[asterisk-users] DUNDI Help
Would like to try setting up dundi with 3-4 asterisk.
But for poc, i would like to try setting up dundi on between 2 asterisk.
I copied the config from DUNDI enterprise SIP with no password. Only thing i
changed is the part where i used regcontext.
on both boxes dundi.conf i have
[mapping]
priv = sipregistrations,0,SIP,[EMAIL PROTECTED],nopartial
i can see both peers on each server:
CLI dundi show peers
EID Host Model AvgTime Status
00:8e:8c:8e:cb:53 10.10.10.XX (S) Symmetric Unavail OK (1 ms)
i can see my extension being added on sipregistrations context
Added extension '136101' priority 1 to sipregistrations
tried a dundi lookup but got no result
dundi lookup [EMAIL PROTECTED]
DUNDi lookup returned no results.
DUNDi lookup completed in 0 ms
here's what's on extensions.conf
; Private DUNDi network
[dundi-priv-canonical]
; Direct numbers
[dundi-priv-customers]
; If you are an ITSP or Reseller, list your customers here.
[dundi-priv-via-pstn]
; If you are freely delivering calls to the PSTN, list them here
[dundi-priv-local]
include = dundi-priv-canonical
include = dundi-priv-customers
include = dundi-priv-via-pstn
[dundi-priv-switch]
; Just a wrapper for the switch
switch = DUNDi/priv
[dundi-priv-lookup]
include = dundi-priv-local
include = dundi-priv-switch
[macro-dundi-priv]
exten = s,1,Goto(${ARG1}|1)
include = dundi-priv-lookup
[diallocal]
exten = _1X,1,Macro(dundi-priv|${EXTEN})
i also tried dialing from my xlite:
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
Macro(SIP/138100-08269548, dundi-priv|136101) in new stack
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
Goto(SIP/138100-08269548, 136101|1) in new stack
[Aug 26 15:58:07] -- Goto (macro-dundi-priv,136101,1)
[Aug 26 15:58:07] == Auto fallthrough, channel 'SIP/138100-08269548' status
is 'UNKNOWN'
any guess what's wrong? Thanks
ron
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[asterisk-users] Asterisk/Other PBX interconnection
Hi all, I would like to interconnect send and receive traffic to another SIP PBX. The only thing I have is the IP of the other SIP PBX. Actually, my Asterisk has an ISDN card so I can communicate with 3G networks. I want to offer to the other PBX the ability to call a 3G devices using my Asterisk 3G. Any advice? Thank you Khaldon ___ -- 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 intercom with asterisk
On Mon, 25 Aug 2008, Jonathan Disher wrote: I am looking to replace the phone system at my father's shop with an Asterisk box and some Cisco phones, but one piece of the implementation is tripping me up. He has two buildings (the office, and the shop proper), separated by about 3-400 yards. Currently with the ancient Meridian system installed, there is a paging intercom (to page employees, etc) on a dedicated extension - play a loud tone, then set up a 2 way channel. Anyone got any ideas, hardware wise, on how I might implement this with an Asterisk system? Do you have some sort of IP connectivity between the sites? 400 yards is a too long for copper cat5, but can be done with fibre, wireless or free-space optics... (which I don't personally recommend!) (And if you haven't IP how are you talking to the phones between sites?) So what's to stop you from putting a Cisco phone into auto-answer mode and calling it via ths Page() application? And if a Cisco can't work in this manner, there are plenty of others that will. Gordon ___ -- 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] Dial Plan Help
Steve Alex thanks for your help. I've got it working perfectly now.
-Jon
- Original Message -
From: Alex Balashov [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, August 24, 2008 9:22 AM
Subject: Re: [asterisk-users] Dial Plan Help
John,
This is the default behaviour anyway. If Dial() is successful,
execution of subsequent priorities in the dial plan for that extension
is not resumed. It'll only fall through to the other priorities if
Dial() fails.
I do, however, suggest supplying a timeout argument to your Dial()s.
-- Alex
Jon Weisman wrote:
I'd like to do the following can someone guide me on how to accomplish
this?
Call comes in via PRI and tries to go out via SIP if for some reason the
ISP
is down and the call can not go out i want it to fail over and send the
same
call through a different PRI.
I was thinking something like this:
exten=_X.,1,Dial(SIP/[EMAIL PROTECTED])
exten=_X.,2,Dial,Zap/g2/${EXTEN}; I only want it to go here if
it
was unable to send the call via SIP (if the first priority failed), but
if
it did go through sip then it should just hangup the call when the person
is
done speaking.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
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[asterisk-users] Limit to the length of string ?
Had an issue recently and it looked like there is a limit to the length of a string... I built up a string dynamically and it seemed to get truncated... Below is 2 lines of console output: I've split this line to make sense: -- Executing NoOp(IAX2/inco1-10969, About to dial IAX2/106200SIP/106200 IAX2/106201SIP/106201 IAX2/106202SIP/106202 IAX2/106203SIP/106203 IAX2/106204SIP/106204 IAX2/106205SIP/106205 IAX2/106206SIP/106206 IAX2/106207SIP/106207 IAX2/106208SIP/106208 IAX2/106209SIP/106209 IAX2/106210SIP/106210 IA) in new stack -- Executing Dial(IAX2/inco1-10969, IAX2/106200SIP/106200 IAX2/106201SIP/106201 IAX2/106202SIP/106202 IAX2/106203SIP/106203 IAX2/106204SIP/106204 IAX2/106205SIP/106205 IAX2/106206SIP/106206 IAX2/106207SIP/106207 IAX2/106208SIP/106208 IAX2/106209SIP/106209 IAX2/106210SIP/106210 IA||wton) in new stack So you can see where it got upset )-: There should have been 2 more lines (for accounts 106211 and 106212) in there. A-Ha... That string is 256 characters long... Now there's a fishy number if ever there was one. So, if this a real limitation? This is 1.2.30 if that makes a difference... Did this limit go away in 1.4 ? Gordon ___ -- 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] Get call status and hangup
Ciao Loic, Hello, I am looking for a way to check if a call could be established with the destination (SIP,IAX,ZAP). So I thought about an application like DIAL but instead it should return a variable and hangup immediately as soon as it gets something that could lead to a valid connection ringing... Is there something like this that could be used in Asterisk or can anyone recommend a different/better solution? I don't know if it's exactly what you want, but you can monitor the status of a given user using the Asterisk Manager Interface (AMI), through the events NewState, PeerStatus, Hangup. More info: http://www.voip-info.org/wiki-Asterisk+manager+API http://www.voip-info.org/wiki/view/asterisk+manager+events HTH, -- Dr. Andrea Spadaccini Multimedia Technologies Institute - MTI S.r.l. Web: www.x-voice.it - Tel: +39 (0) 95 7224945 ___ -- 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 intercom with asterisk
Jonathan Disher wrote: He has two buildings (the office, and the shop proper), separated by about 3-400 yards. Your inter-building distance exceeds ethernet over copper limits, you will need a fiber link. paging intercom (to page employees, etc) on a dedicated extension - Easy to implement, with a Polycom phone you configure an auto-answer extension. On a 501, you have three line appearances so you can configure line app. two to be auto-answer and still use line app. one as normal phone, hang it on the wall for anyone needs to make a call. If you need more volume, take the headset out and amplify it. There is a lot of information on auto-answer configuration, it's a little tricky but solid once you get it. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- 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] Limit to the length of string ?
On Tuesday 26 August 2008 06:49:54 Gordon Henderson wrote: A-Ha... That string is 256 characters long... Now there's a fishy number if ever there was one. So, if this a real limitation? This is 1.2.30 if that makes a difference... Did this limit go away in 1.4 ? Yes, it did. -- 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] Limit to the length of string ?
