Re: [asterisk-users] Asterisk 10.0.0-rc1 Now Available
Hey, There is a way to use the new confbridg without installing the new version of Asterisk, I'm new in using asterisk, my is version 1.6.2 of Asterisk. I would not want just to get into the installation of a new version just for a one commend which I have very great need. Thank you for your help. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Development Team Sent: Thursday, November 10, 2011 6:39 PM To: Asterisk Development Team Subject: [asterisk-users] Asterisk 10.0.0-rc1 Now Available The Asterisk Development Team is pleased to announce the first release candidate of Asterisk 10.0.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ All Asterisk users are encouraged to participate in the Asterisk 10 testing process. Please report any issues found to the issue tracker, https://issues.asterisk.org/jira. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list. All Asterisk users are invited to participate in the #asterisk-testing channel on IRC to work together in testing the many parts of Asterisk. Asterisk 10 is the next major release series of Asterisk. It will be a Standard support release, similar to Asterisk 1.6.2. For more information about support time lines for Asterisk releases, see the Asterisk versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions A short list of features includes: * T.38 gateway functionality has been added to res_fax. * Protocol independent out-of-call messaging support. Text messages not associated with an active call can now be routed through the Asterisk dialplan. SIP and XMPP are supported so far. * New highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8kHz-192kHz (More information available at https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 ) * Addition of video_mode option in confbridge.conf to provide basic video conferencing in the ConfBridge() dialplan application. * Support for defining hints has been added to pbx_lua. * Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB). * Much, much more! A full list of new features can be found in the CHANGES file. http://svnview.digium.com/svn/asterisk/branches/10/CHANGES For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog- 10.0.0-rc1 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MeetMeAdmin()
Hi, I use this command " exten => unmute,n,MeetMeAdmin(room_number,m,${caller})", Caller is the number of the participant that I want to mute, but when the number of participant is not in the room I get a notice in the CLI: "NOTICE[17723]: app_meetme.c:4244 admin_exec: Specified User not found!" And the program continues from there as it should, My question is, is there a way to use that answer in the program? Is there a way to get a result of whether it works or not? Thank you for your help. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Get return code from MeetMeAdmin()? did it possible?
? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eyal Sent: Thursday, November 24, 2011 3:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] MeetMeAdmin() Hi, I use this command " exten => unmute,n,MeetMeAdmin(room_number,m,${caller})", Caller is the number of the participant that I want to mute, but when the number of participant is not in the room I get a notice in the CLI: "NOTICE[17723]: app_meetme.c:4244 admin_exec: Specified User not found!" And the program continues from there as it should, My question is, is there a way to use that answer in the program? Is there a way to get a result of whether it works or not? Thank you for your help. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Talk detection in meetme
Hi, I create Chat room with MEETME and now I have a problem. I want that the host of the room could identify the participants in the room by their speech, so that if a participant uses language the host could kick him from the room. Is there a way to do it? thanks. Eyal Mahalal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Talk detection in meetme
??? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eyal Sent: Tuesday, December 06, 2011 11:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Talk detection in meetme Hi, I create Chat room with MEETME and now I have a problem. I want that the host of the room could identify the participants in the room by their speech, so that if a participant uses language the host could kick him from the room. Is there a way to do it? thanks. Eyal Mahalal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] create table in mysql using asterisk
Hi, I try to create a new table using MYSQL command in asterisk. This is what i write: Query resultid ${connid} CREATE TABLE IF NOT EXISTS "conference_600" ("id" int(11) NOT NULL auto_increment, "channel_id" varchar(40), "number_in_line" int(2), PRIMARY KEY("id")") and this is the warning that i get in the cli: app_addon_sql_mysql.c:383 aMYSQL_query: aMYSQL_query: mysql_query failed. Error: You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near '"conference_600" ("id" int(11) NOT NULL auto_increment, "channel_id" varchar(40)' at line 1 What is the problem do you think? Do I in the direction or have a completely different way to do this? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] create table in mysql using asterisk
