Re: [asterisk-users] Problems with TE110P

On Mon, Apr 02, 2007 at 04:21:45PM +1000, Klaverstyn, David C wrote: > Lspci does show: > 03:02.0 Ethernet controller: Digium, Inc.: Unknown device 0120 (rev 11) > > dmesg | tail > ip_tables: (C) 2000-2002 Netfilter core team > ip_tables: (C) 2000-2002 Netfilter core team > tg3: eth0: Link is up a

Re: [asterisk-users] Best Hardphone (Subjective?)

On 4/2/07, Corporate IT Solutions - Michael Dunne < [EMAIL PROTECTED]> wrote: So subjectively what would be the best Hardphone for a small/medium business with multiple line support, BLF, etc. Does _anyone_ read the archives anymore? This is like a weekly question or something.

RE: [asterisk-users] Problems with TE110P

Lspci does show: 03:02.0 Ethernet controller: Digium, Inc.: Unknown device 0120 (rev 11) dmesg | tail ip_tables: (C) 2000-2002 Netfilter core team ip_tables: (C) 2000-2002 Netfilter core team tg3: eth0: Link is up at 100 Mbps, full duplex. tg3: eth0: Flow control is on for TX and on for RX. lp: dr

[asterisk-users] Best Hardphone (Subjective?)

After working with the Grandstream GXP 2000 series phones, I have decided that I am quite unhappy with their problems, both voice quality, volume, features and others. For their price now, there are plenty of phones to choose from as well. So subjectively what would be the best Hardphone for a

Re: [asterisk-users] Problems with TE110P

On Mon, Apr 02, 2007 at 03:38:18PM +1000, Joel Hill wrote: > Give this a try it fixes a problem we have had with a couple of Via > boxes. > > modprobe wcte11xp > modprobe wcte11xp > ztfcg -vv > zttool > > We found that probing the card twice before running ztcfg helped alot. Are you sure tha

Re: [asterisk-users] Problems with TE110P

On Mon, Apr 02, 2007 at 02:56:30PM +1000, Klaverstyn, David C wrote: > Type in cat /proc/zaptel/* displays > > Span 1: ZTDUMMY/1 "ZTDUMMY/1 1" The driver has not picked up your card. > > > But if I type in > lsmod | grep -i wct > > I get > wcte11xp 26016 0 > wct4xxp

RE: [asterisk-users] Problems with TE110P

Give this a try it fixes a problem we have had with a couple of Via boxes. modprobe wcte11xp modprobe wcte11xp ztfcg -vv zttool We found that probing the card twice before running ztcfg helped alot. Cheers, Joel. Joel Hill Support Engineer Asterisk IT 03 8320 8100 On Mon, 2007-04-02 at

RE: [asterisk-users] Problems with TE110P

Type in cat /proc/zaptel/* displays Span 1: ZTDUMMY/1 "ZTDUMMY/1 1" But if I type in lsmod | grep -i wct I get wcte11xp 26016 0 wct4xxp 221120 0 zaptel184996 3 ztdummy,wcte11xp,wct4xxp -Original Message- From: [EMAIL PROTECTED] [m

Re: [asterisk-users] OT - IP Network Call Recording

Check out Oreka at sourceforge, too.(aka OrkAudio) On 2/15/07, Kristian Kielhofner <[EMAIL PROTECTED]> wrote: On 2/15/07, Cory Andrews <[EMAIL PROTECTED]> wrote: > Apologies in advance as this is not directly Asterisk related, however I > thought I might be able to leverage the experience of pa

[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 4

Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To U

Re: [asterisk-users] Problems with TE110P

On Mon, Apr 02, 2007 at 12:50:18PM +1000, Klaverstyn, David C wrote: > I have a new server using Zaptel 1.2.16 > > Issuing a ztcfg gives the following error: > > ZT_CHANCONFIG failed on channel 1: No such device or address (6) > > > loadzone=au > > defaultzone=au > > span=1,1,0,ccs,hdb3,crc4

