[asterisk-users] realtime outgoing
Dear, is any solution for replacing .call files into the database? best ___ -- 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] announcement server using asterisk
Hi Can I use Asterisk as an announcement server. We want to build announcement server with ISDN PRI card terminating on our server and announcement being fed on the incoming calls. Also is there any ISDN card available for Laptop. regards Sandesh ___ -- 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] Need Help Regarding Asterisk
Hi,I need a bit of help regarding setting up asterisk. I am trying to setup a simple PBX for a small office we have. We just need 4 extensions. I would like to spent as less as possible. The below are possible solutions i can think off. 1.) I use a Digium Card and connect my PSTN line to the asterisk server. Asterisk would then forward the call to particular extension according to whatever user presses. The extensions would be another set up of analog telephone which would receive the call forward by asterisk. What else (HardWare) i would need except asterisk and digium card...? 2.) I use the digium card and connect my PSTN line to asterisk server. Then it forward calls to 4 pc's (ip addr) which has some kind of software installer (soft phone) and it shows like some call is coming and user can recieve call and talk using headphone.. Again what else i need..? Thanks, Preeteesh ___ -- 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] realtime outgoing
On Sat, Jul 26, 2008 at 12:50:21AM -0700, Pezhman Lali wrote: Dear, is any solution for replacing .call files into the database? This is not likely to work. It would require Asterisk to constantly poll the database. But then again, you can easily originate calls from the manager interface if you want to initiate a call from a remote host. -- 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] Need Help Regarding Asterisk
If you want to connect you pstn only, nothing more, and dont forget that FXO is for the lines. FXS for the Phones 2008/7/26 Preetish Kakkar [EMAIL PROTECTED] Hi,I need a bit of help regarding setting up asterisk. I am trying to setup a simple PBX for a small office we have. We just need 4 extensions. I would like to spent as less as possible. The below are possible solutions i can think off. 1.) I use a Digium Card and connect my PSTN line to the asterisk server. Asterisk would then forward the call to particular extension according to whatever user presses. The extensions would be another set up of analog telephone which would receive the call forward by asterisk. What else (HardWare) i would need except asterisk and digium card...? 2.) I use the digium card and connect my PSTN line to asterisk server. Then it forward calls to 4 pc's (ip addr) which has some kind of software installer (soft phone) and it shows like some call is coming and user can recieve call and talk using headphone.. Again what else i need..? Thanks, Preeteesh ___ -- 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] announcement server using asterisk
Yes you can. As for the bri and laptop cards you have xorcom with usb http://www.xorcom.com/index.php/products/astribank/astribank_models__1/bri_astribank_models and various bri--sip gates on the market. 2008/7/26 ballamudi madhulika [EMAIL PROTECTED] Hi Can I use Asterisk as an announcement server. We want to build announcement server with ISDN PRI card terminating on our server and announcement being fed on the incoming calls. Also is there any ISDN card available for Laptop. regards Sandesh ___ -- 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] announcement server using asterisk
ballamudi madhulika wrote: Can I use Asterisk as an announcement server. We want to build announcement server with ISDN PRI card terminating on our server and announcement being fed on the incoming calls. With the right dialplan and scripts or AGIs, I don't see why not. Shouldn't be a problem at all. Also is there any ISDN card available for Laptop. What type of laptop? (PCMCIA or ExpressCard?) What type of ISDN? (BRI/T1/E1?) ___ -- 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] openSUSE Asterisk Packages
Why not compile from source ? 2008/7/25 Andrew Joakimsen [EMAIL PROTECTED] Does anyone know who maintains the asterisk packages in the openSUSE buildservice? They are not updating Zaptel with their kernel updates and I want to get that matter corrected. I submitted to them a bug report but they seem to not care... https://bugzilla.novell.com/show_bug.cgi?id=407408 ... usually within 24 hours a bugreport is assigned or some sort of comment is made. ___ -- 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] openSUSE Asterisk Packages
On Sat, Jul 26, 2008 at 02:32:28PM +0200, Grygoriy Dobrovolskyy wrote: Why not compile from source ? So it works well as a packagee. So it has proper dependencies on the kernel package isntalled. So you could upgrade kernel and relatively easily upgrade Zaptel. So you could use 'rpm -V' to check for files. -- 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] openSUSE Asterisk Packages
On Fri, Jul 25, 2008 at 12:17:55PM -0400, Andrew Joakimsen wrote: Does anyone know who maintains the asterisk packages in the openSUSE buildservice? They are not updating Zaptel with their kernel updates and I want to get that matter corrected. I submitted to them a bug report but they seem to not care... https://bugzilla.novell.com/show_bug.cgi?id=407408 ... usually within 24 hours a bugreport is assigned or some sort of comment is made. To make the issue more aparant, maybe also provide there the output of the following commands: uname -r find /lib/modules -name tor2.ko rpm -ql zaptel-kmp-pae -- 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] announcement server using asterisk
ballamudi madhulika schrieb: Can I use Asterisk as an announcement server. We want to build announcement server with ISDN PRI card terminating on our server and announcement being fed on the incoming calls. Yes. Also is there any ISDN card available for Laptop. Recently I discovered a cool new site called Google. They have lots of information about ISDN cards. :-P Grüße, Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 ___ -- 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] different gains per channel?
