and I need to use 2 of those, with
one guest, but I see that only the 3rd is used.
This is what I've put into my extensions.conf:
---
[trunk]
exten = _7653.,1,SetCallerID(${CALLERID(number)})
exten = _7653.,2,Dial(${PRITRUNK2}/${EXTEN})
exten = _7653.,3,Dial(${PRITRUNK3}/${EXTEN})
exten = _7653.,4,Congestion
--
What's wrong?
What is PRITRUNK1? where is it defined?
How do you know something is wrong? Could you please paste the trace
from the logs/cli when verbosity is set to a high enough value? (e.g: 3)
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il | | a Mutt's
[EMAIL PROTECTED] | | best
ICQ# 16849755 | | friend
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--
.:FaberK:.
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[Asterisk-Users] BT - DSS
Has anyone used BT's DSS services? http://www.bt.com/isdn/isdn2e/extra/dss.htm I have an ISDN 2e comming into my Asterisk and would like to deflect calls when I am busy (or I can't get my HFC-PCI card to run correctly LOL) to my PSTN-IAX VOIP number if the Asterisk doesn't answer. Listening out... Simon ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Running asterisk within screen
Tzafrir Cohen wrote: /usr/bin/screen -L strace -f -o /tmp/trace /usr/sbin/asterisk -v and I have no screen session running and I also have no asterisk CLI to connect to. I can't explain the behaviour and the screenlog is empty. permissions? If that is what you suspect, strace the whole screen session. Can this be as simple as needing to run 'asterisk -c' to keep a console open (which then 'screen' will manage) ? Cheers, Gavin. ___ --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] Re: Asterisk Users Newsgroup
In article [EMAIL PROTECTED], Tomislav Parèina [EMAIL PROTECTED] wrote: Are there any Asterisk Users Newsgroup? For me it's much easier to follow newsgroup then to read all e-mails. Especially with news readers with so many features. So, if anybody knows for any newsgroup that has big discussions about Asterisk, please let me know. I'm not aware of any. Instead, I run INN on my local Linux box and gateway the Asterisk mailing lists into it. I have them set up as moderated groups asterisk.dev, asterisk.users, etc., with the moderator address being the list posting address. I can then read the lists just like newsgroups. If it's of any interest, I can make the scripts and config available. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.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
[Asterisk-Users] Asterisk-users
Dear guys My asterisk is giving some error as bellow with some extention.Have any body received this type of error. CDR updated on SIP/200-a6a5 -- Executing Goto(SIP/200-a6a5, ivr-main|s|3) in new stack -- Goto (ivr-main,s,3) -- Executing BackGround(SIP/200-a6a5, itabs/greeting) in new stack -- Playing 'itabs/greeting' (language 'en') -- Timeout on SIP/200-360e == CDR updated on SIP/200-360e -- Executing Goto(SIP/200-360e, ivr-main|s|3) in new stack -- Goto (ivr-main,s,3) -- Executing BackGround(SIP/200-360e, itabs/greeting) in new stack -- Playing 'itabs/greeting' (language 'en') -- Timeout on SIP/200-093a == CDR updated on SIP/200-093a -- Executing Goto(SIP/200-093a, ivr-main|s|3) in new stack -- Goto (ivr-main,s,3) -- Executing BackGround(SIP/200-093a, itabs/greeting) in new stack -- Playing 'itabs/greeting' (language 'en') -- Timeout on SIP/200-a6a5 == CDR updated on SIP/200-a6a5 -- Executing Goto(SIP/200-a6a5, ivr-main|s|3) in new stack -- Goto (ivr-main,s,3) -- Executing BackGround(SIP/200-a6a5, itabs/greeting) in new stack -- Playing 'itabs/greeting' (language 'en') -- Timeout on SIP/200-360e I am continuesly receive this problem,when taking a call from 200(Phone type is GrandStream).I am very appreciate if any body can help me. Regards Nirukshitha Gamage -- This e-mail and any attachments are intended for the above named recipient(s) only and may be privileged. This message and any attachments has been scanned for viruses and dangerous content by ITABS Lanka Mail Scanner, and is believed to be clean. Although measures have been taken to ensure that this e-mail and attachments are free from any virus we advise that, in keeping with good computing practice, the recipient should ensure they are actually virus free. Please note that this message has been sent over public networks which may not be a 100% secure communications medium and ITABS Lanka cannot be held responsible for its integrity. ___ --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] Codec Problem
Hi, Do you know from where i can buy g723 codec. for g729 i can buy it from digium.com. But Please let me know from where i can get g723 codec. And the codecs purchasing can solved my problem? -- Thank You, Code Lover ___ --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] IAX Configuration
2005/12/2, ram [EMAIL PROTECTED]: if you are using AMP go to trunk and start regitering your account Humm... what I'm trying to do, and what is this thread subject, is to connect asterisk-to-asterisk. Then... I go to trunks, create a new iax trunk, invent some user/password, use the ip of the other asterisk server, etc. Then I go to the other server... I supose I must also create an iax trunk, but... where do I create the user that I placed in the first server in order to validate it? Also I want to link the two asterisk boxes in order to be able to call extensions in each other and use the external lines, then I supose I must place it in from-internal context. -- Alejandro Vargas ___ --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 Problem
Try print EXEC ChanIsAvail IAX2/24\n;
Channel type is IAX2 not IAX
Cheers
2005/12/2, Cyrille Demaret [EMAIL PROTECTED]:
Hi,
I'm running the last CVS asterisk version (I was running an older version
before with the same problem) and I've a problem with agi scripts. Commands
results are not always correct.
I've made a small agi test script that execute ChanIsAvail on an inexistent
extension:
#!/usr/bin/perl
$|=1;
while(STDIN) {
chomp;
last unless length($_);
if (/^agi_(\w+)\:\s+(.*)$/) {
$AGI{$1} = $2;
}
}
# Check
print EXEC ChanIsAvail IAX/24\n;
$result = STDIN;
print VERBOSE \$result\ 0\n;
# Check
print EXEC ChanIsAvail IAX/24\n;
$result = STDIN;
print VERBOSE \$result\ 0\n;
# Check
print EXEC ChanIsAvail IAX/24\n;
$result = STDIN;
print VERBOSE \$result\ 0\n;
Result is :
-- Executing DeadAGI(SIP/200-60d2, b) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/b
-- AGI Script Executing Application: (ChanIsAvail) Options: (IAX/24)
Dec 2 10:29:37 WARNING[15776]: channel.c:2520 ast_request: No channel type
registered for 'IAX'
b: 200 result=-1
-- AGI Script Executing Application: (ChanIsAvail) Options: (IAX/24)
Dec 2 10:29:37 WARNING[15776]: channel.c:2520 ast_request: No channel type
registered for 'IAX'
b: 200 result=1
-- AGI Script Executing Application: (ChanIsAvail) Options: (IAX/24)
Dec 2 10:29:37 WARNING[15776]: channel.c:2520 ast_request: No channel type
registered for 'IAX'
b: 510 Invalid or unknown command
-- AGI Script b completed, returning 0
The first result is ok (-1) but not the second and the third.
Why do I get different results for the same command?