Thanks But that's not the problem, I also tried without the quotes and still the error appears only this time it is like this app_addon_sql_mysql.c:383 aMYSQL_query: aMYSQL_query: mysql_query failed. Error: You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near '' at line 1 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Sharp Sent: Monday, January 09, 2012 9:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] create table in mysql using asterisk On 01/09/2012 02:44 AM, Eyal wrote: > Hi, > I try to create a new table using MYSQL command in asterisk. > This is what i write: > *Query resultid ${connid} CREATE TABLE IF NOT EXISTS "conference_600" > ("id" int(11) NOT NULL auto_increment, "channel_id" varchar(40), > "number_in_line" int(2), PRIMARY KEY("id")")* > and this is the warning that i get in the cli: > *app_addon_sql_mysql.c:383 aMYSQL_query: aMYSQL_query: mysql_query > failed. Error: You have an error in your SQL syntax; check the manual > that corresponds to your MySQL server version for the right syntax to > use near '"conference_600" ("id" int(11) NOT NULL auto_increment, > "channel_id" varchar(40)' at line 1 > * > What is the problem do you think? > Do I in the direction or have a completely different way to do this? That's a MySQL syntax error, not an Asterisk error. However, the solution is to not put quotes around your table and field names. That will make MySQL happy. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Change the caller's phone number
Hi, I have a system that receives calls from clients and directs them to an external phone, before I pass on the client I change the client's phone number to a number that I choose, so that The call recipient knew the call came from our system. But I have a problem with that, not all phone number change some of the anonymous calls stay anonymous and the recipient See in the caller ID display unlisted number. I use this commend: Set(CALLERID(all)="722979797" <722979797>) Is anyone having a similar problem or know what the problem? Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Change the caller's phone number
Thanks but it did not help resolve the situation. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Thursday, January 19, 2012 4:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Change the caller's phone number or this Set(${CALLERID(all)}="722979797" <722979797>) From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Thursday, January 19, 2012 8:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Change the caller's phone number try the following Set(${CALLERID}="722979797" <722979797>) From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Eyal [e...@mcr-m.com] Sent: Thursday, January 19, 2012 8:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Change the caller's phone number Hi, I have a system that receives calls from clients and directs them to an external phone, before I pass on the client I change the client's phone number to a number that I choose, so that The call recipient knew the call came from our system. But I have a problem with that, not all phone number change some of the anonymous calls stay anonymous and the recipient See in the caller ID display unlisted number. I use this commend: Set(CALLERID(all)="722979797" <722979797>) Is anyone having a similar problem or know what the problem? Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] play sound file
Hi, How can I play a sound file from the middle and end it after a certain number of seconds? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] play sound file
Thanks But this is not what I am looking for, in this way I can start the sound file from some point in the file but the callers must hear the file until the end. I need something that allows me to start from some place in the file and end it in some other place in the file (say song from time 01:32 until 01:57), Or Like the controlplayback doing fast-forward but without having to click any key by caller. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nasir Iqbal Sent: Thursday, January 26, 2012 10:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] play sound file check this http://www.voip-info.org/wiki/view/Asterisk+cmd+ControlPlayback Nasir Iqbal ICTBroadcast SMS, Fax and Voice broadcasting solution http://www.ictbroadcast.com/ On Wed, Jan 25, 2012 at 8:29 PM, Eyal wrote: Hi, How can I play a sound file from the middle and end it after a certain number of seconds? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] play sound file
Thanks Any more ideas? אייל מהלל מתכנת IVR משרד: 03-6034293 שלוחה 220 נייד: 054-4793007 פקס: 03-6006081 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Johan Wilfer Sent: Thursday, January 26, 2012 11:24 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] play sound file 2012-01-26 10:11, Eyal skrev: Thanks But this is not what I am looking for, in this way I can start the sound file from some point in the file but the callers must hear the file until the end. I need something that allows me to start from some place in the file and end it in some other place in the file (say song from time 01:32 until 01:57), Or Like the controlplayback doing fast-forward but without having to click any key by caller. You can do that by combining ControlPlayback and use TIMEOUT function. If you don't want the user to be able to use any keys you can use all keys as stop-keys in ControlPlayback and have some logic restart the playback at position ${CPLAYBACKOFFSET}. For more details do: core show application ControlPlayback core show function TIMEOUT /Johan From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nasir Iqbal Sent: Thursday, January 26, 2012 10:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] play sound file check this http://www.voip-info.org/wiki/view/Asterisk+cmd+ControlPlayback Nasir Iqbal ICTBroadcast SMS, Fax and Voice broadcasting solution http://www.ictbroadcast.com/ On Wed, Jan 25, 2012 at 8:29 PM, Eyal wrote: Hi, How can I play a sound file from the middle and end it after a certain number of seconds? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Johan Wilfer email: jo...@jttech.se JT Tech | Developer webb: http://jttech.se -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is there a default dial plan that is not in extention.conf?
