Re: [asterisk-users] SIP to IAX?
On Thu, Sep 11, 2008 at 8:10 PM, C. Chad Wallace [EMAIL PROTECTED] wrote: At 8:29 AM on 11 Sep 2008, John Millican wrote: Not directly on-topic for this list, but I'd not heard of OpenSIPS before, so I had a look at the website. It looks to be a fork of OpenSER. Does that mean OpenSER development has slowed/ceased, or has the OpenSER project itself morphed into OpenSIPS? Regards, Chris via a quick google:OpenSER is now OpenSIPS www.opensips.org OpenSER continues via OpenSIPS A new name, same project Uhhh, I thought that was Kamailio: www.kamailio.net ...I'm confused. Oh no! While not on-topic for this list the OpenSER thing has been confusing lately. Some company has a trademark on OpenSER. The OpenSER project had to change its name to Kamailio (like the Zaptel-DAHDI issue). Around the same time there were some problems on the Kamailio board. There was plenty of activity on the lists, etc but what I took from it is that Bogdan left Kamailio and forked OpenSIPS. I believe he will continue to commit to both (if they give him commit access to Kamailio back) but OpenSIPS primarily exists for his company (Voice System). Basically if you have a support contract with Voice System you should use OpenSIPS. Otherwise you are free to chose either Kamailio or OpenSIPS for whatever reasons you like. There is no OpenSER anymore. If you are confused you can always just SER (the original from iptel/FOKUS) ;). -- Kristian Kielhofner http://blog.krisk.org ___ -- 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] SIP to IAX?
On Tue, Sep 9, 2008 at 3:34 PM, Darren Sessions [EMAIL PROTECTED] wrote: I would suggest using OpenSIPS with Asterisk and bypass IAX all together for this particular application. An OpenSIPS solution will take care of your traveler's NAT issues (and could handle the registrations) while you used Asterisk for voicemail and whatever else. I've personally used this type of general setup in the past with a great deal of success for remote offices and soft-phones on laptops. _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ OpenSIPS/Kamailio will only help if the OP doesn't want to wait for Asterisk 1.6 to mature and would like to traverse these firewalls using SIP TLS over a non-standard port (which still may not work) to proxy back to a standard pre-1.6 SIP TLS Asterisk system. Their best bet is to use some type of VPN to traverse these firewalls/NATs. IPSEC, OpenVPN, etc. -- Kristian Kielhofner http://blog.krisk.org ___ -- 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] VoIP Users Conference today at 12 Noon EDT
Hi all, The usual suspects will be gathering today at 12 EDT. Join us on the VUC if you have the time: Details: http://VoipUsersConference.org PSTN 1(724) 444-7444 and enter 22622# 1# SIP [EMAIL PROTECTED] DTMF 22622# 1# IRC: #voip-users-conference on Freenode.net RSS: http://feeds.feedburner.com/AstUser ___ -- 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] about application Jack and its runtime
Thanks! You're the best! Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- 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] about application Jack and its runtime
Russell! This time it's really a problem: when I use application Jack I get input and output. When I use functionJACK_HOOK with the same options, just copied from the Jack call, I only get one way. the o-option doesn't work. I connect it to my microphone, sstem:capture_1. So nothing fancy. Just changed the dialplan a few times back and forth to make sure it's not JACK, which is going mad. Do you have an idea, what this might be? Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- 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] about application Jack and its runtime
Some addition... Something I find even stranger is that jack_lsp shows, that the asterisk input AND output ports do exist and ARE CORRECTLY connected. So I should get audio from my microphone and still I don't. Hope that helps... Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- 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] Amazing show uptime
xx-montague-gardens*CLI show uptime System uptime: 38 years, 37 weeks, 4 days, 10 hours, 47 minutes, 11 seconds Amazing. Especially considering: [EMAIL PROTECTED]:/var/log uptime 09:58:14 up 18:42, load average: 0.21, 0.09, 0.02 Steve ___ -- 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] Amazing show uptime
On 09:59, Fri 12 Sep 08, Stephen Davies wrote: xx-montague-gardens*CLI show uptime System uptime: 38 years, 37 weeks, 4 days, 10 hours, 47 minutes, 11 seconds Amazing. Especially considering: [EMAIL PROTECTED]:/var/log uptime 09:58:14 up 18:42, load average: 0.21, 0.09, 0.02 Steve Did ntp/rdate set the clock forward for 38 years right after boot ? -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called 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] Amazing show uptime
On 12 Sep 2008, at 09:20, Michiel van Baak wrote: On 09:59, Fri 12 Sep 08, Stephen Davies wrote: xx-montague-gardens*CLI show uptime System uptime: 38 years, 37 weeks, 4 days, 10 hours, 47 minutes, 11 seconds Amazing. Especially considering: [EMAIL PROTECTED]:/var/log uptime 09:58:14 up 18:42, load average: 0.21, 0.09, 0.02 Steve Did ntp/rdate set the clock forward for 38 years right after boot ? I'd guess the battery on your motherboard has died so it is going back to 1970 at boottime. Watchout, because this can also mean that your BIOS is about to loose all settings too which can cause it to forget how to talk to the harddrive :-( T. ___ -- 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] Outside SIP Caller accessing voivemail
Joseph L. Casale wrote: Now that we have voicemail working, people have asked to be able to dial in externally and be able to access their voicemail. My dial plan is You can either setup a context for just checking voice mail or you can use the following option under the voice mail application: core show application voicemail snip 0 - Jump to the 'o' extension in the current dialplan context. * - Jump to the 'a' extension in the current dialplan context. This application will set the following channel variable upon completion: VMSTATUS - This indicates the status of the execution of the VoiceMail application. The possible values are: SUCCESS | USEREXIT | FAILED So, in your voice mail context you'd have: exten = a,1,VoiceMailMain(@sip) exten = a,n,HangUP() When the user presses the *, they'd be dumped into voice mail main, in the sip context. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] Video on Hold?
On Thu, Sep 11, 2008 at 9:15 PM, Gordon Henderson [EMAIL PROTECTED] wrote: On Thu, 11 Sep 2008, Russell Bryant wrote: [EMAIL PROTECTED] wrote: Is the idea to switch to another video source or stay with the callers camera? An option for both would be nice. I could see a help desk placing a caller in que and a 1-2 min video coming on showing some simple video of how to hook it up. What I had in mind was to play a video stream that went along with the on hold audio. I was going to make it so if a video file was found with the same name as the audio file being played, it would play it. I rather naively tried this :) Well, Echo() was echoing back video and Record() was recording video and audio line this, and Playback() was playing it back, so ... Sounds fantastic :) Out of curiosity - what formats are supported? Any news on 3G video? I remember some time ago there was some weird application level support. Does chan_mobile supports video too? Would it be possible to have 3G adapter and interact with it? This just brings Asterisk to new level :) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- 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] Amazing show uptime
On 12 Sep 2008, at 10:13, Tim Panton wrote: I'd guess the battery on your motherboard has died so it is going back to 1970 at boottime. Watchout, because this can also mean that your BIOS is about to loose all settings too which can cause it to forget how to talk to the harddrive :-( Hmm ___ -- 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] Amazing show uptime
On Fri, Sep 12, 2008 at 11:13 AM, Tim Panton [EMAIL PROTECTED] wrote: I'd guess the battery on your motherboard has died so it is going back to 1970 at boottime. Why do hide the truth, Tim? It's much more likely the motherboard traveled back 38 years in time, is it not? r ___ -- 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] BLF call pickup on Linksys SPA932
2008/9/11 Stefan Schmidt [EMAIL PROTECTED]: Steve Davies schrieb: Thanks for that excellent information - Now does anybody know the XML to provision that field? Normally you take the text on the screen Call Pickup Code and replace space with underscore Call_Pickup_Code ua=na *8# /Call_Pickup_Code Unfortunately Call Pickup Code appears twice in the UI, so this does not work :( Thanks, Steve hello, here you can see the xml config of the phone: http://x.x.x.x/admin/spacfg.xml where x.x.x.x is the ip of your phone. iam not at the office so i cant check the right syntax for the pickup string, but iam sure you will find it there. Thanks for that pointer... I knew that! :) but had forgotten it. Sadly, the Linksys firmware is not clever enough to load its own file: Call_Pickup_Code group=Regional/Vertical_Service_Activation_Codes*37/Call_Pickup_Code Call_Pickup_Code group=SPA932/General*98/Call_Pickup_Code When this is loaded, the group= parameter is ignored, and the first value is written twice, making it impossible to provision the 2nd value :( Ho hum. Steve ___ -- 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] Amazing show uptime
On Fri, Sep 12, 2008 at 10:13:11AM +0100, Tim Panton wrote: On 12 Sep 2008, at 09:20, Michiel van Baak wrote: On 09:59, Fri 12 Sep 08, Stephen Davies wrote: xx-montague-gardens*CLI show uptime System uptime: 38 years, 37 weeks, 4 days, 10 hours, 47 minutes, 11 seconds Amazing. Especially considering: [EMAIL PROTECTED]:/var/log uptime 09:58:14 up 18:42, load average: 0.21, 0.09, 0.02 Steve Did ntp/rdate set the clock forward for 38 years right after boot ? I'd guess the battery on your motherboard has died so it is going back to 1970 at boottime. That would be 1980, right? No. Looks like some signed/unsigned int error somewhere. -- 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
[asterisk-users] Executing dialplan after the call normaly ended
Hi, The Dial command has the g option, voip-info.org says: If the g option is specified, and the called party hangs up before the calling party, then Dial continues execution at priority n+1. and this works well. But I need to continue the execution if the caller hangs up first too. What do I need to do? -- Best regards, Gergomailto:[EMAIL PROTECTED] ___ -- 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] Executing dialplan after the call normaly ended
On Fri, Sep 12, 2008 at 2:35 PM, Gergo Csibra [EMAIL PROTECTED] wrote: Hi, The Dial command has the g option, voip-info.org says: If the g option is specified, and the called party hangs up before the calling party, then Dial continues execution at priority n+1. and this works well. But I need to continue the execution if the caller hangs up first too. What do I need to do? Search for h extension Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- 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] show g729 seems to no longer work in latest 1.4 version. What do I use please?