I need to have different gain settings on each channel. Is this easy to achieve? txgain, rxgain and many other parameters are defined on a per-channel basis in zapata.conf, they're not global. Each channel definition channel = x assumes previous definitions of such parameters. Example: txgain=3.0 rxgain=0.0 channel = 1 channel = 2 Both channel 1 and 2 will have the same gains (both previously defined). txgain=3.0 rxgain=0.0 channel = 1 txgain=-4.5 rxgain=0.0 channel = 2 Now you get different gains for channels 1 and 2. Again, as defined before the channel = x definition. Cheers, -- exvito ___ -- 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] announcement server using asterisk
Lol crackup. Having said that here is some help. Don't even think of using a laptop that's just dumb. Next - check out www.voip-info.org you'll find what you need there. Regards, Dean Collins +1-212-203-4357 (Direct) +61-2-9016-5642 (Sydney in-dial) http://www.Cognation.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: Saturday, 26 July 2008 9:57 AM To: Asterisk Users Subject: Re: [asterisk-users] announcement server using asterisk ballamudi madhulika schrieb: Can I use Asterisk as an announcement server. We want to build announcement server with ISDN PRI card terminating on our server and announcement being fed on the incoming calls. Yes. Also is there any ISDN card available for Laptop. Recently I discovered a cool new site called Google. They have lots of information about ISDN cards. :-P Grüße, Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 ___ -- 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] Using manager originate and Dial() once inside dialplan
Hi List, We are trying to make a click 2 call button, we have a PHP script that passes the 1st phone number of the 1st leg to a manager script, that then dials the 1st call, then the 2nd call gets placed inside of Asterisk using a normal dial command. Problem is, we get no status codes, we cannot see if their was a hangup, a answer anything, and also once the callers hangs up, it's killed and cannot execute more commands any dial plan. Any advice on what to do? Is their another way to start a call but from the CLI or something, we just want to dial part 1, if they pick up/press 1 then it dials part 2, then they bridge the call, pretty easy I would think Any help would be great! ___ -- 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] ISDN card available for Laptop (was: Re: announcement server using asterisk)
Philipp Kempgen schrieb: ballamudi madhulika schrieb: Can I use Asterisk as an announcement server. We want to build announcement server with ISDN PRI card terminating on our server and announcement being fed on the incoming calls. Yes. Also is there any ISDN card available for Laptop. Recently I discovered a cool new site called Google. They have lots of information about ISDN cards. :-P Ok, I'll try to post something more helpful. Just pointing you to http://www.google.com/search?q=isdn+pcmcia doesn't make much sense as it depends on what you're looking for. Obviously it can't be too hard to find any ISDN card. I know there are BRI-PCMCIA cards. Not sure about PRI. Even if you could find one I would probably not even give it a try. You should be a bit more specific about what kind of ISDN card you need and what connectors / bus / slots your laptop has to offer. PCMCIA is common but my MacBook doesn't have a PCMCIA slot for example. My PowerBook did. All I wanted to say is that I found card XYZ, can anybody confirm that it's the right one for me / that works with Asterisk / share your experiences is better than I need XY, point me to it (paraphrasing). Grygoriy already named a product from Xorcom (USB) which might be an alternative solution. Grüße, Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 ___ -- 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] Implementing an Asterisk Server behind a MeridianNorstar
On Fri, Jul 25, 2008 at 3:12 PM, Joseph L. Casale [EMAIL PROTECTED] wrote: The question you haven't answered yet, Joseph, is how does your Meridian connect to the PSTN? Is it a T-1 now, or analog? Sorry Jay, I ended up in an offline conversation with someone regarding this. Its on an analogue setup, it has an RJ-21 connector coming from a punchdown block next to it. I think the best way is to use T1 card in the asterisk server and connect to the Meridian with a T1 card as well. The Meridian can gain a T1 interface relatively cheap. I'm still hazy on how a user in the Meridian system would dial an extension for a voip only user on the asterisk server behind it? Thanks! jlc I have done dozens and dozens of jobs like this. You don't want to put Asterisk behind the legacy system, you want to put it in front. Asterisk goes in between the telco and the legacy PBX. If you are getting different advice, then I think the person giving the advice does not have much experience integrating Asterisk with legacy systems. While it may work, it is going to be overly complex and eliminate a huge amount flexibility afforded, than putting Asterisk in the middle. Thanks, Steve 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] Acceptance testing of a new PRI
On Fri, Jul 25, 2008 at 10:57:49PM +0300, Tzafrir Cohen wrote:
On Fri, Jul 25, 2008 at 01:54:21PM -0400, Jay R. Ashworth wrote:
On Thu, Jul 24, 2008 at 04:35:33PM +0300, Tzafrir Cohen wrote:
What's wrong with plain old Zap/NN ?
[test]
exten = _6XXX.,1,Dial(Zap/{EXTEN:1:3}/${EXTEN:4})
Now call 6chan_numnumber-to-dial in context test.
As it happens, Asterisk 1.2 apparently will not recognize 'Zap/01-1' as
the argument to Dial, I get CHANUNAVAIL.
Zap/01-1 ??? How come?
Zap/01 is valid and equivalent to Zap/1 .