Thank you,
Regards,
Cyrille
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Re: [Asterisk-Users] Limiting DID calls
You can use global var and same if condition Cheers 2005/12/2, omadon [EMAIL PROTECTED]: I have 3 DID numbers and one E1. How to limit incoming calls so first DID can accept 10, second 15 and the third 5 councurent calls. 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 -- Giovanni Miano ___ --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] voice problems under 8 concurent calles
- Check int call on IRQ - Check cpu usage Good Luck 2005/12/2, Matt [EMAIL PROTECTED]: hi guys: we suffer strange voice shakings after only 8 concurrent PSTN calls, any one knows why? we use g729, which can be cpu intensive, is this the coz? Matt ___ --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 -- Giovanni Miano ___ --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] RE:how to solve error : cannot find extension context 'from-sip'
hi, I am a newbie to asterisk. I am tryining to connect two sip based soft X-Lite phones to an asterisk server. i made following settings in sip.conf: [general] port=5060 bindaddr=0.0.0.0 allow=all context=bogon-calls [2000] type=friend username=2000 secret=tejas host=dynamic context=from-sip [2001] type=friend username=2001 secret=tejas host=dynamic context=from-sip and made following configuration in extension.conf : exten = 2000,1,Dial(SIP/2000,20) exten = 2001,1,Dial(SIP/2001,20) also i made proper settings in both SIP phones. Now the problem is when i am making call from any of phone at srever i am getting an error : "cannot find extension contex 'from-sip'. I analysed result in ethereal also. packets are comming to server. but server is saying : "Proxy authentication required" . N ow i m not getting where is the exact problem. can any one help me for this problem. Response to this problem is most welcome. my email-id is [EMAIL PROTECTED] thanks tejas Yahoo! Personals Single? There's someone we'd like you to meet. Lots of someones, actually. Yahoo! Personals___ --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] equal priority trunks for balancing
Is there any way to create various trunks with the same priority. I'm interested on usingo 2 trunks, but balancing the usage in both because both has a number of free minutes. If I give preference to one over other, this one will exceed the free limit much before the other. -- Alejandro Vargas ___ --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] change priority by time
I has varios access to pstn but each one has different hours in the day when the calls are free. Then I want to change the priority with the time of deay in order to make asterisk to prefer the one where the calls are free. Is there an easy way to do this? If there is not, I can place a script in cron, but what is the minumun change I must do from this script? Is there some command from the asterisk interface for doing the change on line without modificating the config files? -- Alejandro Vargas ___ --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] RE:how to solve error : cannot find extension context 'from-sip'
Hi Tejas, what context are the extensinos you included below in your extensions.conf file? Tejas Shah wrote: hi, I am a newbie to asterisk. I am tryining to connect two sip based soft X-Lite phones to an asterisk server. i made following settings in sip.conf: [general] port=5060 bindaddr=0.0.0.0 allow=all context=bogon-calls [2000] type=friend username=2000 secret=tejas host=dynamic context=from-sip [2001] type=friend username=2001 secret=tejas host=dynamic context=from-sip and made following configuration in extension.conf : exten = 2000,1,Dial(SIP/2000,20) exten = 2001,1,Dial(SIP/2001,20) also i made proper settings in both SIP phones. Now the problem is when i am making call from any of phone at srever i am getting an error : "cannot find extension contex 'from-sip'. I analysed result in ethereal also. packets are comming to server. but server is saying : "Proxy authentication required" . N ow i m not getting where is the exact problem. can any one help me for this problem. Response to this problem is most welcome. my email-id is [EMAIL PROTECTED] thanks tejas Yahoo! Personals Single? There's someone we'd like you to meet. Lots of someones, actually. Yahoo! Personals ___ --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 begin:vcard fn:Tomas Paseka n:Paseka;Tomas adr:;;PO Box 46;Balgowlah;NSW;2093;Australia email;internet:[EMAIL PROTECTED] tel;work:02 9850 0994 tel;fax:02 9949 1875 tel;home:02 9011 2135 tel;cell:0413 920 074 note:Who actually reads this? x-mozilla-html:TRUE url:http://www.peskey.info/ version:2.1 end:vcard ___ --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] equal priority trunks for balancing
Alejandro,
You could do something like:
[balance]
exten = _X., 1, Random(50:4)
exten = _X., 2, Dial(Zap/g1/${EXTEN})
exten = _X., 3, Congestion
exten = _X., 4, Dial(Zap/g2/${EXTEN})
exten = _X., 5, Congestion
See http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Random
Alistair Cunningham,
Integrics Ltd,
+44 (0)7870 699 479
http://integrics.com/
Alejandro Vargas wrote:
Is there any way to create various trunks with the same priority.
I'm interested on usingo 2 trunks, but balancing the usage in both
because both has a number of free minutes. If I give preference to one
over other, this one will exceed the free limit much before the other.
--
Alejandro Vargas
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Re: [Asterisk-Users] polycom backlight?
Official Polycom view seems to be that you shouldn't work at night :) The phones are crying out for a backlit LCD that only lights when ambient light is low. I have a cheap radio/weather station with a large LCD that does that. ___ --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] sixtel
I have. I have not noticed any major problems with my 800 DIDs or outgoing with them for about a year now. I don't use them very much though -Original Message- From: Bill Michaelson [mailto:[EMAIL PROTECTED] Sent: Thursday, December 01, 2005 2:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] sixtel Just curious... Is there anyone out there who has given this outfit money and actually received any service from them? ___ --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 Problem
Hi,
I've changed that and it's the same problem. I've this problem with all
applications. Results from agi are not correct.
Regards,
Cyrille
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Giovanni
Miano
Envoyé : vendredi 2 décembre 2005 12:52
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [Asterisk-Users] AGI Problem
Try print EXEC ChanIsAvail IAX2/24\n;
Channel type is IAX2 not IAX
Cheers
2005/12/2, Cyrille Demaret [EMAIL PROTECTED]:
Hi,
I'm running the last CVS asterisk version (I was running an older version
before with the same problem) and I've a problem with agi scripts.
Commands
results are not always correct.
I've made a small agi test script that execute ChanIsAvail on an
inexistent
extension:
#!/usr/bin/perl
$|=1;
while(STDIN) {
chomp;
last unless length($_);
if (/^agi_(\w+)\:\s+(.*)$/) {
$AGI{$1} = $2;
}
}
# Check
print EXEC ChanIsAvail IAX/24\n;
$result = STDIN;
print VERBOSE \$result\ 0\n;
# Check
print EXEC ChanIsAvail IAX/24\n;
$result = STDIN;
print VERBOSE \$result\ 0\n;
# Check
print EXEC ChanIsAvail IAX/24\n;
$result = STDIN;
print VERBOSE \$result\ 0\n;
Result is :
-- Executing DeadAGI(SIP/200-60d2, b) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/b
-- AGI Script Executing Application: (ChanIsAvail) Options: (IAX/24)
Dec 2 10:29:37 WARNING[15776]: channel.c:2520 ast_request: No channel
type
registered for 'IAX'
b: 200 result=-1
-- AGI Script Executing Application: (ChanIsAvail) Options: (IAX/24)
Dec 2 10:29:37 WARNING[15776]: channel.c:2520 ast_request: No channel
type
registered for 'IAX'
b: 200 result=1
-- AGI Script Executing Application: (ChanIsAvail) Options: (IAX/24)
Dec 2 10:29:37 WARNING[15776]: channel.c:2520 ast_request: No channel
type
registered for 'IAX'
b: 510 Invalid or unknown command
-- AGI Script b completed, returning 0
The first result is ok (-1) but not the second and the third.
Why do I get different results for the same command?