On Tue, Aug 26, 2008 at 9:01 AM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Tuesday 26 August 2008 06:49:54 Gordon Henderson wrote: A-Ha... That string is 256 characters long... Now there's a fishy number if ever there was one. So, if this a real limitation? This is 1.2.30 if that makes a difference... Did this limit go away in 1.4 ? Yes, it did. -- Tilghman Tilghman, While I appreciate your dedication, I suggest you Unplug during your vacation man! Turn off the laptop, cell phone, and everything else and keep it off. Get some RR. 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] implementing an intercom with asterisk
Jonathan Disher wrote: I am looking to replace the phone system at my father's shop with an Asterisk box and some Cisco phones, but one piece of the implementation is tripping me up. He has two buildings (the office, and the shop proper), separated by about 3-400 yards. Currently with the ancient Meridian system installed, there is a paging intercom (to page employees, etc) on a dedicated extension - play a loud tone, then set up a 2 way channel. Anyone got any ideas, hardware wise, on how I might implement this with an Asterisk system? Thanks, and if this isn't appropriate for this list, if anyone has a better destination for the question, Id be quite appreciative. Hi Jon, how is the existing intercom implemented? Is it on the phones or separate speakers? If it is a paging system wired into the Nortel, you may be able to reuse it via an ATA, If it's done through the phones, try http://www.voip-info.org/wiki/index.php?page=Asterisk+Paging+and+Intercom for ideas That is, once you've bridged the ethernet across the 3-400 yards. You can use fibre, wireless or Thicknet. Fibre is the most robust, wireless is the cheapest and Thicknet, well, good luck getting the parts! :-) regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] sip peering between 2 asterisk
On 26/08/2008 Nhadie wrote: is it possible to peer via IAX, then send SIP calls over the IAX peer? yes indeed. The SIP calls would exist between the phones and the Asterisk servers, and become IAX calls on the interconnect. Tried insecure=very ? Phil ___ -- 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] Limit to the length of string ?
On Tuesday 26 August 2008 08:06:10 Steve Totaro wrote: On Tue, Aug 26, 2008 at 9:01 AM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Tuesday 26 August 2008 06:49:54 Gordon Henderson wrote: A-Ha... That string is 256 characters long... Now there's a fishy number if ever there was one. So, if this a real limitation? This is 1.2.30 if that makes a difference... Did this limit go away in 1.4 ? Yes, it did. While I appreciate your dedication, I suggest you Unplug during your vacation man! Turn off the laptop, cell phone, and everything else and keep it off. Get some RR. That was last week. -- 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] Limit to the length of string ?
On Tue, 26 Aug 2008, Tilghman Lesher wrote: On Tuesday 26 August 2008 06:49:54 Gordon Henderson wrote: A-Ha... That string is 256 characters long... Now there's a fishy number if ever there was one. So, if this a real limitation? This is 1.2.30 if that makes a difference... Did this limit go away in 1.4 ? Yes, it did. OK. Thanks. Now I guess I have to play the 1.4 lottery :) (or will it get fixed in 1.2.31 ;-) Cheers, Gordon ___ -- 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] is shared_lastcall available in 1.4
On Mon, 2008-08-25 at 17:47 -0500, Bob Pierce wrote: I'm glad to hear that you're enjoying Asterisk 1.4! Unfortunately, the shared_lastcall option is only in versions 1.6.0 and up. Does anybody have a workaround for this in 1.4? Or maybe a better question: How stable is 1.6 for production use? Bob ___ -- 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] is shared_lastcall available in 1.4
On Tue, Aug 26, 2008 at 5:14 PM, Bob Pierce [EMAIL PROTECTED] wrote: On Mon, 2008-08-25 at 17:47 -0500, Bob Pierce wrote: I'm glad to hear that you're enjoying Asterisk 1.4! Unfortunately, the shared_lastcall option is only in versions 1.6.0 and up. Does anybody have a workaround for this in 1.4? Or maybe a better question: How stable is 1.6 for production use? I'd say - go for backport instead. shared_lastcall is commited in http://svn.digium.com/view/asterisk/trunk/apps/app_queue.c?r1=86820r2=86985 and it seems that there are no bugfixes for it since. So, backporting should be fairly simple. Also i would suggest subscribing to asterisk-svn and watch for commits to app_queue to not miss any bugfixes to it. Migration to 1.6 could be more time consuming, as there are lot of changes, you will probably have to adjust dialplan, etc. 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] implementing an intercom with asterisk
On Tue, Aug 26, 2008 at 9:22 AM, Drew Gibson [EMAIL PROTECTED] wrote: Jonathan Disher wrote: I am looking to replace the phone system at my father's shop with an Asterisk box and some Cisco phones, but one piece of the implementation is tripping me up. He has two buildings (the office, and the shop proper), separated by about 3-400 yards. Currently with the ancient Meridian system installed, there is a paging intercom (to page employees, etc) on a dedicated extension - play a loud tone, then set up a 2 way channel. Anyone got any ideas, hardware wise, on how I might implement this with an Asterisk system? Thanks, and if this isn't appropriate for this list, if anyone has a better destination for the question, Id be quite appreciative. Hi Jon, how is the existing intercom implemented? Is it on the phones or separate speakers? If it is a paging system wired into the Nortel, you may be able to reuse it via an ATA, If it's done through the phones, try http://www.voip-info.org/wiki/index.php?page=Asterisk+Paging+and+Intercom for ideas That is, once you've bridged the ethernet across the 3-400 yards. You can use fibre, wireless or Thicknet. Fibre is the most robust, wireless is the cheapest and Thicknet, well, good luck getting the parts! :-) regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com http://www.zaptech.net/index.php?copperlinkmodel2172 These work very well. 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] is shared_lastcall available in 1.4
On Tue, 2008-08-26 at 17:30 +0300, Atis Lezdins wrote: I'd say - go for backport instead. shared_lastcall is commited in http://svn.digium.com/view/asterisk/trunk/apps/app_queue.c?r1=86820r2=86985 and it seems that there are no bugfixes for it since. So, backporting should be fairly simple. Also i would suggest subscribing to asterisk-svn and watch for commits to app_queue to not miss any bugfixes to it. Are there any plans to back port this feature into upcoming 1.4 releases? ___ -- 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 connected to the PSTN vs. a commercial solution
Dear all, now I'm using an Asterisk 1.4.13 SIP server with 50 SIP users and it works very well only in an intranet environment (no connections to the PSTN world). But in the near future, we have to plan a telephone system that works in the intranet (voip) and also it must be connected to the PSTN public network with a T1/E1 trunk, with 200 SIP users aproximately. So at first I have to ways to do that: 1- Continue using Asterisk and adding a T1/E1 interface in order to connect to the PSTN 2- Discard Asterisk and buy a commercial solution, because we have the money My questions are: does Asterisk work in the scenario I've described What is the best solution you can recommend to me ??? Thanks in advance, 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] sip peering between 2 asterisk
hi phil, yup i have tried insecure=very as well. but still get forbidden. if i use an IAX trunk, how do i dial a SIP user? e.g. if i define this on iax.conf [asterisk-iax-1] type=peer host=10.10.10.10 can i simply dial like this: exten = _1X,1,Dial(SIP/[EMAIL PROTECTED]) nhadie Phil Thompson wrote: On 26/08/2008 Nhadie wrote: is it possible to peer via IAX, then send SIP calls over the IAX peer? yes indeed. The SIP calls would exist between the phones and the Asterisk servers, and become IAX calls on the interconnect. Tried insecure=very ? Phil ___ -- 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] is shared_lastcall available in 1.4
On Tue, Aug 26, 2008 at 5:39 PM, Bob Pierce [EMAIL PROTECTED] wrote: On Tue, 2008-08-26 at 17:30 +0300, Atis Lezdins wrote: I'd say - go for backport instead. shared_lastcall is commited in http://svn.digium.com/view/asterisk/trunk/apps/app_queue.c?r1=86820r2=86985 and it seems that there are no bugfixes for it since. So, backporting should be fairly simple. Also i would suggest subscribing to asterisk-svn and watch for commits to app_queue to not miss any bugfixes to it. Are there any plans to back port this feature into upcoming 1.4 releases? No, new features are added only in trunk, and released in next major release (1.6). 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] TDM2400P Voice Quality Problem
Hi Shariq - I m facing problem with TDM2400P pstn card. When someone dials, the voice quality is crappyInstead of hearing. Echo cancel almost works, but the callee hear what they describe as a 'background crackle/buzz' coming back when they talk. Crackling noise is usually caused by an unbalanced hybrid or a shared IRQ. Have you used the fxotune tool? This is the first thing you should do with any analog card. If you still have issues after running fxotune, check to see if your card is sharing interrupts with anything else like a network card or disk controller. You can use lspci -vv or cat /proc/interrupts for this. - Noah ___ -- 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] Limit to the length of string ?
