Re: [asterisk-users] conference bridge
Nhadie Ramos wrote: Hi, How can i setup conference when i have 2 asterisk servers? my setup is 2 asterisk servers using realtime, i'm simply using DNS SRV just for redundancy (not really high availability). i have a web interface, wherein i can create extension, conference etc. adding extension is ok, even if ext1 is registered on Asterisk 1 and ext2 is registered on asterisk 2 they will still be able to call each other, but on the conference, e.g. when ext1 dials conference no. 1000 and ext 2 dials conf 1000 also, they will be connected to two different conference room. my meetme is also setup on realtime. how can i set it up in such a way ext on registered on different asterisk server can connect to the same conference room. Build a SIP trunk between them, and have an extension in a dedicated dial plan context on one of them (the one that will host the shared conference room) that automatically dumps the caller into the MeetMe room when dialed from the other Asterisk server. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about stopping Asterisk
On Sat, Jul 19, 2008 at 10:25:10PM -0400, Alex Balashov wrote: Christian wrote: Hi all, I've installed Asterisk 1.6 on my Ubuntu Hardy system and I also used the make config command at the end of the installation so that Asterisk loads at boot. However, I want to disable this now. What is the best way of doing this? Many thanks for any help, Christian This is really an Ubuntu question, but: cd /etc/init.d update-rc.d asterisk remove (no need to cd anywhere, and) update-rc.d -f asterisk remove Alternatively, if you used the init.d script from the package, set: RUNASTERISK=no in /etc/default/asterisk . -- 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Bridge Command/Event in 1.6?
20 jul 2008 kl. 02.55 skrev Douglas Garstang: I just downloaded Asterisk 1.6 beta 9 because I had read that there was a new bridge command. After looking through the doc/* documentation, I see no mention of a bridge application or AMI command. Does it exist? I am trying to take a bridged call, and redirect each to another destination, which I can do with the redirect() AMI command. After doing some dial plan processing, I would like to bridge them back together. How can I do this? The redirect command takes a channel and an extension as an argument, not another channel. Read the CHANGES file: * Added a Bridge action which allows you to bridge any two channels that are currently active on the system. The developer forgot to add documentation to doc/manager_1_1.txt. Adding doc would be helpful. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dialplan Action on Authentication
Morning guys and gals, I'd like to be able to run some code when a device (soft/hardphone) authenticates to Asterisk. I remember reading somewhere that there's the possibility of part of a dialplan can be run when a device authenticates. Does anybody have a pointer to some documentation or some pointers about the context that can be used when a device authenticates/unauthenticates to Asterisk? I'm looking for some actions to be performed on Client Authentication without using a manual authentication (using VMAuthenticate or AgentLogin). Environment: Asterisk: 1.4.20 Clients: Soft (mostly Zoiper) Hardphones (Atcom-530's) using IAX2 Thanks for any pointers, David ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan Action on Authentication
David Ashwood wrote: Morning guys and gals, I’d like to be able to run some code when a device (soft/hardphone) authenticates to Asterisk. I remember reading somewhere that there’s the possibility of part of a dialplan can be run when a device authenticates. Does anybody have a pointer to some documentation or some pointers about the context that can be used when a device authenticates/unauthenticates to Asterisk? There is no such possibility. SIP registration, challenge and authentication are all internal protocol events, not calls. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan Action on Authentication
Ok - thanks for the prompt answer Alex. I thought something might be available under the associated context connected with the IAX registration. So the only approach dealing with registrations would be a script running listening to manager events? Regards, David -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: 20 July 2008 12:22 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dialplan Action on Authentication David Ashwood wrote: Morning guys and gals, I'd like to be able to run some code when a device (soft/hardphone) authenticates to Asterisk. I remember reading somewhere that there's the possibility of part of a dialplan can be run when a device authenticates. Does anybody have a pointer to some documentation or some pointers about the context that can be used when a device authenticates/unauthenticates to Asterisk? There is no such possibility. SIP registration, challenge and authentication are all internal protocol events, not calls. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan Action on Authentication
On Sun, Jul 20, 2008 at 11:31 AM, David Ashwood [EMAIL PROTECTED] wrote: Ok - thanks for the prompt answer Alex. I thought something might be available under the associated context connected with the IAX registration. So the only approach dealing with registrations would be a script running listening to manager events? If you configured Asterisk to use realtime you could set up a database trigger such that when Asterisk updated the FullContact field, i.e. registered, you could process some logic. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan Action on Authentication