And yet, feeding it to Dial didn't work, and stripping the 0 off did.
I'm on 1.2 if that makes a diff.
So I guess I need finally to end up with
exten = _88XX1NXXNXX,1,AGI(call_log.agi,${EXTEN})
exten = _88XX1NXXNXX,2,Dial(Zap/${EXTEN:2:2}-1/${EXTEN:4},30,o)
Err.. that's not mine. It seems like a dial-by-span syntax.
Just remove the '-1' .
Well, it worked, but ok, I'll take it off.
Now to figure out how to do it across IAX channels from one server to
another.
Which I have, but I haven't tested it yet.
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)
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Re: [asterisk-users] Acceptance testing of a new PRI
On Sat, Jul 26, 2008 at 01:09:16AM +0300, Tzafrir Cohen wrote: On Fri, Jul 25, 2008 at 05:52:20PM -0400, Al Baker wrote: Quote seems like a dial-by-span syntax. What is Dial-by-span ? Zap/span-num-channel-in-span Hmmm. Zap/2 here means the second Zap timeslot on the machine, as does Zap/2-1, using all PRI's on Digium and Sangoma cards. I would have *expected* that it might behave the way you suggest, but it appears not to. Unless it has something to do with the way my zaptel presents the spans to Asterisk... 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] CME/Asterisk Voicemail Problems
I am having problems with CME transferring calls that are busy or noan to
voicemail which is on Asterisk. I have used the no supplementary-service sip
moved-temporarily
and no supplementary-service sip refer commands but when an outside call is
transferred to voicemail it just goes to a busy tone. Internal calls go to
voicemail fine. Does anyone have any suggestions or experience integrating CME
with Asterisk for voicemail?
I have the following setup in extensions_custom.conf (I am using Trixbox):
[cme-vmail]
include = attendant
;messages button
exten = 999,1,Background(silence/1)
exten = 999,2,VoicemailMain(${CALLERID(num)})
exten = 999,3,Hangup
;busy message
exten = 998,1,NoOp,${CALLERID(num)}
exten = 998,2,NoOp,${CALLERID(rdnis)}
exten = 998,3,Playback(silence/1)
exten = 998,4,VoiceMail(b${CALLERID(rdnis)})
exten = 998,5,Hangup
;unavaiable message
exten = 997,1,NoOp,${CALLERID(num)}
exten = 997,2,NoOp,${CALLERID(rdnis)}
exten = 997,3,Playback(silence/1)
exten = 997,4,VoiceMail(u${CALLERID(rdnis)})
exten = 997,5,Hangup
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Re: [asterisk-users] Acceptance testing of a new PRI
On Sat, Jul 26, 2008 at 01:14:14PM -0400, Jay R. Ashworth wrote: On Sat, Jul 26, 2008 at 01:09:16AM +0300, Tzafrir Cohen wrote: On Fri, Jul 25, 2008 at 05:52:20PM -0400, Al Baker wrote: Quote seems like a dial-by-span syntax. What is Dial-by-span ? Zap/span-num-channel-in-span Hmmm. Zap/2 here means the second Zap timeslot on the machine, as does Zap/2-1, using all PRI's on Digium and Sangoma cards. I would have *expected* that it might behave the way you suggest, but it appears not to. Unless it has something to do with the way my zaptel presents the spans to Asterisk... Right. This is not supported. And you get there a warning: zt_request: Unknown option '-' As the '-' is parsed as a channel option (like 'r' or 'c'). Time to fix voip-info. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Acceptance testing of a new PRI
On Sat, Jul 26, 2008 at 01:12:42PM -0400, Jay R. Ashworth wrote:
On Fri, Jul 25, 2008 at 10:57:49PM +0300, Tzafrir Cohen wrote:
On Fri, Jul 25, 2008 at 01:54:21PM -0400, Jay R. Ashworth wrote:
On Thu, Jul 24, 2008 at 04:35:33PM +0300, Tzafrir Cohen wrote:
What's wrong with plain old Zap/NN ?
[test]
exten = _6XXX.,1,Dial(Zap/{EXTEN:1:3}/${EXTEN:4})
Now call 6chan_numnumber-to-dial in context test.
As it happens, Asterisk 1.2 apparently will not recognize 'Zap/01-1' as
the argument to Dial, I get CHANUNAVAIL.
Zap/01-1 ??? How come?
Zap/01 is valid and equivalent to Zap/1 .
And yet, feeding it to Dial didn't work, and stripping the 0 off did.
I'm on 1.2 if that makes a diff.
I've used this extensively since 1.0, FWIW.
Looking at the code: the paarsing is done by sscanf. Maybe it does not
consider a number with a leading 0 as a number?
What error/warning do you get when trying to use Zap/01 ?