On Tuesday 26 August 2008 09:08:58 Gordon Henderson wrote: On Tue, 26 Aug 2008, Tilghman Lesher wrote: On Tuesday 26 August 2008 06:49:54 Gordon Henderson wrote: A-Ha... That string is 256 characters long... Now there's a fishy number if ever there was one. So, if this a real limitation? This is 1.2.30 if that makes a difference... Did this limit go away in 1.4 ? Yes, it did. OK. Thanks. Now I guess I have to play the 1.4 lottery :) (or will it get fixed in 1.2.31 ;-) It will not get changed in 1.2 at all, sorry. That's a major architectural change, for a branch which has been end-of-lifed and is only eligible for security fixes, not new features. You could manually up the various buffers (there's more than one which affects this!) and recompile, if you were so inclined. 1.4 uses a completely different method of sizing the buffer, which allows for a seemingly limitless length of argments. One additional caveat, though: the buffer used for variable substitution is only 8k, so there's an additional limitation there, as well. -- 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
[asterisk-users] te410p remains in red-alarm
I have a TE410P in a Dell 2650 running in production on an older redhat distribution. The various packages have gotten old and I can no longer been able to build asterisk on this machine. I have prepaired another 2650 running SUSE Enterprise 10.1 sp2 (my workplace standard) to replace it with. When I swap in the card in my new machine Asterisk starts without error, however the spans remain in red alarm and will pass no calls. Ztcfg indicates the correct signalling... being found for the zap channels. On the SUSE box with the wct4xxp module loaded (and the card NOT installed) loads the following: lsmod wct4xxp 411904 0 firmware_class 25984 1 wct4xxp zaptel203140 1 wct4xxp crc_ccitt 18432 1 zaptel I am suspecting that echo cancelling firmware is being loaded (I don't have an echo cancellation module installed on the TE410P) and I'm not sure if any firmware should be loaded for the te410p – or how to tell if it is or if I can prevent it from being loaded. Lsmod on the working redhat box is simply, wct4xxp89472 96 zaptel184224 240 [wct4xxp] I have also attempted to do a make distclean and make menuselect with without [*] 1. FIRMWARE-OCT6114-064 [*] 2. FIRMWARE-OCT6114-128 ...but perhaps part is not being cleaned out. Or is this firmware required for this card (and not the echo cancellation) with the current module? Thanks, John ___ -- 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] X100P Card in OFFHOOK state
After I make a call o n the Zaptel Card X100P FXO moduleit remains offhook state as shown here... Signalling Type: FXS Kewlstart Radio: 0re2uk*CLI Owner: None*CLI Real: Nonek*CLI Callwait: NoneI Threeway: NoneI Confno: -12uk*CLI Propagated Conference: -1 Real in conference: 0 DSP: noore2uk*CLI Relax DTMF: noCLI Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: noLI Pulse phone: noLI Echo Cancellation: 128 taps, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Offhook -- Sometimes it still takes a new call while in this state and sometimes rejects it... How to correct it such that after I hangup a call it goes back to onhook state... reloading wcfxo module using modprobe clears the issue ___ -- 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] is shared_lastcall available in 1.4
On Tue, 2008-08-26 at 17:53 +0300, Atis Lezdins wrote: Are there any plans to back port this feature into upcoming 1.4 releases? No, new features are added only in trunk, and released in next major release (1.6). So what would be involved in back porting this feature for our system? Do I simply follow the diff from the link you provided and apply the highlighted changes to the app_queue.c file in my Asterisk source directory before recompiling? ___ -- 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 connected to the PSTN vs. a commercial solution
Hi Alejandro - Dear all, now I'm using an Asterisk 1.4.13 SIP server with 50 SIP users and it works very well only in an intranet environment (no connections to the PSTN world). But in the near future, we have to plan a telephone system that works in the intranet (voip) and also it must be connected to the PSTN public network with a T1/E1 trunk, with 200 SIP users aproximately. So at first I have to ways to do that: 1- Continue using Asterisk and adding a T1/E1 interface in order to connect to the PSTN This is exactly what asterisk was designed to do. 2- Discard Asterisk and buy a commercial solution, because we have the money My questions are: does Asterisk work in the scenario I've described Yes. I've used it in just the way you describe in a number of production environments with great success. What is the best solution you can recommend to me ??? Get what you WANT. Both Asterisk and commercial solutions will probably work well for you (just be sure to use quality hardware). With asterisk you get great flexibility and expandability. With a commercial solution you get less of that, but you get to blame someone else if the system fails. Talk to management. What do THEY want? As has been discussed here before, nobody ever got fired for buying Cisco, but that doesn't mean Cisco is any better than any other vendor, including Digium/Asterisk. Find out what the needs of your company are and get the system that best fits those needs. - Noah ___ -- 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] Call transfer over IAX trunk
Hi Andrea - I have two asterisk servers, an IAX trunk between and some SIP users registered to each server. The scenario is this: user A, registered to PBX 1, calls user B, registered to PBX 2. Then A wants to transfer the call using the features.conf method (in my case, **), but is unable to do this. What flags do you have in your Dial() statement? If you want both parties to be able to transfer with the features.conf transfer, you need to have 'Tt' in your dial statement, like this: Dial(IAX2/user:[EMAIL PROTECTED]/exten,20,Tt) - Noah ___ -- 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] Limit to the length of string ?
Gordon Henderson wrote: On Tue, 26 Aug 2008, Tilghman Lesher wrote: On Tuesday 26 August 2008 06:49:54 Gordon Henderson wrote: A-Ha... That string is 256 characters long... Now there's a fishy number if ever there was one. So, if this a real limitation? This is 1.2.30 if that makes a difference... Did this limit go away in 1.4 ? Yes, it did. OK. Thanks. Now I guess I have to play the 1.4 lottery :) (or will it get fixed in 1.2.31 ;-) Is there a maximum string length for use with the legacy interface chan_string? Does it depend on the type of cup used? Does styrofoam give better range than paper? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] DUNDI Help
Ron,
What does the peers section in dundi.conf look like?
On Tue, Aug 26, 2008 at 3:00 AM, ronald ramos [EMAIL PROTECTED] wrote:
Would like to try setting up dundi with 3-4 asterisk.
But for poc, i would like to try setting up dundi on between 2 asterisk.
I copied the config from DUNDI enterprise SIP with no password. Only thing i
changed is the part where i used regcontext.
on both boxes dundi.conf i have
[mapping]
priv = sipregistrations,0,SIP,[EMAIL PROTECTED],nopartial
i can see both peers on each server:
CLI dundi show peers
EID HostModel AvgTime Status
00:8e:8c:8e:cb:5310.10.10.XX (S) Symmetric Unavail OK (1 ms)
i can see my extension being added on sipregistrations context
Added extension '136101' priority 1 to sipregistrations
tried a dundi lookup but got no result
dundi lookup [EMAIL PROTECTED]
DUNDi lookup returned no results.
DUNDi lookup completed in 0 ms
here's what's on extensions.conf
; Private DUNDi network
[dundi-priv-canonical]
; Direct numbers
[dundi-priv-customers]
; If you are an ITSP or Reseller, list your customers here.