Thanks Shaun___ -- 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] Dial function, and no telephone line fixed in the fxo port
Hi List; First of all, how can I know that the telephone line is not fixed in the fxo port? Then, if the Dial function used to place a call via the zaptel (via the fxo port), and no telephone line was fixed in the fxo, can I have any returned error to know that the telephone line is not fixed in the fxo? Last issue: if there any allarm (in any file, or via some script) to give notice that the telephone line now is disconnected from the fxo port? All of this because my customer always keep removing the telephone line from the fxo and use it for some other purposes, we need to control this by giving allarms, indications or something to let the use understand the reason of the mistake, specially that people who remove the line, they do not say that they removed it (they say, it is connected). Any advise? Regards Bilal ___ -- 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] Executing dialplan after the call normaly ended
On Fri, Sep 12, 2008 at 7:35 AM, Gergo Csibra [EMAIL PROTECTED] wrote: Hi, The Dial command has the g option, voip-info.org says: If the g option is specified, and the called party hangs up before the calling party, then Dial continues execution at priority n+1. and this works well. But I need to continue the execution if the caller hangs up first too. What do I need to do? -- Best regards, Gergomailto:[EMAIL PROTECTED] You can use the h extension to continue. Retaining channel variables is iffy. I have an application where I use the h extension to execute System() which still has the channel variables from the call, to create a .call file and drop it into the /outgoing/ directory. It works perfectly. I will probably modify it slightly to create the .call file in a tmp dir and then create a cron job to move it since this seems to be a Best Practice but it works fine the way it is for low volume usage. The above was the only solution to be able to Dial after hangup, while keeping the channel variables. Obviously, Dial on the h exten will create a nasty loop. Goto() another context worked but the OK but dial gets Cancel, I suspect because there is only one leg of a call. Using a local channel for dial, I lost channel variables. I tried using Goto() in the h extension which worked fine but it seems that channel variables disappear after the first exten priority after the h exten. Thanks, Steve Totaro ___ -- 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 1.6.0rc6 make menuselect failed.
Sean Bright wrote: Thomas Kenyon wrote: In trying to upgrade my test machine from 1.6.0beta9 to 1.6.0rc6 when I try to make menuseletc I get the following error. This is using gcc 4.1, libgtk 2.0, on an intel Core2Duo machine running an up to date Debian etch. Asterisk builds okay (not tried running it yet) menuselect_gtk.c: In function ârun_menuâ: menuselect_gtk.c:311: warning: implicit declaration of function Would you mind opening a bug in mantis (http://bugs.digium.com/) and include the config.log in your asterisk source directory as well as the one in the menuselect sub-directory as attachments to the bug? I've seen this problem crop up before and I would like to get it worked out. Thanks, Thanks, I've opened it, id 0013472 http://bugs.digium.com/view.php?id=13472 ___ -- 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] Amazing show uptime
2008/9/12 randulo [EMAIL PROTECTED] On Fri, Sep 12, 2008 at 11:13 AM, Tim Panton [EMAIL PROTECTED] wrote: I'd guess the battery on your motherboard has died so it is going back to 1970 at boottime. Why do hide the truth, Tim? It's much more likely the motherboard traveled back 38 years in time, is it not? Why don't you guys believe that my Asterisk has just been up for 38 years? asterisk -rx 'show version' Asterisk 0.0.1 built by root @ uunet!olsa99!cstat on a ENIAC running No OS At All on 1968-09-11 16:56:34 UTC Steve ___ -- 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 and Fedora 9
The best way I can think of is: wget http://ftp.digium.com/pub/asterisk/asterisk-1.4.21.2.tar.gz tar -zxvf asterisk-1.4.21.2.tar.gz cd asterisk-1.4.21.2 ./configure make menuselect (You don't have to select anything) make make install make samples Pascal Bruno wrote: I am about to install Asterisk on a Fedora 9 box, but i see with yum, they only have Asterisk 1.6 beta in the package repos which I didn't really want to install until they have a stable release. Does anybody know or have a good and easy way to install Asterisk 1.4 on fedora 9? Thank you. ___ -- 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] Amazing show uptime
Stephen Davies wrote: Why don't you guys believe that my Asterisk has just been up for 38 years? Because Mark was born in 1977 and he's 31. http://en.wikipedia.org/wiki/Mark_Spencer Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] Amazing show uptime
Maybe that robot in his office doubles as a time machine. ___ Chris Hoff Telecommunications Administrator SEI LLC Voice +1 701 298 8865 Ext 2189 Mobile +1 701 361 5976 Fax +1 701 298 8860 Email [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Friday, September 12, 2008 8:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Amazing show uptime Stephen Davies wrote: Why don't you guys believe that my Asterisk has just been up for 38 years? Because Mark was born in 1977 and he's 31. http://en.wikipedia.org/wiki/Mark_Spencer Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] Amazing show uptime
Nominated for dumbest thread ever On Fri, Sep 12, 2008 at 9:34 AM, Christopher Hoff [EMAIL PROTECTED] wrote: Maybe that robot in his office doubles as a time machine. ___ Chris Hoff Telecommunications Administrator SEI LLC Voice +1 701 298 8865 Ext 2189 Mobile +1 701 361 5976 Fax +1 701 298 8860 Email [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Friday, September 12, 2008 8:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Amazing show uptime Stephen Davies wrote: Why don't you guys believe that my Asterisk has just been up for 38 years? Because Mark was born in 1977 and he's 31. http://en.wikipedia.org/wiki/Mark_Spencer Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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 ___ -- 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] Amazing show uptime
On Fri, Sep 12, 2008 at 3:24 PM, Doug Lytle [EMAIL PROTECTED] wrote: Stephen Davies wrote: Why don't you guys believe that my Asterisk has just been up for 38 years? Because Mark was born in 1977 and he's 31. Which proves the time travel explanation! ___ -- 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] about application Jack and its runtime
On Thu, Sep 11, 2008 at 08:11:09PM -0500, Russell Bryant wrote: The Jack application acts as an endpoint for a call. A bit of nomenclature: is Jack the name of an Asterisk application? Or are you referring to JACK, the Jack Audio Connection Kit, whose name is all-caps, directly? And if not, of course, is Jack something that connects JACK to Asterisk? And why should I know all of this already? :-) 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
Re: [asterisk-users] about application Jack and its runtime
Julien Claassen wrote: Something I find even stranger is that jack_lsp shows, that the asterisk input AND output ports do exist and ARE CORRECTLY connected. So I should get audio from my microphone and still I don't. Hope that helps... Can you share the dialplan that you're using? That may help me understand the audio path involved .. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- 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] PRI auto-configure - continued from DEV list
Tzafrir Cohen wrote: I usually configure the entire span of 24 channels (23 B + 1 D) and only the turned up channels go into service. This is good for a couple of reasons. Also note that Zaptel will anyway reserve all the 24 (for T1) or 31 (for E1) Zaptel channels for the span. So the Zaptel channel numbers will not change whether the span is fractional or full. What do you mean by reserve? Seriously, I'm trying to get a good grasp. I have always assumed that the signal presented by the Adtran TSU120e appears as a full 24 channels. But it was not clear to me how those channels are transformed on the TDM side of the fork, if at all, by the Adtran. I supposed they might be remapped within the frames. But thinking (out loud) about it some more, I realize that remapping any of the channel positions would likely invalidate some references embedded within the Q.931 data stream on the D channel, vastly complicating the process by requiring the Adtran to be aware of the content structure at a protocol layer that would otherwise be unnecessary. So I suppose that it almost certainly does not remap these channels. In fact, the nature of this animal is such that I suppose for a PRI, each entire frame could be passed to the TDM side unmodified and it would work just fine, with the PBX ignore the IP channels. And following this same line of reasoning, the zaptel code would have little need to be told through its configuration which B channels are available because such information is implicitly available via Q.931 - and thus the channels specifications in zaptel.conf serve only to restrict usage. Have I got this right? Steve, Thanks, I like this idea, and I appreciate the tip. I will try it. Meanwhile, I'm finding from others' comments that it is extremely common to find the D channel on 24, which is primarily what concerned me - and my inability to divine this precisely in my case led to my suggestion/inquiry on the dev list. I've seen enough docs that indicate that the D channel could be anywhere in the group, also implying that it's not unlikely to be at 13 or 6, IIRC. I have visions of sitting in a lonely room repeatedly editing zaptel/zapata.conf and smacking it again, and again... Please give a list of variables. At least the ones you can think of. I guess you are referring to variables in the broadest sense, as I was, so to wit... Having never attached asterisk to a T1, I have no working reference system, and I don't have a personal finite checklist of completion items. So not knowing what I don't know is the biggest variable! But I have placed configuration info in redfone.com, zaptel.conf and zapata.conf (see below). I have built the ethmf module and it loads, and I can observe a stream of data on the designated ethernet interface with tcpdump. It is a bidirectional stream of fixed length blocks that look something like what I might expect, but I have been unable to decipher any content upon superficial inspection. I am supposing it is functioning correctly, but it's