Re: [asterisk-users] Dial command for SIP driver with To-header config

Thanks Matt, I adjusted my code to trim the URI scheme. Regards, Nitesh On Tue, Apr 26, 2016 at 2:45 PM, Matthew Jordan wrote: > > On Fri, Apr 22, 2016 at 11:04 AM, Nitesh Bansal > wrote: > >> Hello, >> >> I'm using the following Dial command

Re: [asterisk-users] Dial command for SIP driver with To-header config

On Fri, Apr 22, 2016 at 11:04 AM, Nitesh Bansal wrote: > Hello, > > I'm using the following Dial command syntax: > Dial*(SIP/peer/exten!sip:x...@xyz.com *), the SIP URI > after the '!' mark should be set as To-URI in outgoing INVITE > from Asterisk. >

[asterisk-users] Dial command for SIP driver with To-header config

Hello, I'm using the following Dial command syntax: Dial*(SIP/peer/exten!sip:x...@xyz.com *), the SIP URI after the '!' mark should be set as To-URI in outgoing INVITE from Asterisk. It works, but problem is that To-URI formatting is a bit messed up, It looks as follows:

Re: [asterisk-users] Dial command: channel type detection

Some of my users connect to my asterisk box using SIP, other using iax (in users.conf, I set "hasiax=yes" for those users). How do I detect which protocol some user is using ? I cannot find any variable which contains that information. Reason is: I need this information for the Dial() command

[asterisk-users] Dial command: channel type detection

Hi, Some of my users connect to my asterisk box using SIP, other using iax (in users.conf, I set "hasiax=yes" for those users). How do I detect which protocol some user is using ? I cannot find any variable which contains that information. Reason is: I need this information for the Dial()

[asterisk-users] Dial command

I am wondering if its possible to have sometime like this: exten 100 = Dial (g/08039269311) where g would be a group of SIP extensions and i would be parsing/hard coding the PSTN numbers into it, so when i dial extension 100, it passes the call to a group of SIP service provider extensions. i

Re: [asterisk-users] Dial command

this command will not work. what is your main purpose? do u need to have a conference with a group of sip phones? best On Tue, Feb 15, 2011 at 3:13 PM, ayodele abejide ayodeleabej...@hotmail.com wrote: I am wondering if its possible to have sometime like this: exten 100 = Dial

Re: [asterisk-users] Dial command

A. (CCNA) +2348039269311 Before long, paying for a phone call will be as alien as paying for email From: l...@lopl.net Date: Tue, 15 Feb 2011 16:14:22 +0330 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dial command this command will not work. what is your main purpose? do u

Re: [asterisk-users] Dial command

On Tue, Feb 15, 2011 at 01:06:16PM +, ayodele abejide wrote: I want to trunk outbound calls through a sip provider to PSTN, and i want to write a script to parse the PSTN numbers, so when say extension 100 is dialled it just starts to dial PSTN numbers through the SIP provider, so it justs

[asterisk-users] Dial command with r parameter - no ring tone

Hi, Any ideas why? If I leave it out - there is ring tone passed through. Using g729 codec. Sip based call...___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] Dial command with r parameter - no ring tone

On 6 Mar 2009, at 08:58, Shaun Wingrin wrote: Hi, Any ideas why? If I leave it out - there is ring tone passed through. Using g729 codec. Sip based call... What version of asterisk, what licenses, what endpoints, what transcoding? S ___ --

[asterisk-users] Dial command

When I call an extension on my Asterisk system, and the extension is unplugged, I just get silence for the 30 seconds (Dial command ring time) before it goes to voice mail. How can I get around this? Michael ___ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Dial command

Michael schrieb: When I call an extension on my Asterisk system, and the extension is unplugged, I just get silence for the 30 seconds (Dial command ring time) before it goes to voice mail. How can I get around this? qualify=yes in sip.conf? Philipp Kempgen --

Re: [asterisk-users] Dial Command Option D Early Bridged

How do other applications, such a the automated dialers from telemarketers, reliably detect when the call has been answered ? I thought this sort of basic functionality that had been around for quite awhile. Jared Smith wrote: On Thu, 2008-06-12 at 16:43 +0800, tcchan wrote: However, in