[dundi-priv-via-pstn]
; If you are freely delivering calls to the PSTN, list them here
[dundi-priv-local]
include = dundi-priv-canonical
include = dundi-priv-customers
include = dundi-priv-via-pstn
[dundi-priv-switch]
; Just a wrapper for the switch
switch = DUNDi/priv
[dundi-priv-lookup]
include = dundi-priv-local
include = dundi-priv-switch
[macro-dundi-priv]
exten = s,1,Goto(${ARG1}|1)
include = dundi-priv-lookup
[diallocal]
exten = _1X,1,Macro(dundi-priv|${EXTEN})
i also tried dialing from my xlite:
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
Macro(SIP/138100-08269548, dundi-priv|136101) in new stack
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
Goto(SIP/138100-08269548, 136101|1) in new stack
[Aug 26 15:58:07] -- Goto (macro-dundi-priv,136101,1)
[Aug 26 15:58:07] == Auto fallthrough, channel 'SIP/138100-08269548'
status is 'UNKNOWN'
any guess what's wrong? Thanks
ron
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Re: [asterisk-users] DUNDI Help
Hi Bruce,
my apologies, but the error was because of the key.
i just run keys init on the CLI and it works,
question on regcontext though, i set it to sipregistrations, how often does an
extension be added to the context sipregistrations and for how long will it
stay there? i'm looking at dialplan show sipregistration, sometimes i only see
one extension there. even though i know i have 4 ip phones registered to the
asterisk.
TIA
Ron
--- On Tue, 8/26/08, Bruce Reeves [EMAIL PROTECTED] wrote:
From: Bruce Reeves [EMAIL PROTECTED]
Subject: Re: [asterisk-users] DUNDI Help
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Date: Tuesday, August 26, 2008, 6:23 PM
Ron,
What does the peers section in dundi.conf look like?
On Tue, Aug 26, 2008 at 3:00 AM, ronald ramos [EMAIL PROTECTED]
wrote:
Would like to try setting up dundi with 3-4 asterisk.
But for poc, i would like to try setting up dundi on between 2 asterisk.
I copied the config from DUNDI enterprise SIP with no password. Only thing
i
changed is the part where i used regcontext.
on both boxes dundi.conf i have
[mapping]
priv = sipregistrations,0,SIP,[EMAIL PROTECTED],nopartial
i can see both peers on each server:
CLI dundi show peers
EID HostModel AvgTime Status
00:8e:8c:8e:cb:5310.10.10.XX (S) Symmetric Unavail OK (1 ms)
i can see my extension being added on sipregistrations context
Added extension '136101' priority 1 to sipregistrations
tried a dundi lookup but got no result
dundi lookup [EMAIL PROTECTED]
DUNDi lookup returned no results.
DUNDi lookup completed in 0 ms
here's what's on extensions.conf
; Private DUNDi network
[dundi-priv-canonical]
; Direct numbers
[dundi-priv-customers]
; If you are an ITSP or Reseller, list your customers here.
[dundi-priv-via-pstn]
; If you are freely delivering calls to the PSTN, list them here
[dundi-priv-local]
include = dundi-priv-canonical
include = dundi-priv-customers
include = dundi-priv-via-pstn
[dundi-priv-switch]
; Just a wrapper for the switch
switch = DUNDi/priv
[dundi-priv-lookup]
include = dundi-priv-local
include = dundi-priv-switch
[macro-dundi-priv]
exten = s,1,Goto(${ARG1}|1)
include = dundi-priv-lookup
[diallocal]
exten = _1X,1,Macro(dundi-priv|${EXTEN})
i also tried dialing from my xlite:
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
Macro(SIP/138100-08269548, dundi-priv|136101) in
new stack
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
Goto(SIP/138100-08269548, 136101|1) in new stack
[Aug 26 15:58:07] -- Goto (macro-dundi-priv,136101,1)
[Aug 26 15:58:07] == Auto fallthrough, channel
'SIP/138100-08269548'
status is 'UNKNOWN'
any guess what's wrong? Thanks
ron
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Bruce Reeves, dCAp
EUS Networks
Office: 212-624-5943
Web: www.euscorp.com
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Re: [asterisk-users] implementing an intercom with asterisk
On Aug 26, 2008, at 2:27 AM, Gordon Henderson wrote: Do you have some sort of IP connectivity between the sites? 400 yards is a too long for copper cat5, but can be done with fibre, wireless or free-space optics... (which I don't personally recommend!) The current plan is wireless bridge + directional antennae. That wasn't the problem I needed to solve. (And if you haven't IP how are you talking to the phones between sites?) So what's to stop you from putting a Cisco phone into auto-answer mode and calling it via ths Page() application? This is an industrial environment. I'm looking for a slightly less expensive (and hopefully more robust) device - whether an intercom unit + ATA or a magic black box that does everything I want and has a power plug and an ethernet jack. Dedicating a $175 cisco phone to this is overkill, IMO. I had given thought to this, it is a backup plan, but again, I'd like to get something perhaps less expensive to the function. -j ___ -- 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 intercom with asterisk
On Aug 26, 2008, at 5:34 AM, Chris Mason (Lists) wrote: Jonathan Disher wrote: He has two buildings (the office, and the shop proper), separated by about 3-400 yards. Your inter-building distance exceeds ethernet over copper limits, you will need a fiber link. Fiber would be great, if I could bury it, which I can't. I can string it on poles, but I don't want the phone system to go down every time a thunderstorm blows a tree limb into the cable (and, being South Carolina, that would be often). Wireless is probably the way to roll. -j ___ -- 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] DUNDI Help
It is added when a phone registers, or re-registers. Depending on the
timing of the registrations and any restarts on the asterisk process
it may take some time for phones to re-register.
On Tue, Aug 26, 2008 at 2:10 PM, ronald ramos [EMAIL PROTECTED] wrote:
Hi Bruce,
my apologies, but the error was because of the key.
i just run keys init on the CLI and it works,
question on regcontext though, i set it to sipregistrations, how often does
an extension be added to the context sipregistrations and for how long will
it stay there? i'm looking at dialplan show sipregistration, sometimes i
only see one extension there. even though i know i have 4 ip phones
registered to the asterisk.
TIA
Ron
--- On Tue, 8/26/08, Bruce Reeves [EMAIL PROTECTED] wrote:
From: Bruce Reeves [EMAIL PROTECTED]
Subject: Re: [asterisk-users] DUNDI Help
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Date: Tuesday, August 26, 2008, 6:23 PM
Ron,
What does the peers section in dundi.conf look like?
On Tue, Aug 26, 2008 at 3:00 AM, ronald ramos [EMAIL PROTECTED]
wrote:
Would like to try setting up dundi with 3-4 asterisk.
But for poc, i would like to try setting up dundi on between 2 asterisk.
I copied the config from DUNDI enterprise SIP with no password. Only thing
i
changed is the part where i used regcontext.
on both boxes dundi.conf i have
[mapping]
priv = sipregistrations,0,SIP,[EMAIL PROTECTED],nopartial
i can see both peers on each server:
CLI dundi show peers
EID HostModel AvgTime Status
00:8e:8c:8e:cb:5310.10.10.XX (S) Symmetric Unavail OK (1 ms)
i can see my extension being added
on sipregistrations context
Added extension '136101' priority 1 to sipregistrations
tried a dundi lookup but got no result
dundi lookup [EMAIL PROTECTED]
DUNDi lookup returned no results.
DUNDi lookup completed in 0 ms
here's what's on extensions.conf
; Private DUNDi network
[dundi-priv-canonical]
; Direct numbers
[dundi-priv-customers]
; If you are an ITSP or Reseller, list your customers here.