validity is still a variable to me, albeit only a small source of doubt. Basic info such as alarm state is definitely getting transmitted, as zttool and the asterisk app are able to detect state changes... When I move the DSX-1 cable from the Nortel box (which works for actual phone calls, so this is not a variable) and I plug it into the redfone TDMoE box, the LED goes from yellow to green, implying that it sees the data (I guess). Similarly, zttool tells me there are no alarms and that I have the number of channels configured as specified in my configuration. It has thus far only indicated that 0 are active, which based on googling, I suppose means 0 live calls established. Now it seems that the only configuration that causes asterisk to start without complaint has been with the D channel on 24. I'll omit detail on this for the moment. Now I am at a point where I can use the pri command to get status. With the cable out, I see this: left*CLI pri show spans PRI span 1/0: Provisioned, In Alarm, Down, Active and with the cable connected, I see this: left*CLI pri show span 1 Primary D-channel: 24 Status: Provisioned, Down, Active Switchtype: National ISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T309 Timer: -1 T313 Timer: 4000 N200 Counter: 3 Note that this cable is ordinarily attached to a Nortel PBX which is fully functioning with the T1 service. Perusing the net, I've decided that the Down status is what I must understand and correct. So the variable is the meaning of Down. Other clues seem to indicate that my box is sending stuff down the line, but hearing nothing in return. But I haven't seen any messages that elaborate. For example, the pri command provides certain trace options which yields stuff
Re: [asterisk-users] about application Jack and its runtime
Jay R. Ashworth wrote: A bit of nomenclature: is Jack the name of an Asterisk application? Or are you referring to JACK, the Jack Audio Connection Kit, whose name is all-caps, directly? And if not, of course, is Jack something that connects JACK to Asterisk? Sorry for the confusion. There is a JACK() application, and JACK_HOOK() function, which both connect Asterisk to JACK, the Jack Audio Connection Kit. And why should I know all of this already? :-) You should be psychic. It's a new 1.6 thing, though, so I don't expect many people to already know all about it. As posted earlier, more info here ... http://www.russellbryant.net/blog/2008/01/13/jack-interfaces-for-asterisk/ -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- 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 and Fedora 9
Ok very good, how about for the asterisk addonds and sounds? Can you provide me the commands to get, build and install for the 1.4.21 version? Thanks a lot guys. On Fri, Sep 12, 2008 at 6:07 AM, MFH [EMAIL PROTECTED] wrote: The best way I can think of is: wget http://ftp.digium.com/pub/asterisk/asterisk-1.4.21.2.tar.gz tar -zxvf asterisk-1.4.21.2.tar.gz cd asterisk-1.4.21.2 ./configure make menuselect (You don't have to select anything) make make install make samples Pascal Bruno wrote: I am about to install Asterisk on a Fedora 9 box, but i see with yum, they only have Asterisk 1.6 beta in the package repos which I didn't really want to install until they have a stable release. Does anybody know or have a good and easy way to install Asterisk 1.4 on fedora 9? Thank you. ___ -- 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 ___ -- 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] about application Jack and its runtime
On Fri, Sep 12, 2008 at 09:04:57AM -0500, Russell Bryant wrote: Jay R. Ashworth wrote: A bit of nomenclature: is Jack the name of an Asterisk application? Or are you referring to JACK, the Jack Audio Connection Kit, whose name is all-caps, directly? And if not, of course, is Jack something that connects JACK to Asterisk? Sorry for the confusion. There is a JACK() application, and JACK_HOOK() function, which both connect Asterisk to JACK, the Jack Audio Connection Kit. Ok; that's rather what I thought. And why should I know all of this already? :-) You should be psychic. It's a new 1.6 thing, though, so I don't expect many people to already know all about it. Got it. As posted earlier, more info here ... http://www.russellbryant.net/blog/2008/01/13/jack-interfaces-for-asterisk/ I'll check it out. Any chance you or someone could chime in one more time on my TBCT thread? 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] Extension not found
Dear All,
I have the following scenario...When a customer dial 111 number a beep
message will iplay in order to record and playback his voice...Else he'll be
routed to another call flow as you can see in the context below:
[a2billing]
exten = _X.,1,Gotoif($[${EXTEN} = 111] ? custom-recordme,111,1)
exten = _X.,2,DeadAGI,a2billing.php
exten = _X.,3,Wait,2
exten = _X.,4,Hangup
But i have the following error when trying to dial 111:
[Sep 12 14:16:32] WARNING[30978]: pbx.c:2483 __ast_pbx_run: Channel
'SIP/michofr-093833e0' sent into invalid extension '111' in context '
custom-recordme', but no invalid handler
Any help?
Regards
___
-- 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] Read one or X DTMF
Hi all I'm just having a problem now and I don't have any idea how to do this. It is pretty simple. When a customer calls, to speed up the navigation in the dialplan, I want something like Welcome. Please enter your 10 digit customer number or press * to register So, I want to read up to 10 digits, and if the user press *, I want to go to the next extension. Do you have an idea ?? ___ -- 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] asterisk 16 and zapata
Asterisk 1.6 installed with last zaptel... On cli, when typing zap show channels, I get No such command 'zap show channels' (type 'help zap show' for other possible commands) Help doesn't help, of course... I have a zaptel conf on the /etc/asterisk... Any Idea? Olivier ___ -- 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] PRI auto-configure - continued from DEV list
On Fri, Sep 12, 2008 at 09:53:48AM -0400, Bill Michaelson wrote: Tzafrir Cohen wrote: I usually configure the entire span of 24 channels (23 B + 1 D) and only the turned up channels go into service. This is good for a couple of reasons. Also note that Zaptel will anyway reserve all the 24 (for T1) or 31 (for E1) Zaptel channels for the span. So the Zaptel channel numbers will not change whether the span is fractional or full. What do you mean by reserve? Seriously, I'm trying to get a good grasp. I have always assumed that the signal presented by the Adtran TSU120e appears as a full 24 channels. But it was not clear to me how those channels are transformed on the TDM side of the fork, if at all, by the Adtran. I supposed they might be remapped within the frames. But thinking (out loud) about it some more, I realize that remapping any of the channel positions would likely invalidate some references embedded within the Q.931 data stream on the D channel, vastly complicating the process by requiring the Adtran to be aware of the content structure at a protocol layer that would otherwise be unnecessary. So I suppose that it almost certainly does not remap these channels. In fact, the nature of this animal is such that I suppose for a PRI, each entire frame could be passed to the TDM side unmodified and it would work just fine, with the PBX ignore the IP channels. And following this same line of reasoning, the zaptel code would have little need to be told through its configuration which B channels are available because such information is implicitly available via Q.931 - and thus the channels specifications in zaptel.conf serve only to restrict usage. Have I got this right? The configuration in zaptel.conf is applied to channels before layer 2 is up. So I guess your question could be rephrased as: do we really need to specify 'bchan' in zaptel.conf? Any way to avoid that? I do need to configure the D channel explicitly. But any way to discover the D channels? -- 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] asterisk 16 and zapata
On Fri, Sep 12, 2008 at 11:07 AM, hh174 [EMAIL PROTECTED] wrote: Asterisk 1.6 installed with last zaptel... On cli, when typing zap show channels, I get No such command 'zap show channels' (type 'help zap show' for other possible commands) Help doesn't help, of course... I have a zaptel conf on the /etc/asterisk... Any Idea? Olivier Zaptel is not loading. What do you see in /var/log/messages? What is the output of ztcfg -vv? Have you recently upgraded the kernel? If so, you need to rebuild and install zaptel against the new kernel. Thanks, Steve Totaro ___ -- 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 16 and zapata
On Fri, Sep 12, 2008 at 11:09 AM, Steve Totaro [EMAIL PROTECTED] wrote: On Fri, Sep 12, 2008 at 11:07 AM, hh174 [EMAIL PROTECTED] wrote: Asterisk 1.6 installed with last zaptel... On cli, when typing zap show channels, I get No such command 'zap show channels' (type 'help zap show' for other possible commands) Help doesn't help, of course... I have a zaptel conf on the /etc/asterisk... Any Idea? Olivier Zaptel is not loading. What do you see in /var/log/messages? What is the output of ztcfg -vv? Have you recently upgraded the kernel? If so, you need to rebuild and install zaptel against the new kernel. Thanks, Steve Totaro Another thought since I don't use 1.6 but could the correct syntax be core zap show channels? Thanks, Steve Totaro ___ -- 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] Amazing show uptime
2008/9/12 Doug Lytle [EMAIL PROTECTED] Stephen Davies wrote: Why don't you guys believe that my Asterisk has just been up for 38 years? Because Mark was born in 1977 and he's 31. Oh dear. Maybe this will help: ;-) :-) Steve ___ -- 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] about application Jack and its runtime
Hello Russell!
Certainly, here's the shortened dialplan:
exten = NUM,1,System(ast_picker ring.wav)
exten = NUM,2,Answer()
exten = NUM,3,GotoIf($[${SYSTEMSTATUS} = SUCCESS]?4:7)
exten = \
NUM,4,Set(JACK_HOOK(manipulate,i(sstem:playback_1)o(system:capture_1)=on)
exten = NUM,5,System(ast_connect)
exten = NUM,6,Goto(8)
exten = NUM,7,VoiceMail(MBOX)
exten = NUM,8,Hangup()
A few notes: ast_picker is my ringing application. It returns 0 exit-code
when someone connected to its socket and -1 (255) otherwise.
ast_connect connects the asterisk jack-output port to system:playback_2 for
stereo sound, it's more pleasant. The Wait(15) part of the plan is still a
problem. I don't know what to call, to keep the call alive until one of the
parties decides to hangup, me with soft hangup CHANNEL for CLI or the other
one in the usual way.