Re: [asterisk-users] Problems with TE110P

On Sun, Apr 01, 2007 at 11:34:20PM -0400, C F wrote: > > On 4/1/07, Klaverstyn, David C <[EMAIL PROTECTED]> wrote: > >I have a new server using Zaptel 1.2.16 > > > > > > > >Issuing a ztcfg gives the following error: > > > >ZT_CHANCONFIG failed on channel 1: No such device or address (6) > > Looks

RE: [asterisk-users] Trigger and Email in Dial Plan

Put this in the incoming context for that number called. Exten => s,1,Wait(1) Exten => s.2.System("mail -s 'Smitty called from ${CALLERID(all)' [EMAIL PROTECTED]") Exten => s,3,Congestion From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of R

Re: [asterisk-users] Trigger and Email in Dial Plan

On Sun, Apr 01, 2007 at 08:35:23PM -0700, Robert DeVries wrote: > I have a friend traveling overseas. I want to allow him to call a number > which will give him a busy signal (so no charge), but will then send me an > email that he has called. > > I know how to use a call file to trigger a call (

Re: [asterisk-users] Asterisk hangs up SIP call after 6 200 retransmits

>One potential reason could be that the ACK request being sent to Asterisk is >malformed. Notice >"branch=0" in the top Via. This should start with "z9hG4bK" >magic cookie since the INVITE was an RFC >3261 transaction. >While "branch=0" is valid in RFC 2543, I don't think an INVITE can start-of

[asterisk-users] Trigger and Email in Dial Plan

I have a friend traveling overseas. I want to allow him to call a number which will give him a busy signal (so no charge), but will then send me an email that he has called. I know how to use a call file to trigger a call (I created a callback system for myself when I traveled overseas a few mon

Re: [asterisk-users] Problems with TE110P

Looks like a UDEV problem On 4/1/07, Klaverstyn, David C <[EMAIL PROTECTED]> wrote: I have a new server using Zaptel 1.2.16 Issuing a ztcfg gives the following error: ZT_CHANCONFIG failed on channel 1: No such device or address (6) Issuing the command lsmod | grep -i wct results in:

RE: [asterisk-users] Problems with TE110P

I forgot to mention that my /etc/zaptel.conf file contains: loadzone=au defaultzone=au span=1,1,0,ccs,hdb3,crc4 bchan=1-10 unused=11-15,17-31 dchan=16 I have a new server using Zaptel 1.2.16 Issuing a ztcfg gives the following error: ZT_CHANCONFIG failed on channel 1: No

Re: [asterisk-users] Polycom and Asterisk

Flawless as far as I know. Bruce Reeves wrote: Matt, I am running Polycom 2.1 on both 1.4 and 1.2 svn releases without any problems. What kind of issues did you experience? On 3/28/07, *Mike Hammett * <[EMAIL PROTECTED] > wrote: I was previously having an iss

Re: [asterisk-users] Call dies when I press *

Is this while using queues? Mike Diehl wrote: Hi all, I've trying to fix a problem. If I'm in a call and I press the * key, the call goes silent but doesn't hang up. I need to be able to send the * key for various IVR's that I interact with. Since I thought this was related to the feature

Re: [asterisk-users] Dialplan Streaming

Madplay Doug Garstang wrote: Oh poo. No one seems to know. :( Doug Garstang wrote: All, Is there a dial plan command that can stream uncompressed audio from another source? I see there's an MP3Player command that can stream, but I assume that plays MP3's, which means it has to decode them.

[asterisk-users] Problems with TE110P

I have a new server using Zaptel 1.2.16 Issuing a ztcfg gives the following error: ZT_CHANCONFIG failed on channel 1: No such device or address (6) Issuing the command lsmod | grep -i wct results in: wcte11xp 26016 0 wct4xxp 221120 0 zaptel

Re: [asterisk-users] Sponsored development - Monodirectional audio handling

Edoardo Serra wrote: > The purpose of thisi implementation is to deal with some carriers that > give us the call as ANSWERED when the called party is still ringing. > Our billing software is billing the user (and the carrier is billing us) > even with unsuccessful calls. How about you try a dif

Re: [asterisk-users] [MACRO-SCREEN] and MACRO_RESULT

The logic of the macro is totally opposite of what it should be. I do recall sending a corrected version of the script to someone a while back, it might be on the mailing lists archive. However, there is an option for the Dial() command to do exactly what you wish p: This option enables screenin

Re: [asterisk-users] Re: Sponsored development - Monodirectional audio handling

And also you should post it on the voip-info wiki there is a page just with bounties. On 3/31/07, Edoardo Serra <[EMAIL PROTECTED]> wrote: Salvatore Giudice ha scritto: > You could put a bounty on this. You may find someone who will be willing to > write this for money. My Bounty for that featu

[asterisk-users] RE: On Topic: Cheapest Asterisk USB Key?