[dundi-priv-via-pstn]
; If you are freely delivering calls to the PSTN, list them here
[dundi-priv-local]
include = dundi-priv-canonical
include = dundi-priv-customers
include = dundi-priv-via-pstn
[dundi-priv-switch]
; Just a wrapper for the switch
switch = DUNDi/priv
[dundi-priv-lookup]
include =
dundi-priv-local
include = dundi-priv-switch
[macro-dundi-priv]
exten = s,1,Goto(${ARG1}|1)
include = dundi-priv-lookup
[diallocal]
exten = _1X,1,Macro(dundi-priv|${EXTEN})
i also tried dialing from my xlite:
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
Macro(SIP/138100-08269548, dundi-priv|136101) in
new stack
[Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
Goto(SIP/138100-08269548, 136101|1) in new stack
[Aug 26 15:58:07] -- Goto (macro-dundi-priv,136101,1)
[Aug 26 15:58:07] == Auto fallthrough, channel
'SIP/138100-08269548'
status is 'UNKNOWN'
any guess what's wrong? Thanks
ron
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--
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Bruce Reeves, dCAp
EUS Networks
Office: 212-624-5943
Web: www.euscorp.com
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*
Bruce Reeves, dCAp
EUS Networks
Office: 212-624-5943
Web: www.euscorp.com
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Re: [asterisk-users] implementing an intercom with asterisk
On Tue, 26 Aug 2008, Jonathan Disher wrote: On Aug 26, 2008, at 2:27 AM, Gordon Henderson wrote: Do you have some sort of IP connectivity between the sites? 400 yards is a too long for copper cat5, but can be done with fibre, wireless or free-space optics... (which I don't personally recommend!) The current plan is wireless bridge + directional antennae. That wasn't the problem I needed to solve. Good luck there then... (don't use Wi-Fi - go for something more robust!) (And if you haven't IP how are you talking to the phones between sites?) So what's to stop you from putting a Cisco phone into auto-answer mode and calling it via ths Page() application? This is an industrial environment. I'm looking for a slightly less expensive (and hopefully more robust) device - whether an intercom unit + ATA or a magic black box that does everything I want and has a power plug and an ethernet jack. Dedicating a $175 cisco phone to this is overkill, IMO. I had given thought to this, it is a backup plan, but again, I'd like to get something perhaps less expensive to the function. What about a $40 Grandstream BT200? (Or whatever they are where you are - I can buy them for under £40 in the UK) Or if there are going to be phones on every desk, just double them up as intercoms... I know zilch about Ciscos, but this can be done with Grandstreams and Snoms, so I imagine that Ciscos being a market leader will have this capability as a matter of course ... Gordon___ -- 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 intercom with asterisk
Some one already touched on this, but my guess is the Nortel system is sending the page signal out to an actual paging system and the speakers are in the remote building or the page port on the Nortel is running over cat 3 copper to the other building. in either case tie it in to the Asterisk system via SIP ATA or FXO port on the box. I have done a number of these setups with an extra FXO port connected to a bogen or viking system, even page pac. On Tue, Aug 26, 2008 at 3:02 PM, Jonathan Disher [EMAIL PROTECTED] wrote: On Aug 26, 2008, at 2:27 AM, Gordon Henderson wrote: Do you have some sort of IP connectivity between the sites? 400 yards is a too long for copper cat5, but can be done with fibre, wireless or free-space optics... (which I don't personally recommend!) The current plan is wireless bridge + directional antennae. That wasn't the problem I needed to solve. (And if you haven't IP how are you talking to the phones between sites?) So what's to stop you from putting a Cisco phone into auto-answer mode and calling it via ths Page() application? This is an industrial environment. I'm looking for a slightly less expensive (and hopefully more robust) device - whether an intercom unit + ATA or a magic black box that does everything I want and has a power plug and an ethernet jack. Dedicating a $175 cisco phone to this is overkill, IMO. I had given thought to this, it is a backup plan, but again, I'd like to get something perhaps less expensive to the function. -j ___ -- 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 -- * Bruce Reeves, dCAp EUS Networks Office: 212-624-5943 Web: www.euscorp.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] Limit to the length of string ?
On Tue, 26 Aug 2008, Drew Gibson wrote: Gordon Henderson wrote: On Tue, 26 Aug 2008, Tilghman Lesher wrote: On Tuesday 26 August 2008 06:49:54 Gordon Henderson wrote: A-Ha... That string is 256 characters long... Now there's a fishy number if ever there was one. So, if this a real limitation? This is 1.2.30 if that makes a difference... Did this limit go away in 1.4 ? Yes, it did. OK. Thanks. Now I guess I have to play the 1.4 lottery :) (or will it get fixed in 1.2.31 ;-) Is there a maximum string length for use with the legacy interface chan_string? Does it depend on the type of cup used? Does styrofoam give better range than paper? You need a layer one technology interface that's firm enough to stop the string being pulled out, (via a washer/knot interface) yet flexible enough to pickup vibrations and transfer them down the layer 2 technology... (Good parcel string) In tests, with Heinz Baked bean tins, we found they worked very well over distances of several metres... Gordon ___ -- 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] Limit to the length of string ?
On 26 Aug 2008, at 18:33, Drew Gibson wrote: Is there a maximum string length for use with the legacy interface chan_string? Does it depend on the type of cup used? Does styrofoam give better range than paper? regards, Drew DTMF modes include: as audio, tugging on the string correct number of times, or holding up correct digits. ___ -- 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] Atlanta Asterisk User's Group Conference Tonight Tuesday, August 26th at 7PM EDT
AAUG Meeting - Tuesday, August 26th at 7PM EDT August 26th, 2008 Here’s how to participate on Tuesday, August 26th at 7PM EDT. The audio portion of the program will be available via a conference bridge at 1-404-492-8060. The shared desktop is available using a Java enabled browser at “http://callin.xelatec.com/vnc” with a password of “aretta”. Our agenda will be general announcements, news, a list of upcoming presentations and John Mullinex will tell us what he’s learned from a sample of the new Sangoma USB FXO device. Also we will describe the popular methods to dial into the conference bridge directly via SIP or IAX and not through the telephone network! Connecting to the Atlanta Asterisk Users Group Conference Bridge via the Internet The following methods enable you to connect to the conference bridge over the Internet without using a public telephone number or line. The first method is to configure routes in your Asterisk server dialplan. For example, the following entry in /etc/asterisk/extensions_custom.conf and context [from-internal-custom] (or the equivalent file and context for your Asterisk system) routes calls dialed to extension 6345 to the AtlAUG Conference Bridge via a SIP connection over the Internet. exten = 6345,1,Set(CALLERID(all)=YourName6785551234) exten = 6345,n,Dial(SIP/[EMAIL PROTECTED]) And this next entry in the same file routes calls to extension 6446 to the Conference Bridge but via an IAX2 trunk. exten = 6346,1,Set(CALLERID(all)=YourName6785551234) exten = 6346,n,Dial(IAX2/[EMAIL PROTECTED]/2284) You can also use the free Zoiper (http://www.zoiper.com) softphone client on your computer to connect directly to the Conference Bridge without going through an Asterisk server. Of course you must first have Zoiper installed and then add a new Zoiper IAX account with Account name ‘AtlaugConf’, Server Hostname ‘pbx.aretta.net’, Username ‘guest’, and no password or other information. Select Show Advanced Options for that account and uncheck “Register on startup’. Apply the new account and click OK. Then from the main user interface, select the new account, go off hook and dial ‘2284′. That should connect you to the conference. Finally, for those really brave souls, you can also connect using the ITAD number of ‘2284*455′. ITAD details are available at “http://www.freenum.org/cookbook/”. http://atlaug.com/blog/ ___ -- 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] Atlanta Asterisk User's Group Conference Tonight Tuesday, August 26th at 7PM EDT