Thank you,
Regards,
Cyrille
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--
Giovanni Miano
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Re: [Asterisk-Users] [EMAIL PROTECTED] isdn
2005/11/30, Francesco Peeters [EMAIL PROTECTED]: Yep, tried APIC, NOAPIC, ACPI=OFF, etc. (capitals only for clarity!) but to no avail! As soon as both share the same IRQ, the zaphfc driver stops passing data to asterisk... It is supposed to when you are using APIC, you should obtain many interrupts and the devices will obtain new interrupts with values grater than 16 which is the maxymun in a normal PC without APIC. Chech this lspc, you will see: smbus in irq17 sound card in irq18 usb controllers in irq 20, 21 and 23 4 ethernet cards, in irqs 16, 18 and 18 (it seems one of them are sharing irq with other) Your problem could be the isdn cards are not apic compatible. # lspci -v 00:00.0 Host bridge: Silicon Integrated Systems [SiS] 746 Host (rev 10) Subsystem: Unknown device 1849:0746 Flags: bus master, medium devsel, latency 0 Memory at d000 (32-bit, non-prefetchable) [size=64M] Capabilities: [c0] AGP version 3.0 00:01.0 PCI bridge: Silicon Integrated Systems [SiS] SG86C202 (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 32 Bus: primary=00, secondary=01, subordinate=02, sec-latency=32 I/O behind bridge: 9000-9fff Memory behind bridge: cfd0-cfef Prefetchable memory behind bridge: afa0-cfbf 00:02.0 ISA bridge: Silicon Integrated Systems [SiS] SiS963 [MuTIOL Media IO] (rev 25) Flags: bus master, medium devsel, latency 0 00:02.1 SMBus: Silicon Integrated Systems [SiS] SiS961/2 SMBus Controller Flags: medium devsel, IRQ 17 I/O ports at 0c00 [size=32] 00:02.5 IDE interface: Silicon Integrated Systems [SiS] 5513 [IDE] (prog-if 80 [Master]) Subsystem: Unknown device 1849:5513 Flags: bus master, medium devsel, latency 128 I/O ports at ff00 [size=16] 00:02.7 Multimedia audio controller: Silicon Integrated Systems [SiS] Sound Controller (rev a0) Subsystem: Unknown device 1849:7012 Flags: bus master, medium devsel, latency 32, IRQ 18 I/O ports at dc00 [size=256] I/O ports at d800 [size=128] Capabilities: [48] Power Management version 2 00:03.0 USB Controller: Silicon Integrated Systems [SiS] USB 1.0 Controller (rev 0f) (prog-if 10 [OHCI]) Subsystem: Unknown device 1849:7001 Flags: bus master, medium devsel, latency 32, IRQ 20 Memory at cfffd000 (32-bit, non-prefetchable) [size=4K] 00:03.1 USB Controller: Silicon Integrated Systems [SiS] USB 1.0 Controller (rev 0f) (prog-if 10 [OHCI]) Subsystem: Unknown device 1849:7001 Flags: bus master, medium devsel, latency 32, IRQ 21 Memory at cfffe000 (32-bit, non-prefetchable) [size=4K] 00:03.2 USB Controller: Silicon Integrated Systems [SiS] USB 2.0 Controller (prog-if 20 [EHCI]) Subsystem: Unknown device 1849:7001 Flags: bus master, medium devsel, latency 32, IRQ 23 Memory at c000 (32-bit, non-prefetchable) [size=4K] Capabilities: [50] Power Management version 2 00:04.0 Ethernet controller: Silicon Integrated Systems [SiS] SiS900 PCI Fast Ethernet (rev 90) Subsystem: Unknown device 1849:8201 Flags: bus master, medium devsel, latency 32, IRQ 19 I/O ports at d400 [size=256] Memory at cfffc000 (32-bit, non-prefetchable) [size=4K] Expansion ROM at fffe [disabled] [size=128K] Capabilities: [40] Power Management version 2 00:0a.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL-8139/8139C/8139C+ (rev 10) Subsystem: Realtek Semiconductor Co., Ltd. RT8139 Flags: bus master, medium devsel, latency 32, IRQ 18 I/O ports at d000 [size=256] Memory at cfffbf00 (32-bit, non-prefetchable) [size=256] Expansion ROM at [disabled] [size=64K] Capabilities: [50] Power Management version 2 00:0b.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL-8139/8139C/8139C+ (rev 10) Subsystem: Realtek Semiconductor Co., Ltd. RT8139 Flags: bus master, medium devsel, latency 32, IRQ 19 I/O ports at cc00 [size=256] Memory at cfffbe00 (32-bit, non-prefetchable) [size=256] Expansion ROM at [disabled] [size=64K] Capabilities: [50] Power Management version 2 00:0c.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL-8139/8139C/8139C+ (rev 10) Subsystem: Kingmax Technology Inc: Unknown device 0203 Flags: bus master, medium devsel, latency 32, IRQ 16 I/O ports at c800 [size=256] Memory at cfffbd00 (32-bit, non-prefetchable) [size=256] Expansion ROM at fffe [disabled] [size=128K] Capabilities: [50] Power Management version 2 01:00.0 VGA compatible controller: ATI Technologies Inc RV280 [Radeon 9200 SE] (rev 01) (prog-if 00 [VGA]) Subsystem: C.P. Technology Co. Ltd CN-AG92E Flags: bus master, 66Mhz, medium devsel, latency 32, IRQ 16 Memory at b000
Re: [Asterisk-Users] change priority by time
Alejandro Vargas wrote: I has varios access to pstn but each one has different hours in the day when the calls are free. Then I want to change the priority with the time of deay in order to make asterisk to prefer the one where the calls are free. Is there an easy way to do this? If there is not, I can place a script in cron, but what is the minumun change I must do from this script? Is there some command from the asterisk interface for doing the change on line without modificating the config files? -- Alejandro Vargas ___ --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 You can use GotoIf() to select different access depending on the time. Check out the documentation at http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip page 123. or http://www.voip-info.org/wiki-Asterisk+cmd+GotoIfTimefor more information. Dan ___ --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] Error on using queue.
My aterisk is working now. I had some spelling
mistakes in queues.conf.
Thanks for your help.
- Original Message -
From:
Dov Bigio
To: gc ; Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, December 01, 2005 12:22
PM
Subject: Re: [Asterisk-Users] Error on
using queue.
How is your agents.conf ? How is your login in
extensions.conf?
- Original Message -
From:
gc
To: Dov Bigio ; Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, December 01, 2005 2:53
PM
Subject: Re: [Asterisk-Users] Error on
using queue.
Thanks. I made change to joinempty=yes. And now
I can hear the music on hold. But it would not ring the agent even if I
login agent in. When I run show queue command under CLI, I got these
messages:
queue1 has
1 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:2,
SL:0.0% within 0s Members:
Agent/555997 (Unavailable) has taken no calls
yet Agent/555998 (Unavailable) has
taken no calls yet
It seems that something wrong with my config
file, it did not login any agent.
- Original Message -
From:
Dov Bigio
To: Asterisk Users Mailing List
- Non-Commercial Discussion
Sent: Thursday, December 01, 2005
8:33 AM
Subject: Re: [Asterisk-Users] Error
on using queue.
If you are using 1.2, it might be the
joinempty and leavewhenempty parameters.
Their default are different than the 1.0.x
releases
- Original Message -
From:
gc
To: Asterisk Users Mailing
List - Non-Commercial Discussion
Sent: Thursday, December 01, 2005
11:27 AM
Subject: [Asterisk-Users] Error on
using queue.
I am trying to use * as ACD server for our
sip proxy.
I first dial 55 to login 98
as ACD agent it worked fine and then when I dialed 98,
I got these messages from *
CLI:
-- Executing
Answer("SIP/98-f718", "") in new stack --
Executing Ringing("SIP/98-f718", "") in new
stack -- Executing Wait("SIP/98-f718",
"2") in new stack -- Executing
Queue("SIP/98-f718", "queue1") in new stackNov 30 16:54:12
WARNING[7579]: app_queue.c:3078 queue_exec: Unable to join queue
'queue1' -- Executing
Hangup("SIP/98-f718", "") in new stack == Spawn
extension (default, 99, 5) exited non-zero on
'SIP/5025155598-f718'
Can anybody tell me what cause this
problem?
