Re: [asterisk-users] Replace music-on-hold on MeetMe with ringing sound
Date: Mon, 23 Jun 2008 08:00:08 -0400 From: David Backeberg [EMAIL PROTECTED] Subject: Re: [asterisk-users] Replace music-on-hold on MeetMe with ringing sound To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 MeetMe() provides very useful tones when a caller is added to a MeetMe room. That is, if you're using the musiconhold option, the agent would hear music, immediately followed by two tones, and then they would be bridged to the client. Perhaps you're running MeetMe() with those join tones disabled? Check out the docs for MeetMe. I think it's option capital i, as in Iberia. On Mon, Jun 23, 2008 at 12:19 AM, Cosmin Prund [EMAIL PROTECTED] wrote: Hello. It's been a while since I last posted (probably because my * works just fine). I'm working on something to replace call queues in my own application-specific way and I'm using MeetMe rooms to bridge agents and clients and do other things. When an agent needs to be bridged with a client I'll first put the agent in the MeetMe room and when I have confirmation that the agent is in the MeetMe room I'll send the client to the same room. My agent gets to hear music on hold while it's the only one in the conference room (it takes 1 or 2 seconds for the client to be put in the same room). Is it possible to make the agent here ringing (or replace the music on hold with a recording of ringing)? At the moment I'm telling agents when the music stops playing you're talking to the client but that just doesn't sound right and it's a bit fiddely because music on hold is music and music has pauses. One can imediatelly tell the ringing is done but they might need a few extra seconds to realise the music has stoped. On the other hand the client has no such problem since he/she hears ringing just before they get bridged to the MeetMe room. Any ideas? Thanks! -- Cosmin Prund I have the same problem and wish as the original poster, but couldn't find any information about this. The M option for meetme (play musiconhold) doesn't seem to have any switches to change MOH class. I've also checked voip-info.org variable list to see if there was any variable I could set before entering the meetme, but nothing there either from what I could find :( Does anyone have a solution to this, other than to replace the sound files in the default MOH directory? Thanks, Best regards, Tobias ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6, t.38 and zoiper - t38_udptl or t38pt_udptl ?
2008/12/10 [EMAIL PROTECTED] [EMAIL PROTECTED] This should be sufficient to get it to work from zoiper to zoiper. http://asteriskguru.org/tutorials/zoiper2zoiperfaxt38.html If you would still experience any issues, please send us a packet capture + a description of the setup. Fine ! I'll try it ASAP. Thanks Cheers and good luck! Zoa Olivier wrote: Hello, 2008/12/5 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] I will publish a tutorial in the beginning of next week about how to configure Zoiper and Asterisk to do t.38 together. Zoa. Where will you publish this tuto ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DIY IP hardphone reference design
Hi, I am interested in building my own DIY IP hardphone to connect to Asterisk for my personal usage. Does anyone know of any good reference design in guidance me on how to build one? I am capable of building it from even raw material or circuit design if I can get some info on how to start. Thanks for all your help. Regards, Mark ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID provider in Sweden
On Wed, 10 Dec 2008, Gideon Hack wrote: Hi Gordon, DID World Wide (see http://www.didww.com/virtual_numbers/Sweden) has the DIDs that you require. And they can forward to IAX if that is preferable to you. Thanks. I was actually hoping I'd find a Swedish company, but I'll pass this and the other on to my customer (who's in Sweden and wants to pay in Swedish money) Cheers, Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIY IP hardphone reference design
mark morreny wrote: Hi, I am interested in building my own DIY IP hardphone to connect to Asterisk for my personal usage. Does anyone know of any good reference design in guidance me on how to build one? I am capable of building it from even raw material or circuit design if I can get some info on how to start. There is a reference design application note, and an evaluation board, for every IP phone chip on the market. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WARNING[19164] chan_sip.c: Forbidden - wrong password on authentication for INVITE to 'unknown sip:[EMAIL PROTECTED]