That's quite interesting. You can get the microdrives cheaper than $50. We recently purchased 2 GB microdrives for $17. Try contacting this company: IPMedia Asia Co. Ltd PO Box 2074 Northbrook Illinois United States 60065 Tel: (886 2) 85227000 Ext : 107 (1 847) 6565759 Fax: (886 2) 66021000

RE: [asterisk-users] [MACRO-SCREEN] and MACRO_RESULT

-Original Message- From: [EMAIL PROTECTED] on behalf of Philipp Kempgen Sent: Sun 4/1/2007 4:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] [MACRO-SCREEN] and MACRO_RESULT Andy Hester wrote: > exten => s,n,Set(TIMEOUT(response=10)) Should

Re: [asterisk-users] [MACRO-SCREEN] and MACRO_RESULT

Andy Hester wrote: > exten => s,n,Set(TIMEOUT(response=10)) Should be exten => s,n,Set(TIMEOUT(response)=10) Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://w

[asterisk-users] [MACRO-SCREEN] and MACRO_RESULT

I am following the example at http://www.voip-info.org/wiki/view/Asterisk+tips+findme but I find that no matter what, the call is connected. Can anyone confirm that config is working for them? Any suggestions appreciated. I need to transfer calls to a list of cell phones, ring all of them, al

Re: [asterisk-users] Speed Dial Application in *

On Sun, 1 Apr 2007, Dovid B wrote: You can try putting out a bounty to make a new app in asterisk or set a variable in the asterisk db. Why a new app? That way you get "forced" into doing it the app way, (like FollowMe in 1.4 which, if I understand the wiki page on it, doesn't do anything re

[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 3

Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To U

[asterisk-users] Add/remove international prefix

I have 2 asterisk servers located in different countries. When there is an outside call incoming in either country the phones ring in both; but the problem is understanding where the call origin is! The incoming number has always no international prefix if it is from the some country where the cal

RE: On Topic: Cheapest Asterisk USB Key? (was: Re: [asterisk-users]Off Topic: Open Source USB Softphone)

> > Here's a flipside of this subject: what is the absolute > cheapest Linux > device that can be connected to a PC's USB port? That has just enough > power for a minimal Asterisk server running on it. The Asterisk just > maintains a CDR database on its Flash memory, which it periodically > s

[asterisk-users] RE: On Topic: Cheapest Asterisk USB Key?

I need a USB microprocessor *device* on which the Linux and Asterisk will run (even if very slowly), not just a storage drive from which to run it on the PC. MonteVista is a good distro, though there are other "minimal" embedded distros, of which I've already got one selected. The CDR usage

RE: [asterisk-users] Weird extension behavior

From: Mark Hennessy <[EMAIL PROTECTED]> Date: Sun, 01 Apr 2007 06:15:40 -0400 Hi, I'm using Asterisk with two Cisco 7960 phones using SIP. I'm seeing the following weird behavior: SIP Phome 1 is extension 4002 SIP Phone 2 is extension 4003 I call 4002 from 4003 and that works fine. I call 4003 f

[asterisk-users] Asterisk 1.2 and res_perl - lock that leads to weird behaviour

Hi guys, as I wrote in a previous thread I was experiencing dropped audio (apparently randomly) and SIP + IAX peers getting REACHABLE / UNREACHABLE without reason, servers were in the same LAN. Investingating deeply in the problem I also noticed that 'show channels' command on the CLI, so