Btw.: This could just make a feature of application Jack, allow for more
then one port to be connected, perhaps like this:
Jack(i(system:playback_1,system:playback_2)o(system:capture_1))
I tried to do the ast_connect from the script I use to place and receive my
calls now, trouble is, I don't know, when the call is established.
Kindest regards
Julien
Music was my first love and it will be my last (John Miles)
FIND MY WEB-PROJECT AT:
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
=== AND MY PERSONAL PAGES AT: ===
http://www.juliencoder.de
___
-- 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] SCCP port numbers used for audio stram?
I have a 7921 wireless phone working with Asterisk, and I want to tighten the wide open port range of my IPTABLES now. I tried allowing only SCCP port (2000) in/out and found that my audio was gone. A quick look at my iptables message shows source port 15886 and dest port 25968 used: FORWARD - Drop: IN=eth1 OUT=eth2 SRC=172.31.253.4 DST=172.31.254.102 LEN=200 TOS=0x18 PREC=0xA0 TTL=63 ID=0 DF PROTO=UDP SPT=15886 DPT=25968 LEN=180 Can anyone tell my 1. which port range I have to open for the audio stream? 2. Is there a way to force SCCP and the phone to use a different port range for audio? Thanks MD ___ -- 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] Amazing show uptime
Stephen Davies wrote: Because Mark was born in 1977 and he's 31. Oh dear. Maybe this will help: ;-) :-) I knew you were joking, maybe I should have added a :=P Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] Which internet phone protocol best to choose
Hello! I'll classify the subject. :-) I have a nasty firewall, I don't have to much power over. It's javascript based in configuration and I can't use any graphical browser. The only other person at my home, doesn't know too much about computers. So I know, from experience, that SIP is ugly in that way, what with freely negotiating port for the real audio stream. So which protocol/technique best to choose to communicate with the rest of the world. Something the glossy windows phones understand, something that's still rather common and if possible, something you can join free of charge. Any suggestion is welcome. If the possibilities are still so numerous: I believe a lot of my friends trust in X-Lite and other apps, gamers also like to use. (little kids :-) ) Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- 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] SCCP port numbers used for audio stram?
On Fri, Sep 12, 2008 at 11:19 AM, OCG Technical Support [EMAIL PROTECTED] wrote: I have a 7921 wireless phone working with Asterisk, and I want to tighten the wide open port range of my IPTABLES now. I tried allowing only SCCP port (2000) in/out and found that my audio was gone. A quick look at my iptables message shows source port 15886 and dest port 25968 used: FORWARD - Drop: IN=eth1 OUT=eth2 SRC=172.31.253.4 DST=172.31.254.102 LEN=200 TOS=0x18 PREC=0xA0 TTL=63 ID=0 DF PROTO=UDP SPT=15886 DPT=25968 LEN=180 Can anyone tell my 1. which port range I have to open for the audio stream? 2. Is there a way to force SCCP and the phone to use a different port range for audio? Thanks MD SCCP (like SIP, MGCP, etc) uses RTP for audio transport. You will need to modify rtp.conf to change the port range Asterisk uses. -- Kristian Kielhofner http://blog.krisk.org ___ -- 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 1.6.0rc6 make menuselect failed.
Thomas Kenyon wrote: Sean Bright wrote: Thomas Kenyon wrote: In trying to upgrade my test machine from 1.6.0beta9 to 1.6.0rc6 when I try to make menuseletc I get the following error. This is using gcc 4.1, libgtk 2.0, on an intel Core2Duo machine running an up to date Debian etch. Asterisk builds okay (not tried running it yet) menuselect_gtk.c: In function ârun_menuâ: menuselect_gtk.c:311: warning: implicit declaration of function Would you mind opening a bug in mantis (http://bugs.digium.com/) and include the config.log in your asterisk source directory as well as the one in the menuselect sub-directory as attachments to the bug? I've seen this problem crop up before and I would like to get it worked out. Thanks, Thanks, I've opened it, id 0013472 http://bugs.digium.com/view.php?id=13472 For the archives, this has been resolved and the fix will appear in the next 1.6.0 release candidate (if applicable) and official release. Thanks, -- Sean Bright [EMAIL PROTECTED] ___ -- 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] SCCP port numbers used for audio stram?
SCCP (aka Skinny), H323, MGCP, and SIP all use the RTP protocol for audio. For all signalling protocols (except maybe H323) use rtp.conf for the RTP ports. OCG Technical Support wrote: I have a 7921 wireless phone working with Asterisk, and I want to tighten the wide open port range of my IPTABLES now. I tried allowing only SCCP port (2000) in/out and found that my audio was gone. A quick look at my iptables message shows source port 15886 and dest port 25968 used: FORWARD - Drop: IN=eth1 OUT=eth2 SRC=172.31.253.4 DST=172.31.254.102 LEN=200 TOS=0x18 PREC=0xA0 TTL=63 ID=0 DF PROTO=UDP SPT=15886 DPT=25968 LEN=180 Can anyone tell my 1. which port range I have to open for the audio stream? 2. Is there a way to force SCCP and the phone to use a different port range for audio? Thanks MD ___ -- 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 -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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] PRI TBCT - Practical Experience, Anybody?
Jay R. Ashworth wrote:
On Mon, Sep 08, 2008 at 11:28:13AM -0500, Matthew Fredrickson wrote:
For DMS100's version of TBCT, called RLT, one leg *must* be inbound and
the other *must* be outbound. No other combination is going to work.
This is explicitly mentioned in the protocol in RLT.
Ok.
Just found this in my archive.
Matt: should I assume that this implies that if my switch is provisioned
for NI2, and my Asterisk is set to DMS, that things aren't going to work
well at all? :-) (Outbound calls, FWIW, seem to work fine like that...)
Probably not. You can obviously try this out, but don't be surprised if
this doesn't work. You usually want to have your switchtype (which
likewise sets the version of TBCT which is used) set to the same thing
that the other end is provisioned to be.
Ok. I've run a simple test:
exten = 727xxx,1,Dial(${TRUNKY}/727yyy,,r)
exten = 727xxx,2,Hangup
Where TRUNKY is a group that points to the same T-1 on which the calls
are coming in.
And what I get is:
-- Accepting call from '727zzz' to '727xxx' on channel 0/1, span 4
-- Executing Dial(Zap/73-1, Zap/g3/727yyy||r) in new stack
-- Requested transfer capability: 0x10 - 3K1AUDIO
-- Called g3/7276471274
-- Zap/74-1 is proceeding passing it to Zap/73-1
-- Zap/74-1 is ringing
-- Zap/74-1 answered Zap/73-1
-- Attempting native bridge of Zap/73-1 and Zap/74-1
-- Channel 0/1, span 4 got hangup request, cause 16
-- Hungup 'Zap/74-1'
== Spawn extension (default, 727xxx, 1) exited non-zero on 'Zap/73-1'
(I think I got all those numbers sanitized properly.)
And yes, the call went through, and had the CNID of the originating
phone, as I want.
So, since I can't tell from the logs -- no timestamps -- I have to guess
from when the messages show up, but I can't tell if the attempted native
bridge is *succeeding*. How would I know that it had? We do
*successful* ones in other contexts, and I don't recall seeing a
'success' message on those.
Will I actually need to do PRI debug on that span to tell?
Or will seeing hangup messages while I'm still talking be the solution?
Seeing hangup messages on the console while the audio path remains
indicates success :-)
--
Matthew Fredrickson
Digium, Inc.
___
-- 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] Read one or X DTMF
Hi, Am Freitag, den 12.09.2008, 11:03 -0400 schrieb Ruddy Gbaguidi: Hi all I'm just having a problem now and I don't have any idea how to do this. It is pretty simple. When a customer calls, to speed up the navigation in the dialplan, I want something like Welcome. Please enter your 10 digit customer number or press * to register So, I want to read up to 10 digits, and if the user press *, I want to go to the next extension. Do you have an idea ?? You can use the read application to get some digits. This application returns the number a user entered in a variable. If the user enters '*' the variable is set to an empty string. You can than proceed in Your dialplan. To distinguish the answers, You can use the function len. The read application is able to play a audio file. (see the doc with 'core show application read') One little hint: If You start a new thread, create a new message instead of using an old one. Your question is now part of the thread about application jack and its runtime, what is probably not what You want. Maybe some people ignore Your mail, because they are not interessted in jack... Regards, 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] Read one or X DTMF
In article [EMAIL PROTECTED], Ruddy Gbaguidi [EMAIL PROTECTED] wrote: Hi all I'm just having a problem now and I don't have any idea how to do this. It is pretty simple. When a customer calls, to speed up the navigation in the dialplan, I want something like Welcome. Please enter your 10 digit customer number or press * to register So, I want to read up to 10 digits, and if the user press *, I want to go to the next extension. Do you have an idea ?? One possibility: [getnumber] exten = s,1,Background(please-enter-num-or-star) exten = s,n,Waitexten(30) exten = *,1,Goto(register,s,1) exten = _X*,1,Goto(register,s,1) exten = _XX*,1,Goto(register,s,1) exten = _XXX*,1,Goto(register,s,1) exten = _*,1,Goto(register,s,1) exten = _X*,1,Goto(register,s,1) exten = _XX*,1,Goto(register,s,1) exten = _XXX*,1,Goto(register,s,1) exten = _*,1,Goto(register,s,1) exten = _X*,1,Goto(register,s,1) exten = _XX,1,Do whatever exten = _XX,n,You want to do with exten = _XX,n,A 10-digit customer number [register] exten = s,1,Start registration process Hope that helps Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- 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] SCCP - max lines per phone limit
I'm setting up a 7921 and now want to add a second line to the phone. In my SCCP.conf file I have: autologin = 235,299 However, on reloading SCCP the phone fails to login to the second line with this error: [Sep 12 12:46:49] WARNING[12224]: sccp_actions.c:185 sccp_handle_register: SEP001BD457F8B1: Failed to autolog into 299: Max available lines phone limit reached 299 Is there a setting to tell Asterisk how many lines to permit per phone? (The 7921 should allow for 6 lines according to the manual) Thanks MD ___ -- 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] PRI TBCT - Practical Experience, Anybody?