Hi Aldo, sorry not having posted it. ::)) You have to set canreinvite=no. Giorgio Incantalupo Aldo Alexander Leyva Alvarado wrote: What parameter??? 2008/1/17 gincantalupo [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Hi Olle, that was a phone misconfigurationa parameter had a wrong value. The message has disappeared and now the phone seems to work! Thank you! Giorgio Johansson Olle E wrote: 10 jan 2008 kl. 16.48 skrev gincantalupo: Hi, I'm using an Asterisk 1.2.18 box with a remote Snom 360. My Snom always rings but sometimes (it happens randomly!) no voice is passing thru (2 ways). Asterisk CLI shows this warning: Jan 10 10:03:26 WARNING[19164] chan_sip.c: Forbidden - wrong password on authentication for INVITE to 'unknown sip:[EMAIL PROTECTED] I have already set localnet and externip parameters inside the general section of my sip.conf: localnet = 192.168.4.0/24 http://192.168.4.0/24 externip = xx.xx.xx.xxx Is there anybody who knows how to solve this problem? The error message has nothing to do with no voice is passing thru. The error message clearly indicates that you have bad credentials for an INVITE. In order for anyone to help you, you need to reveal more about the setup and the involved parties in the communication. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ Giorgio Incantalupo, mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID provider in Sweden
Hi Gordon, Take a look at http://www.cellip.com/ //Peter Gordon Henderson wrote: On Wed, 10 Dec 2008, Gideon Hack wrote: Hi Gordon, DID World Wide (see http://www.didww.com/virtual_numbers/Sweden) has the DIDs that you require. And they can forward to IAX if that is preferable to you. Thanks. I was actually hoping I'd find a Swedish company, but I'll pass this and the other on to my customer (who's in Sweden and wants to pay in Swedish money) Cheers, Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] a problem on Ubuntu with Asterisk
Have a nice day, Scott Berry E-mail: [EMAIL PROTECTED] I am studying out of the book Asterisk: The Future of Telephony on Chapter 4, and right now for practicing using the built in Debian version of Asterisk for Ubuntu. I am however having some problem where I cannot do asterisk -r and hook up to the asterisk CLI. I have checked to see that /var/run/asterisk/asterisk.ctl is available which it is. I have also set up the zaptel.conf, zapata.conf and also the extensions.conf as specified in the book. The error I get is: Unable to connect to asterisk remote (does /var/run/asterisk/asterisk.ctl exist? Yes it certainly does. Any help would be appreciated. if need be i would be happy to send my extensions.conf, zaptel.conf, and zapata.conf to the lisOne other question I think I am correct on this but not sure does zaptel.conf and zapata.conf go in to /etc? Thanks for all the help. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a problem on Ubuntu with Asterisk
Scott Berry wrote: Have a nice day, Scott Berry E-mail: [EMAIL PROTECTED] I am studying out of the book Asterisk: The Future of Telephony on Chapter 4, and right now for practicing using the built in Debian version of Asterisk for Ubuntu. I am however having some problem where I cannot do asterisk -r and hook up to the asterisk CLI. I have checked to see that /var/run/asterisk/asterisk.ctl is available which it is. I have also set up the zaptel.conf, zapata.conf and also the extensions.conf as specified in the book. The error I get is: Unable to connect to asterisk remote (does /var/run/asterisk/asterisk.ctl exist? Yes it certainly does. Any help would be appreciated. if need be i would be happy to send my extensions.conf, zaptel.conf, and zapata.conf to the lisOne other question I think I am correct on this but not sure does zaptel.conf and zapata.conf go in to /etc? Thanks for all the help. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Permissions problems? sudo asterisk -r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a problem on Ubuntu with Asterisk
You've checked that another asterisk is running (ps -ef|grep asterisk)? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Berry Sent: Wednesday, December 10, 2008 9:06 AM To: Asterisk Users Subject: [asterisk-users] a problem on Ubuntu with Asterisk Have a nice day, Scott Berry E-mail: [EMAIL PROTECTED] I am studying out of the book Asterisk: The Future of Telephony on Chapter 4, and right now for practicing using the built in Debian version of Asterisk for Ubuntu. I am however having some problem where I cannot do asterisk -r and hook up to the asterisk CLI. I have checked to see that /var/run/asterisk/asterisk.ctl is available which it is. I have also set up the zaptel.conf, zapata.conf and also the extensions.conf as specified in the book. The error I get is: Unable to connect to asterisk remote (does /var/run/asterisk/asterisk.ctl exist? Yes it certainly does. Any help would be appreciated. if need be i would be happy to send my extensions.conf, zaptel.conf, and zapata.conf to the lisOne other question I think I am correct on this but not sure does zaptel.conf and zapata.conf go in to /etc? Thanks for all the help. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIY IP hardphone reference design
On Wed, Dec 10, 2008 at 5:59 AM, mark morreny [EMAIL PROTECTED] wrote: Hi, I am interested in building my own DIY IP hardphone to connect to Asterisk for my personal usage. Does anyone know of any good reference design in guidance me on how to build one? I am capable of building it from even raw material or circuit design if I can get some info on how to start. Thanks for all your help. Regards, Mark Voted most odd question posted to the list. If it were me, I would start with a USB sound card (think Magicjack) -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a problem on Ubuntu with Asterisk