Hi, I have a trivial peace of dialplan for exten 100. I try to change it to _1XX and the asterisk act according to a different (Default??) dial plan and not the one I want? Is that possible? Where is the other dialplan sits? In my extention.conf I can't see something that look like what asterisk is dialing. How can I trace\debug my dialplan? Thanks, Eyal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a default dial plan that is not in extention.conf?
Thank you all, This is what I see after CLI> dialplan show 1...@default : '100' => hint: SIP/100&IAX2/100 [pbx_config] 1. Dial(${HINT}) [pbx_config] '_1XX' => 1. Playback(digits/4) [pbx_config] >From where come the 2 first lines?? I only have the third one as the only one under my default context at extention.conf. And what is [pbx_config]? Thanks Eyal -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Friday, June 25, 2010 4:05 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Is there a default dial plan that is not in extention.conf? On Fri, Jun 25, 2010 at 02:25:38PM +0300, Eyal Goltzman wrote: > Hi, > > > > I have a trivial peace of dialplan for exten 100. I try to change it to _1XX > and the asterisk act according to a different (Default??) dial plan and not > the one I want? Is that possible? Where is the other dialplan sits? In my > extention.conf I can't see something that look like what asterisk is > dialing. > > How can I trace\debug my dialplan? To see where it comes from, run in the Asterisk CLI: dialplan show or: dialplan show @ Here is a partial output from 'dialplan show' here, that shows all of them (but is normally overly long) [ Context 'app_queue_gosub_virtual_context' created by 'app_queue' ] 's' =>1. NoOp() [app_queue] [ Context 'parkedcalls' created by 'features' ] '700' => 1. Park() [features] [ Context 'app_dial_gosub_virtual_context' created by 'app_dial' ] 's' =>1. NoOp() [app_dial] [ Context 'from-pstn' created by 'pbx_config' ] '_X.' => 1. Answer() [pbx_config] 2. Playback(demo-instruct)[pbx_config] 3. Hangup() [pbx_config] [ Context 'ael-dundi-e164' created by 'pbx_ael' ] 's' =>1. MSet(LOCAL(exten)=${ARG1}) [pbx_ael] 2. Goto(${exten},1) [pbx_ael] 3. Return() [pbx_ael] 'pbx_config' is dialplan that was generated from your extensions.conf. 'pbx_ael' is dialplan that was generated from extensions.ael. Various other modules include their own minor dialplan snippets. 'dialplan show @' also resolves various 'include=>' directives. If you had: [local] include => phones exten => 120,1,Dial(SIP/trunk/123456) [phones] exten => 100,1,Dial(SIP/phone1) the 'dialplan show local' would show the equivalent of include => phones exten => 120,1,Dial(SIP/trunk/123456) whereas 'dialplan show 1...@local would show the actual (equivalent of) exten => 100,1,Dial(SIP/phone1) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - www.avg.com Version: 9.0.830 / Virus Database: 271.1.1/2961 - Release Date: 06/24/10 21:35:00 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] unwanted entries created in dialplan from users.conf, how can I get rid of it?