Re: [asterisk-users] Dial Command Option D Early Bridged

On Tue, 2008-06-24 at 05:54 -0400, Al Baker wrote: How do other applications, such a the automated dialers from telemarketers, reliably detect when the call has been answered ? I thought this sort of basic functionality that had been around for quite awhile. For digital connections (such as

[asterisk-users] Dial Command Option D Early Bridged

Dear All, The documentation of the Dial Command, says the following about Option D: D([called][:calling]) - Send the specified DTMF strings *after* the called party has answered, but before the call gets bridged. However, in my experience, the timing the call get bridged is not

[asterisk-users] Dial command and its g option

I need to execute an action after a call is hangup. I just see the command Dial has an option for that, the g option. I configure the dial command as exten = s,n,Dial(SIP/100,100,Ttg) How should I add the line which the command will be executed after the dial command in this example? I don`t

Re: [asterisk-users] Dial command and its g option

just add as many extensions as you want under the Dial command extension keeping the extension number same: exten = s,n,Dial(SIP/100,100,Ttg) exten = s,n,Application here On Thu, Jun 12, 2008 at 3:25 PM, voip crazy [EMAIL PROTECTED] wrote: I need to execute an action after a call is hangup.

Re: [asterisk-users] Dial command and its g option

[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, June 12, 2008 12:25 PM Subject: [asterisk-users] Dial command and its g option I need to execute an action after a call is hangup. I just see the command Dial has an option

Re: [asterisk-users] Dial Command Option D Early Bridged

On Thu, 2008-06-12 at 16:43 +0800, tcchan wrote: However, in my experience, the timing the call get bridged is not consistance, Do you happen to be calling out over an analog phone line? In the case of dialing out an analog line, we have no easy way of knowing when the far-end has answered the

Re: [asterisk-users] Dial command and its g option

snip BUT if the two legs hangup, you have to use DEADAGI on the h extension.. Quick note, he doesn't necessarily have to use DeadAGI unless it's an AGI being called. He just has to make sure he defines the h extension in that context and set up the same executions as the post-dial

[asterisk-users] Dial Command with the L switch again

Hello, Is there some way to play different files to caller and callee when using the L-switch in the dial command? I can only send the same sound files to them with the available options, but it would be nice to have a workaround with this functionality. On a sidenote, Digium developers tells me

[asterisk-users] Dial() Command Parameter L Overflow?

I have two Asterisk Systems. One on of those, when I execute this: Dial(SIP/teleglobe-007931d0, SIP/[EMAIL PROTECTED]|60|oL(400752:6:3)) ... It causes Asterisk to immediately read out the time limit of the call (66,792 minutes), as soon as the other end answers, even though we aren't

Re: [asterisk-users] Dial() Command Parameter L Overflow?

On Wed, 19 Sep 2007, Douglas Garstang wrote: Anyone got any idea what might be causing this? Maybe one was compiled in 32 bit mode and an Integer value is overflowing? How do I check this? This is certainly possible, but you might save yourself some time debugging by posting this to the

Re: [asterisk-users] Dial() command h and H options for SIP channel

23 feb 2007 kl. 14.06 skrev ast guy: Hi, Just need to confirm whether dial() command provided options h: Allow the callee to hang up by dialing * H: Allow the caller to hang up by dialing * work for SIP channels as well ? Yes, Asterisk is a multiprotocol Open Source PBX. Those functions

[asterisk-users] Dial() command h and H options for SIP channel

Hi, Just need to confirm whether dial() command provided options h: Allow the callee to hang up by dialing * H: Allow the caller to hang up by dialing * work for SIP channels as well ? -ag ___ --Bandwidth and Colocation provided by Easynews.com --

Re: [asterisk-users] Dial() command h and H options for SIP channel

On 18:06, Fri 23 Feb 07, ast guy wrote: Hi, Just need to confirm whether dial() command provided options h: Allow the callee to hang up by dialing * H: Allow the caller to hang up by dialing * work for SIP channels as well ? yes. It's a function in asterisk call thingie, not in the sip

[asterisk-users] Dial command option D(digits)

I am using asterisk 1.2.9.1. I had been using the option D to send some dtmf tones after the call is answered. This doesnt seem to be working for me now. I am using and IAX2 connection from one machine to another. my extensions.conf has: exten= 57,1,Dial(IAX2/boxa_to_boxb/597,,tD(101))