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Need Help Regarding Asterisk
But how would my calls be transferred to extension phones from asterisk server. Would i need to connect those phones to Digium card as well. What i mean is would digium card have a main extension where i would connect main pstn line and other 3 port where i would connect another phone line which would be used as extension is that so?? On 7/26/08, Grygoriy Dobrovolskyy [EMAIL PROTECTED] wrote: If you want to connect you pstn only, nothing more, and dont forget that FXO is for the lines. FXS for the Phones 2008/7/26 Preetish Kakkar [EMAIL PROTECTED] Hi,I need a bit of help regarding setting up asterisk. I am trying to setup a simple PBX for a small office we have. We just need 4 extensions. I would like to spent as less as possible. The below are possible solutions i can think off. 1.) I use a Digium Card and connect my PSTN line to the asterisk server. Asterisk would then forward the call to particular extension according to whatever user presses. The extensions would be another set up of analog telephone which would receive the call forward by asterisk. What else (HardWare) i would need except asterisk and digium card...? 2.) I use the digium card and connect my PSTN line to asterisk server. Then it forward calls to 4 pc's (ip addr) which has some kind of software installer (soft phone) and it shows like some call is coming and user can recieve call and talk using headphone.. Again what else i need..? Thanks, Preeteesh ___ -- 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 -- Thanks, Preetish Kakkar (+919818187724) www.successivesoftwares.com ___ -- 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] Cisco Call Manager to Asterisk conversion
Al, I managed a CCM/Unity setup for more than 5 years and saying that Asterisk is less reliable is in my experience a stretch. I will agree that Cisco makes excellent gateways, routers and switches and their pstn gateways are top notch but I was never a fan of CCM and especially Unity. We were plagued with failures on the Unity server - mostly because of the use of Microsoft Exchange. During this same time, I ran Asterisk as a conference bridge, voicemail server and fax to email server and never had the issues I had with CCM, and did much more for much less in terms of $ as well. For the most part, CCM did what it was tasked for, but I think that Asterisk has reached a point where it can compete at the reliability level and support can for Asterisk via the open source community and commercial vendors is acceptable for my needs which is why I am choosing to switch rather than continue on the upgrade cycle with CCM. I think Asterisk is the future for the IP-PBX realm, both homegrown and commercial, and CCM is a dead end. On Fri, Jul 25, 2008 at 12:02 PM, Al Baker [EMAIL PROTECTED] wrote: Quote I need to replace a cisco call manager with an asterisk box. WHY ? You want your TELCO to be LESS Reliable with LESS SUPPORT Grygoriy Dobrovolskyy wrote: Search someone in local area, remote configuration of server is possible but configuring the phones is more difficult, you need someone to load firmwares, ect 2008/7/24 Chad Whitten [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: I need to replace a cisco call manager with an asterisk box. Phones are cisco 7940 and 7910. I know the 40's can use SIP but the 7910's have to use the skinny/sccp driver. Its been quite awhile since I did anything with asterisk, so I am looking for some assistance with the configuration and am willing to pay. Its a basic setup, 30+ phones, incoming lines via PRI, 1 dial plan for incoming and outgoing - nothing fancy there, voicemail for each phone and DID number for each phone. -- Chad Whitten [EMAIL PROTECTED] mailto:[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 ___ -- 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 -- Chad Whitten Metro Network Solutions (601) 366-6630 Phone (601) 366-6066 Fax (601) 842-6804 Cellular [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] Acceptance testing of a new PRI
On Sat, Jul 26, 2008 at 09:34:29PM +0300, Tzafrir Cohen wrote:
On Sat, Jul 26, 2008 at 01:12:42PM -0400, Jay R. Ashworth wrote:
On Fri, Jul 25, 2008 at 10:57:49PM +0300, Tzafrir Cohen wrote:
On Fri, Jul 25, 2008 at 01:54:21PM -0400, Jay R. Ashworth wrote:
On Thu, Jul 24, 2008 at 04:35:33PM +0300, Tzafrir Cohen wrote:
What's wrong with plain old Zap/NN ?
[test]
exten = _6XXX.,1,Dial(Zap/{EXTEN:1:3}/${EXTEN:4})
Now call 6chan_numnumber-to-dial in context test.
As it happens, Asterisk 1.2 apparently will not recognize 'Zap/01-1' as
the argument to Dial, I get CHANUNAVAIL.
Zap/01-1 ??? How come?
Zap/01 is valid and equivalent to Zap/1 .
And yet, feeding it to Dial didn't work, and stripping the 0 off did.
I'm on 1.2 if that makes a diff.
I've used this extensively since 1.0, FWIW.
Looking at the code: the paarsing is done by sscanf. Maybe it does not
consider a number with a leading 0 as a number?
What error/warning do you get when trying to use Zap/01 ?
Chanunavail/Congestion.
Here, let me go get the exact message...
==88
-- Executing AGI(SIP/101cathy-b7619990, call_log.agi|880116142154432)
in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
+ CALL LOG START : |1216995262.36|SIP/101cathy-b7619990|880116142154432|SIP|
7274514974|2008-07-25 10:14:22
-- AGI Script call_log.agi completed, returning 0
-- Executing Dial(SIP/101cathy-b7619990, Zap/01-1/16142154432|30|o) in
new stack
Jul 25 10:14:22 NOTICE[25497]: app_dial.c:1076 dial_exec_full: Unable to create
channel of type 'Zap' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing NoOp(SIP/101cathy-b7619990, CHANUNAVAIL) in new stack
-- Executing NoOp(SIP/101cathy-b7619990, Hangup Cause: 0) in new stack
-- Executing Hangup(SIP/101cathy-b7619990, ) in new stack
== Spawn extension (default, 880116142154432, 5) exited non-zero on 'SIP/101
cathy-b7619990'
==88
Copied and pasted. I later extended the rules, as you saw, to have a
special rule for 880X, and it worked just fine.