On Tue, Aug 26, 2008 at 05:10:35PM -0400, Asterisk wrote: The shared desktop is available using a Java enabled browser at ???http://callin.xelatec.com/vnc??? with a password of ???aretta???. Of course you must first have Zoiper installed and then add a new Zoiper IAX account with Account name ???AtlaugConf???, Server Hostname ???pbx.aretta.net???, Username ???guest???, and no password or other information. Select Show Advanced Options for that account and uncheck ???Register on startup???. Apply the new account and click OK. Then from the main user interface, select the new account, go off hook and dial ???2284???. That should connect you to the conference. Finally, for those really brave souls, you can also connect using the ITAD number of ???2284*455???. ITAD details are available at ???http://www.freenum.org/cookbook/???. Please don't post to mailing lists in non-7bit-ASCII unless you really have no other choice? That's how Mutt rendered your message here. I *think* those ???'s represent smart-quotes, but I really can't tell... and I'm probably not alone. 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] implementing an intercom with asterisk
We've done a similar thing at a metal worker here by running xlite on a pc set to auto answer and with the speaker out of the pc connected to an amplifier which runs to the speakers. One way paging though. Sorry for the top post, doesn't let me comment inline. On 8/27/08, Bruce Reeves [EMAIL PROTECTED] wrote: Some one already touched on this, but my guess is the Nortel system is sending the page signal out to an actual paging system and the speakers are in the remote building or the page port on the Nortel is running over cat 3 copper to the other building. in either case tie it in to the Asterisk system via SIP ATA or FXO port on the box. I have done a number of these setups with an extra FXO port connected to a bogen or viking system, even page pac. On Tue, Aug 26, 2008 at 3:02 PM, Jonathan Disher [EMAIL PROTECTED] wrote: On Aug 26, 2008, at 2:27 AM, Gordon Henderson wrote: Do you have some sort of IP connectivity between the sites? 400 yards is a too long for copper cat5, but can be done with fibre, wireless or free-space optics... (which I don't personally recommend!) The current plan is wireless bridge + directional antennae. That wasn't the problem I needed to solve. (And if you haven't IP how are you talking to the phones between sites?) So what's to stop you from putting a Cisco phone into auto-answer mode and calling it via ths Page() application? This is an industrial environment. I'm looking for a slightly less expensive (and hopefully more robust) device - whether an intercom unit + ATA or a magic black box that does everything I want and has a power plug and an ethernet jack. Dedicating a $175 cisco phone to this is overkill, IMO. I had given thought to this, it is a backup plan, but again, I'd like to get something perhaps less expensive to the function. -j ___ -- 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 -- * Bruce Reeves, dCAp EUS Networks Office: 212-624-5943 Web: www.euscorp.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 -- Sent from Gmail for mobile | mobile.google.com Matt Riddell Director VentureVoIP ___ -- 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 connected to the PSTN vs. a commercial solution
Asterisks greatest strength is that it's a highly flexible platform that let's you pretty much do anything. It's downside, is that it's a highly flexible platform that let's you pretty much do anything. In other words, the quality of what you are trying to do depends on the quality and volume of the development and testing. If you want something that just works and if you want somebody to be willing to answer the phone and fix the problems that _will_ happen, then I recommend looking at the commercial products. Ironically, many of those commercial products use Asterisk. That's okay. The key is that they have vetted and developed it for you. The reason I bring this up, is that asking if Asterisk supports T1/E1 interfaces to do PSTN implies that you might not have done a lot of research. You have a lot left to go if you want to roll-your-own solution. Please forgive me if my impression is wrong about this. John Alejandro Cabrera Obed wrote: Dear all, now I'm using an Asterisk 1.4.13 SIP server with 50 SIP users and it works very well only in an intranet environment (no connections to the PSTN world). But in the near future, we have to plan a telephone system that works in the intranet (voip) and also it must be connected to the PSTN public network with a T1/E1 trunk, with 200 SIP users aproximately. So at first I have to ways to do that: 1- Continue using Asterisk and adding a T1/E1 interface in order to connect to the PSTN 2- Discard Asterisk and buy a commercial solution, because we have the money My questions are: does Asterisk work in the scenario I've described What is the best solution you can recommend to me ??? Thanks in advance, 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 ___ -- 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] app_jack and calling with pc only
Hello everyone! Sorry, if the whole task is silly, I'm new to this. I have my newly installed asterisk (1.6.0-beta9) and my AVM Fritz a1 card. I have a simple German isdn line and I have a microphone, headphones and a running JACKd (JACK Aduio Connection Kit). The question: Can I (mis)use my asterisk CLI interface to make and recieve calls coming in/going out via the ISDN-card, while using my soundcard I/Os under JACK as a phone? Why I'm doing this and not use another app: 1. I'm blind, I LOVE my console/commandline 2. I tried linphone with SIP, didn't work. JACK crashed and the firewall is in the way. 3. The others don't have JACK and I need my JACK running (soundcard too big for the simple ALSA stuff and I'm a musician often in need of JACK's services. So asterisk seems to offer all I need. I know it's meant as a SERVER, but with all this horse-power: Is a simple client so far of the track? 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] sip peering between 2 asterisk
Nhadie
Can you copy and paste your sip.conf settings for those two servers?? i think
there is a problem with your settings..
regards
Tarek Sawah
Date: Tue, 26 Aug 2008 09:00:52 +0800 From: [EMAIL PROTECTED] To:
asterisk-users@lists.digium.com Subject: Re: [asterisk-users] sip peering
between 2 asterisk Hi Tariq, Tnx for your reply. Tried adding the
deny/permit but still gave me the same result. I still have these error,
handle_response_invite: Failed to authenticate on INVITE regards,
nhadie Tariq .. wrote: im not sure this will help but i did the same
settings you mentioned and added my lines and it worked.. you need
some sort of authentication between the Asterisk boxes.. and the easiest
way to do it is to do it like this[asterisk-2] type=peer host
10.20.30.2 *** i will assume that you have the = sign after the host
context=from-remote-asterisk insecure=port,invite deny=0.0.0.0/0.0.0.0
permit=10.20.30.2/0.0.0.0and do the same on the other server and
you are done.. test it and let me know how did it go ... salam
Tarek Sawah http://www.tareksawah.com/
Date: Mon, 25 Aug 2008 21:06:51 +0800 From: [EMAIL PROTECTED] To:
asterisk-users@lists.digium.com Subject: [asterisk-users] sip peering
between 2 asterisk Hi, I have 2 asterisk on 2 separate
location: sip.conf of asterisk-1 [asterisk-2]
type=peer host 10.20.30.2 context=from-remote-asterisk
insecure=port,invite sip.conf of asterisk-2 [asterisk-1]
type=peer host 10.20.30.1 context=from-remote-asterisk
insecure=port,invite extensions.conf on asterisk-1
[from-remote-asterisk] exten = _1X,1,Dial(SIP/${EXTEN}) exten
= _1X,n,Hangup extensions.conf on asterisk-2
[from-remote-asterisk] exten = _1X,1,Dial(SIP/${EXTEN}) exten
= _1X,n,Hangup when i am registered on asterisk-1 i
called an extension on asterisk-2, this is what happens; ip
phone --INVITE-- asterisk-1 asterisk-1 --407 Proxy Authentication
Required-- ip phone ip phone --ACK-- asterisk-1 ip phone
--INVITE-- asterisk-1 asterisk-1 --Trying-- ip phone since
the extension is on asterisk-2, asterisk -1 will will send invite to
asterisk-2 asterisk-1 --INVIITE-- asterisk-2 asterisk-2 --407
Proxy Authentication Required-- asterisk-1 asterisk-1 --ACK--
asterisk-2 asterisk-1 --Forbidden-- ip phone (this part i don't get,
after sending ACK to asterisk-2 it suddenly send Forbidden to IP phone)
it seems like, asterisk-2 still trying to authenticate the IP phone
even though it was already authenticated on asterisk-1. on
asterisk-1 this is a NOTICE on the console: [Aug 25 21:00:30] --
Called [EMAIL PROTECTED] [Aug 25 21:00:30] NOTICE[840]: chan_sip.c:12322
handle_response_invite: Failed to authenticate on INVITE what
could i be doing wrong? having insecure=port,invite i think should not
authenticate calls from the other asterisk anymore, at least that's how
i understand it. regards, nhadie
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[asterisk-users] FreeTDS Versions?