On Fri, Sep 12, 2008 at 10:56:40AM -0500, Matthew Fredrickson wrote: Will I actually need to do PRI debug on that span to tell? Or will seeing hangup messages while I'm still talking be the solution? Seeing hangup messages on the console while the audio path remains indicates success :-) Then, as I suspected, I'm failing. I need to confirm that it's actually provisioned with the carrier, and which switchtype I'm really on. Can *you* confirm, off hand, that 1.2 would do TBCT at *all*? Someone on IRC thinks it wouldn't. -- j -- 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] Encrypted IP phone compatible with Asterisk
Dear, I'm looking for IP phones (directly connected to the RJ-45 port from my LAN) that support any level of encryption for use with an Asterisk 1.4 SIP server we have. What branch and type can I use What is the encryption mechanism I can have with this equipments ??? Greetings Alejandro ___ -- 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] PRI TBCT - Practical Experience, Anybody?
Jay R. Ashworth wrote: On Fri, Sep 12, 2008 at 10:56:40AM -0500, Matthew Fredrickson wrote: Will I actually need to do PRI debug on that span to tell? Or will seeing hangup messages while I'm still talking be the solution? Seeing hangup messages on the console while the audio path remains indicates success :-) Then, as I suspected, I'm failing. I need to confirm that it's actually provisioned with the carrier, and which switchtype I'm really on. Can *you* confirm, off hand, that 1.2 would do TBCT at *all*? Someone on IRC thinks it wouldn't. It will only attempt it for DMS100 switchtype. You must have 1.4 libpri for any other switchtype. Matthew Fredrickson ___ -- 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] Setup speed dials on Cisco 7921
I've added lines like this: speeddial = 123,test speeddial = 260,Bob in the [device] section for my 7921, but the speed dials do NOT appear on the menu (click right from the main screen). Am I missing something obvious here? Thanks MD ___ -- 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] Read one or X DTMF
Thanks for the hint. Sorry about that. If I use your soution, I cannot make any difference between a user pressing * and a user that reach the timeout because he didn't enter any digit. In both cases, I will have an empty string Karsten Wemheuer wrote: Hi, Am Freitag, den 12.09.2008, 11:03 -0400 schrieb Ruddy Gbaguidi: Hi all I'm just having a problem now and I don't have any idea how to do this. It is pretty simple. When a customer calls, to speed up the navigation in the dialplan, I want something like Welcome. Please enter your 10 digit customer number or press * to register So, I want to read up to 10 digits, and if the user press *, I want to go to the next extension. Do you have an idea ?? You can use the read application to get some digits. This application returns the number a user entered in a variable. If the user enters '*' the variable is set to an empty string. You can than proceed in Your dialplan. To distinguish the answers, You can use the function len. The read application is able to play a audio file. (see the doc with 'core show application read') One little hint: If You start a new thread, create a new message instead of using an old one. Your question is now part of the thread about application jack and its runtime, what is probably not what You want. Maybe some people ignore Your mail, because they are not interessted in jack... Regards, 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 Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- 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] Read one or X DTMF
Hi thanks for the hint. That will works I think. But now, if I'm in an AGI script and I want to stay in there and don't want to jump from an extension to other in the dialplan, how can I do it ?? Tony Mountifield wrote: In article [EMAIL PROTECTED], Ruddy Gbaguidi [EMAIL PROTECTED] wrote: Hi all I'm just having a problem now and I don't have any idea how to do this. It is pretty simple. When a customer calls, to speed up the navigation in the dialplan, I want something like Welcome. Please enter your 10 digit customer number or press * to register So, I want to read up to 10 digits, and if the user press *, I want to go to the next extension. Do you have an idea ?? One possibility: [getnumber] exten = s,1,Background(please-enter-num-or-star) exten = s,n,Waitexten(30) exten = *,1,Goto(register,s,1) exten = _X*,1,Goto(register,s,1) exten = _XX*,1,Goto(register,s,1) exten = _XXX*,1,Goto(register,s,1) exten = _*,1,Goto(register,s,1) exten = _X*,1,Goto(register,s,1) exten = _XX*,1,Goto(register,s,1) exten = _XXX*,1,Goto(register,s,1) exten = _*,1,Goto(register,s,1) exten = _X*,1,Goto(register,s,1) exten = _XX,1,Do whatever exten = _XX,n,You want to do with exten = _XX,n,A 10-digit customer number [register] exten = s,1,Start registration process Hope that helps Cheers Tony Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- 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] PRI TBCT - Practical Experience, Anybody?
On Fri, Sep 12, 2008 at 12:12:56PM -0500, Matthew Fredrickson wrote: Can *you* confirm, off hand, that 1.2 would do TBCT at *all*? Someone on IRC thinks it wouldn't. It will only attempt it for DMS100 switchtype. You must have 1.4 libpri for any other switchtype. Will libpri 1.4 work with asterisk 1.2? 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
Re: [asterisk-users] SCCP - max lines per phone limit
On 12:51, Fri 12 Sep 08, OCG Technical Support wrote: I'm setting up a 7921 and now want to add a second line to the phone. In my SCCP.conf file I have: autologin = 235,299 However, on reloading SCCP the phone fails to login to the second line with this error: [Sep 12 12:46:49] WARNING[12224]: sccp_actions.c:185 sccp_handle_register: SEP001BD457F8B1: Failed to autolog into 299: Max available lines phone limit reached 299 You are better off asking on the chan_sccp mailinglist. Asterisk has chan_skinny which works differently in assigning lines to devices. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called 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] Extension not found
Hi Michel,
Am Freitag, den 12.09.2008, 17:41 +0300 schrieb michel freiha:
Dear All,
I have the following scenario...When a customer dial 111 number a beep
message will iplay in order to record and playback his voice...Else
he'll be routed to another call flow as you can see in the context
below:
[a2billing]
exten = _X.,1,Gotoif($[${EXTEN} = 111] ? custom-recordme,111,1)
exten = _X.,2,DeadAGI,a2billing.php
exten = _X.,3,Wait,2
exten = _X.,4,Hangup
But i have the following error when trying to dial 111:
[Sep 12 14:16:32] WARNING[30978]: pbx.c:2483 __ast_pbx_run: Channel
'SIP/michofr-093833e0' sent into invalid extension '111' in context '
custom-recordme', but no invalid handler
The above dialplan sends the call into context custom-recordme with
extension 111 and to priority 1, if the caller dials 111. For further
help we would need that context too.
Regards,
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] Setup speed dials on Cisco 7921
On 13:15, Fri 12 Sep 08, OCG Technical Support wrote: I've added lines like this: speeddial = 123,test speeddial = 260,Bob in the [device] section for my 7921, but the speed dials do NOT appear on the menu (click right from the main screen). Am I missing something obvious here? chan_skinny or chan_sccp ? -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called 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] echo cancellation problem with dahdi
I am having problems with echo cancel using dahdi and latest (as of Saturday) version of asterisk 1.4. The problem only occurs between zap and sip or iax. The far end gets an echo. I can even get it by calling my own analog phone hooked up to an ata! Zap to Zap is just fine. Here is my /etc/asterisk/chan_dahdi.conf . Any assistance would be appreciated. ; ; Zapata telephony interface ; ; Configuration file ; ; You need to restart Asterisk to re-configure the Zap channel ; CLI reload chan_zap.so ; will reload the configuration file, ; but not all configuration options are ; re-configured during a reload. [trunkgroups] ; ; Trunk groups are used for NFAS or GR-303 connections. ; ; Group: Defines a trunk group. ;group = trunkgroup,dchannel[,backup1...] ; ;trunkgroup is the numerical trunk group to create ;dchannelis the zap channel which will have the ;d-channel for the trunk. ;backup1 is an optional list of backup d-channels. ; ;trunkgroup = 1,24,48 ; ; Spanmap: Associates a span with a trunk group ;spanmap = zapspan,trunkgroup[,logicalspan] ; ;zapspan is the zap span number to associate ;trunkgroup is the trunkgroup (specified above) for the mapping ;logicalspan is the logical span number within the trunk group to use. ;if unspecified, no logical span number is used. ; ;spanmap = 1,1,1 ;spanmap = 2,1,2 ;spanmap = 3,1,3 ;spanmap = 4,1,4 ;[channels] ; ; Default language ; ;language=en ; ; Default context ; ;context=default ; ; Switchtype: Only used for PRI. ; ; national: National ISDN 2 (default) ; dms100: Nortel DMS100 ; 4ess: ATT 4ESS ; 5ess: Lucent 5ESS ; euroisdn: EuroISDN ; ni1:Old National ISDN 1 ; ;switchtype=national ; ; Some switches (ATT especially) require network specific facility IE ; supported values are currently 'none', 'sdn', 'megacom', 'accunet' ; ;nsf=none ; ; PRI Dialplan: Only RARELY used for PRI. ; ; unknown:Unknown ; private:Private ISDN ; local: Local ISDN ; national: National ISDN ; international: International ISDN ; ;pridialplan=national ; ; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's numbering plan) ; ; unknown:Unknown ; private:Private ISDN ; local: Local ISDN ; national: National ISDN ; international: International ISDN ; ;prilocaldialplan=national ; ; PRI callerid prefixes based on the given TON/NPI (dialplan) ; This is especially needed for euroisdn E1-PRIs ; ; sample 1 for Germany ;internationalprefix = 00 ;nationalprefix = 0 ;localprefix = 0711 ;privateprefix = 07115678 ;unknownprefix = ; ; sample 2 for Germany ;internationalprefix = + ;nationalprefix = +49 ;localprefix = +49711 ;privateprefix = +497115678 ;unknownprefix = ; ; PRI resetinterval: sets the time in seconds between restart of unused channels, defaults to 3600 ; minimum 60 seconds ; some PBXs don't like channel restarts. so set the interval to a very long interval e.g. 1 ; or 'never' to disable *entirely*. ; ;resetinterval = 3600 ; ; Overlap dialing mode (sending overlap digits) ; ;overlapdial=yes ; ; PRI Out of band indications. ; Enable this to report Busy and Congestion on a PRI using out-of-band ; notification. Inband indication, as used by Asterisk doesn't seem to work ; with all telcos. ; ; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT ; inband: Signal Busy/Congestion using in-band tones ; ; priindication = outofband ; ; ISDN Timers ; All of the ISDN timers and counters that are used are configurable. Specify ; the timer name, and its value (in ms for timers) ; ; pritimer = t200,1000 ; pritimer = t313,4000 ; ; ; Signalling method (default is fxs). Valid values: ; em: E M ; em_w:E M Wink ; featd: Feature Group D (The fake, Adtran style, DTMF) ; featdmf: Feature Group D (The real thing, MF (domestic, US)) ; featb: Feature Group B (MF (domestic, US)) ; fxs_ls: FXS (Loop Start) ; fxs_gs: FXS (Ground Start) ; fxs_ks: FXS (Kewl Start) ; fxo_ls: FXO (Loop Start) ; fxo_gs: FXO (Ground Start) ; fxo_ks: FXO (Kewl Start) ; pri_cpe: PRI signalling, CPE side ; pri_net: PRI signalling, Network side ; gr303fxoks_net: GR-303 Signalling, FXO Loopstart, Network side ; gr303fxsks_cpe: GR-303 Signalling, FXS Loopstart, CPE side ; sf: SF (Inband Tone) Signalling ; sf_w: SF Wink ; sf_featd: SF Feature Group D (The fake, Adtran style, DTMF) ; sf_featdmf: SF Feature Group D (The real thing, MF (domestic, US)) ; sf_featb: SF Feature Group B (MF (domestic, US)) ; The following are used for Radio interfaces: ; fxs_rx: Receive audio/COR on an FXS kewlstart interface (FXO at the channel bank) ; fxs_tx: Transmit audio/PTT on an FXS loopstart interface (FXO at the channel bank) ; fxo_rx: Receive audio/COR on an FXO