Not sure what to tell you, but it seems to be that.
Note that I have not *yet* taken the -1 off the end, so it cannot be
that.
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)
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Re: [asterisk-users] Acceptance testing of a new PRI
On Sat, Jul 26, 2008 at 09:32:03PM +0300, Tzafrir Cohen wrote: Zap/2 here means the second Zap timeslot on the machine, as does Zap/2-1, using all PRI's on Digium and Sangoma cards. I would have *expected* that it might behave the way you suggest, but it appears not to. Unless it has something to do with the way my zaptel presents the spans to Asterisk... Right. This is not supported. And you get there a warning: zt_request: Unknown option '-' As the '-' is parsed as a channel option (like 'r' or 'c'). Time to fix voip-info. Except that that is what Asterisk is giving *us*: -- Local/[EMAIL PROTECTED],1 answered Zap/73-1 -- IAX2/VICIast26-19 answered Zap/73-1 -- Zap/11-1 is ringing -- Zap/11-1 answered SIP/101cathy-0824cda0 As nearly as I can discern, those are messages where the Zap channel ide is being generated by Asterisk, based on no particular configuration we gave it (there are lots of others, but they could just be repeating an argument they were passed; mostly Application messages). We do in fact, see that zt_request message, but it's not like we made *up* the whole 73-1 thing... :-) 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] Acceptance testing of a new PRI
On Sat, Jul 26, 2008 at 03:28:10PM -0400, Jay R. Ashworth wrote:
On Sat, Jul 26, 2008 at 09:34:29PM +0300, Tzafrir Cohen wrote:
On Sat, Jul 26, 2008 at 01:12:42PM -0400, Jay R. Ashworth wrote:
On Fri, Jul 25, 2008 at 10:57:49PM +0300, Tzafrir Cohen wrote:
On Fri, Jul 25, 2008 at 01:54:21PM -0400, Jay R. Ashworth wrote:
On Thu, Jul 24, 2008 at 04:35:33PM +0300, Tzafrir Cohen wrote:
What's wrong with plain old Zap/NN ?
[test]
exten = _6XXX.,1,Dial(Zap/{EXTEN:1:3}/${EXTEN:4})
Now call 6chan_numnumber-to-dial in context test.
As it happens, Asterisk 1.2 apparently will not recognize 'Zap/01-1'
as
the argument to Dial, I get CHANUNAVAIL.
Zap/01-1 ??? How come?
Zap/01 is valid and equivalent to Zap/1 .
And yet, feeding it to Dial didn't work, and stripping the 0 off did.
I'm on 1.2 if that makes a diff.
I've used this extensively since 1.0, FWIW.
Looking at the code: the paarsing is done by sscanf. Maybe it does not
consider a number with a leading 0 as a number?
What error/warning do you get when trying to use Zap/01 ?
Chanunavail/Congestion.
Here, let me go get the exact message...
==88
-- Executing AGI(SIP/101cathy-b7619990, call_log.agi|880116142154432)
in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
+ CALL LOG START :
|1216995262.36|SIP/101cathy-b7619990|880116142154432|SIP|
7274514974|2008-07-25 10:14:22
-- AGI Script call_log.agi completed, returning 0
-- Executing Dial(SIP/101cathy-b7619990, Zap/01-1/16142154432|30|o) in
new stack
Why do you keep adding that -1?
Try Zap/01
Though I tried originating a call to Zap/04 and Zap/04-1 and both worked
well here (1.4). With the -1 I got the warning I mentioned above about
the unknown option.
Jul 25 10:14:22 NOTICE[25497]: app_dial.c:1076 dial_exec_full: Unable to
create
channel of type 'Zap' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing NoOp(SIP/101cathy-b7619990, CHANUNAVAIL) in new stack
-- Executing NoOp(SIP/101cathy-b7619990, Hangup Cause: 0) in new stack
-- Executing Hangup(SIP/101cathy-b7619990, ) in new stack
== Spawn extension (default, 880116142154432, 5) exited non-zero on 'SIP/101
cathy-b7619990'
==88
Copied and pasted. I later extended the rules, as you saw, to have a
special rule for 880X, and it worked just fine.
Not sure what to tell you, but it seems to be that.
Note that I have not *yet* taken the -1 off the end, so it cannot be
that.