The followings are my configuration
files:
extensions.conf:
[default]
;For incoming call to ring into the
queue.exten= 99,1,Answerexten=
99,2,Ringingexten= 99,3,Wait(2)exten=
99,4,Queue(queue1)exten=
99,5,Hangup
;Agent loginexten =
55,1,AgentCallBackLogin(|[EMAIL PROTECTED]);Agent
logoutexten = 55,1,AgentCallBackLogin(|1)
exten =
97,1,Dial(SIP/97)exten =
98,1,Dial(SIP/98)
agents.conf:
[Agent1]agent =
97,,Gary1agent = 98,,Gary2
queues.conf:
[queue1]musiconhold =
defaultstrategy=ringalltimeout=15retry=5wrapuptime=0maxlen
= 0announce-frequency = 0announce-holdtime = nomember =
Agent1/555997member = Agent1/555998
sip.conf:
port=5060bindaddr=192.168.111.11context=defaultallow=ulaw
[97]type=friendusername=97insecure=verycanreinvite=nocontext=defaulthost=192.168.111.2
[98]type=friendusername=98insecure=verycanreinvite=nocontext=defaulthost=192.168.111.2
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Re: [Asterisk-Users] WG: App_rxfax problem
Ouch ... error while writing audio data: : Broken pipe If you are talking about the Ouch message, yes lots of people have seen the error and its usually the result of some misconfiguration in one of your files (likely zapata.conf). Correct me if I'm wrong, but isn't that message from mpg123 itself? It appears in the binary (via strings), and I've seen it at non-asterisk times too. AFAIK it comes up whenever the parent application (asterisk in this case) quits without closing it properly (hence, broken pipe). As such, this means that the above error simply shows that asterisk crashed (which they presumably already knew), and has nothing to do with the problem itself... Andrew -- Linux supports the notion of a command line or a shell for the same reason that only children read books with only pictures in them. Language, be it English or something else, is the only tool flexible enough to accomplish a sufficiently broad range of tasks. -- Bill Garrett ___ --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] Soft Phone IP
Hello to all, i have now two questions the first is, anybody know some software to emulate a ip phone? or a soft phone ip to work with asterisk in other computer ? and the other is how i do to install 1 rpm from my cd rom? i accessed with the root password but i navigate for all the directories and dont locate the cdrom directory in /mnt, how i do that?? thanks to everybody __ Visita http://www.tutopia.com y comienza a navegar más rápido en Internet. Tutopia es Internet para todos. ___ --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
AW: [Asterisk-Users] WG: App_rxfax problem
But i have this in astewrisk log: Dec 1 15:01:08 VERBOSE[27950] logger.c: [app_rxfax.so] Dec 1 15:01:08 WARNING[27950] loader.c: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handler Dec 1 15:01:08 WARNING[27950] loader.c: Loading module app_rxfax.so failed! -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Andrew Furey Gesendet: Freitag, 2. Dezember 2005 14:36 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [Asterisk-Users] WG: App_rxfax problem Ouch ... error while writing audio data: : Broken pipe If you are talking about the Ouch message, yes lots of people have seen the error and its usually the result of some misconfiguration in one of your files (likely zapata.conf). Correct me if I'm wrong, but isn't that message from mpg123 itself? It appears in the binary (via strings), and I've seen it at non-asterisk times too. AFAIK it comes up whenever the parent application (asterisk in this case) quits without closing it properly (hence, broken pipe). As such, this means that the above error simply shows that asterisk crashed (which they presumably already knew), and has nothing to do with the problem itself... Andrew -- Linux supports the notion of a command line or a shell for the same reason that only children read books with only pictures in them. Language, be it English or something else, is the only tool flexible enough to accomplish a sufficiently broad range of tasks. -- Bill Garrett ___ --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] version 1.2 with chan_bluetooth
Dan, Thanks - that helps... Now when I run it, I hear my headset ring or beep for the incoming call and when I answer it I dont get any audio. I have a kensington dongle and a plantronics head set. Jerry Hi Jerry, - Original Message - From: "Jerry Geis" geisj at pagestation.com To: asterisk-users at lists.digium.com Sent: Thursday, December 01, 2005 10:56 PM Subject: [Asterisk-Users] version 1.2 with chan_bluetooth Any body gotten this to compile chan_bluetooth under 1.2? WHat steps did you take. You only need to change version number in the Asterisk tree. include/asterisk/version.h change #define ASTERISK_VERSION_NUM 00 in #define ASTERISK_VERSION_NUM 010200 ... then it will compile ok. Anyway, I cannot make it work with an Ericsson or SonyEricsson phone,... Best regards, Dan ___ --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] IP Phones
wath ip hardware phones are the recomended to work with asterisk? or any phone work fine? __ Visita http://www.tutopia.com y comienza a navegar más rápido en Internet. Tutopia es Internet para todos. ___ --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] Polycom DTMF after connection not working
On a polycom 600 which is working perfectly otherwise, I am unable to use DTMF with IVR or such - not even to dialout of a Sipura setup elsewhere. Other phones (analogue connected to ATA) are accepted. I suspectthe phone is not using rfc2833 but I don't know how to specify that it should useit (not available on the http configuration). ___ --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] voipbuster
On Fri, 2005-12-02 at 10:24 +0100, Francesco Peeters wrote: [snip] (The 0031x are set up in this manner to avoid Cellphone (0031-6.) and Premium (0031-8. 0031-9.) numbers.) Afaik 0031-8. are freephone numbers, not premium. Regards, Patrick ___ --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] DTMF is choppy on the receive
I'm currently using X-Ten as a softphone and I've been having issues with dialing into IVR's. It seems that my DTMF passes in chirps and not clear tones. Any solutions? ___ --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] ISDN card Sirrix.PCI4S0
I have bout ISDN 4-port BRI Sirrix.PCI4S0 card and I'm unable to make it work. I have followed instructions that you get with drivers but I'm unable to start Asterisk ([chan_sirrix.so] Program exited with code 01. Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe). This is what I have done so far. I have installed Fedora Core 4 (Kernel 2.6.11) with web server, MySQL database and Development tools. Then I have installed esnacc compiler. Then I have installed Zaptel, Asterisk, Asterisk-sounds and Asterisk-addons. I have copied my configuration files to /etc/asterisk/. Then I have tried to install Sirrix PCI drivers. To make dev I needed to delete 7155 line in sirrix-pci/asterisk/chan_sirrix.c (support staff told me that - It's needed for FC4). Everything has been done without any error. Now, when I try to start asterisk I get following error. [chan_sirrix.so] Program exited with code 01. Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe Have I done something wrong? Seams that support staff from Sirrix (which have been weary useful till now) doesn't know how to solve this one. I would appreciate any hint or help. Thank you for your time. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)393447 e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Re: Asterisk fax
On Sat, Nov 26, 2005 at 10:37:44AM -0500, Tom Rymes wrote: More specifically, you can make it work using an ATA or a TDM400P card with an fxs port, but it is not likely to be reliable. If you TDM400P (FXS) and some ISDN quad bri Card works fine for approximately 10 months in a small company. No VoIP involved. Threre is no guaranty, but it seems to be reliable enough. If the ISDN card has one port in NT mode you could / should use some a/b converter connected to this port. -- Stefan Tichy [EMAIL PROTECTED] ___ --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] voipbuster
On Fri, December 2, 2005 15:08, Patrick said: On Fri, 2005-12-02 at 10:24 +0100, Francesco Peeters wrote: [snip] (The 0031x are set up in this manner to avoid Cellphone (0031-6.) and Premium (0031-8. 0031-9.) numbers.) Afaik 0031-8. are freephone numbers, not premium. Regards, Patrick You're right from a user's point of view, of course, but because I have had issues with dialing 0800 numbers before, I am keeping them out of the VoipBuster range... :-) From a provider's POV even 0800 numbers are premium numbers, because the owner of that number pays extra to use them. -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --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] what is your echo solution
Hi Just wandering what solution worked to eliminate echo on your setup. I am trying every solutions I can find on the wiki and none is working perfectly. We have asterisk 1.2.0 3 x digium TDM400P 30 Snom320 + 5 Snom360 For now the best setup I have is using Mark2 Echo cancel. Thanks Patrick ___ --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] [EMAIL PROTECTED] isdn
On Fri, December 2, 2005 14:00, Alejandro Vargas said: 2005/11/30, Francesco Peeters [EMAIL PROTECTED]: Yep, tried APIC, NOAPIC, ACPI=OFF, etc. (capitals only for clarity!) but to no avail! As soon as both share the same IRQ, the zaphfc driver stops passing data to asterisk... It is supposed to when you are using APIC, you should obtain many interrupts and the devices will obtain new interrupts with values grater than 16 which is the maxymun in a normal PC without APIC. Chech this lspc, you will see: smbus in irq17 sound card in irq18 usb controllers in irq 20, 21 and 23 4 ethernet cards, in irqs 16, 18 and 18 (it seems one of them are sharing irq with other) Your problem could be the isdn cards are not apic compatible. It's most likely the MoBo that is at fault... It *is* one of the earlier MoBo's with APIC capabilities, and IIRC the damn' thing has shared IRQ lines for the two bottom PCI slots! :-o Many people use older (surplus) hardware for their * server, and it is in such cases that one can run in to things like above madness. It's just a general warning that may or may not apply to your situation. At least you cannot say I didn't warn you ! ;-p -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --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] IAX Configuration
2005/12/2, Ishanka Anuradha Ranasooriya [EMAIL PROTECTED]: Hi all, I have configured two asterisk Boxes.Then I need to communicate these asterisk boxes via the IAX.It is better if you can help me to configure two boxes to communicate via asterisk I've found the answer: http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers -- Alejandro Vargas ___ --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] what is your echo solution
Patrick Fortin wrote: Just wandering what solution worked to eliminate echo on your setup. I am trying every solutions I can find on the wiki and none is working perfectly. I have been (since 1 1/2 weeks) using the ECHO_CAN_MG2. We have got a different setup (quadBRI, 12 GXP-2000's), and untill now it seems to be much better. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queue and agent transfer
Hello, I have a queue for incoming calls with some agents (defined as Agents) using iax2 softphone. I would like to use the Attended transfer (ATXFER) feature, however app_queue cannot handle it (I guess because it is not a channel). For this reason I put a Local channel in between with /n option. This way ATXFER works perfectly. The problem is that when the call has been transferred, the originally dialed agent does not get freed up - this is from some point of view logical. So the queue sees the agent as engaged/busy, while he/she is not anymore. Does anybody have any idea how to free up the not used agent? (transferer) Thank you in advance! Kind regards, Tamas ___ --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 Problem
I thing u cant use ChanIsAvail with exec command
... as use EXEC DIAL(SIP/40) .. it isnt work
2005/12/2, Cyrille Demaret [EMAIL PROTECTED]:
Hi,
I've changed that and it's the same problem. I've this problem with all
applications. Results from agi are not correct.