On Wed, Dec 10, 2008 at 09:05:34AM -0600, Scott Berry wrote: Have a nice day, Scott Berry E-mail: [EMAIL PROTECTED] I am studying out of the book Asterisk: The Future of Telephony on Chapter 4, and right now for practicing using the built in Debian version of Asterisk for Ubuntu. What version of Ubuntu? (and thus: what version of Asterisk) cat /etc/ubuntu_version # or something similar dpkg -l asterisk I am however having some problem where I cannot do asterisk -r and hook up to the asterisk CLI. I have checked to see that /var/run/asterisk/asterisk.ctl is available which it is. I have also set up the zaptel.conf, zapata.conf and also the extensions.conf as specified in the book. The error I get is: Unable to connect to asterisk remote (does /var/run/asterisk/asterisk.ctl exist? Yes it certainly does. Any help would be appreciated. if need be i would be happy to send my extensions.conf, zaptel.conf, and zapata.conf to the lisOne other question I think I am correct on this but not sure does zaptel.conf and zapata.conf go in to /etc? Is asterisk running? ps aux | grep asterisk If it isn't: what do you see in /var/log/asterisk/messages -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID provider in Sweden
On Wed, 10 Dec 2008, Peter Lindquist wrote: Hi Gordon, Take a look at http://www.cellip.com/ Ah! Thanks! I'll pass it on. Gordon //Peter Gordon Henderson wrote: On Wed, 10 Dec 2008, Gideon Hack wrote: Hi Gordon, DID World Wide (see http://www.didww.com/virtual_numbers/Sweden) has the DIDs that you require. And they can forward to IAX if that is preferable to you. Thanks. I was actually hoping I'd find a Swedish company, but I'll pass this and the other on to my customer (who's in Sweden and wants to pay in Swedish money) Cheers, Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a problem on Ubuntu with Asterisk
Have you do just asterisk before try to reconect to cli?? Try asterisk - to see if is crashing. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rodrigo Gonzalez Sent: miércoles, 10 de diciembre de 2008 01:12 p.m. To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] a problem on Ubuntu with Asterisk Scott Berry wrote: Have a nice day, Scott Berry E-mail: [EMAIL PROTECTED] I am studying out of the book Asterisk: The Future of Telephony on Chapter 4, and right now for practicing using the built in Debian version of Asterisk for Ubuntu. I am however having some problem where I cannot do asterisk -r and hook up to the asterisk CLI. I have checked to see that /var/run/asterisk/asterisk.ctl is available which it is. I have also set up the zaptel.conf, zapata.conf and also the extensions.conf as specified in the book. The error I get is: Unable to connect to asterisk remote (does /var/run/asterisk/asterisk.ctl exist? Yes it certainly does. Any help would be appreciated. if need be i would be happy to send my extensions.conf, zaptel.conf, and zapata.conf to the lisOne other question I think I am correct on this but not sure does zaptel.conf and zapata.conf go in to /etc? Thanks for all the help. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Permissions problems? sudo asterisk -r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Information from ESET Smart Security, version of virus signature database 3679 (20081209) __ The message was checked by ESET Smart Security. http://www.eset.com __ Information from ESET Smart Security, version of virus signature database 3679 (20081209) __ The message was checked by ESET Smart Security. http://www.eset.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a problem on Ubuntu with Asterisk
A bit of clarification on my previous answer that may help with some of these replies: Asterisk is theoretically launched at start up or some other time with /usr/sbin/asterisk -g causes core dump -vvv causes verbosity level to be set When you try to run your command, it should always include -r connect to the running instance of asterisk -vv set verbosity to at least 2 -c open the console interface to this instance (otherwise you're just viewing the console) So if you run asterisk -vvv then you shut down asterisk with a ctrl-c If you run asterisk -vvc then you can do a stop now. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sebastian Sent: Wednesday, December 10, 2008 9:34 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] a problem on Ubuntu with Asterisk Have you do just asterisk before try to reconect to cli?? Try asterisk - to see if is crashing. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rodrigo Gonzalez Sent: miércoles, 10 de diciembre de 2008 01:12 p.m. To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] a problem on Ubuntu with Asterisk Scott Berry wrote: Have a nice day, Scott Berry E-mail: [EMAIL PROTECTED] I am studying out of the book Asterisk: The Future of Telephony on Chapter 4, and right now for practicing using the built in Debian version of Asterisk for Ubuntu. I am however having some problem where I cannot do asterisk -r and hook up to the asterisk CLI. I have checked to see that /var/run/asterisk/asterisk.ctl is available which it is. I have also set up the zaptel.conf, zapata.conf and also the extensions.conf as specified in the book. The error I get is: Unable to connect to asterisk remote (does /var/run/asterisk/asterisk.ctl exist? Yes it certainly does. Any help would be appreciated. if need be i would be happy to send my extensions.conf, zaptel.conf, and zapata.conf to the lisOne other question I think I am correct on this but not sure does zaptel.conf and zapata.conf go in to /etc? Thanks for all the help. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Permissions problems? sudo asterisk -r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Information from ESET Smart Security, version of virus signature database 3679 (20081209) __ The message was checked by ESET Smart Security. http://www.eset.com __ Information from ESET Smart Security, version of virus signature database 3679 (20081209) __ The message was checked by ESET Smart Security. http://www.eset.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a problem on Ubuntu with Asterisk
On Wed, Dec 10, 2008 at 01:33:39PM -0200, Sebastian wrote: Have you do just asterisk before try to reconect to cli?? Try asterisk - to see if is crashing. This runs asterisk as root (instead of as the user asterisk). In the worst case it might write some files as root (e.g. the log files) that the user asterisk will not be able to overwrite. asterisk -U asterisk Or: /etc/init.d/asterisk debug -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a problem on Ubuntu with Asterisk
Scott, Login as root user and start asterisk by typing asterisk and then give command asterisk -r Amit Mehta Cell: +91 9898340962 On Wed, Dec 10, 2008 at 8:35 PM, Scott Berry [EMAIL PROTECTED] wrote: Have a nice day, Scott Berry E-mail: [EMAIL PROTECTED] I am studying out of the book Asterisk: The Future of Telephony on Chapter 4, and right now for practicing using the built in Debian version of Asterisk for Ubuntu. I am however having some problem where I cannot do asterisk -r and hook up to the asterisk CLI. I have checked to see that /var/run/asterisk/asterisk.ctl is available which it is. I have also set up the zaptel.conf, zapata.conf and also the extensions.conf as specified in the book. The error I get is: Unable to connect to asterisk remote (does /var/run/asterisk/asterisk.ctl exist? Yes it certainly does. Any help would be appreciated. if need be i would be happy to send my extensions.conf, zaptel.conf, and zapata.conf to the lisOne other question I think I am correct on this but not sure does zaptel.conf and zapata.conf go in to /etc? Thanks for all the help. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Func_ODBC question
if you are using MYSQL.. why don't you query your DB directly from Asterisk ?
the following example is something i use with my servers
[ivr1-cont]exten = 7700,1,Answer
exten = 7700,n,MYSQL(Connect connid 127.0.0.1 root rootpass TarekDB)exten =
7700,n,MYSQL(Query resultid_2 ${connid} SELECT q_name FROM tbl_ivr ORDER BY
RAND( ) LIMIT 1 )exten = 7700,n,MYSQL(Fetch fetchid1 ${resultid_2}
question)exten = 7700,n,Read(A1,ivr1/${question})exten =
7700,n,MYSQL(Disconnect ${connid})
-- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria:
+963 944 618286 USA: +1 347 562 2308
From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Tue, 9 Dec 2008 15:45:59
-0200Subject: [asterisk-users] Func_ODBC question
Hi I have
On func_odbc
[EXEC]
readhandle=ressqlserver
writehandle=ressqlserver
readsql=${ARG1}
writesql=${ARG1}
I’m trying an update on dialplan:
exten= 141,3,Set(dummy=${ODBC_EXEC(UPDATE Tabla set campo = ${EXTEN})})
On Cli:
WARNING[3579]: func_odbc.c:353 acf_odbc_read: Error -1 in FETCH [UPDATE Tabla
set campo = 4356]
Any idea why is this??
The query works fine, I just wanto to know if the warning can cause any problem
to me.
Thanks!!
Sebastian
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[asterisk-users] Execute AGI after answered Dial() has ended
Hello, I am googling for a while but google seems to be broken today, no solution yet :D I have a PHP script which needs to be started after Dial() has ended. If there is no answer, busy, etc., it is not a problem, because dialplan continues after the Dial() application, but when the call is established (i call macro in Dial() with AGI execution, working ok) and after the call ends, dialplan execution stops on the Dial(). But I need dialplan to continue after call end and execute the AGI script. Is there any way how to do it? Thanks for help, mt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Execute AGI after answered Dial() has ended
Use the h extension and execute DeadAGI. On Wed, 2008-12-10 at 18:21 +0100, Martin Tirsel wrote: Hello, I am googling for a while but google seems to be broken today, no solution yet :D I have a PHP script which needs to be started after Dial() has ended. If there is no answer, busy, etc., it is not a problem, because dialplan continues after the Dial() application, but when the call is established (i call macro in Dial() with AGI execution, working ok) and after the call ends, dialplan execution stops on the Dial(). But I need dialplan to continue after call end and execute the AGI script. Is there any way how to do it? Thanks for help, mt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Execute AGI after answered Dial() has ended