Hello, When I call "dialplan reload" I can see the following lines: == Parsing '/etc/asterisk/extensions.conf': == Found -- Registered extension context 'default' (0x8a72410) in local table 0x8a679d0; registrar: pbx_config -- Added extension '_1XX' priority 1 to default (0x8a72410) . . . == Parsing '/etc/asterisk/users.conf': == Found -- Added extension '100' priority -1 to default (0xb731b4e8) -- Added extension '100' priority 1 to default (0xb731b4e8) -- Added extension '101' priority -1 to default (0xb731b4e8) -- Added extension '101' priority 1 to default (0xb731b4e8) That result in a dialplan that look like that: [ Context 'default' created by 'pbx_config' ] '100' => hint: SIP/100&IAX2/100 [pbx_config] 1. Dial(${HINT})[pbx_config] '_1XX' => 1. Playback(digits/4) [pbx_config] I don't want those 2 first line to be there only the _1XX. How do I get rid of it? Why it is added to the dial plan automatically? extention.conf look like this: [default] exten => _1XX,1,Playback(digits/4) users.cong look like this: [100] username = 100 transfer = yes mailbox = 100 call-limit = 100 type = peer fullname = Polycom registersip = no host = dynamic callgroup = 1 type = peer context = default cid_number = 100 hasvoicemail = no threewaycalling = no hasdirectory = no callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = no secret = 100 nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 autoprov = no macaddress = linenumber = 1 LINEKEYS = 1 disallow = all allow = ulaw,gsm Thanks, Eyal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Where can I find a minimal set of empty configuration files (SIP only)?
Hello, After installing and learning Asterisk I found myself with a need for a minimal set of empty configuration files with only the "must have" stuff in order to setup a SIP only machine, is there a place to find it? Thanks, Eyal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using AMI Originate to call 2 outside numbers and connect them
Hello, Can I use AMI Originate to call 2 outside numbers (SIP) and connect them? How? Thanks Eyal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using AMI Originate to call 2 outside numbers and connect them
Thanks Tzafrir, If I use the other channel as the extension parameter it will do the work for me also? Eyal -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Saturday, July 03, 2010 2:15 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Using AMI Originate to call 2 outside numbers and connect them On Sat, Jul 03, 2010 at 01:33:25AM +0300, eyal goltzman wrote: > Hello, > > Can I use AMI Originate to call 2 outside numbers (SIP) and connect them? > How? Originate one channel to the application Dial to dial to the other channel? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - www.avg.com Version: 9.0.830 / Virus Database: 271.1.1/2978 - Release Date: 07/02/10 21:35:00 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Trunk configuration problem - fromdomain
Hello, I'm trying to register to my provider sip trunk, I got from him an host IP (a.b.c.d) to connect to and my provider recognize me based on the fixed IP (x.y.z.w) he gave me (no need for username and password) In the sip.conf I add: [mytrunk] type=friend insecure=no host=a.b.c.d fromdomain=x.y.z.w qualify=3600 nat=no ; change to yes if you are behind NAT bindport=5060 bindaddr=0.0.0.0 context=default disallow=all allow=ulaw allow=alaw Now, my asterisk resides in my internal network (10.100.101.107) and in the SIP requests that sent to the provider I can see (via a sniffer) that the "From" and "Contact" fields have - sip:aster...@10.100.101.107 and not the x.y.z.w I expected to see as a result of the fromdomain=x.y.z.w. Any idea? Thanks, Eyal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to change the IP in the SIP contact header