[asterisk-users] Asterisk 1.2 and res_perl - lock that leads to weird behaviour

Hi guys, as I wrote in a previous thread I was experiencing dropped audio (apparently randomly) and SIP + IAX peers getting REACHABLE / UNREACHABLE without reason, servers were in the same LAN. Investingating deeply in the problem I also noticed that 'show channels' command on the CLI, somet

RE: On Topic: Cheapest Asterisk USB Key? (was: Re: [asterisk-users] Off Topic: Open Source USB Softphone)

Try installing Monte Vista http://www.mvista.com/ on the usb stick. It will be a lot cleaner than taking a standard server distribution of linux and stripping out all the unwanted kernel modules. Monte Vista is an embedded linux that should be able to boot your server off a 128mb usb stick with As

RE: [asterisk-users] Security on long distance calls

Using caller id to authenticate anyone is asking for a toll fraud problem. 4-digit pins really are not a good idea either. Try putting your operators and your users in different contexts. If you have specific numbers you don't want the users to be able to dial, then create patterns for those nu

Re: [asterisk-users] Re: SIP/IAX peers UNREACHABLE and audio loss

Hi guys. I have the same problem and donnow why. I set up Nat=yes and Qulify=yes ... and notthin' happens. So..do you think it's related to a perl function? If u find the sol , tell us please. THANKS. El dom, 01-04-2007 a las 18:02 +0200, Edoardo Serra escribió: > Hi guys, > I think I g

Re: [asterisk-users] Announcement: Asterisk Service Provider Edition v1.0 Beta

Olle E Johansson wrote: > Asterisk SPE Nice. ;) Was SpitShare developed by project 0401? Didn't read carefully. Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> htt

Re: [asterisk-users] Announcement: Asterisk Service Provider Editionv1.0 Beta

For immediate release, April 1st 2007 On behalf of the Asterisk development team and project 0401 Project 0401. You don't need to make it an abvious that its an April Fool's joke. ___ --Bandwidth and Colocation provided by Easynews.com -- ast

[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 2

Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To U

[asterisk-users] Re: SIP/IAX peers UNREACHABLE and audio loss

Hi guys, I think I got the point of the problem. I guess it's related to a lock in res_perl (which we use to do lcr, billing, ecc...) I'll open another thread for that Tnx for hep Regards Edoardo Serra WeBRainstorm S.r.l. Edoardo Serra ha scritto: Hi all, I'm having a problem

Re: [asterisk-users] Announcement: Asterisk Service Provider Edition v1.0 Beta

Wow i need a tftp client to download it now . Nice April 1 joke :P . ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-us

[asterisk-users] Announcement: Asterisk Service Provider Edition v1.0 Beta

The Asterisk Developer Team is proud to announce the Asterisk SPE v1.0 Beta Release for immediate download on tftp.digium.com. The SPE has been developed as a joint project between Digium, the Asterisk Company, Voop, the European Asterisk Dialtone provider and the Asterisk community. The A

Re: [asterisk-users] Linking incoming calls

Ronaldo Zacarias Afonso wrote: Hi all, I just want to know how I can make sure that incoming calls to my asterisk server are being treated by [incoming] section of extension.conf file. Thanks in advance. You want to do this kind of check on a consistent basis or are you just checking to make

[asterisk-users] jiaxclient run error

Hi, I have a problem when compile jiaxclient-0.0.6. Mine enviroment is: WindowsXP Cygwin I compile iaxclient library and afterwards I compile jiaxclient: ./configure make make sign make install Everything is ok with the compile and install - generated files: jiaxc__x86.jar jiaxc__x86_md5.j

Re: [asterisk-users] Linking incoming calls

You can put in something like exten => s,x,Noop("Yup it's working right") and look at the cli to see if it comes up. - Original Message - From: "Ronaldo Zacarias Afonso" <[EMAIL PROTECTED]> To: Sent: Sunday, April 01, 2007 4:28 PM Subject: [asterisk-users] Linking incoming calls Hi

Re: [asterisk-users] Anyone here have opinion on the Linksys SPA-400?