Regards,
Cyrille
-Message d'origine-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Giovanni
Miano
Envoyé: vendredi 2 décembre 2005 12:52
À: Asterisk Users Mailing List - Non-Commercial Discussion
Objet: Re: [Asterisk-Users] AGI Problem
Try print EXEC ChanIsAvail IAX2/24\n;
Channel type is IAX2 not IAX
Cheers
2005/12/2, Cyrille Demaret [EMAIL PROTECTED]:
Hi,
I'm running the last CVS asterisk version (I was running an older version
before with the same problem) and I've a problem with agi scripts.
Commands
results are not always correct.
I've made a small agi test script that execute ChanIsAvail on an
inexistent
extension:
#!/usr/bin/perl
$|=1;
while(STDIN) {
chomp;
last unless length($_);
if (/^agi_(\w+)\:\s+(.*)$/) {
$AGI{$1} = $2;
}
}
# Check
print EXEC ChanIsAvail IAX/24\n;
$result = STDIN;
print VERBOSE \$result\ 0\n;
# Check
print EXEC ChanIsAvail IAX/24\n;
$result = STDIN;
print VERBOSE \$result\ 0\n;
# Check
print EXEC ChanIsAvail IAX/24\n;
$result = STDIN;
print VERBOSE \$result\ 0\n;
Result is :
-- Executing DeadAGI(SIP/200-60d2, b) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/b
-- AGI Script Executing Application: (ChanIsAvail) Options: (IAX/24)
Dec 2 10:29:37 WARNING[15776]: channel.c:2520 ast_request: No channel
type
registered for 'IAX'
b: 200 result=-1
-- AGI Script Executing Application: (ChanIsAvail) Options: (IAX/24)
Dec 2 10:29:37 WARNING[15776]: channel.c:2520 ast_request: No channel
type
registered for 'IAX'
b: 200 result=1
-- AGI Script Executing Application: (ChanIsAvail) Options: (IAX/24)
Dec 2 10:29:37 WARNING[15776]: channel.c:2520 ast_request: No channel
type
registered for 'IAX'
b: 510 Invalid or unknown command
-- AGI Script b completed, returning 0
The first result is ok (-1) but not the second and the third.
Why do I get different results for the same command?
Thank you,
Regards,
Cyrille
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--
Giovanni Miano
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Re: [Asterisk-Users] Queue and agent transfer
This must be similar to a problem I have seen here. Some times the main operator's phone will stop ringing when a call comes in on the queue while the other phones still ring. I have to reset her phone which causes a re-login to get it working again. It must stop after she does an attended transfer. The normal blind transfers are what she does mostly. - James Tamas wrote: Hello, I have a queue for incoming calls with some agents (defined as Agents) using iax2 softphone. I would like to use the Attended transfer (ATXFER) feature, however app_queue cannot handle it (I guess because it is not a channel). For this reason I put a Local channel in between with /n option. This way ATXFER works perfectly. The problem is that when the call has been transferred, the originally dialed agent does not get freed up - this is from some point of view logical. So the queue sees the agent as engaged/busy, while he/she is not anymore. Does anybody have any idea how to free up the not used agent? (transferer) Thank you in advance! Kind regards, Tamas ___ --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] Queue and agent transfer
One simple way to overcome this problem would do to make an attended transfer to check whether the receiving person is available and willing to take the call, and then an unattended transfer to discharge the operator of the call. l. On Fri, 02 Dec 2005 16:21:39 +0100, James Armstrong [EMAIL PROTECTED] wrote: This must be similar to a problem I have seen here. Some times the main operator's phone will stop ringing when a call comes in on the queue while the other phones still ring. I have to reset her phone which causes a re-login to get it working again. It must stop after she does an attended transfer. The normal blind transfers are what she does mostly. - James Tamas wrote: Hello, I have a queue for incoming calls with some agents (defined as Agents) using iax2 softphone. I would like to use the Attended transfer (ATXFER) feature, however app_queue cannot handle it (I guess because it is not a channel). For this reason I put a Local channel in between with /n option. This way ATXFER works perfectly. The problem is that when the call has been transferred, the originally dialed agent does not get freed up - this is from some point of view logical. So the queue sees the agent as engaged/busy, while he/she is not anymore. Does anybody have any idea how to free up the not used agent? (transferer) Thank you in advance! Kind regards, Tamas ___ --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 -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --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] Linksys SPA-941 Admin Guide
we should be getting a limited number in a couple of weeks time. Proper stocks will be arriving in January - www.provu.com Paul. Senad Jordanovic wrote: [EMAIL PROTECTED] wrote: There is a review on the homepage at http://voipspeak.net It has been available for a few weeks, it is much nicer than the 841! Who has it for sale in UK? ___ --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] sip invite timeouts
Is there a way in asterisk to configure a sip invite timeout ? It seems to be about 30 seconds right now which is too long. I would like to have asterisk return congestion if a host does not respond to an invite within 5 seconds. ___ --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] Re: Asterisk Users Newsgroup
I am using http://www.gmane.com/ with my newsreader. You still have to be a list member to post. You can then turn on the vacation option in the list manager to stop receiving emails. -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- Tomislav Parèina [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Are there any Asterisk Users Newsgroup? For me it's much easier to follow newsgroup then to read all e-mails. Especially with news readers with so many features. So, if anybody knows for any newsgroup that has big discussions about Asterisk, please let me know. -- Tomislav Parèina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)393447 e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] sip invite timeouts
Matthew Simpson wrote: Is there a way in asterisk to configure a sip invite timeout ? It seems to be about 30 seconds right now which is too long. I would like to have asterisk return congestion if a host does not respond to an invite within 5 seconds. Asterisk 1.2 will use a T1 timer (retransmit) based on the 'qualify' time, if have that turned on for the peer. The INVITE will be transmitted a total of six times (per the RFC, IIRC). If your peer is _close_ and responds quickly to qualify packets, then the total INVITE timeout could be a second or two at most. There is an open feature request to make the T1 timer adjustable on a per-peer basis, but nobody has had the time to implement it yet. ___ --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] Originate calls but can't receive them on a SIP trunk
Hi list, I have a problem with a SIP trunk on my * box: I can originate calls but I cant receive them. The * box is behind a modem-router and as a private address. I think about a NAT problem but I dont know how to resolve it. I have included some debug and configuration. The trunk is registered as shown by sip show registry command: Host Username Refresh State sip.myprovider.fr:5060 08704412XX 105 Registered It appears UNREACHABLE when I execute sip show peers Name/username Host Dyn Nat ACL Mask Port Status cmm_sip/0870441 213.186.61.81 N 255.255.255.255 5060 UNREACHABLE I have configured sip_additionnal.conf with parameters send by my provider : register=08704412XX:[EMAIL PROTECTED] myprovider.fr/08704412XX [cmm_sip] type=friend username=08704412XX secret=5496 context=default host=sip. myprovider.fr permit=sip. myprovider.fr qualify=yes disallow=all allow=g729 I also include a copy of debug sip peers : Sip read: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 193.252.35.74:5060;branch=z9hG4bK14ed2d36 From: Unknown sip:[EMAIL PROTECTED];tag=as740c103b To: sip:213.186.61.81;tag=as051709c0 Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: sip:213.186.61.81 Accept: application/sdp Content-Length: 0 11 headers, 0 lines Destroying call '[EMAIL PROTECTED]' Retransmitting #3 (no NAT): OPTIONS sip:213.186.61.81 SIP/2.0 Via: SIP/2.0/UDP 192.168.8.251:5060;branch=z9hG4bK14ed2d36 From: Unknown sip:[EMAIL PROTECTED];tag=as740c103b To: sip:213.186.61.81 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Fri, 02 Dec 2005 16:17:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 ___ --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] dial-out and variable inheritance problems
Hello,
extensions.conf:
[mytest-in]
exten = 1,1,NoOp(${MYVAR1})
exten = 1,n,Wait(20)
exten = 1,n,Hangup()
[mytest-out]
exten = 1,1,NoOp(${MYVAR1})
exten = 1,n,Dial(Zap/g1/06111,10,H|g)
my test dial.out file:
Channel: Local/[EMAIL PROTECTED]
MaxRetries: 0
Context: mytest-in
Extension: 1
Priority: 1
Set: __MYVAR1=hello
The result:
-- Attempting call on Local/[EMAIL PROTECTED] for [EMAIL PROTECTED]:1
(Retry 1)
-- Executing NoOp(Local/[EMAIL PROTECTED],2, hello) in new stack
-- Executing Dial(Local/[EMAIL PROTECTED],2,
Zap/g1/06111|10|H|g) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/06111
-- Zap/1-1 is proceeding passing it to Local/[EMAIL PROTECTED],2
Dec 2 17:28:18 NOTICE[13079]: channel.c:2412 __ast_request_and_dial:
Don't know what to do with control frame 15
-- Zap/1-1 answered Local/[EMAIL PROTECTED],2
Channel Local/[EMAIL PROTECTED],1 was answered.