Hello, I'm trying to use a SIP trunk service and the provider ask me to have the IP address of the contact header as my public IP and not as my private one, how can I do it? See attached the SIP invite where a.b.c.d is the SIP server IP and x.y.z.w is my public address: sipINVITE sip:144@ a.b.c.d SIP/2.0 Via: SIP/2.0/UDP 10.100.101.107:5060;branch=z9hG4bK76d52819;rport Max-Forwards: 70 From: "Polycom" ;tag=as7435100b To: Contact: Call-ID: 08116cf06661dc091de10c1b3315d...@84.94.96.110 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Mon, 05 Jul 2010 15:49:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 292 v=0 o=root 1812163927 1812163927 IN IP4 10.100.101.107 s=Asterisk PBX 1.6.1.20 c=IN IP4 10.100.101.107 t=0 0 m=audio 18848 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to change the IP in the SIP contact header
Yes, I tried and it did not solve the problem, Thanks From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jamie A. Stapleton Sent: Monday, July 05, 2010 9:05 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] How to change the IP in the SIP contact header Have you tried setting externip= In the [general] of your sip.conf? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eyal Goltzman Sent: Monday, July 05, 2010 1:58 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] How to change the IP in the SIP contact header Hello, I'm trying to use a SIP trunk service and the provider ask me to have the IP address of the contact header as my public IP and not as my private one, how can I do it? See attached the SIP invite where a.b.c.d is the SIP server IP and x.y.z.w is my public address: sipINVITE sip:144@ a.b.c.d SIP/2.0 Via: SIP/2.0/UDP 10.100.101.107:5060;branch=z9hG4bK76d52819;rport Max-Forwards: 70 From: "Polycom" ;tag=as7435100b To: Contact: Call-ID: 08116cf06661dc091de10c1b3315d...@84.94.96.110 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Mon, 05 Jul 2010 15:49:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 292 v=0 o=root 1812163927 1812163927 IN IP4 10.100.101.107 s=Asterisk PBX 1.6.1.20 c=IN IP4 10.100.101.107 t=0 0 m=audio 18848 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv No virus found in this incoming message. Checked by AVG - www.avg.com Version: 9.0.830 / Virus Database: 271.1.1/2978 - Release Date: 07/05/10 09:36:00 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How can get user inputs from called party after dial?
Hi, I want to dial a party, play him a message and wait for his input, i.e. DTMF digits and use them to control the rest of the dial plan. How do I do it? If I use Dial it will not return until the end of the call, isn't it? Thanks, Eyal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How can get user inputs from called party after dial?
Thanks, but I'm missing something here, the dial command is where? I need to do something like: Dial(1234) Read(1 digit) DoSomthing(based on digit from 1234) And as far as I understand the Dial start the call and only come back (ig you use the g option) after call finished. Eyal From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce Sent: Saturday, July 10, 2010 9:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How can get user inputs from called party after dial? You need read(): http://www.voip-info.org/wiki/view/Asterisk+cmd+Read It's as easy as: exten => s,n,Read(variable,,11) exten => s,n,NoOp(${variable}) Above will take up to 11 digits input by user and will display it back in NoOP on Asterisk CLI. -Bruce On Sat, Jul 10, 2010 at 2:16 PM, eyal goltzman wrote: Hi, I want to dial a party, play him a message and wait for his input, i.e. DTMF digits and use them to control the rest of the dial plan. How do I do it? If I use Dial it will not return until the end of the call, isn't it? Thanks, Eyal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - www.avg.com Version: 9.0.830 / Virus Database: 271.1.1/2991 - Release Date: 07/10/10 09:36:00 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How can get user inputs from called party after dial?