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)
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--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Acceptance testing of a new PRI
On Sat, Jul 26, 2008 at 10:38:07PM +0300, Tzafrir Cohen wrote: [ quoting me ] Chanunavail/Congestion. Here, let me go get the exact message... ==88 -- Executing AGI(SIP/101cathy-b7619990, call_log.agi|880116142154432) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi + CALL LOG START : |1216995262.36|SIP/101cathy-b7619990|880116142154432|SIP| 7274514974|2008-07-25 10:14:22 -- AGI Script call_log.agi completed, returning 0 -- Executing Dial(SIP/101cathy-b7619990, Zap/01-1/16142154432|30|o) in new stack Why do you keep adding that -1? Because, as I noted in my other message, *ASTERISK KEEPS ADDING IT*. :-) Try Zap/01 Though I tried originating a call to Zap/04 and Zap/04-1 and both worked well here (1.4). With the -1 I got the warning I mentioned above about the unknown option. Sure. But did *the call go out*? Jul 25 10:14:22 NOTICE[25497]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing NoOp(SIP/101cathy-b7619990, CHANUNAVAIL) in new stack -- Executing NoOp(SIP/101cathy-b7619990, Hangup Cause: 0) in new stack -- Executing Hangup(SIP/101cathy-b7619990, ) in new stack == Spawn extension (default, 880116142154432, 5) exited non-zero on 'SIP/101 cathy-b7619990' ==88 Copied and pasted. I later extended the rules, as you saw, to have a special rule for 880X, and it worked just fine. Not sure what to tell you, but it seems to be that. Note that I have not *yet* taken the -1 off the end, so it cannot be that. See? I *knew* I mentioned it. Note that Mike Cargile at VICIdial looked over that dialplan, and he didn't seem to have a problem with the -1; I'm pretty sure it's in the VICIdial standard dialplans. 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] Acceptance testing of a new PRI
On Sat, Jul 26, 2008 at 03:32:34PM -0400, Jay R. Ashworth wrote: On Sat, Jul 26, 2008 at 09:32:03PM +0300, Tzafrir Cohen wrote: Zap/2 here means the second Zap timeslot on the machine, as does Zap/2-1, using all PRI's on Digium and Sangoma cards. I would have *expected* that it might behave the way you suggest, but it appears not to. Unless it has something to do with the way my zaptel presents the spans to Asterisk... Right. This is not supported. And you get there a warning: zt_request: Unknown option '-' As the '-' is parsed as a channel option (like 'r' or 'c'). Time to fix voip-info. Except that that is what Asterisk is giving *us*: -- Local/[EMAIL PROTECTED],1 answered Zap/73-1 -- IAX2/VICIast26-19 answered Zap/73-1 -- Zap/11-1 is ringing -- Zap/11-1 answered SIP/101cathy-0824cda0 Trying to dial to Zap/11-1 instead of to Zap/11 is like trying to SIP/101cathy-0824cda0 installed of dialing to SIP/101cathy -- 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] Acceptance testing of a new PRI
On Sat, Jul 26, 2008 at 03:46:28PM -0400, Jay R. Ashworth wrote: On Sat, Jul 26, 2008 at 10:38:07PM +0300, Tzafrir Cohen wrote: [ quoting me ] Chanunavail/Congestion. Here, let me go get the exact message... ==88 -- Executing AGI(SIP/101cathy-b7619990, call_log.agi|880116142154432) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi + CALL LOG START : |1216995262.36|SIP/101cathy-b7619990|880116142154432|SIP| 7274514974|2008-07-25 10:14:22 -- AGI Script call_log.agi completed, returning 0 -- Executing Dial(SIP/101cathy-b7619990, Zap/01-1/16142154432|30|o) in new stack Why do you keep adding that -1? Because, as I noted in my other message, *ASTERISK KEEPS ADDING IT*. :-) Try Zap/01 Though I tried originating a call to Zap/04 and Zap/04-1 and both worked well here (1.4). With the -1 I got the warning I mentioned above about the unknown option. Sure. But did *the call go out*? Jul 25 10:14:22 NOTICE[25497]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing NoOp(SIP/101cathy-b7619990, CHANUNAVAIL) in new stack -- Executing NoOp(SIP/101cathy-b7619990, Hangup Cause: 0) in new stack -- Executing Hangup(SIP/101cathy-b7619990, ) in new stack == Spawn extension (default, 880116142154432, 5) exited non-zero on 'SIP/101 cathy-b7619990' ==88 Copied and pasted. I later extended the rules, as you saw, to have a special rule for 880X, and it worked just fine. Not sure what to tell you, but it seems to be that. Note that I have not *yet* taken the -1 off the end, so it cannot be that. See? I *knew* I mentioned it. Note that Mike Cargile at VICIdial looked over that dialplan, and he didn't seem to have a problem with the -1; I'm pretty sure it's in the VICIdial standard dialplans. You can replace the '-1' with 'X56456456', '_123123' or 'p0'. It would be likewise (in)valid, give a warning regarding invalid option but dial anyway. -- 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] Need Help Regarding Asterisk
Preetish Kakkar wrote: But how would my calls be transferred to extension phones from asterisk server. Would i need to connect those phones to Digium card as well. What i mean is would digium card have a main extension where i would connect main pstn line and other 3 port where i would connect another phone line which would be used as extension is that so?? On 7/26/08, *Grygoriy Dobrovolskyy* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: If you want to connect you pstn only, nothing more, and dont forget that FXO is for the lines. FXS for the Phones 2008/7/26 Preetish Kakkar [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Hi,I need a bit of help regarding setting up asterisk. I am trying to setup a simple PBX for a small office we have. We just need 4 extensions. I would like to spent as less as possible. The below are possible solutions i can think off. 1.) I use a Digium Card and connect my PSTN line to the asterisk server. Asterisk would