Does any have some good experience with the various freetds variants? Is 0.64 better or worse than 0.82? I know that to use 0.82 you have to use ODBC, since libtds.a is not long installed. Which is more stable? I plan on using it for CDR, realtime and func_odbc. I'm connecting to SQL Server. I've had a few crashes with 0.82, I think, and I haven't used 0.64. Norman Franke Answering Service for Directors, Inc. www.myasd.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
[asterisk-users] [OT] Re: sip peering between 2 asterisk
Tariq .. schrieb:
i think there is a problem with your settings..
Date: Tue, 26 Aug 2008 09:00:52 +0800 From: [EMAIL PROTECTED] To:
asterisk-users@lists.digium.com Subject: Re: [asterisk-users] sip peering
between 2 asterisk Hi Tariq, Tnx for your reply. Tried adding the
deny/permit but still gave me the same result. I still have these error,
handle_response_invite: Failed to authenticate on INVITE regards,
nhadie Tariq .. wrote: im not sure this will help but i did the same
settings you mentioned and added my lines and it worked.. you need
some sort of authentication between the Asterisk boxes.. and the easiest
way to do it is to do it like this[asterisk-2] type=peer host
10.20.30.2 *** i will assume that you have the = sign after the host
context=from-remote-asterisk insecure=port,invite
deny=0.0.0.0/0.0.0.0 permit=10.20.30.2/0.0.0.0and do the same on
the other server and you are done.. test it and let me know how did it
go ... salam Tarek Sawah
http://www.tareksawah.com/
Date: Mon, 25 Aug 2008 21:06:51 +0800 From: [EMAIL
PROTECTED] To: asterisk-users@lists.digium.com
Subject: [asterisk-users] sip peering between 2 asterisk
Hi, I have 2 asterisk on 2 separate location:
sip.conf of asterisk-1 [asterisk-2]
type=peer host 10.20.30.2
context=from-remote-asterisk insecure=port,invite
sip.conf of asterisk-2 [asterisk-1] type=peer
host 10.20.30.1 context=from-remote-asterisk
insecure=port,invite extensions.conf on asterisk-1
[from-remote-asterisk] exten =
_1X,1,Dial(SIP/${EXTEN}) exten = _1X,n,Hangup
extensions.conf on asterisk-2
[from-remote-asterisk] exten =
_1X,1,Dial(SIP/${EXTEN}) exten = _1X,n,Hangup
when i
am registered on asterisk-1 i called an extension on asterisk-2, this is
what happens; ip phone --INVITE-- asterisk-1 asterisk-1 --407
Proxy Authentication Required-- ip phone ip phone --ACK-- asterisk-1
ip phone --INVITE-- asterisk-1 asterisk-1 --Trying-- ip phone
since the extension is on asterisk-2, asterisk -1 will will send invite
to asterisk-2 asterisk-1 --INVIITE-- asterisk-2 asterisk-2
--407 Proxy Authentication Required-- asterisk-1 asterisk-1 --ACK--
asterisk-2 asterisk-1 --Forbidden-- ip phone (this part i don't get,
after sending ACK to asterisk-2 it suddenly send Forbidden to IP phone)
it seems like, asterisk-2 still trying to authenticate the IP phone
even though it was already authenticated on asterisk-1. on
asterisk-1 this is a NOTICE on the console: [Aug 25 21:00:30] --
Called [EMAIL PROTECTED] [Aug 25 21:00:30] NOTICE[840]: chan_si
p.c:12322 handle_response_invite: Failed to authenticate on INVITE
what could i be doing wrong? having insecure=port,invite i think should
not authenticate calls from the other asterisk anymore, at least that's
how i understand it. regards, nhadie
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Tariq, sorry to say but there is a problem with *your* settings.
The quoted text comes out as a single concatenated string without
any line breaks. Please try to fix how your e-mail client sents
messages or else nobody will be able to follow the discussion.
Thanks.
--
Philipp Kempgen
http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied -
[asterisk-users] Need application, CID number match list to call cell phone
Hi All, I received a request for a special application and need some guidance. Cust has there own Asterisk PBX with SIP phones, pretty standard setup. They want an after hours application that checks inbound caller ID numbers and matches them to a list, say 5 to 10 numbers of special VIP customers, if there is a match on the list, then forward the call straight to a cell phone, instead of ringing local extension and then to voicemail. The customer also wants to be able to manage this VIP list and the call forward cell phone number themselves, so it needs to be configured, numbers added and deleted, through a web page on the PBX. So I'm thinking I need a dialplan app that has to interface with a MySQL database that holds the list of numbers, so I can build a webpage to add/delete the numbers. Any ideas would be much appreciated. Thanks. JR - JR Richardson Engineering for the Masses ___ -- 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 application, CID number match list to call cell phone
Hey JR, Is this a one VIP to one cell number match? Or is it on VIP to multiple cells? On Tue, Aug 26, 2008 at 7:28 PM, JR Richardson [EMAIL PROTECTED] wrote: Hi All, I received a request for a special application and need some guidance. Cust has there own Asterisk PBX with SIP phones, pretty standard setup. They want an after hours application that checks inbound caller ID numbers and matches them to a list, say 5 to 10 numbers of special VIP customers, if there is a match on the list, then forward the call straight to a cell phone, instead of ringing local extension and then to voicemail. The customer also wants to be able to manage this VIP list and the call forward cell phone number themselves, so it needs to be configured, numbers added and deleted, through a web page on the PBX. So I'm thinking I need a dialplan app that has to interface with a MySQL database that holds the list of numbers, so I can build a webpage to add/delete the numbers. Any ideas would be much appreciated. Thanks. JR - JR Richardson Engineering for the Masses ___ -- 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 -- * Bruce Reeves, dCAp EUS Networks Office: 212-624-5943 Web: www.euscorp.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
[asterisk-users] Codec and CPU load
Hi as maximum link capacity could be calculated using codecs and channel types so , regarding the CPU and processors load , Is there any formula or (any relations could help ) that can give the maximum CPU load (mainly processor and RAM ) or scalability average using asterisk channels , codecs , applications …. Ayman _ See what people are saying about Windows Live. Check out featured posts. http://www.windowslive.com/connect?ocid=TXT_TAGLM_WL_connect2_082008___ -- 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] Atlanta Asterisk User's Group Conference Tonight Tuesday, August 26th at 7PM EDT
Jay R. Ashworth wrote: On Tue, Aug 26, 2008 at 05:10:35PM -0400, Asterisk wrote: The shared desktop is available using a Java enabled browser at ???http://callin.xelatec.com/vnc??? with a password of ???aretta???. Of course you must first have Zoiper installed and then add a new Zoiper IAX account with Account name ???AtlaugConf???, Server Hostname ???pbx.aretta.net???, Username ???guest???, and no password or other information. Select Show Advanced Options for that account and uncheck ???Register on startup???. Apply the new account and click OK. Then from the main user interface, select the new account, go off hook and dial ???2284???. That should connect you to the conference. Finally, for those really brave souls, you can also connect using the ITAD number of ???2284*455???. ITAD details are available at ???http://www.freenum.org/cookbook/???. Please don't post to mailing lists in non-7bit-ASCII unless you really have no other choice? That's how Mutt rendered your message here. I *think* those ???'s represent smart-quotes, but I really can't tell... and I'm probably not alone. Cheers, -- jra Now, Jay... it's the global telecom world! Not everyone speaks ASCII-only. That's a little bit like standing in the United Nations and complaining that not everyone speaks 'murican. ;) That said, you're correct. They're smart-quotes. I'm guessing it was copy-pasted from a Word doc or some such. N. ___ -- 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 application, CID number match list to call cell phone
On Tuesday 26 August 2008 19:28:17 JR Richardson wrote:
I received a request for a special application and need some guidance.
Cust has there own Asterisk PBX with SIP phones, pretty standard
setup.
They want an after hours application that checks inbound caller ID
numbers and matches them to a list, say 5 to 10 numbers of special VIP
customers, if there is a match on the list, then forward the call
straight to a cell phone, instead of ringing local extension and then
to voicemail.
The customer also wants to be able to manage this VIP list and the
call forward cell phone number themselves, so it needs to be
configured, numbers added and deleted, through a web page on the PBX.