use deadagi on the h extension maybe? Cheers Geraint 2008/12/10 Martin Tirsel [EMAIL PROTECTED] Hello, I am googling for a while but google seems to be broken today, no solution yet :D I have a PHP script which needs to be started after Dial() has ended. If there is no answer, busy, etc., it is not a problem, because dialplan continues after the Dial() application, but when the call is established (i call macro in Dial() with AGI execution, working ok) and after the call ends, dialplan execution stops on the Dial(). But I need dialplan to continue after call end and execute the AGI script. Is there any way how to do it? Thanks for help, mt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Execute AGI after answered Dial() has ended
2008/12/10 Martin Tirsel [EMAIL PROTECTED] Hello, I am googling for a while but google seems to be broken today, no solution yet :D I have a PHP script which needs to be started after Dial() has ended. If there is no answer, busy, etc., it is not a problem, because dialplan continues after the Dial() application, but when the call is established (i call macro in Dial() with AGI execution, working ok) and after the call ends, dialplan execution stops on the Dial(). But I need dialplan to continue after call end and execute the AGI script. Is there any way how to do it? Thanks for help, mt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users you can try whit the g option to dial. David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR Design
Steve Murphy wrote: Just to be pedantic, how would src_cid be different from the clid field that cdr's have now? and the same with src_exten vs. src; A simple example might help to let this sink into my brain. murf The main thing is that the originating number shouldn't be linked to the callerid. This way you can do things like allow no callerid while maintaining billing integrity. Here is the CDR columns for one one of my providers that exhibits this: *Field Number* *Field Name* *Description* *Type* *Length* *Example* 1 SwitchBatchNbr Sequential, positive integer assigned to each CDR file imported into the system Numeric Long Integer 5594 2 RecNbr Sequential, positive integer assigned to each CDR within a CDR file. Together with the SwitchBatchNbr, a unique combination. Numeric Long Integer 2354 3 SwitchNbr Unique number identifying the switch from which the CDR was processed or assigned Numeric Integer 13 4 CustNbr The unique, numeric number assigned to a customer Numeric Long Integer 1025 5 AuthCode The authorization code used in the call. Can be the Switch/Trunk combination (dedicated), ANI, Travel Card, 800 number, DID. Numeric Float 2145551212 6 AcctCd The Account Code dialed with the CDR Numeric Long Integer 2331 7 CallMMDD Call date at time of answer (MMDD format) Numeric Long Integer 20020131 8 CallHHMMSS Call time at time of answer (HHMMSS format) Numeric Long Integer 205618 9 DestNbr Destination Phone Number Char 18 2145551212 10 DialedNumber Dialed Number Char 18 12145551212 11 ThirdPartyNbr Third Party Number Char 18 2145551212 12 DestCity Destination city name Char 15 Dallas 13 DestState Destination state name Char 2 TX 14 DestOCN Destination OCN Char 4 9100 15 DestLata Destination LATA Numeric integer 552 16 IntraInter Flag indicating jurisdiction: 1=Intralata, 2=Intrastate, 3=Interstate, 4=Canada, 5=Intl, Mexico Numeric Integer 1 17 CallType Flag indicating type of call. See Appendix A: Call Type Codes. Char 3 OE 18 DurMinutes The rounded, billable duration of a rated call. Detailed to a tenth of a minute. Numeric Decimal 10,4 1.5000 19 CustRev The revenue computed for the CDR Numeric Decimal 10,4 0.0168 20 Surchrg The surcharge amount for the CDR Numeric Decimal 10,4 0. 21 OrigNbr Originating Phone Number Char 18 2145551212 22 OrigCity Originating City Char 15 DIR ASST 23 OrigState Originating State Char 2 TX 24 OrigOCN Originating OCN Char 4 9100 25 OrigLata Originating LATA Numeric Integer 552 26 SiteNbr Info digit assigned to CDR. Currently, Site Numbers: 7, 25, 27, 29, 70 are considered payphone Numeric Integer 0 27 SiteSurChrg Charge associated with payphone use as determined by the SiteNbr Numeric Decimal 10,4 . 28 ExtractSeqNbr Number used to designate a batch of CDR's that were extracted. If not used, value will be NULL. Numeric Integer 156 -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [EMAIL PROTECTED] ___ --
Re: [asterisk-users] Improving Asterisk french prompts