Hi David, Am Sonntag, den 20.07.2008, 11:57 +0200 schrieb David Ashwood: Morning guys and gals, I’d like to be able to run some code when a device (soft/hardphone) authenticates to Asterisk. I remember reading somewhere that there’s the possibility of part of a dialplan can be run when a device authenticates. Does anybody have a pointer to some documentation or some pointers about the context that can be used when a device authenticates/unauthenticates to Asterisk? I’m looking for some actions to be performed on Client Authentication without using a manual authentication (using VMAuthenticate or AgentLogin). As Alex said, it is impossible to do this from dialplan. But maybe it is possible for You to use the manager API. On the manager interface there is an event fired, whenever a peer (SIP or IAX) registers. So it should be possible to logon to the manager interface, wait for the event and do some action. If You want go back to the daiplan, you can originate a call to a local channel when the event occurs. HTH, Karsten ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan Action on Authentication
Hi Karsten, Thanks - that's just the approach I've taken and appears to be the most direct approach. I've a simple php script that wraps up a telnet interface with a little parsing and, while it needs more debugging exception handling, it appears to be working. Thanks, David -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Karsten Wemheuer Sent: 20 July 2008 14:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dialplan Action on Authentication Hi David, Am Sonntag, den 20.07.2008, 11:57 +0200 schrieb David Ashwood: Morning guys and gals, I?d like to be able to run some code when a device (soft/hardphone) authenticates to Asterisk. I remember reading somewhere that there?s the possibility of part of a dialplan can be run when a device authenticates. Does anybody have a pointer to some documentation or some pointers about the context that can be used when a device authenticates/unauthenticates to Asterisk? I?m looking for some actions to be performed on Client Authentication without using a manual authentication (using VMAuthenticate or AgentLogin). As Alex said, it is impossible to do this from dialplan. But maybe it is possible for You to use the manager API. On the manager interface there is an event fired, whenever a peer (SIP or IAX) registers. So it should be possible to logon to the manager interface, wait for the event and do some action. If You want go back to the daiplan, you can originate a call to a local channel when the event occurs. HTH, Karsten ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about stopping Asterisk
On 2008-07-20 at 10:49 Tzafrir Cohen wrote: On Sat, Jul 19, 2008 at 10:25:10PM -0400, Alex Balashov wrote: Christian wrote: Hi all, I've installed Asterisk 1.6 on my Ubuntu Hardy system and I also used the make config command at the end of the installation so that Asterisk loads at boot. However, I want to disable this now. What is the best way of doing this? Many thanks for any help, Christian This is really an Ubuntu question, but: cd /etc/init.d update-rc.d asterisk remove (no need to cd anywhere, and) update-rc.d -f asterisk remove Alternatively, if you used the init.d script from the package, set: RUNASTERISK=no in /etc/default/asterisk . -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir Many thanks for that info. I don't have anything in /etc/default/asterisk so I will have to use the first method. What script are you refering to? How can I install that instead? Since I only want to do this temporary. Best regards and thanks, Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 48, Issue 58
Hola, Estoy de vacaciones hasta el 1 de Agosto. Para dar soporte sobre la centralita de telefonia: [EMAIL PROTECTED] Perdonen las molestias. Ruth Llaneza Lapausa - Tecnico de VoIP. [EMAIL PROTECTED] Tlf: 902 199 384 Mildmac SA � www.mildmac.es � [EMAIL PROTECTED] C/ Hnos. Garc�a Noblejas 41, 6� planta. 28037 - Madrid Tlf: +34 91 501 33 02 Fax: +34 91 501 57 45 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Explication for ast_safe_system
Never noticed the c is next to the n, however thinking about it, the c is to the f what the n is to the j. which might make for an easy mistake. I guess if you put only your right hand on the keyboard, and mistake the f for the j on your right index finger that it could happen easyly. On Sat, Jul 19, 2008 at 7:41 AM, Eric Dantie [EMAIL PROTECTED] wrote: Can someone please explain the reason on the following code (in asterisk.c, function ast_safe_system()): /* Close file descriptors and launch system command */ for (x = STDERR_FILENO + 1; x 4096; x++) close(x); Why to close so many descriptors? Thanks in advance Éric ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Magnetic door locks
Yes I have done it thanks to mikesendman just put it on an fxs port: http://www.sandman.com/pdf/page40.pdf I believe its the universal ring relay. Call him he'll help you. On Thu, Jul 17, 2008 at 8:43 AM, c james [EMAIL PROTECTED] wrote: I have an opportunity to interface asterisk with a security system to open their magnetic door locks. The security system needs a dry contact close upon activation to signal the door. Has anyone done this before? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue() AGI Bug ?