Re: [asterisk-users] Amazing show uptime
Does your box run on the Mr. Fusion power supply? Doug Lytle wrote: Stephen Davies wrote: Because Mark was born in 1977 and he's 31. Oh dear. Maybe this will help: ;-) :-) I knew you were joking, maybe I should have added a :=P 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
Re: [asterisk-users] Read one or X DTMF
Hi Ruddy, Am Freitag, den 12.09.2008, 13:22 -0400 schrieb Ruddy Gbaguidi: Thanks for the hint. Sorry about that. If I use your soution, I cannot make any difference between a user pressing * and a user that reach the timeout because he didn't enter any digit. In both cases, I will have an empty string You can use the variable EPOCH to get a timestamp before and after execution of the read application. If the difference of the two values evaluates to the timeout, the user enters nothing. Otherwise the user enters '*#' or directly the #-key without anything more. I don't know how to distinguish this two cases. Regards, 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] Setup speed dials on Cisco 7921
Chan_sccp again... From what I read chan_sccp is the successor to chan_skinny. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: September 12, 2008 2:08 PM To: Asterisk Users List Subject: Re: [asterisk-users] Setup speed dials on Cisco 7921 On 13:15, Fri 12 Sep 08, OCG Technical Support wrote: I've added lines like this: speeddial = 123,test speeddial = 260,Bob in the [device] section for my 7921, but the speed dials do NOT appear on the menu (click right from the main screen). Am I missing something obvious here? chan_skinny or chan_sccp ? -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called 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 ___ -- 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] Transfer via AMI
I have a call between two people. I know their channel identifier. I want to trasfer a call away from one person and pass it to another person. To start, let's talk about a blind transfer. My system places both outgoing calls to people and bridges them together (cheaper, works via AGI). Action: Redirect Channel: prospect ExtraChannel: 0 Exten: SIP/transfer_to Context: default Priority: 1 So that works just fine. I'm having an issue however that when the person who was orginally talking decides to hang up his call, Asterisk disconnects the other line as well, as if the ownership of that line is still controled by the orginal process. I'd love to solve that problem. Maybe putting the SIP/transfer_to into the ExtraChannel and then transfering them to a conference room. Suggestions welcome. Could also be that AGI maintains control of any channels it creates and when the main calling line dies, it kills all the others even if they've been transfered away. Okay, in the end, I'd like this to be assisted transfer. Place the party on hold, call another party, and then bridge the two together. Whenever a channel is taken away from the current person, the call status is returned and my AGI script can continue. So I think it should be fine. Has anyone done anything like this? Any pointers would be great. PS: (update since I wrote this original message a while back), via the web, you click a link. That creates a CALL file which calls your number. Once connected, it passes it to an extension that spawns an AGI program. That AGI program looks in the database for the number you wanted to call and places that phone call. You than chat with that person and decide that you're done with that call and want to go onto your next phone call. I use the Asterisk Manager Interface (AMI) to perform a Redirect on the person you're talking to. Doing this causes the AGI script to continue. -- Nicholas Blasgen [EMAIL PROTECTED] 408.497.9796 (c) ___ -- 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] Setup speed dials on Cisco 7921
On 15:37, Fri 12 Sep 08, OCG Technical Support wrote: Chan_sccp again... From what I read chan_sccp is the successor to chan_skinny. No, it's a fork that never contribute back anything to asterisk. The last year there have been activity in chan_skinny again, and I can say it works ok for my home system now. There are some interesting patches on the bugtracker, and I know that at least wedhorn is putting effort into chan_skinny to make it even better. The biggest problem with chan_sccp is that there are already around 4 different branches of it, and they all go another way. None of them is as close to the asterisk development as chan_skinny. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: September 12, 2008 2:08 PM To: Asterisk Users List Subject: Re: [asterisk-users] Setup speed dials on Cisco 7921 On 13:15, Fri 12 Sep 08, OCG Technical Support wrote: I've added lines like this: speeddial = 123,test speeddial = 260,Bob in the [device] section for my 7921, but the speed dials do NOT appear on the menu (click right from the main screen). Am I missing something obvious here? chan_skinny or chan_sccp ? -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called 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 ___ -- 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 -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called 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] SIp Signalling
Is there a way to force asterisk to take care only of sip signaling without forcing it to take care of rtp traffic? ___ -- 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 and Fedora 9
http://www.taylortelephone.com/asterisk/ There are install scripts for Centos 5 Asterisk 1.4. They should work just fine on FC9. If you have a problem just email me. Jonn _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pascal Bruno Sent: Friday, September 12, 2008 9:14 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk and Fedora 9 Ok very good, how about for the asterisk addonds and sounds? Can you provide me the commands to get, build and install for the 1.4.21 version? Thanks a lot guys. On Fri, Sep 12, 2008 at 6:07 AM, MFH [EMAIL PROTECTED] wrote: The best way I can think of is: wget http://ftp.digium.com/pub/asterisk/asterisk-1.4.21.2.tar.gz tar -zxvf asterisk-1.4.21.2.tar.gz cd asterisk-1.4.21.2 ./configure make menuselect (You don't have to select anything) make make install make samples Pascal Bruno wrote: I am about to install Asterisk on a Fedora 9 box, but i see with yum, they only have Asterisk 1.6 beta in the package repos which I didn't really want to install until they have a stable release. Does anybody know or have a good and easy way to install Asterisk 1.4 on fedora 9? Thank you. ___ -- 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 ___ -- 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 16 and zapata
Asterisk 1.6rc4 will only use dahdi. I just went though this on my test system. Jonn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, September 12, 2008 10:12 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk 16 and zapata On Fri, Sep 12, 2008 at 11:09 AM, Steve Totaro [EMAIL PROTECTED] wrote: On Fri, Sep 12, 2008 at 11:07 AM, hh174 [EMAIL PROTECTED] wrote: Asterisk 1.6 installed with last zaptel... On cli, when typing zap show channels, I get No such command 'zap show channels' (type 'help zap show' for other possible commands) Help doesn't help, of course... I have a zaptel conf on the /etc/asterisk... Any Idea? Olivier Zaptel is not loading. What do you see in /var/log/messages? What is the output of ztcfg -vv? Have you recently upgraded the kernel? If so, you need to rebuild and install zaptel against the new kernel. Thanks, Steve Totaro Another thought since I don't use 1.6 but could the correct syntax be core zap show channels? Thanks, Steve Totaro ___ -- 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 and Fedora 9
Pascal Bruno [EMAIL PROTECTED] writes: Ok very good, how about for the asterisk addonds and sounds? Can you provide me the commands to get, build and install for the 1.4.21 version? Thanks a lot guys. If you can't figure that out on your own, you really should stick with the distribution-provided packages. But hey, if you love dealing with dependencies by hand and you don't mind not having a clean upgrade path, feel free to avoid the package manager. /Benny ___ -- 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 16 and zapata
Should have been 1.6.0rc6. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonn R Taylor Sent: Friday, September 12, 2008 4:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk 16 and zapata Asterisk 1.6rc4 will only use dahdi. I just went though this on my test system. Jonn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, September 12, 2008 10:12 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk 16 and zapata On Fri, Sep 12, 2008 at 11:09 AM, Steve Totaro [EMAIL PROTECTED] wrote: On Fri, Sep 12, 2008 at 11:07 AM, hh174 [EMAIL PROTECTED] wrote: Asterisk 1.6 installed with last zaptel... On cli, when typing zap show channels, I get No such command 'zap show channels' (type 'help zap show' for other possible commands) Help doesn't help, of course... I have a zaptel conf on the /etc/asterisk... Any Idea? Olivier Zaptel is not loading. What do you see in /var/log/messages? What is the output of ztcfg -vv? Have you recently upgraded the kernel? If so, you need to rebuild and install zaptel against the new kernel. Thanks, Steve Totaro Another thought since I don't use 1.6 but could the correct syntax be core zap show channels? Thanks, Steve Totaro ___ -- 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 ___ -- 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 and Fedora 9