On Wednesday 10 December 2008 01:27:17 Olivier wrote: When listening to the time and date a voicemail message was received, you can hear french sentences like : message reçu à vingt-et-un heure (message received at twenty and one hour) It should be (in whatever french flavour): message reçu à vingt-et-une heure In short, instead of: -- SIP/7533-094fdbb8 Playing 'digits/at.gsm' (language 'fr') -- SIP/7533-094fdbb8 Playing 'digits/20.gsm' (language 'fr') -- SIP/7533-094fdbb8 Playing 'digits/et.gsm' (language 'fr') -- SIP/7533-094fdbb8 Playing 'digits/1.gsm' (language 'fr') -- SIP/7533-094fdbb8 Playing 'digits/oclock.gsm' (language 'fr') it should be something like: -- SIP/7533-094fdbb8 Playing 'digits/at.gsm' (language 'fr') -- SIP/7533-094fdbb8 Playing 'digits/20.gsm' (language 'fr') -- SIP/7533-094fdbb8 Playing 'digits/et.gsm' (language 'fr') -- SIP/7533-094fdbb8 Playing 'digits/une.gsm' (language 'fr') -- -- SIP/7533-094fdbb8 Playing 'digits/oclock.gsm' (language 'fr') I didn't check but the same problem should arise with 01h00. Is this seen as a feature or can we provide enhancements ? If it's clearly wrong, then certainly, it's a candidate for fixing in 1.4. I would invite you to work with Clod Patry (JunK-Y on IRC), as he is our French (Canadian) translator and is most responsible for the prompts as they are today. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk video
Hi All, Got some problem with asterisk video, i'm testing an eyebeam and a grandstream video phone. call from grandstream to eyebeam works ok, video shows up. but calls from eyebeam to grandstream there's no video, but audio is ok. Regards, Nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Registry Problems
Stefan I tried this and now I get this: -- ast_get_srv: SRV lookup for '_sip._udp.grimlock.vtnoc.net' mapped to host galvatron.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.megatron.vtnoc.net' mapped to host grimlock.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.grimlock.vtnoc.net' mapped to host optimusprime.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.galvatron.vtnoc.net' mapped to host grimlock.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.optimusprime.vtnoc.net' mapped to host megatron.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.grimlock.vtnoc.net' mapped to host megatron.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.megatron.vtnoc.net' mapped to host galvatron.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.galvatron.vtnoc.net' mapped to host galvatron.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.megatron.vtnoc.net' mapped to host optimusprime.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.megatron.vtnoc.net' mapped to host galvatron.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.grimlock.vtnoc.net' mapped to host megatron.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.optimusprime.vtnoc.net' mapped to host galvatron.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.galvatron.vtnoc.net' mapped to host galvatron.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.megatron.vtnoc.net' mapped to host grimlock.vtnoc.net, port 5060 The connections are all over the place and I still don't get DTMF to pass. Any other suggestions? Stefan Schmidt wrote: Brent Vrieze schrieb: Here is what happens: 1. Asterisk verifies connection to the server and we get this. (CLI output) -- ast_get_srv: SRV lookup for '_sip._udp.galvatron.vtnoc.net' mapped to host galvatron.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.galvatron.vtnoc.net' mapped to host optimusprime.vtnoc.net, port 5060 -- ast_get_srv: SRV lookup for '_sip._udp.galvatron.vtnoc.net' mapped to host megatron.vtnoc.net, port 5060 It jumps around from server to server all the time. hello, you should add the second and third server to your user.conf as a friend too, they just use something like a load balancer but you only accept calls from one of their 3 servers. so i think what happens is that if a call comes from one of the server you didnt authorize it just get an error like authorization required and then fallback to your boss cell number. maybe you could trace this with sip debug. best regards. steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Parked Extension Variable
Hello, When I execute parkandannounce() in the dialplan, is the extension that the call is parked to stored in a variable? I would like to send it to an AGI script but can't seem to figure out where the 'announcer' gets its information. Thanks Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Parked Extension Variable
According to lists.digium.com/pipermail/asterisk-dev/2006-March/019516.html
the value is stored in ${PARKEDAT}
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Gibbons
Sent: Wednesday, December 10, 2008 1:02 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Parked Extension Variable
Hello,
When I execute parkandannounce() in the dialplan, is the extension that the
call is parked to stored in a variable? I would like to send it to an AGI
script but can't seem to figure out where the 'announcer' gets its
information.
Thanks
Dave
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[asterisk-users] SendImage() to Polycom ip550 or ip670
I tried really quickly the other day to send an image to these phones from the dialplan like this: exten = 2821,n,SendImage(/var/lib/asterisk/images/asterisk-intro) or exten = 2821,n,SendImage(asterisk-intro) It didn't work for me. Should this work? Is anyone else using this with Polycom Phones? Bob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Parked Extension Variable
snip
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Danny Nicholas
According to lists.digium.com/pipermail/asterisk-dev/2006-March/019516.html
the value is stored in ${PARKEDAT}
/snip
*grin*
I guess I deserved that.
Thanks for checking.