On Sun, 1 Apr 2007 16:14:32 +0300, Dovid B wrote: >- Original Message - >From: "Michael Graves" <[EMAIL PROTECTED]> >To: "Asterisk Users Mailing List - Non-Commercial Discussion" > >Sent: Sunday, April 01, 2007 3:59 PM >Subject: [asterisk-users] Anyone here have opinion on the Linksys S

[asterisk-users] Linking incoming calls

Hi all, I just want to know how I can make sure that incoming calls to my asterisk server are being treated by [incoming] section of extension.conf file. Thanks in advance. Ronaldo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-u

Re: [asterisk-users] Question on Priorities

Thank you - Got it now. Makes everything look a lot cleaner :) On Sun, 01 Apr 2007 11:58:20 +0200 Philipp Kempgen <[EMAIL PROTECTED]> wrote: > --[ UxBoD ]-- wrote: > > > [inbound-sip] > > exten => uxbod(u1),1,Dial(sip/1001,20,t) > > exten => uxbod,n,PlayBack(uxbod) > > exten => uxbod,n,Hangup()

Re: [asterisk-users] Anyone here have opinion on the Linksys SPA-400?

- Original Message - From: "Michael Graves" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Sunday, April 01, 2007 3:59 PM Subject: [asterisk-users] Anyone here have opinion on the Linksys SPA-400? I may have need for a small multi-port FXO i/

Re: On Topic: Cheapest Asterisk USB Key? (was: Re: [asterisk-users] Off Topic: Open Source USB Softphone)

Probably not exactly what you're looking for but Astlinux runs on Gumstix. Would be suitable for prototyping. Michael On Sun, 01 Apr 2007 09:08:17 -0400, Matthew Rubenstein wrote: > Here's a flipside of this subject: what is the absolute cheapest Linux >device that can be connected to a P

On Topic: Cheapest Asterisk USB Key? (was: Re: [asterisk-users] Off Topic: Open Source USB Softphone)

Here's a flipside of this subject: what is the absolute cheapest Linux device that can be connected to a PC's USB port? That has just enough power for a minimal Asterisk server running on it. The Asterisk just maintains a CDR database on its Flash memory, which it periodically submits over

[asterisk-users] Anyone here have opinion on the Linksys SPA-400?

I may have need for a small multi-port FXO i/f. Anyone have hands-on experience with this linksys device? On past projects I've used SPA-3000 & Digium TDM400 and found them unsatisfactory. Michael ___ --Bandwidth and Colocation provided by Easynews

Re: [asterisk-users] DTMF Corruption Problem in 1.4.2 for SIP RFC2833 plz halp

I actually meant to take that out before copy and pasting. It was just a zany option I tried to see if things would get better but they didn't. I will post a bug later this week about this. On 3/30/07, Kevin P. Fleming <[EMAIL PROTECTED]> wrote: Justin Tunney wrote: > rfc2833compensate=yes W

RE: [asterisk-users] Multi-Level Queue

How about using one queue to provide the caller with music and position info, then delayed dialling on the mobiles: [queue] timeout=30 retry=0 joinempty=yes member => SIP/201 member => SIP/202 member => SIP/203 member => SIP/204 member => Local/@delaydial member => Local/@delaydial [delaydial] e

[asterisk-users] Weird extension behavior

Hi, I'm using Asterisk with two Cisco 7960 phones using SIP. I'm seeing the following weird behavior: SIP Phome 1 is extension 4002 SIP Phone 2 is extension 4003 I call 4002 from 4003 and that works fine. I call 4003 from 4002, and it rings locally to 4002, never gets to 4003. I'm able to send a

Re: [asterisk-users] h323

Did you compile H.323 for asterisk and then make install asterisk ? - Original Message - From: "Pezhman Lali" <[EMAIL PROTECTED]> To: Sent: Wednesday, March 28, 2007 4:30 PM Subject: [asterisk-users] h323 hi After compiling and installing pwlib and openh323 , the asterisk, give the f

Re: [asterisk-users] Meetme question

Create two seperate extensions. One for the admin and one for the regular users that go to the same room. The issue you will have then is that the admin will have to call in first to create the dynamic room. - Original Message - From: "Adrian Marsh" <[EMAIL PROTECTED]> To: Sent: Satu