then forward the call to particular extension according to whatever user presses. The extensions would be another set up of analog telephone which would receive the call forward by asterisk. What else (HardWare) i would need except asterisk and digium card...? 2.) I use the digium card and connect my PSTN line to asterisk server. Then it forward calls to 4 pc's (ip addr) which has some kind of software installer (soft phone) and it shows like some call is coming and user can recieve call and talk using headphone.. Again what else i need..? Thanks, Preeteesh ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 http://www.api-digital.com/ -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Preetish Kakkar (+919818187724) www.successivesoftwares.com http://www.successivesoftwares.com ___ -- 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 This should help; http://www.voip-info.org/wiki/view/Asterisk+hardware+recommendations -- Powered by Gentoo GNU/LINUX http://www.linuxcrazy.com ___ -- 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] Acceptance testing of a new PRI
On Sat, Jul 26, 2008 at 10:48:54PM +0300, Tzafrir Cohen wrote: Except that that is what Asterisk is giving *us*: -- Local/[EMAIL PROTECTED],1 answered Zap/73-1 -- IAX2/VICIast26-19 answered Zap/73-1 -- Zap/11-1 is ringing -- Zap/11-1 answered SIP/101cathy-0824cda0 Trying to dial to Zap/11-1 instead of to Zap/11 is like trying to SIP/101cathy-0824cda0 installed of dialing to SIP/101cathy So, clearly, I'm not smart enough; precisely what are the semantics of the 'Something' in Technology/Channel-Something? 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] Acceptance testing of a new PRI
On Sat, Jul 26, 2008 at 3:53 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sat, Jul 26, 2008 at 03:46:28PM -0400, Jay R. Ashworth wrote: On Sat, Jul 26, 2008 at 10:38:07PM +0300, Tzafrir Cohen wrote: [ quoting me ] Chanunavail/Congestion. Here, let me go get the exact message... ==88 -- Executing AGI(SIP/101cathy-b7619990, call_log.agi|880116142154432) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi + CALL LOG START : |1216995262.36|SIP/101cathy-b7619990|880116142154432|SIP| 7274514974|2008-07-25 10:14:22 -- AGI Script call_log.agi completed, returning 0 -- Executing Dial(SIP/101cathy-b7619990, Zap/01-1/16142154432|30|o) in new stack Why do you keep adding that -1? Because, as I noted in my other message, *ASTERISK KEEPS ADDING IT*. :-) Try Zap/01 Though I tried originating a call to Zap/04 and Zap/04-1 and both worked well here (1.4). With the -1 I got the warning I mentioned above about the unknown option. Sure. But did *the call go out*? Jul 25 10:14:22 NOTICE[25497]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing NoOp(SIP/101cathy-b7619990, CHANUNAVAIL) in new stack -- Executing NoOp(SIP/101cathy-b7619990, Hangup Cause: 0) in new stack -- Executing Hangup(SIP/101cathy-b7619990, ) in new stack == Spawn extension (default, 880116142154432, 5) exited non-zero on 'SIP/101 cathy-b7619990' ==88 Copied and pasted. I later extended the rules, as you saw, to have a special rule for 880X, and it worked just fine. Not sure what to tell you, but it seems to be that. Note that I have not *yet* taken the -1 off the end, so it cannot be that. See? I *knew* I mentioned it. Note that Mike Cargile at VICIdial looked over that dialplan, and he didn't seem to have a problem with the -1; I'm pretty sure it's in the VICIdial standard dialplans. You can replace the '-1' with 'X56456456', '_123123' or 'p0'. It would be likewise (in)valid, give a warning regarding invalid option but dial anyway. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir If you want to test inbound and fill all of your channels, you could post something creative on Craigslist and then put them all in a queue with MOH that would keep them on the line. Or you could make a dialplan that takes the inbound caller ID and turn around and dial it. Do that with one of your DIDs and you should fill all your channels pretty quickly. Anyways, with a PRI, when I see the channels come up and I can dial out and in, I have never had an issue with a particular channel. 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
[asterisk-users] Visual Dial Plan
I just stumbled across this on youtube. Does any on the list us it? This is the first I've heard over it. http://www.youtube.com/watch?v=H1j5OrgL1og Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 (New York) +61-2-9016-5642 (Sydney) http://www.Cognation.net http://www.Cognation.net/profile ___ -- 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] Need Help Regarding Asterisk
On Sat, Jul 26, 2008 at 11:40 AM, Preetish Kakkar [EMAIL PROTECTED] wrote: But how would my calls be transferred to extension phones from asterisk server. Would i need to connect those phones to Digium card as well. What i mean is would digium card have a main extension where i would connect main pstn line and other 3 port where i would connect another phone line which would be used as extension is that so?? Preetish, There are many ways to do this. The phone line(s) will need to be connected to FXO interfaces. You can get single units or cards to do 4 or more lines. If you have no analog (normal) telephones that need to be used, you will usually want to purchase SIP phones or use software. If you do have analog phones, you want one FXS interface per phone to connect to asterisk. Once you have the hardware hooked up, asterisk will route the calls according to the dialplan you create. Take a look at a site like http://voipsupply.com or a more local equivalent if you can find one that sells this stuff. It's a good way to see what is currently available and get an idea of prices. /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] Acceptance testing of a new PRI