Dave
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Re: [asterisk-users] SendImage() to Polycom ip550 or ip670
Two things - Polycom phones require specific images sizes (480x180 pixels I thinkg) and the sip.cfg has to allow presentation of images. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob Pierce Sent: Wednesday, December 10, 2008 1:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SendImage() to Polycom ip550 or ip670 I tried really quickly the other day to send an image to these phones from the dialplan like this: exten = 2821,n,SendImage(/var/lib/asterisk/images/asterisk-intro) or exten = 2821,n,SendImage(asterisk-intro) It didn't work for me. Should this work? Is anyone else using this with Polycom Phones? Bob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk video
On Thu, 11 Dec 2008, Nhadie wrote: Hi All, Got some problem with asterisk video, i'm testing an eyebeam and a grandstream video phone. call from grandstream to eyebeam works ok, video shows up. but calls from eyebeam to grandstream there's no video, but audio is ok. It's a Video Codec issue. Tell eyebeam not to use H263p then it'll work. At least that's how I got a Grandstram to talk to eyebeam. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Park buttons on Polycom IP501/601
Is anyone using fixed Park buttons (some of the ones on the left side of the screen) on a Polycom phone? Here's what I mean: - Call is received and parked, by the user pressing an unlit park button (e.g. 701) and the call is parked there. - The call can be picked up at any other extension by pressing the flashing park 701 button. - Once the call has been picked up, the 701 park slot is idle and the light goes off. For a small site, only a couple of Park buttons would be needed. Can you give an example of how to do this? Thanks, S. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Parked Extension Variable
Use the docs, Luke.
dev-1*CLI core show application parkandannounce
dev-1*CLI
-= Info about application 'ParkAndAnnounce' =-
[Synopsis]
Park and Announce
[Description]
ParkAndAnnounce(announce:template,timeout,dial[,return_context]):
Park a call into the parkinglot and announce the call to another channel.
announce template: Colon-separated list of files to announce. The word
PARKED
will be replaced by a say_digits of the extension in
which
the call is parked.
timeout: Time in seconds before the call returns into the return
context.
dial: The app_dial style resource to call to make the
announcement. Console/dsp calls the console.
return_context:The goto-style label to jump the call back into after
timeout. Default priority+1.
The variable ${PARKEDAT} will contain the parking extension into which the
call was placed. Use with the Local channel to allow the dialplan to make
use of this information.
David Gibbons wrote:
Hello,
When I execute parkandannounce() in the dialplan, is the extension that the
call is parked to stored in a variable? I would like to send it to an AGI
script but can't seem to figure out where the 'announcer' gets its
information.
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[asterisk-users] AST-2008-012: Remote crash vulnerability in IAX2
Asterisk Project Security Advisory - AST-2008-012 ++ | Product| Asterisk| |--+-| | Summary| Remote crash vulnerability in IAX2 | |--+-| | Nature of Advisory | Remote Crash| |--+-| |Susceptibility| Remote Unauthenticated Sessions | |--+-| | Severity | Major | |--+-| |Exploits Known| No | |--+-| | Reported On | November 22, 2008 | |--+-| | Reported By |Jon Leren Scho/pzinsky | |--+-| | Posted On | | |--+-| | Last Updated On| December 9, 2008| |--+-| | Advisory Contact | Mark Michelson mmichelson AT digium DOT com | |--+-| | CVE Name | | ++ ++ | Description | There is a possibility to remotely crash an Asterisk | | | server if the server is configured to use realtime IAX2 | | | users. The issue occurs if either an unknown user| | | attempts to authenticate or if a user that uses hostname | | | matching attempts to authenticate. | | | | | | The problem was due to a broken function call to | | | Asterisk's realtime configuration API. | ++ ++ | Resolution| The function calls in question have been fixed. | ++ ++ | Affected Versions| || | Product | Release Series | | |-++-| | Asterisk Open Source | 1.2.x | 1.2.26-1.2.30.3 | |-++-| | Asterisk Open Source | 1.4.x | Unaffected | |-++-| | Asterisk Open Source | 1.6.x | Unaffected | |-++-| | Asterisk Addons | 1.2.x | Unaffected | |-++-| | Asterisk Addons | 1.4.x | Unaffected | |-++-| | Asterisk Addons | 1.6.x | Unaffected | |-++-| |Asterisk Business Edition| A.x.x | Unaffected | |-++-| |Asterisk Business Edition| B.x.x | B.2.3.5-B.2.5.5 | |-++-| |Asterisk Business Edition| C.x.x | Unaffected | |-++-| | AsteriskNOW | 1.5 | Unaffected | |-++-| | s800i
[asterisk-users] Asterisk 1.2.30.4 released
The Asterisk.org development team has released Asterisk version 1.2.30.4. This release is available for immediate download from http://downloads.digium.com/. This update for Asterisk includes a security fix for chan_iax2. Please see the associated security advisory for more details: http://downloads.digium.com/pub/security/AST-2008-012.pdf . This security issue affects only the 1.2 series. Thank you for your continued support of Asterisk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G729 licenses
Hi, The way I understand this (http://www.voip-info.org/wiki-Asterisk+G.729+Licensing) is that a call from a G729 enabled phone to Asterisk, then to a ulaw SIP provider would take one license, and would show up as 1/1 (one encoder and one decoder). So, in short, if all my calls were from outside to a G729 enabled phone and vice versa, I would reach the limit at 30/30, NOT 15/15. Right? I am asking because show g729 was near 15/15 and I started seeing codec unknown messages in my CLI, and I sure am only using g729 for all registered phones. Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Softphone recommendation