Thank you Bruce, I think we are not on the same page. I have an AMI script that issue an originate command, after one channel is connected I'm in my dialplan at extensions_custom.conf (I use FreePBX). Now I'm issuing a Dial command to the another party that when he pick up the phone I play for him a message (using the A option in the Dial command) and then want to wait for his input, this is the case. Eyal From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce Sent: Saturday, July 10, 2010 9:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How can get user inputs from called party after dial? You need to do some reading :-) I will give you a quick teach here. At the end of file /etc/asterisk/extensions_custom.conf (if you are running FreePBX) OR in /etc/asterisk/extensions.conf (if you are running vanilla Asterisk) add this: [first-Dialplan] exten => s,1,Answer exten => s,n,Playback(Welcome) exten => s,n,Read(numb,,10) exten => s,n,NoOp(${numb}) And send your inbound route to context first-Dialplan so that it's triggered when a call comes in. Then on terminal do a "asterisk -r" and you will see the NoOp show the DTMF number entered. From there on you can do anything you want with the variable ${numb} If any part of above is unclear to you, you must consult your friend, google, for examples of Asterisk dialplan. -Bruce On Sat, Jul 10, 2010 at 2:38 PM, Eyal Goltzman wrote: Thanks, but I'm missing something here, the dial command is where? I need to do something like: Dial(1234) Read(1 digit) DoSomthing(based on digit from 1234) And as far as I understand the Dial start the call and only come back (ig you use the g option) after call finished. Eyal From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce Sent: Saturday, July 10, 2010 9:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How can get user inputs from called party after dial? You need read(): http://www.voip-info.org/wiki/view/Asterisk+cmd+Read It's as easy as: exten => s,n,Read(variable,,11) exten => s,n,NoOp(${variable}) Above will take up to 11 digits input by user and will display it back in NoOP on Asterisk CLI. -Bruce On Sat, Jul 10, 2010 at 2:16 PM, eyal goltzman wrote: Hi, I want to dial a party, play him a message and wait for his input, i.e. DTMF digits and use them to control the rest of the dial plan. How do I do it? If I use Dial it will not return until the end of the call, isn't it? Thanks, Eyal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - www.avg.com Version: 9.0.830 / Virus Database: 271.1.1/2991 - Release Date: 07/10/10 09:36:00 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - www.avg.com Version: 9.0.830 / Virus Database: 271.1.1/2991 - Release Date: 07/10/10 09:36:00 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How can get user inputs from called party after dial?
Thank you Bruce, In the below example you sent the dialplan will stop after Dial. I found the solution to my problem in the M option of the Dial command that let you run a macro BEFORE the parties are connected and continue the dialplan based on the MACRO_RESULT. Thanks for your help, Eyal From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce Sent: Saturday, July 10, 2010 9:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How can get user inputs from called party after dial? For dial you do this: [first-Dialplan] exten => s,1,Answer exten => s,n,Dial(SIP/provider/111222) exten => s,n,Playback(Welcome) exten => s,n,Read(numb,,10) exten => s,n,NoOp(${numb}) -Bruce On Sat, Jul 10, 2010 at 2:51 PM, bruce bruce wrote: You need to do some reading :-) I will give you a quick teach here. At the end of file /etc/asterisk/extensions_custom.conf (if you are running FreePBX) OR in /etc/asterisk/extensions.conf (if you are running vanilla Asterisk) add this: [first-Dialplan] exten => s,1,Answer exten => s,n,Playback(Welcome) exten => s,n,Read(numb,,10) exten => s,n,NoOp(${numb}) And send your inbound route to context first-Dialplan so that it's triggered when a call comes in. Then on terminal do a "asterisk -r" and you will see the NoOp show the DTMF number entered. From there on you can do anything you want with the variable ${numb} If any part of above is unclear to you, you must consult your friend, google, for examples of Asterisk dialplan. -Bruce On Sat, Jul 10, 2010 at 2:38 PM, Eyal Goltzman wrote: Thanks, but I'm missing something here, the dial command is where? I need to do something like: Dial(1234) Read(1 digit) DoSomthing(based on digit from 1234) And as far as I understand the Dial start the call and only come back (ig you use the g option) after call finished. Eyal From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce Sent: Saturday, July 10, 2010 9:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How can get user inputs from called party after dial? You need read(): http://www.voip-info.org/wiki/view/Asterisk+cmd+Read It's as easy as: exten => s,n,Read(variable,,11) exten => s,n,NoOp(${variable}) Above will take up to 11 digits input by user and will display it back in NoOP on Asterisk CLI. -Bruce On Sat, Jul 10, 2010 at 2:16 PM, eyal goltzman wrote: Hi, I want to dial a party, play him a message and wait for his input, i.e. DTMF digits and use them to control the rest of the dial plan. How do I do it? If I use Dial it will not return until the end of the call, isn't it? Thanks, Eyal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - www.avg.com Version: 9.0.830 / Virus Database: 271.1.1/2991 - Release Date: 07/10/10 09:36:00 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - www.avg.com Version: 9.0.830 / Virus Database: 271.1.1/2991 - Release Date: 07/10/10 09:36:00 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users