-- Executing NoOp(Local/[EMAIL PROTECTED],1, ) in new stack
-- Executing Wait(Local/[EMAIL PROTECTED],1, 20) in new stack
== Spawn extension (mytest-out, 1, 2) exited non-zero on
'Local/[EMAIL PROTECTED],2'
As you can see, the MYVAR1 variable did not inherit, which breaks my
dial-out application. This way it worked well for a long time, however
an upgrade to recent HEAD version broke things. Unfortunately I don't
know which version worked fine, I didn't use this feature for a long
time and just now I upgraded...
Anybody else experiencing this problem?
Regards,
Tamas
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Re: [Asterisk-Users] WG: App_rxfax problem
Ouch ... error while writing audio data: : Broken pipe If you are talking about the Ouch message, yes lots of people have seen the error and its usually the result of some misconfiguration in one of your files (likely zapata.conf). Correct me if I'm wrong, but isn't that message from mpg123 itself? It appears in the binary (via strings), and I've seen it at non-asterisk times too. AFAIK it comes up whenever the parent application (asterisk in this case) quits without closing it properly (hence, broken pipe). As such, this means that the above error simply shows that asterisk crashed (which they presumably already knew), and has nothing to do with the problem itself... The above is one case, however there are lots of other cases where asterisk is still running, the Ouch messages continuously scrolls, and killing asterisk with a -9 becomes necessary. Try misconfiguring a PRI and see what you get. ;) Rich ___ --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] Meetme option 'b'
Hi;
I've been looking for an arbitrary way of discovering when the last
user has left a Meetme conference...
It occurred to me that I could launch an agi script to keep watch over
the conference and do something when the user count reaches zero... And
of course, I can do that directly from the dialplan.
But I was looking at app_meetme, and the docs say:
* 'b' — run AGI script specified in ${MEETME_AGI_BACKGROUND}
o Default: conf-background.agi (Note: This does not work
with non-Zap channels in the same conference)
I can't see anything in the code to explain this; does anyone understand
why it might be ?
jd
--
John Daragon [EMAIL PROTECTED]
argv[0] limited
Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK
v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127
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Re: [Asterisk-Users] what is your echo solution
Just wandering what solution worked to eliminate echo on your setup. I am trying every solutions I can find on the wiki and none is working perfectly. We have asterisk 1.2.0 3 x digium TDM400P 30 Snom320 + 5 Snom360 For now the best setup I have is using Mark2 Echo cancel. I'm running cvs-head from about a week ago with all stock parameters including echo can. Zapata.conf looks like: context=inbound-bus-dialin signalling=fxs_ks group=1 echocancel=yes echotraining=800 echocancelwhenbridged=yes usecallerid=no hidecallerid=no callwaiting=no callwaitingcallerid=no threewaycalling=no rxgain=5 txgain=0 I'm about 7db from the central office, so the gains have been tweeked to work with those pstn losses. It _seems_ like folks with snom phones and a TDM card have more issues with echo then others. Could be wrong though. I'm using Cisco 7960's. ___ --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] Re: Asterisk-Users Digest, Vol 17, Issue 11
Hello AllI'm bought VoiceTronix Card (Openswitch), it's bad card and resaller (www.telephonyware.com) give me are old card (for one year old). than, after that, my card is fault. I didn't received any help from telephoneware or voicetronix. I don't like voicetronix and telephoneware. i notice to all people in room to know voicetronix's services.Please don't pay anything cards from telephonyware and voicetronixBest regards Yahoo! Personals Single? There's someone we'd like you to meet. Lots of someones, actually. Try Yahoo! Personals___ --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] Sangoma Asterisk at home
Guys, I'm curious if it's possible to asterisk at home and the sangoma T1 cards together. I realize asteriskathome is traditionally used for at home, but I'd like to use it in a small office with a T1 and our hardware is a Sangoma card. I know all I need to do to get the sangoma working is recompile the zaptel but I can't seem to find the source, etc on the server after asteriskathome installs. Jess ___ --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] Sangoma Asterisk at home
Jess Coburn wrote: Guys, I'm curious if it's possible to asterisk at home and the sangoma T1 cards together. I realize asteriskathome is traditionally used for at home, but I'd like to use it in a small office with a T1 and our hardware is a Sangoma card. I know all I need to do to get the sangoma working is recompile the zaptel but I can't seem to find the source, etc on the server after asteriskathome installs. Jess ___ --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 I use the regular asterisk at home.. with a tdm card and a sangoma card... we have an small home business and we use a manager we develop for it.. excelents results www.cripiland.com/screenshots/manager3.jpg ___ --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] Sangoma Asterisk at home
Thanks Saul, What you do to get the Sangoma to install and how'd you go about compiling the zaptel source did you just download zaptel and extra RPMs? I'm by no means a linux guru... Jess On 12/2/05, Saul Diaz [EMAIL PROTECTED] wrote: Jess Coburn wrote: Guys, I'm curious if it's possible to asterisk at home and the sangoma T1 cards together. I realize asteriskathome is traditionally used for at home, but I'd like to use it in a small office with a T1 and our hardware is a Sangoma card. I know all I need to do to get the sangoma working is recompile the zaptel but I can't seem to find the source, etc on the server after asteriskathome installs. Jess ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usersI use the regular asterisk at home.. with a tdm card and a sangoma card... we have an small home business and we use a manager we developfor it.. excelents resultswww.cripiland.com/screenshots/manager3.jpg ___--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] Sangoma Asterisk at home
@home by no means means it just for the home - its Asterisk nothing more, nothing less. I don't think the @home designation was meant to limit it by perception. I read somewhere it was called @home for another reason, anyone know more? RegardsRobOn 12/2/05, Jess Coburn [EMAIL PROTECTED] wrote: Thanks Saul, What you do to get the Sangoma to install and how'd you go about compiling the zaptel source did you just download zaptel and extra RPMs? I'm by no means a linux guru... Jess On 12/2/05, Saul Diaz [EMAIL PROTECTED] wrote: Jess Coburn wrote: Guys, I'm curious if it's possible to asterisk at home and the sangoma T1 cards together. I realize asteriskathome is traditionally used for at home, but I'd like to use it in a small office with a T1 and our hardware is a Sangoma card. I know all I need to do to get the sangoma working is recompile the zaptel but I can't seem to find the source, etc on the server after asteriskathome installs. Jess ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I use the regular asterisk at home.. with a tdm card and a sangoma card... we have an small home business and we use a manager we developfor it.. excelents results www.cripiland.com/screenshots/manager3.jpg ___--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 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
[Asterisk-Users] Music on Hold Error
Title: Music on Hold Error Can anyone help with; Dec 2 12:11:56 WARNING[2562]: res_musiconhold.c:421 spawn_mp3: Found no files in '/var/lib/asterisk/mohmp3' Dec 2 12:11:56 WARNING[2562]: res_musiconhold.c:488 monmp3thread: Unable to spawn mp3player Dec 2 12:20:16 WARNING[2562]: res_musiconhold.c:421 spawn_mp3: Found no files in '/var/lib/asterisk/mohmp3' Dec 2 12:20:16 WARNING[2562]: res_musiconhold.c:488 monmp3thread: Unable to spawn mp3player Dec 2 12:28:36 WARNING[2562]: res_musiconhold.c:421 spawn_mp3: Found no files in '/var/lib/asterisk/mohmp3' Dec 2 12:28:36 WARNING[2562]: res_musiconhold.c:488 monmp3thread: Unable to spawn mp3player David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net * PLEASE NOTE THAT EFFECTIVE DEC 1,2005 MY TELEPHONE NUMBER WILL CHANGE * NEW !!! Tel: (519) 963-3020 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --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] Re: Meetme option 'b'
In article [EMAIL PROTECTED], John Daragon [EMAIL PROTECTED] wrote:
Hi;
I've been looking for an arbitrary way of discovering when the last
user has left a Meetme conference...
It occurred to me that I could launch an agi script to keep watch over
the conference and do something when the user count reaches zero... And
of course, I can do that directly from the dialplan.
But I was looking at app_meetme, and the docs say:
* 'b' run AGI script specified in ${MEETME_AGI_BACKGROUND}
o Default: conf-background.agi (Note: This does not work
with non-Zap channels in the same conference)
I can't see anything in the code to explain this; does anyone understand
why it might be ?