You wont need things like PHP, MySQL, etc but you do need some of the other things otherwise you'll get errors. And while I run these as automated batches, I suggest you take my commands and do them one line at a time. Keep an eye out for errors. yum -y install kernel kernel-devel ntp yum -y install subversion gcc gcc-c++ libtermcap-devel bison yum -y update ntpdate time.apple.com cd /usr/src svn checkout http://svn.digium.com/svn/zaptel/branches/1.4 zaptel svn checkout http://svn.digium.com/svn/asterisk/branches/1.4 asterisk svn checkout http://svn.digium.com/svn/asterisk-addons/branches/1.4 asterisk-addons cd zaptel; ./configure; make; make install; make config; cd .. cd asterisk; ./configure; make; make install; make samples; cd .. cd asterisk-addons; ./configure --with-mysqlclient=/usr; make; make install; make samples; cd .. On Fri, Sep 12, 2008 at 2:02 PM, Jonn R Taylor [EMAIL PROTECTED]wrote: http://www.taylortelephone.com/asterisk/ There are install scripts for Centos 5 Asterisk 1.4. They should work just fine on FC9. If you have a problem just email me. Jonn -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Pascal Bruno *Sent:* Friday, September 12, 2008 9:14 AM *To:* [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Asterisk and Fedora 9 Ok very good, how about for the asterisk addonds and sounds? Can you provide me the commands to get, build and install for the 1.4.21 version? Thanks a lot guys. On Fri, Sep 12, 2008 at 6:07 AM, MFH [EMAIL PROTECTED] wrote: The best way I can think of is: wget http://ftp.digium.com/pub/asterisk/asterisk-1.4.21.2.tar.gz tar -zxvf asterisk-1.4.21.2.tar.gz cd asterisk-1.4.21.2 ./configure make menuselect (You don't have to select anything) make make install make samples Pascal Bruno wrote: I am about to install Asterisk on a Fedora 9 box, but i see with yum, they only have Asterisk 1.6 beta in the package repos which I didn't really want to install until they have a stable release. Does anybody know or have a good and easy way to install Asterisk 1.4 on fedora 9? Thank you. ___ -- 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 ___ -- 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 -- Nicholas Blasgen [EMAIL PROTECTED] 408.497.9796 (c) ___ -- 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 and Fedora 9
Thanks Jonn!!! On Fri, Sep 12, 2008 at 2:02 PM, Jonn R Taylor [EMAIL PROTECTED]wrote: http://www.taylortelephone.com/asterisk/ There are install scripts for Centos 5 Asterisk 1.4. They should work just fine on FC9. If you have a problem just email me. Jonn -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Pascal Bruno *Sent:* Friday, September 12, 2008 9:14 AM *To:* [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Asterisk and Fedora 9 Ok very good, how about for the asterisk addonds and sounds? Can you provide me the commands to get, build and install for the 1.4.21 version? Thanks a lot guys. On Fri, Sep 12, 2008 at 6:07 AM, MFH [EMAIL PROTECTED] wrote: The best way I can think of is: wget http://ftp.digium.com/pub/asterisk/asterisk-1.4.21.2.tar.gz tar -zxvf asterisk-1.4.21.2.tar.gz cd asterisk-1.4.21.2 ./configure make menuselect (You don't have to select anything) make make install make samples Pascal Bruno wrote: I am about to install Asterisk on a Fedora 9 box, but i see with yum, they only have Asterisk 1.6 beta in the package repos which I didn't really want to install until they have a stable release. Does anybody know or have a good and easy way to install Asterisk 1.4 on fedora 9? Thank you. ___ -- 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 ___ -- 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 and Fedora 9
It's like the same except you wget a different package and I don't think you have a menuselect option and you do it before you compile asterisk. For addons I think there might be some configuration if you are planning to use the database stuff which I don't use. The sounds come with the asterisk install and the menuselect allows you to decide which sounds you want. But, if compiling is foreign to you as someone points out maybe you should not take that route. I don't agree that upgrading is difficult with this process though. Pascal Bruno wrote: Ok very good, how about for the asterisk addonds and sounds? Can you provide me the commands to get, build and install for the 1.4.21 version? Thanks a lot guys. On Fri, Sep 12, 2008 at 6:07 AM, MFH [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: The best way I can think of is: wget http://ftp.digium.com/pub/asterisk/asterisk-1.4.21.2.tar.gz tar -zxvf asterisk-1.4.21.2.tar.gz cd asterisk-1.4.21.2 ./configure make menuselect (You don't have to select anything) make make install make samples Pascal Bruno wrote: I am about to install Asterisk on a Fedora 9 box, but i see with yum, they only have Asterisk 1.6 beta in the package repos which I didn't really want to install until they have a stable release. Does anybody know or have a good and easy way to install Asterisk 1.4 on fedora 9? Thank you. ___ -- 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 ___ -- 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] OpenStage20 Problem
Hi, is anyone Siemens OpenStage 20 SIP phone connected to asterisk 1.4 ? Since V1 R4.11.0 the phone shows Number unavailable each time an outgoing call gets connected. To users this looks like an error message. It is a bit confusing. This problem did not occur when V1 R3 was used, but this had a lot of bugs. -- Stefan Tichy ( asterisk2 at pi4tel dot de ) ___ -- 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] SIp Signalling
Il Neofita wrote: Is there a way to force asterisk to take care only of sip signaling without forcing it to take care of rtp traffic? Yes. The canonical way is to enable canreinvite=yes on both SIP peers (incoming and outgoing legs), which will cause Asterisk to send a new INVITE within the dialog that has updated SDP information corresponding to both endpoints. The more interesting option is newer -- directrtpsetup=yes in sip.conf. This will cause Asterisk to behave more like a proxy does with respect to media and simply pass the SDP payloads as received to both endpoints without pivoting the media stream toward itself at any time, unless explicitly forced to do so (i.e. generating music on hold or IVR messages). Both approaches come with the caveat that the endpoints must be able to address each other directly, so it can't be that one endpoint is behind NAT on a private network that only Asterisk can see and the other endpoint cannot. But if that's taken care of, or you have a far-end NAT traversal solution in place to go with it, then you can do media release on Asterisk. -- Alex -- 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] Read one or X DTMF
In article [EMAIL PROTECTED], Ruddy Gbaguidi [EMAIL PROTECTED] wrote: Hi thanks for the hint. That will works I think. But now, if I'm in an AGI script and I want to stay in there and don't want to jump from an extension to other in the dialplan, how can I do it ?? Ah, you didn't say anything about AGI, so I gave you a solution just using the dialplan. If you are writing an AGI program to do this, you might just as well have a loop around a WAIT FOR DIGIT command and check each digit as it comes, collecting numeric digits until you have ten of them, or jumping to the registration section if you get a *. Cheers Tony Tony Mountifield wrote: In article [EMAIL PROTECTED], Ruddy Gbaguidi [EMAIL PROTECTED] wrote: Hi all I'm just having a problem now and I don't have any idea how to do this. It is pretty simple. When a customer calls, to speed up the navigation in the dialplan, I want something like Welcome. Please enter your 10 digit customer number or press * to register So, I want to read up to 10 digits, and if the user press *, I want to go to the next extension. Do you have an idea ?? One possibility: [getnumber] exten = s,1,Background(please-enter-num-or-star) exten = s,n,Waitexten(30) exten = *,1,Goto(register,s,1) exten = _X*,1,Goto(register,s,1) exten = _XX*,1,Goto(register,s,1) exten = _XXX*,1,Goto(register,s,1) exten = _*,1,Goto(register,s,1) exten = _X*,1,Goto(register,s,1) exten = _XX*,1,Goto(register,s,1) exten = _XXX*,1,Goto(register,s,1) exten = _*,1,Goto(register,s,1) exten = _X*,1,Goto(register,s,1) exten = _XX,1,Do whatever exten = _XX,n,You want to do with exten = _XX,n,A 10-digit customer number [register] exten = s,1,Start registration process Hope that helps Cheers Tony Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- 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 -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- 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] [FreeBSD 6.3/Ports] Make does nothing
Hello I updated the Ports collection to compile the latest Asterisk, but after running make config, make just returns without doing anything: = # pkg_version -v | grep asterisk asterisk-1.4.20.1_1needs updating (port has 1.4.21.2_3) ^C # cd /usr/ports/net/asterisk # make # = There's nothing in /var/log/messages that would explain why this happens. FWIW, Asterisk is currently running on this host, but until I type make deinstall ; make reinstall, I guess it shouldn't be a problem. Any idea why this is happening? Thank you. ___ -- 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] Which internet phone protocol best to choose
http://www.voip-info.org/wiki-IAX http://www.voip-info.org/wiki-IAX+versus+SIP http://www.voip-info.org/wiki/view/Asterisk+IAX+clients Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- 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] Which internet phone protocol best to choose
The short answer is SIP. Stefan Gofferje wrote: http://www.voip-info.org/wiki-IAX http://www.voip-info.org/wiki-IAX+versus+SIP http://www.voip-info.org/wiki/view/Asterisk+IAX+clients Terve, Stefan -- 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] Which internet phone protocol best to choose
I would have said the short answer is IAX :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: September 12, 2008 7:31 PM To: Asterisk Users List Subject: Re: [asterisk-users] Which internet phone protocol best to choose The short answer is SIP. Stefan Gofferje wrote: http://www.voip-info.org/wiki-IAX http://www.voip-info.org/wiki-IAX+versus+SIP http://www.voip-info.org/wiki/view/Asterisk+IAX+clients Terve, Stefan -- 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] Which internet phone protocol best to choose