The something is generated by Asterisk at the time the call is created. You should never add it, since you don't control that call instance info. In fact, you should almost never care about the call instance string. The -1 means first instance of a call on this channel, a -2 would be seen in you answer a 2nd call for call waiting. Jay R. Ashworth wrote: On Sat, Jul 26, 2008 at 10:48:54PM +0300, Tzafrir Cohen wrote: Except that that is what Asterisk is giving *us*: -- Local/[EMAIL PROTECTED],1 answered Zap/73-1 -- IAX2/VICIast26-19 answered Zap/73-1 -- Zap/11-1 is ringing -- Zap/11-1 answered SIP/101cathy-0824cda0 Trying to dial to Zap/11-1 instead of to Zap/11 is like trying to SIP/101cathy-0824cda0 installed of dialing to SIP/101cathy So, clearly, I'm not smart enough; precisely what are the semantics of the 'Something' in Technology/Channel-Something? Cheers, -- jra -- 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] Need Help Regarding Asterisk
On Sat, Jul 26, 2008 at 6:12 PM, randulo [EMAIL PROTECTED] wrote: On Sat, Jul 26, 2008 at 11:40 AM, Preetish Kakkar [EMAIL PROTECTED] wrote: But how would my calls be transferred to extension phones from asterisk server. Would i need to connect those phones to Digium card as well. What i mean is would digium card have a main extension where i would connect main pstn line and other 3 port where i would connect another phone line which would be used as extension is that so?? Preetish, There are many ways to do this. The phone line(s) will need to be connected to FXO interfaces. You can get single units or cards to do 4 or more lines. If you have no analog (normal) telephones that need to be used, you will usually want to purchase SIP phones or use software. If you do have analog phones, you want one FXS interface per phone to connect to asterisk. Once you have the hardware hooked up, asterisk will route the calls according to the dialplan you create. Take a look at a site like http://voipsupply.com or a more local equivalent if you can find one that sells this stuff. It's a good way to see what is currently available and get an idea of prices. /r Personally, I would suggest something like this unless you want to do something special. http://cgi.ebay.com/Toshiba-Strata-DK424-Phone-System-w-10-Phones-Wrrnty_W0QQitemZ120285800045QQihZ002QQcategoryZ11908QQssPageNameZWDVWQQrdZ1QQcmdZViewItem Here is an Ebay/Microsoft hack/trick. Go to Ebay, find something you want with a reasonable buy it now price, they must also accept PayPal. Select watch this item In a new browser, open this url http://www.live.com/ type wii (good luck for me so far different words get different results. In the new window, you will see something like, Wii- www.ebay.comLive Search cashback Buy Wii. You may get 20% off with PayPal if eligible. Just click on the Live Search Icon which will take you back to ebay, now go to your watched items and click buy it now. You should see the % discount at the bottom of the page. I think the limit is based on $1,000, I have seen the % discount as high as 35% and as low as 10%. I got a great deal on a repeater just over $1k with 35% back so in 90 days I will get $350 back. 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] Need Help Regarding Asterisk
On Sat, Jul 26, 2008 at 7:05 PM, Steve Totaro [EMAIL PROTECTED] wrote: On Sat, Jul 26, 2008 at 6:12 PM, randulo [EMAIL PROTECTED] wrote: On Sat, Jul 26, 2008 at 11:40 AM, Preetish Kakkar [EMAIL PROTECTED] wrote: But how would my calls be transferred to extension phones from asterisk server. Would i need to connect those phones to Digium card as well. What i mean is would digium card have a main extension where i would connect main pstn line and other 3 port where i would connect another phone line which would be used as extension is that so?? Preetish, There are many ways to do this. The phone line(s) will need to be connected to FXO interfaces. You can get single units or cards to do 4 or more lines. If you have no analog (normal) telephones that need to be used, you will usually want to purchase SIP phones or use software. If you do have analog phones, you want one FXS interface per phone to connect to asterisk. Once you have the hardware hooked up, asterisk will route the calls according to the dialplan you create. Take a look at a site like http://voipsupply.com or a more local equivalent if you can find one that sells this stuff. It's a good way to see what is currently available and get an idea of prices. /r Personally, I would suggest something like this unless you want to do something special. http://cgi.ebay.com/Toshiba-Strata-DK424-Phone-System-w-10-Phones-Wrrnty_W0QQitemZ120285800045QQihZ002QQcategoryZ11908QQssPageNameZWDVWQQrdZ1QQcmdZViewItem Here is an Ebay/Microsoft hack/trick. Go to Ebay, find something you want with a reasonable buy it now price, they must also accept PayPal. Select watch this item In a new browser, open this url http://www.live.com/ type wii (good luck for me so far different words get different results. In the new window, you will see something like, Wii- www.ebay.comLive Search cashback Buy Wii. You may get 20% off with PayPal if eligible. Just click on the Live Search Icon which will take you back to ebay, now go to your watched items and click buy it now. You should see the % discount at the bottom of the page. I think the limit is based on $1,000, I have seen the % discount as high as 35% and as low as 10%. I got a great deal on a repeater just over $1k with 35% back so in 90 days I will get $350 back. Thanks, Steve Totaro Actually, this might suit your desires better. http://www.rowetel.com/ucasterisk/store.html#ip04 You might want to look at Xlite for a free softphone or pickup some SIP phones. I will be obtaining one for testing myself pretty soon. 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