Re: [Asterisk-Users] Dial Command Reference for SIP channel

On Fri, 12 May 2006, Dave Morrow [EMAIL PROTECTED] wrote Hi all.  I was reading a sample config someone had posted relating to call forwarding, and in it, they use a Dial command with components that I cannot find any reference to.   Can someone point me to a reference which could explain the

Re: [Asterisk-Users] Dial Command Reference for SIP channel

Dave Morrow wrote: Hi all. I was reading a sample config someone had posted relating to call forwarding, and in it, they use a Dial command with components that I cannot find any reference to. Can someone point me to a reference which could explain the difference between

Re: [Asterisk-Users] Dial Command Reference for SIP channel

Doug Lytle wrote: Dave Morrow wrote: Hi all. I was reading a sample config someone had posted relating to call forwarding, and in it, they use a Dial command with components that I cannot find any reference to. Can someone point me to a reference which could explain the difference between

[Asterisk-Users] Dial Command Reference for SIP channel

Hi all. I was reading a sample config someone had posted relating to call forwarding, and in it, they use a Dial command with components that I cannot find any reference to. Can someone point me to a reference which could explain the difference between Dial(SIP/100|20|Ttr,,wW) and

[Asterisk-Users] Dial command

I have an ZAP extension number 222 which is connected instead to a phone to a FXS/FXO converter and from there to a CDMA gateway. To dial my mobile phone I use: 222 (wait 2 seconds) 09123456789 I cannot figure out how to write this into the dialplan as a default number! 222 as above I will

[Asterisk-Users] Dial command

I have an ZAP extension number 222 which is connected instead to a phone to a FXS/FXO converter and from there to a CDMA gateway. To dial my mobile phone I use: 222 (wait 2 seconds) 09123456789 I cannot figure out how to write this into the dialplan as a default number! 222 as above I will

RE: [Asterisk-Users] Dial command

Subject: [Asterisk-Users] Dial command I have an ZAP extension number 222 which is connected instead to a phone to a FXS/FXO converter and from there to a CDMA gateway. To dial my mobile phone I use: 222 (wait 2 seconds) 09123456789 I cannot figure out how to write this into the dialplan

Re: [Asterisk-Users] Dial command to connect two channels and bypass asterisk server

thanks for the information Peter, its really helpful. Also Ihave one more question - do you have any idea how many such simultaneous calls can an asterisk server handle (say running od 2.6Ghz, 1GB Ram, fedora machine)? Thanks, Nitin On 2/13/06, Peter Fern [EMAIL PROTECTED] wrote: You can enable

RE: [Asterisk-Users] Dial command to connect two channels and bypass asterisk server

this is not usefull for public enviroment. clients behind nat does not work... turby From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nitin GuptaSent: Tuesday, February 14, 2006 10:51 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users

RE: [Asterisk-Users] Dial command to connect two channels and bypassasterisk server

AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] Dial command to connect two channels and bypassasterisk server this is not usefull for public enviroment. clients behind nat does not work... turby From: [EMAIL PROTECTED] [mailto

Re: [Asterisk-Users] Dial command to connect two channels and bypassasterisk server

Wai Wu wrote: If Asterisk is in the public network, it will work. The problem is when Asterisk is behind NAT and one of the client is also behind the same NAT. No, it won't. If one of the clients is behind a NAT firewall, you cannot tell that firewall to start accepting media directly from the

Re: [Asterisk-Users] Dial command to connect two channels and bypassasterisk server

Unless you use SIP ALG (Application Layer Gateway) like the module in netfilter to set the expectations? correct? RegardsOn 2/14/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: Wai Wu wrote: If Asterisk is in the public network, it will work. The problem is when Asterisk is behind NAT and one of

Re: [Asterisk-Users] Dial command to connect two channels and bypassasterisk server

Moises Silva wrote: Unless you use SIP ALG (Application Layer Gateway) like the module in netfilter to set the expectations? correct? If that exists on every NAT firewall in the path to both clients, yes. ___ --Bandwidth and Colocation provided by

RE: [Asterisk-Users] Dial command to connect two channelsand bypassasterisk server

] [mailto:[EMAIL PROTECTED] Behalf Of Kevin P. Fleming Sent: Tuesday, February 14, 2006 11:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dial command to connect two channelsand bypassasterisk server Wai Wu wrote: If Asterisk is in the public network