Hi Folks, Had a quick search through the archives for softphones and cannot see any recommended ones. My question is what recommended free softphones are out there that can be used with Asterix? I don't really know how many are out there. Is anyone currently using a softphone with Asterix and if so which one and how do you find it? I'm only interested in ones that I can download and use for free. Not interested in any commercial ones that require licenses. Thanks+Regards George ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone recommendation
X-lite from CounterPath work with Asterisk. No g729 support on the free version. If u plan to use ulaw will work perfectly. Best regards, Chris Hariga --Original Message-- From: Georgecooldude Sender: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Softphone recommendation Sent: Dec 10, 2008 7:35 PM Hi Folks, Had a quick search through the archives for softphones and cannot see any recommended ones. My question is what recommended free softphones are out there that can be used with Asterix? I don't really know how many are out there. Is anyone currently using a softphone with Asterix and if so which one and how do you find it? I'm only interested in ones that I can download and use for free. Not interested in any commercial ones that require licenses. Thanks+Regards George ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sent from my BlackBerry® smartphone with SprintSpeed ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Video conference with Asterisk
Dear all, I've read that it's possible to set up Asterisk 1.4 with video using H.263 and H.264 video codecs. Now I'm using just SIP in order to have voice over IP. My question are: 1) Do both SIP video and voice work OK simultaneously in Asterisk 1.4 ??? 2) What is the best SIP video+voice free clients for Windows and Linux ??? Thanks in advance. Alejandro___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone recommendation
Hi Folks, Had a quick search through the archives for softphones and cannot see any recommended ones. My question is what recommended free softphones are out there that can be used with Asterix? I don't really know how many are out there. Is anyone currently using a softphone with Asterix and if so which one and how do you find it? I'm only interested in ones that I can download and use for free. Not interested in any commercial ones that require licenses. Zopier works well and supports both IAX and SIP. Works on Windows, Mac and Linux. http://www.zoiper.com/ Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 licenses
So, in short, if all my calls were from outside to a G729 enabled phone and vice versa, I would reach the limit at 30/30, NOT 15/15. If you had 30 licenses, yes the limit would be when you needed either 30 decoders or 30 encoders. i.e. 1/30 would max you out. -M+ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 licenses
Thank you for the sanity check! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Darnell Sent: Wednesday, December 10, 2008 22:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] G729 licenses So, in short, if all my calls were from outside to a G729 enabled phone and vice versa, I would reach the limit at 30/30, NOT 15/15. If you had 30 licenses, yes the limit would be when you needed either 30 decoders or 30 encoders. i.e. 1/30 would max you out. -M+ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk video
Hi, I've tried that also, i remove h263+ and leave only h263. i prioritize h263 on grandstream aslo. does it work bothways on your side? for mine it works if i call from grandstream to eyebeam, but eyebeam to grandstream does not. Thanks! Regards Nhadie Gordon Henderson wrote: On Thu, 11 Dec 2008, Nhadie wrote: Hi All, Got some problem with asterisk video, i'm testing an eyebeam and a grandstream video phone. call from grandstream to eyebeam works ok, video shows up. but calls from eyebeam to grandstream there's no video, but audio is ok. It's a Video Codec issue. Tell eyebeam not to use H263p then it'll work. At least that's how I got a Grandstram to talk to eyebeam. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CallingCard Applications
I want to build my own calling card system on Asterisk. I looked at this page - http://www.voipinfo.org/wiki/view/CallingCard+Applications and it has listed some applications that I thought could help speed up the development process though the link down the bottom doesn't work. Does anyone know of any AGI etc applications to build a Calling Card system on Asterisk? Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Pickup (*8) / Attended forward and CallerID
Hi, Since we're moving from a legacy PABX that has been serving one of our customers for more than 15 years, we'd like this process to require no human habits change among the users. Software: Asterisk 1.4.22 Hardware: Polycom phones (mainly 430/601) Here are the problems: We did configure call groups, pickup groups, ... - When someone picks up a call from another person, the display of his phone only shows *8 and not the original CallerID. - When doing an attended transfer, the callerid of the original caller (A calls B, then B forwards to C = We want to show C the original callerid somewhere on his phone's screen). - When using the blind transfer feature, the CallerID is fine. I know this has already been discussed in 2006 (from digium's BTS), and would like to know if this situation did change, or not. Is it still considered as features ? Is it considered as bugs ? Will it be implemented in another way in some future release ? ...? Thanks Laurent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