The docs state that the AGI is run when the caller is connected but this does not appear to be true with 1.4.21.1 What I see is 1) caller enters queue 2) agent is found for call 3) agent1's call begins to ring 4) AGI is executed 5) agent does not answer the call before timeout, call goes to next agent 6) agent2 answers call but the AGI has already run Expected behaviour 1) caller enters queue 2) agent is found for call 3) agent1's call begins to ring 4) agent does not answer the call before timeout, call goes to next agent 5) agent2 answers call but the AGI has already run 6) AGI is executed I need the AGI to run when the actual call is connected to an agent as my AGI is tracking which agent took the call to then fire of a jabber message to that agent giving them them the url to access the caller's account page. Currently the message is going to agent1 and agent2 who actually takes the call never sees the message -- Will Tatam *** Unite against human rights abuse in the 'war on terror' http://www.unsubscribe-me.org Amnesty International ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.21.1 SIP failing, requiring reboot
David Nedved wrote: Interestingly enough, I've had my Grandstream suffering from the same problem since I upgraded to 1.4.20, although my config is static rather than realtime. I'd actually written it off to typical Grand-heap-of-$#!+-stream behaviour. :) I didn't say because I wanted my original email to limit itself to facts I was sure of, but I think my SIP problems started with 1.4.20 as well. I'm fairly sure 1.4.19 was solid... going back today. It looks like someone at bugs.digium has found what it was, so a fix should be coming soon. PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT Astricon/Digium Beach Ball Mailing
On Sat, Jul 19, 2008 at 03:40:46AM -0400, Alex Balashov wrote: Steve Totaro wrote: I post this not to put down Digium, the thought was nice, I wish I could play with my Digium beach ball, but Digium should know about it if it was common. Postage alone was costly. I mean this without a hint of sarcasm or derision toward you or Digium, but: Award for ... most bewildering asterisk-users list post ever! :-) Makes perfect sense to me. Matt F had one of them at our local users group meetup last week. They forgot to put the city on them, though at least they had the full date. They did seem kind of cheaply made... Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] New Bridge App/AMI Command in Asterisk 1.6?
I just downloaded Asterisk 1.6 beta 9 because I had read that there was a new bridge command. After looking through the doc/* documentation, I see no mention of a bridge application or AMI command. Does it exist? I am trying to take a bridged call, and redirect each to another destination, which I can do with the redirect() AMI command. After doing some dial plan processing, I would like to bridge them back together. How can I do this? The redirect command takes a channel and an extension as an argument, not another channel. Doug. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Required an Auto Dialing Solution
Hello Dears We are providing route testing services to a Calling Card company. We need an Auto Dialing solution to test A - Z destinations for a carrier. We need functionality to feed carrier details and upload CSV file containing test numbers of all destinations. Reports should have following details. 1) Calls connected and attended by IVR or Human voice. 2) Calls not connected. (Its shows that carrier route is not working) 3) Calls ringed but not attended. 3) Calls connected during ringing (Its shows false billing of Carrier) Please contact if you can provide this solution. Regards, -- Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email: [EMAIL PROTECTED] MSN: [EMAIL PROTECTED] Gmail: [EMAIL PROTECTED] Skype: kashif.naeem 302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