To explain which part? That it doesn't work with non-Zap channels?
For Zap channels, the mixing is automatically done at the driver level
once MeetMe has told the driver which channels to mix.
For a non-Zap channel, a proxy Zap channel (pseudo) is created to
participate in the driver-level mix. The meetme thread on the channel
then enters a loop to copy audio back and forth between the non-Zap
channel and the proxy pseudo-channel.
When an AGI background script is specified, it runs INSTEAD OF the
copying loop mentioned above. Therefore there is nothing to move the
audio to and from the non-Zap channel.
Hope this helps!
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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RE: [Asterisk-Users] what is your echo solution
Before 1.2.0 I used Mark2 with AGGRESSIVE turned on I would recommend switching to KB or MG in 1.2.0, we have done this with very good results (using KB now) Jared Armstrong -Original Message- From: Patrick Fortin [mailto:[EMAIL PROTECTED] Sent: Friday, December 02, 2005 9:17 AM To: asterisk-users-lists.digium.com Subject: [Asterisk-Users] what is your echo solution Hi Just wandering what solution worked to eliminate echo on your setup. I am trying every solutions I can find on the wiki and none is working perfectly. We have asterisk 1.2.0 3 x digium TDM400P 30 Snom320 + 5 Snom360 For now the best setup I have is using Mark2 Echo cancel. Thanks Patrick ___ --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] Music on Hold Error
Dave Morrow wrote: Can anyone help with; Dec 2 12:11:56 WARNING[2562]: res_musiconhold.c:421 spawn_mp3: Found no files in '/var/lib/asterisk/mohmp3' Dec 2 12:11:56 WARNING[2562]: res_musiconhold.c:488 monmp3thread: Unable to spawn mp3player Dec 2 12:20:16 WARNING[2562]: res_musiconhold.c:421 spawn_mp3: Found no files in '/var/lib/asterisk/mohmp3' Dec 2 12:20:16 WARNING[2562]: res_musiconhold.c:488 monmp3thread: Unable to spawn mp3player Dec 2 12:28:36 WARNING[2562]: res_musiconhold.c:421 spawn_mp3: Found no files in '/var/lib/asterisk/mohmp3' Dec 2 12:28:36 WARNING[2562]: res_musiconhold.c:488 monmp3thread: Unable to spawn mp3player s Unfortunately, you're the only one who can fix this. Put some mp3 files in the directory that is mentioned in the error messages. D -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.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
[Asterisk-Users] v1.2 and cdr badly written
Has anyone encountered 'bad' cdr logging in * 1.2? Since upgrading to 1.2 (bristuffed) and asterisk-addons 1.2, sometimes the clid is 'messed' up. I use AMP to look at the reports, but when I look in the cdr database, it's the same, here's an example: 2/12/2005 15:06:02 Tech: ÀB ÀB 2 ext-group Zap/1-1 SIP/211-f379 Dial SIP/209SIP/200SIP/210SIP/211|30|tr 582 582 ANSWERED 3 asterisk-2559-1133532362.58 So, why would the numver be saved as ÀB ?? It's a correct number, but gets written incorrect.. another one: 2/12/2005 16:07:50 ÀB ÀB s aa_4 Zap/1-1 Hangup 3 2 ANSWERED 3 asterisk-2559-1133536070.98 Any suggestions on this? cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with a Company or Site for a DEMO. AYUDA con una empresa para una DEMO
(ENGLISH VERSION AT THE END) Hola lista: Requiero saber si alguien tiene un cliente o empresa donde se encuentren montado algun Asterisk como PBX de tamaño mediano (al menos unas 50 extensiones). Esto para dar una demostracion a un cliente mio que esta interesado en invertir en Asterisk. Les pido que todos los que conoscan algo asi, me contacten ya sea a este correo fuera de la lista o a mi correo arabe AT xmarts.com.mx Gracias. Hi List: I want to now if any one have a client or company with a asterisk with more or less 50 extensiones, This for a demostration that i need to give to a client who want to invert on Asterisk. I ask to all if have some one like this contact me out of the list, to this email or to arabe AT xmarts.com.mx 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
[Asterisk-Users] hint priority in AEL?
Hello, I am trying to convert my hint priorities from the old style: exten = 2130,hint,SIP/0146472130 to Asterisk Extension Language (AEL) style. I haven't found anything in the docs, wiki or examples about it. How should I do it? -- Sigs have been known to cause cancer in California. ___ --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] Re: Meetme option 'b'
Tony Mountifield wrote:
In article [EMAIL PROTECTED], John Daragon [EMAIL PROTECTED] wrote:
Hi;
I've been looking for an arbitrary way of discovering when the last
user has left a Meetme conference...
It occurred to me that I could launch an agi script to keep watch over
the conference and do something when the user count reaches zero... And
of course, I can do that directly from the dialplan.
But I was looking at app_meetme, and the docs say:
* 'b' — run AGI script specified in ${MEETME_AGI_BACKGROUND}
o Default: conf-background.agi (Note: This does not work
with non-Zap channels in the same conference)
I can't see anything in the code to explain this; does anyone understand
why it might be ?
To explain which part? That it doesn't work with non-Zap channels?
For Zap channels, the mixing is automatically done at the driver level
once MeetMe has told the driver which channels to mix.
For a non-Zap channel, a proxy Zap channel (pseudo) is created to
participate in the driver-level mix. The meetme thread on the channel
then enters a loop to copy audio back and forth between the non-Zap
channel and the proxy pseudo-channel.
When an AGI background script is specified, it runs INSTEAD OF the
copying loop mentioned above. Therefore there is nothing to move the
audio to and from the non-Zap channel.
Hope this helps!
It does, indeed ! Thanks for the succinct explanation.
I owe you a beer.
jd
--
John Daragon [EMAIL PROTECTED]
argv[0] limited
Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK
v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127
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Re: [Asterisk-Users] WG: App_rxfax problem
How could a CVS update fix an error you have made during installation? Steve René Enskat [Teamware GmbH] wrote: so is there a solution in the next cvs udpate? *Von:* René Enskat [Teamware GmbH] [mailto:[EMAIL PROTECTED] *Gesendet:* Donnerstag, 1. Dezember 2005 14:47 *An:* 'asterisk-users@lists.digium.com' *Betreff:* WG: App_rxfax problem I just sent the error in full log: Dec 1 15:01:08 VERBOSE[27950] logger.c: [app_rxfax.so]Dec 1 15:01:08 WARNING[27950] loader.c: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handler Dec 1 15:01:08 WARNING[27950] loader.c: Loading module app_rxfax.so failed! *Von:* René Enskat [Teamware GmbH] [mailto:[EMAIL PROTECTED] *Gesendet:* Donnerstag, 1. Dezember 2005 08:35 *An:* 'asterisk-users@lists.digium.com' *Betreff:* App_rxfax problem When i load the fax modules into the asterisk i got this errors but compile was ok! I have the latest cvs head [res_musiconhold.so] = (Music On Hold Resource) == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' == Registered application 'StartMusicOnHold' == Registered application 'StopMusicOnHold' [app_rxfax.so]Warning, flexibel rate not heavily tested! Warning, flexibel rate not heavily tested! Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe ___ --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] Is a BUG ? Hints and incominglimit
Could you send it patch please. On 11/30/05, Paradise Dove [EMAIL PROTECTED] wrote: btw, i've patched this part of code and now its working fine for me.i'm going to upload it.Paradise Dove On 11/30/05, Kevin Hanson [EMAIL PROTECTED] wrote: Paradise Dove wrote: Yes with version 1.2. I have tried already with call-limit and the same. i agree with you, it seems to be a bug which i've submited before (bug #5281) but it's now closed by bug marshals!It's not closed.It's suspended waiting input from you: Closing until the appropriate debug/trace output can be provided. On 10/30 you said you were still trying to get the debug output. Cheers, Kevin ___ --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] hint priority in AEL?