So I'm thinking I need a dialplan app that has to interface with a
MySQL database that holds the list of numbers, so I can build a
webpage to add/delete the numbers.
Any ideas would be much appreciated.
Sounds like a good use of func_odbc, something along the lines of:
func_odbc.conf:
[APPROVED]
dsn=asterisk-mysql
read=SELECT COUNT(*) FROM approved_table WHERE callerid='${ARG1}'
extensions.conf:
exten = foo,1,GotoIf(${ODBC_APPROVED(${CALLERID(num)})}?callout)
exten = foo,n,Voicemail(foo,u)
And then your web app is pretty easily just a frontend to your database table.
--
Tilghman
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Re: [asterisk-users] Need application, CID number match list to call cell phone
Hi JR, This may help you - we were using it to route calls from friends through the IVR so they hit us directly. You'll have to modify it to suit your dialplan, but it should be a good starting point. http://www.voipphreak.ca/2006/11/26/asterisk-14-php-rolodex-howto-script/ Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JR Richardson Sent: Tuesday, August 26, 2008 8:28 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Need application, CID number match list to call cell phone Hi All, I received a request for a special application and need some guidance. Cust has there own Asterisk PBX with SIP phones, pretty standard setup. They want an after hours application that checks inbound caller ID numbers and matches them to a list, say 5 to 10 numbers of special VIP customers, if there is a match on the list, then forward the call straight to a cell phone, instead of ringing local extension and then to voicemail. The customer also wants to be able to manage this VIP list and the call forward cell phone number themselves, so it needs to be configured, numbers added and deleted, through a web page on the PBX. So I'm thinking I need a dialplan app that has to interface with a MySQL database that holds the list of numbers, so I can build a webpage to add/delete the numbers. Any ideas would be much appreciated. Thanks. JR - JR Richardson Engineering for the Masses ___ -- 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] Limit to the length of string ?
Is there a maximum string length for use with the legacy interface chan_string? Does it depend on the type of cup used? Does styrofoam give better range than paper? regards, Drew A lighter material for the cup will give better dynamic range than a heavier one, at the expense of durability. PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk connected to the PSTN vs. a commercial solution
Asterisks greatest strength is that it's a highly flexible platform that let's you pretty much do anything. It's downside, is that it's a highly flexible platform that let's you pretty much do anything. In other words, the quality of what you are trying to do depends on the quality and volume of the development and testing. That's one of the best statements about deploying asterisk that I've yet read. 1) Research Research Research 2) Plan Plan Plan 3) Build/Implement 4) Test Test Test Test 5) Deploy If you don't feel like doing steps 1, 2, and 4, then go with a commercial solution where they've already done those things for you. You'll likely sacrifice flexibility, but those things are taken care of (or should be) by the vendor. - Noah ___ -- 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] X100P Card in OFFHOOK state
Any pointers on this one? --- On Tue, 8/26/08, Jay Ray [EMAIL PROTECTED] wrote: From: Jay Ray [EMAIL PROTECTED] Subject: [asterisk-users] X100P Card in OFFHOOK state To: asterisk-users@lists.digium.com Date: Tuesday, August 26, 2008, 12:24 PM After I make a call o n the Zaptel Card X100P FXO moduleit remains offhook state as shown here... Signalling Type: FXS Kewlstart Radio: 0re2uk*CLI Owner: None*CLI Real: Nonek*CLI Callwait: NoneI Threeway: NoneI Confno: -12uk*CLI Propagated Conference: -1 Real in conference: 0 DSP: noore2uk*CLI Relax DTMF: noCLI Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: noLI Pulse phone: noLI Echo Cancellation: 128 taps, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Offhook -- Sometimes it still takes a new call while in this state and sometimes rejects it... How to correct it such that after I hangup a call it goes back to onhook state... reloading wcfxo module using modprobe clears the issue ___ -- 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] X100P Card in OFFHOOK state
El mar, 26-08-2008 a las 19:46 -0700, Jay Ray escribió: Any pointers on this one? --- On Tue, 8/26/08, Jay Ray [EMAIL PROTECTED] wrote: From: Jay Ray [EMAIL PROTECTED] Subject: [asterisk-users] X100P Card in OFFHOOK state To: asterisk-users@lists.digium.com Date: Tuesday, August 26, 2008, 12:24 PM After I make a call o n the Zaptel Card X100P FXO moduleit remains offhook state as shown here... Signalling Type: FXS Kewlstart Radio: 0re2uk*CLI Owner: None*CLI Real: Nonek*CLI Callwait: NoneI Threeway: NoneI Confno: -12uk*CLI Propagated Conference: -1 Real in conference: 0 DSP: noore2uk*CLI Relax DTMF: noCLI Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: noLI Pulse phone: noLI Echo Cancellation: 128 taps, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Offhook -- Sometimes it still takes a new call while in this state and sometimes rejects it... How to correct it such that after I hangup a call it goes back to onhook state... reloading wcfxo module using modprobe clears the issue Sounds like your card is not detecting the busy tone, try adding the following line at your zapata.conf file: busydetect=yes busycount=6 Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 USA : 1 360 968 1701 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting smime.p7s Description: S/MIME cryptographic signature ___ -- 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] X100P Card in OFFHOOK state
It would be clearer if it said Hookstate (FXS ports only): Offhook i.e. the state information is not valid for FXO ports. Jay Ray wrote: Any pointers on this one? --- On Tue, 8/26/08, Jay Ray [EMAIL PROTECTED] wrote: From: Jay Ray [EMAIL PROTECTED] Subject: [asterisk-users] X100P Card in OFFHOOK state To: asterisk-users@lists.digium.com Date: Tuesday, August 26, 2008, 12:24 PM After I make a call o n the Zaptel Card X100P FXO moduleit remains offhook state as shown here... Signalling Type: FXS Kewlstart Radio: 0re2uk*CLI Owner: None*CLI Real: Nonek*CLI Callwait: NoneI Threeway: NoneI Confno: -12uk*CLI Propagated Conference: -1 Real in conference: 0 DSP: noore2uk*CLI Relax DTMF: noCLI Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: noLI Pulse phone: noLI Echo Cancellation: 128 taps, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Offhook -- Sometimes it still takes a new call while in this state and sometimes rejects it... How to correct it such that after I hangup a call it goes back to onhook state... reloading wcfxo module using modprobe clears the issue ___ -- 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 -- 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
[asterisk-users] Asterisk for calling no of users
Hi ,
I am new user of asterisk.
here is my environment which is setup on Suse linux 10.0.
zaptel-1.4.11
libpri-1.4.7
asterisk-1.4.21.2
E1 Line.
and i have configured extension.conf,zapata.conf and able to make the
outgoing call from call files and originate command and incoming call also
working fine.
but accoring to our requiremnet most imporment thing is to get user msisdn
for billing and CDR generation.
1) when i trying to print MSISDN with ${CALLERID(num)},it is printing
sometimes msisdn(mobile no) and sometimes only blank.
2) I am using following command line to make call but it will try to connect
to channel 1 or any available channel .and if it will be busy it will not
try again and call files will keep trying to call customer even though
customer disconnect the call.
originate Zap/1/MSISDN extension @incoming
originate Zap/g0/MSISDN extension @incoming
I wanted to configure system for telemarketing.
System - takes no from database - make out going call - choose
appropriate channel if busy than add it into queue.
Please let me know what is the best way to configure above approach as
originate command/call files are not giving me call back for success/failed.
3) what is the best way to track above system ?
4) I read about voicexml...is it how most of the commercial IVR configured ?
Please let me know best free open source software for voicexml and i tried
http://www.i6net.com/.
waiting for your reply.
Thanks Regards,
Samir.
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[asterisk-users] Digium Coffee anyone? PCI Expresso? WTF?
I'll be that none of the other coffee makers can handle anywhere NEAR 60 voice channels, and don't get me started about HPEC! http://www1.shopzilla.com/8N_-_cat_id--13050802__oid--680459759 ___ -- 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