[Asterisk-Users] Dial command to connect two channels and bypass asterisk server

Hi I was wondering if its possible to make Dial commandbridge two channels and after bridgingbypass asterisk, so that the voice doesn't need to pass through my asterisk server. For e.g., I have a user dialed in and he verifies himself and then dials an international extension, after the call

Re: [Asterisk-Users] Dial command to connect two channels and bypass asterisk server

You can enable this on a per-peer basis with: sip peers: canreinvite=yes iax peers: notransfer=no Check the iax.conf.sample and sip.conf.sample files for usage. Nitin Gupta wrote: Hi I was wondering if its possible to make Dial command bridge two channels and after bridging bypass asterisk,

[Asterisk-Users] Dial command exits non-zero

Why does my dial command exit non-zero when the calling party hangs up? I am using a t1 with the following configuration: /*from the zaptel.conf */ pan=1,1,0,esf,b8zs em=1-24 = /*from zapata.conf*/ [channels] language=en signalling=em_w ; change signalling to featd when

Re: [Asterisk-Users] Dial command not executing following priority when caller hangs up

If it worked with 1.0.7 then it was a bug, it should *not* work. The g will only work if one statys on the phone, from show application dial: g- Proceed with dialplan execution at the current extension if the destination channel hangs up. If the origianl channel hangsup as well,

[Asterisk-Users] Dial command not executing following priority when caller hangs up

Hi, I'm using Asterisk 1.2.1 on Sarge. it seems like if I call a phone and nobody answers, asterisk does not jump to the next priority...it freezes. Take a look at this: exten = 777,1,NoOp(before) exten = 777,2,Dial(SIP/7|60|g) exten = 777,3,NoOp(after) priority 3 is never executed but this

[Asterisk-Users] Dial Command Doesn't return Correctly! (Bug?)

I have a really really weird problem here! I'm trying to dial a proxy server, and when that fails, dial another, and finally take some action such as play a sound file. I found that if the proxy server isn't up, Asterisk returns a CONGESTION on the channel, and if the proxy server is up but

[Asterisk-Users] Dial command not working at CLI

Hi everybody, I was once able to type in Dial 200@context and it worked on my other computer, but on this computer it says No such command 'dial'. Somebody told me that it had something to do with soundcard. I activated load = chan_oss.so in modules.conf. But when I type show modules, I don't

Re: [Asterisk-Users] Dial command not working at CLI

To have dial command woking at the cli you have to have the soundcard working and the channel for that. Have you got sound to work on the machines at all? Best regards jan --On Sunday, November 13, 2005 05:59:57 AM -0500 Zeeshan [EMAIL PROTECTED] wrote: Hi everybody, I was once able to

RE: [Asterisk-Users] Dial command not working at CLI

How to make sound card work. Shouldn't Linux find it automatically? Zeeshan A Zakaria -Original Message- From: Jan Saell [mailto:[EMAIL PROTECTED] Sent: Sunday, November 13, 2005 7:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dial

RE: [Asterisk-Users] Dial command not working at CLI

[mailto:[EMAIL PROTECTED] Sent: Sunday, November 13, 2005 7:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dial command not working at CLI To have dial command woking at the cli you have to have the soundcard working and the channel for that. Have

Re: [Asterisk-Users] Dial command not working at CLI

: [Asterisk-Users] Dial command not working at CLI To have dial command woking at the cli you have to have the soundcard working and the channel for that. Have you got sound to work on the machines at all? Best regards jan --On Sunday, November 13, 2005 05:59:57 AM -0500 Zeeshan

Re: [Asterisk-Users] Dial command not working at CLI

Zeeshan wrote: Hi everybody, I was once able to type in Dial 200@context and it worked on my other computer, but on this computer it says No such command 'dial'. Somebody told me that it had something to do with soundcard. I activated load = chan_oss.so in modules.conf. But when I type show

Re: [Asterisk-Users] Dial command in extensions

Kevin Bockman wrote: Patrick wrote: is there anyway to make the dial command return and execute the next line in the dial plan after the channel hangs up? Try with h (for hangup): exten = 1234,1,Dial... exten = 1234,h,... He actually meant the 'h' exten and not priority: exten =