Louis-David Mitterrand wrote: to Asterisk Extension Language (AEL) style. I haven't found anything in the docs, wiki or examples about it. I don't believe hints are supported in AEL at this time. ___ --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] MeetMe with the V (video) option
Dean Collins wrote: who's done it? and how much money are they talking about? I've been looking to pay for something like that for a while. -Original Message- From: Neil Stratford [mailto:[EMAIL PROTECTED] Sent: 24 November 2005 09:30 To: John Martin; [EMAIL PROTECTED] Subject: Re: Fwd: [Asterisk-Dev] Chan_sip: video capabilities, call bandwidth and RTCP Hi John/Matt, I am replying to your emails to the Asterisk-Dev mailing list concerning Asterisk and video support. I agree with many of the points that you both raised and would really like to see the support for video in Asterisk improved - it does work, but there are limitations today and I would like to see Asterisk leading the way. (My background is academic/commercial research in the area of multimedia QoS.) I would like to make you aware of some projects that I have been working on, and to ask if you would be interested in helping to fund any future development to see these projects to completion or to start new projects in this area. While I don't think they are ratified, most video UAs support the draft RFCs: - levin-mmusic-xml-media-control-02 - INFO fast updates and Asterisk *should* already support this RFC - I implemented it earlier this year and it is in CVS, and in 1.2. Unfortunately I have just noticed that a minor typo was introduced into the XML when it was integrated into CVS, so it doesn't currently work - which is why you probably didn't realize it existed. It is a single character typo and I'll be feeding the patch into the bug tracker. (from later in your email) - Other fast update mechanisms (the H.261 RTP FU for instance) I also implemented this, but it didn't make it into CVS. If you think it is still important I can revive that code. Multiparty Video Conferencing: I have 95% of a working solution for multi-party video conferencing in Asterisk (based on app_conference). You can test the current version by calling [EMAIL PROTECTED] It currently allows for up to 10 callers per conference, and switches the displayed video when you send DTMF - press 1 for caller 1 etc. (This is running on a test server - if it is not up or has an error, please let me know.) I am currently looking for additional funding to complete this work and enable me to recover some of the development costs so that we can release it as open source. Asterisk h263 file format generation: I have written a couple of modules for GStreamer (www.gstreamer.net) which allow it to generate Asterisk format h263 files. With a GStreamer command line you can now convert video files from other formats into h263 and wav files for playback using Asterisk. I believe that the modules are now in the latest CVS version of GStreamer, but if you would like patches to 0.9 let me know. RTSP Streaming integration: This is a new project which I may have funding to complete already, but if you are interested we may be able to accelerate development. H324m/SIP gateway: This is something that many people are interested in, but there has been little progress. I would really like to drive this project forward. If you are interested in any of these projects, or are looking for any other development work (or collaboration), please do not hesitate to contact me. We are based just outside of Cambridge in the UK. Thanks Neil Stratford -- Neil Stratford | Vipadia Limited | +44 1223 858 111 | sip:[EMAIL PROTECTED] | www.vipadia.com -- 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
Re: [Asterisk-Users] Hint: how to include dialplan files from remote systems
On Thu, Dec 01, 2005 at 06:51:54PM -0800, John Todd wrote: #exec /usr/bin/curl -s http://webserver.domain.com/privatefiles/username-to-numbers /etc/asterisk/username-to-numbers #include username-to-numbers Nice. However, what happens if curl takes longer than expected? your reload waits for it. And what if you get a broken copy? I figure you should generally fetch to a temporary file and only replace the working copy if the download was successful. Something like (untested): filename=`mktemp` destination=/etc/asterisk/username-to-numbers wget -q -O$filename cp $filename $destination rm $filename This still won't report errors up, and won't do any single sanity check, but you get my point. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend Yes, I didn't put extensive error-checking into the example so other curl options like MaxTime and ConnectTimeout were not used. Some fiddling with curl's options can start to significantly limit the threats due to timeout/delay errors. As for corrupted files, there is a whole different parsing requirement there that I think is outside of an easy scope. Another whole shell script/perl program would probably be required to check the integrity of the file to ensure it was proper Asterisk format. Bah - too much work. :-) Additionally, you use wget in your follow-up code. I originally used wget in my example, but hit a wall when testing it. For some inexplicable reason, it is the case that on RH9 and Astlinux installations that wget and Asterisk #exec don't mix. It simply doesn't work, even when called from inside a shell script. It doesn't make any sense, but that is what happened. I even had a bug open on it - http://bugs.digium.com/view.php?id=5833 I have _no_ idea why this is the case, but I'm assuming it has something to do with environment values. Use curl with #exec to fetch files, or if someone can figure out why wget doesn't work I'd be interested in hearing about it. JT ___ --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] DIAXY to DIAXY problems
Hi list: I'm having problem with some DIAXY ATA FROM DIGIUM, I have 3 of them in different points, all of them register to a central asterisk server. If i call from any of the ATA's to Asterisk or Asterisk's to ATAs. But when any ATA's want to talk to another ATA's.. TheATA's rings, but when the call is establish it hangups...And at the CLI i don'thave any error. Any Idea ??? 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] hint priority in AEL?
On Fri, Dec 02, 2005 at 12:05:08PM -0600, Kevin P. Fleming wrote: Louis-David Mitterrand wrote: to Asterisk Extension Language (AEL) style. I haven't found anything in the docs, wiki or examples about it. I don't believe hints are supported in AEL at this time. Thanks for the heads-up. -- I had no wish to arrive, but I had to do my utmost, in order to arrive. -- Samuel Beckett, The Unnamable ___ --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] ael questions
I was experimenting with ael and first thing I tried to do was move the inclusions for the default context form the extensions.conf file to the extensions.ael file Can a context that is defined in extensions.conf be included by the ael parser? Just asking in case anyone has already discovered this. There is a page for ael started on the wiki and I thought I might update it once I am sure. I am going to convert one test server entirely to ael just for the sake of learning and testing. I was also wondering about ael/realtime interaction. If such a thing exists and anybody has implemented it please let me know. That is another answer I can add to the wiki page. ___ --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] Kernel upgrade causes ztdummy to refuse to run
Hello all, I recently upgraded the kernel on one of the phone servers I have at home (dual Xeon 2.4) from 2.6.11 to 2.6.14 in the usual way, copying the .config file across and building the new kernel. Now ztdummy is refusing to run, and gives the following errors in dmesg: ztdummy: Unknown symbol rtc_register ztdummy: Unknown symbol rtc_unregister ztdummy: Unknown symbol rtc_control Which I assume to mean something about the RTC config has changed. The only options I can find relating to this in the kernel config are: Enhanced Real Time Clock Support Generic /dev/rtc emulation Enabling Enhanced RTC Support, rebuilding the kernel and restarting allows ztdummy to load, but dmesg is loaded with messages about rtc: lost some interrupts at 1024Hz. Thousands of 'em every few seconds. I also notice asterisk spiking between 5 and 25% CPU even with no calls or other activity going on. The second option (generic /dev/rtc) doesn't seem to affect the ztdummy error at all. I know ztdummy definitely worked fine with 2.6.11 and I wasn't getting rtc errors in the logs. Nothing has changed on the hardware. Any suggestions gratefully appreciated. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip invite timeouts
At 10:16 AM -0600 12/2/05, Kevin P. Fleming wrote: Matthew Simpson wrote: Is there a way in asterisk to configure a sip invite timeout ? It seems to be about 30 seconds right now which is too long. I would like to have asterisk return congestion if a host does not respond to an invite within 5 seconds. Asterisk 1.2 will use a T1 timer (retransmit) based on the 'qualify' time, if have that turned on for the peer. The INVITE will be transmitted a total of six times (per the RFC, IIRC). If your peer is _close_ and responds quickly to qualify packets, then the total INVITE timeout could be a second or two at most. There is an open feature request to make the T1 timer adjustable on a per-peer basis, but nobody has had the time to implement it yet. I'll throw my $.02 in here, since this has recently bitten me but in the opposite direction, so it's worth putting up for people to find this data in the archives... We have connections between Asterisk servers and SER proxies that have qualify= enabled. These boxes sit right next to each other, so the RTT is sometimes less than 12ms. Using the qualify= results as T1, this means that the TOTAL time that an INVITE can exist is 768ms (Timer B = 64*T1) and retransmissions of INVITEs happen if the SER proxy does not respond with a 100 Trying or other valid response within 24ms (retransmit delay = 2*T1). Considering that there are databases, etc. are firing on my SER proxy, it takes sometimes quite a bit of time before an answer is generated for the actual dialing result. In worst-case scenarios (i.e.: ENUM on SER) I would get all six retransmits from Asterisk, and then the call would fail before the lookups were complete. Therefore, to fix the problem it is necessary to have SER respond with a 100 Trying response immediately. This is not a problem between two Asterisk servers, as Asterisk always sends a 100 Trying reply on an INVITE. Feh. SIP trying to be TCP. I'll be glad to see the eventual tune-ability of T1 and other timers on a per-peer basis. JT ___ --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