Hello! IAX I can basically understand, although I wasn't aware in the slightest, that other standard softphones supported it. But why SIP? Correct me if I'm wrong. there's a standard SIP-port. Then you send out the request to talk, then server and client negotiate a port for the audio stream to use. That precisely is, what is bothersome, if you have a difficult firewall. In short: I am up for the longer answer. :-) Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- 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] Which internet phone protocol best to choose
On Friday 12 September 2008 18:31:23 Alex Balashov wrote: The short answer is SIP. Stefan Gofferje wrote: http://www.voip-info.org/wiki-IAX http://www.voip-info.org/wiki-IAX+versus+SIP http://www.voip-info.org/wiki/view/Asterisk+IAX+clients The longer and more accurate answer is that it depends on what you're doing and what your goals are. If you need to be able to pierce firewalls and handle NAT easily, especially in areas where a government-controlled telephone monopoly might be hostile to VoIP, then IAX2 is for you. If, on the other hand, interoperability and choice of phones are what you need, then SIP is pretty much a no-brainer. Of course, nothing prevents you from using SIP for phones and IAX2 for backbone, either. Notably missing to this discussion are the MGCP and H.323 protocols, which probably aren't much use to you, unless you need to connect to certain VoIP providers who haven't yet upgraded their equipment to the 21st century. -- Tilghman ___ -- 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] Which internet phone protocol best to choose
Alex Balashov schrieb: The short answer is SIP. Maybe not behind a firewall which you don't have control over. IAX is a single-port-protocol and as such much less problematic with firewalls and NAT. Read the second link in my previous mail. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- 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] Which internet phone protocol best to choose
Julien Claassen schrieb: IAX I can basically understand, although I wasn't aware in the slightest, that other standard softphones supported it. They don't. Well - it depends, what you see as standard. There are very good multi-platform combined SIP/IAX clients like Zoiper. But Zoiper is not as popular as X-Lite because it wasn't adopted by lots of providers yet. In short: I am up for the longer answer. :-) My short answer contained links to pretty long explanations and a list of clients. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- 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] Which internet phone protocol best to choose
Hello Stefan! Sorry for the miss-understanding. I didn't refer to your mail about IAX, but about the one sayng SIP. I read your links and it seems I'll delve into it. I'll try to quote next time. I hate doing this, it always looks a bit unorganised, while writing... :-( Kindest regards and thanks! Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- 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] Which internet phone protocol best to choose
The OP asked, if I recall, about the protocol which is likely to be supported rather universally by softphones and a wide variety of clients. That is not a feature of IAX. Tilghman Lesher wrote: On Friday 12 September 2008 18:31:23 Alex Balashov wrote: The short answer is SIP. Stefan Gofferje wrote: http://www.voip-info.org/wiki-IAX http://www.voip-info.org/wiki-IAX+versus+SIP http://www.voip-info.org/wiki/view/Asterisk+IAX+clients The longer and more accurate answer is that it depends on what you're doing and what your goals are. If you need to be able to pierce firewalls and handle NAT easily, especially in areas where a government-controlled telephone monopoly might be hostile to VoIP, then IAX2 is for you. If, on the other hand, interoperability and choice of phones are what you need, then SIP is pretty much a no-brainer. Of course, nothing prevents you from using SIP for phones and IAX2 for backbone, either. Notably missing to this discussion are the MGCP and H.323 protocols, which probably aren't much use to you, unless you need to connect to certain VoIP providers who haven't yet upgraded their equipment to the 21st century. -- 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] Which internet phone protocol best to choose
On Fri, Sep 12, 2008 at 7:43 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Friday 12 September 2008 18:31:23 Alex Balashov wrote: The short answer is SIP. Stefan Gofferje wrote: http://www.voip-info.org/wiki-IAX http://www.voip-info.org/wiki-IAX+versus+SIP http://www.voip-info.org/wiki/view/Asterisk+IAX+clients The longer and more accurate answer is that it depends on what you're doing and what your goals are. If you need to be able to pierce firewalls and handle NAT easily, especially in areas where a government-controlled telephone monopoly might be hostile to VoIP, then IAX2 is for you. If, on the other hand, interoperability and choice of phones are what you need, then SIP is pretty much a no-brainer. Of course, nothing prevents you from using SIP for phones and IAX2 for backbone, either. Notably missing to this discussion are the MGCP and H.323 protocols, which probably aren't much use to you, unless you need to connect to certain VoIP providers who haven't yet upgraded their equipment to the 21st century. -- Tilghman I think the most notably missing solution is OpenVPN and SIP. One port for the tunnel, encrypted traffic, benefits of IAX as far as firewalls and hostile governments (BTW, IAX2 is not as obscure as it once was, therefore, the hostile government argument is not as anywhere as strong as a VPN). Since you will be running SIP over the VPN, you get the interoperability that SIP provides. I am sure you could pretty quickly find someone to offer you the gateway side of the VPN for a small charge, or a virtual hosted server should do fine. I have not looked but there may be some VoIP providers that offer or would accommodate OpenVPN tunnels. Thanks, Steve Totaro ___ -- 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] Which internet phone protocol best to choose
Really? I thought both IAX and SIP are, at 3 characters apiece, equally short. However, if you get into IAX2, then yes... SIP is definitely a shorter answer. N. Alex Balashov wrote: The short answer is SIP. Stefan Gofferje wrote: http://www.voip-info.org/wiki-IAX http://www.voip-info.org/wiki-IAX+versus+SIP http://www.voip-info.org/wiki/view/Asterisk+IAX+clients Terve, Stefan ___ -- 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] Read one or X DTMF
But user just needs to enter * instead of *# We are doing this because 80% of the callers already have an account, so, instead of playing : If you have an account, press 1, if not press 2 we prefer to play Enter you account now or press * if you don't have any Karsten Wemheuer wrote: Hi Ruddy, Am Freitag, den 12.09.2008, 13:22 -0400 schrieb Ruddy Gbaguidi: Thanks for the hint. Sorry about that. If I use your soution, I cannot make any difference between a user pressing * and a user that reach the timeout because he didn't enter any digit. In both cases, I will have an empty string You can use the variable EPOCH to get a timestamp before and after execution of the read application. If the difference of the two values evaluates to the timeout, the user enters nothing. Otherwise the user enters '*#' or directly the #-key without anything more. I don't know how to distinguish this two cases. Regards, 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 Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- 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] Append String to CIDNAME
Hello I've been trying to add a string to CIDNAME for incoming calls from
PSTN to tag calls so I know how to answer more appropriately. I have tried
numerous combinations to no avail and hope someone can point me in the right
direction. My context from extensions.conf is listed below.
[xxx-xxx-_incoming]
exten = s,1,Set(CALLERID(name)=${CALLERIDNAME} AppropriateTag)
exten = s,n,Dial(${SIPPHONE},20,r)
exten = s,n,Voicemail(${VMBOX1},u)
exten = s,102,Voicemail(${VMBOX1},b)
All calls however just come in marked as Appropriate Tag xxx-xxx-
Thanks in advance for the help.
___
-- 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] Read one or X DTMF
Thanks for your help. This can be add to Read command as feature Tony Mountifield wrote: In article [EMAIL PROTECTED], Ruddy Gbaguidi [EMAIL PROTECTED] wrote: Hi thanks for the hint. That will works I think. But now, if I'm in an AGI script and I want to stay in there and don't want to jump from an extension to other in the dialplan, how can I do it ?? Ah, you didn't say anything about AGI, so I gave you a solution just using the dialplan. If you are writing an AGI program to do this, you might just as well have a loop around a WAIT FOR DIGIT command and check each digit as it comes, collecting numeric digits until you have ten of them, or jumping to the registration section if you get a *. Cheers Tony Tony Mountifield wrote: In article [EMAIL PROTECTED], Ruddy Gbaguidi [EMAIL PROTECTED] wrote: Hi all I'm just having a problem now and I don't have any idea how to do this. It is pretty simple. When a customer calls, to speed up the navigation in the dialplan, I want something like Welcome. Please enter your 10 digit customer number or press * to register So, I want to read up to 10 digits, and if the user press *, I want to go to the next extension. Do you have an idea ?? One possibility: [getnumber] exten = s,1,Background(please-enter-num-or-star) exten = s,n,Waitexten(30) exten = *,1,Goto(register,s,1) exten = _X*,1,Goto(register,s,1) exten = _XX*,1,Goto(register,s,1) exten = _XXX*,1,Goto(register,s,1) exten = _*,1,Goto(register,s,1) exten = _X*,1,Goto(register,s,1) exten = _XX*,1,Goto(register,s,1) exten = _XXX*,1,Goto(register,s,1) exten = _*,1,Goto(register,s,1) exten = _X*,1,Goto(register,s,1) exten = _XX,1,Do whatever exten = _XX,n,You want to do with exten = _XX,n,A 10-digit customer number [register] exten = s,1,Start registration process Hope that helps Cheers Tony Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- 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 Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- 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