Re: [asterisk-users] Question on Priorities

--[ UxBoD ]-- wrote: > [inbound-sip] > exten => uxbod(u1),1,Dial(sip/1001,20,t) > exten => uxbod,n,PlayBack(uxbod) > exten => uxbod,n,Hangup() > exten => uxbod,u1+101,PlayBack(uxbod) > exten => uxbod,n,VoiceMail([EMAIL PROTECTED],s) > exten => uxbod,n,Hangup() > > but when I do a extensions reloa

[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 1

Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To U

Re: [asterisk-users] REG : H.323 Configurations with Asterisk

1) Good luck with H.323 on asterisk. 2) Should be handled by Asterisk it's self. - Original Message - From: "Anisha Kumar" <[EMAIL PROTECTED]> To: Sent: Friday, March 30, 2007 7:03 AM Subject: FW: [asterisk-users] REG : H.323 Configurations with Asterisk Hi , I am new to Asterisk

Re: [asterisk-users] Security on long distance calls

Or you can do exten => _011.,1,Authenticate(1234) exten => _011.,2,Dial(SIP/[EMAIL PROTECTED]) Also this is a bit more complicated but you can do it by sip extension. If CID of phone = phone that is allowed then let it go out. This will be hard considering you will have to make a gotoif for

Re: [asterisk-users] Speed Dial Application in *

You can try putting out a bounty to make a new app in asterisk or set a variable in the asterisk db. Exten => _*XX,1,nswer exten => _*XX,2,Set(speeddial=${EXTEN:1}) exten => _*XX,3,Goto(setspeeddial,s,1) [speeddial] eexten => s,1,Set(TIMEOUT(digit)=3) exten => s,2,Set(TIMEOUT(response)=3) exten =

Re: [asterisk-users] Sponsored development - Monodirectional audiohandling

Try posting this on the developers list. - Original Message - From: "Edoardo Serra" <[EMAIL PROTECTED]> To: Sent: Saturday, March 31, 2007 6:42 PM Subject: [asterisk-users] Sponsored development - Monodirectional audiohandling Hi Guys, we're needing a special implementation on Ast

Re: [asterisk-users] Speed Dial Application in *

On 30 mar 2007, at 14.19, Chris Nighswonger wrote: Hi all, Is there a "speed dial" type application in *? The NEC PBX we currently use has a feature which allows any phone to access a system-wide speed dail database simply by keying the speed-dial number and pressing the 'redial' key from any

Re: [asterisk-users] CallerID + Name

On 29 mar 2007, at 23.01, Rob Schall wrote: We have the caller id with name option enabled with our provider, however, our polycom 501 phones will only display the number of the incoming call. Is there a way to see the callerid name from the cli when the call is coming in (like a print in th

Re: [asterisk-users] Question on Priorities

Okay, I have changed it too :- [inbound-sip] exten => uxbod(u1),1,Dial(sip/1001,20,t) exten => uxbod,n,PlayBack(uxbod) exten => uxbod,n,Hangup() exten => uxbod,u1+101,PlayBack(uxbod) exten => uxbod,n,VoiceMail([EMAIL PROTECTED],s) exten => uxbod,n,Hangup() but when I do a extensions reload I get

Re: [asterisk-users] Question on Priorities

Cool. That is nice and clean :) Many thanks. On Sat, 31 Mar 2007 23:32:45 -0700 "Yuan LIU" <[EMAIL PROTECTED]> wrote: > >From: "Rizwan Hisham" <[EMAIL PROTECTED]> > >Date: Sat, 31 Mar 2007 17:01:51 +0500 > > > >[inbound-sip] > >exten => uxbod,1,Dial(sip/1001,20,jt) > >exten => uxbod,n,Hangup > >

Re: [asterisk-users] Question on Priorities

Yuan LIU wrote: >> From: "Rizwan Hisham" <[EMAIL PROTECTED]> >> Date: Sat, 31 Mar 2007 17:01:51 +0500 >> >> [inbound-sip] >> exten => uxbod,1,Dial(sip/1001,20,jt) >> exten => uxbod,n,Hangup >> >> exten => uxbod,102,PlayBack(uxbod) >> exten => uxbod,103,VoiceMail([EMAIL PROTECTED],s) >> exten => ux