[Asterisk-Users] Dial command in extensions

hi folks. is there anyway to make the dial command return and execute the next line in the dial plan after the channel hangs up? suppose i want to do something like this: exten = 1234,1,dial(SIP/1234) exten = 1234,2,do something but when the dial command hangs up normally, line 2 won't get

Re: [Asterisk-Users] Dial command in extensions

On Mon, 2005-10-17 at 17:04 -0700, Edwin Lam wrote: hi folks. is there anyway to make the dial command return and execute the next line in the dial plan after the channel hangs up? suppose i want to do something like this: exten = 1234,1,dial(SIP/1234) exten = 1234,2,do something

Re: [Asterisk-Users] Dial command in extensions

Edwin Lam wrote: is there anyway to make the dial command return and execute the next line in the dial plan after the channel hangs up? Try g: exten = 1234,1,dial(SIP/1234,,g) exten = 1234,2,do something g: When the called party hangs up, exit to execute more commands in the current

Re: [Asterisk-Users] Dial command in extensions

Patrick wrote: is there anyway to make the dial command return and execute the next line in the dial plan after the channel hangs up? Try with h (for hangup): exten = 1234,1,Dial... exten = 1234,h,... He actually meant the 'h' exten and not priority: exten = h,1,blah but that would

Re: [Asterisk-Users] Dial command nor progressing on Zap channels

Already have that.. On 8/27/05, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote: Eric Bishop wrote: Hi all, Our Asterisk box sends calls outbound via either SIP (through our VoIP provider) or an E1 PRI (directly connected via a TE410P). When we dial a number that is engaged via

[Asterisk-Users] Dial command nor progressing on Zap channels

Hi all, Our Asterisk box sends calls outbound via either SIP (through our VoIP provider) or an E1 PRI (directly connected via a TE410P). When we dial a number that is engaged via our VoIP provider we get the following on the Asterisk console (numbers and IP addresses changed to protect the

Re: [Asterisk-Users] Dial command nor progressing on Zap channels

Eric Bishop wrote: Hi all, Our Asterisk box sends calls outbound via either SIP (through our VoIP provider) or an E1 PRI (directly connected via a TE410P). When we dial a number that is engaged via our VoIP provider we get the following on the Asterisk console (numbers and IP addresses changed

[Asterisk-Users] Dial command problem(VOIP+*+TDM400P+Legacy PBX)

Hello, I just setup the Asterisk to integrate with Panasonic legacy PBX. Config as followings, PSTN -- PanasonicPBX--TDM400P(FXO)--AsteriskPC-- Internet * is for AA / Voicemail and VOIP in/out Currently the AA / Voicemail function for incoming PSTN callsare working well. My problem is

[Asterisk-Users] Dial command announcement

The Dial command can be made to make an announcement to the called party before channel is bridged. Is it possible to make that announcement a Festival command in some way. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com

RE: [Asterisk-Users] Dial Command M(x) Option

-Original Message- From: Shaun Tierney [mailto:[EMAIL PROTECTED] Sent: 02 December 2004 22:37 To: Asterisk Users Subject: [Asterisk-Users] Dial Command M(x) Option http://lists.digium.com/pipermail/asterisk-users/2004-October/ 065540.html I never did find a solution

RE: [Asterisk-Users] Dial Command M(x) Option

http://bugs.digium.com/bug_view_page.php?bug_id=0002905 This email and any attached files are confidential and copyright protected. If you are not the addressee, any dissemination, distribution or copying of this communication is strictly prohibited. Unless otherwise expressly agreed in

RE: [Asterisk-Users] Dial Command M(x) Option

- Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Dial Command M(x) Option http://bugs.digium.com/bug_view_page.php?bug_id=0002905 This email and any attached files are confidential and copyright protected. If you are not the addressee, any dissemination, distribution or copying

RE: [Asterisk-Users] Dial Command M(x) Option

- Non-Commercial Discussion Subject: RE: [Asterisk-Users] Dial Command M(x) Option What version of Asterisk should I be applying this patch to? The patch command doesn't seem to be working. I think because the dates on the files in Asterisk 1.0.2 don't match the dates in the diff file. Any

[Asterisk-Users] Dial Command M(x) Option

http://lists.digium.com/pipermail/asterisk-users/2004-October/065540.html I saw this post about the M(x) option for the Dial command, but I could not find a reply questions posed here. I am wanting to pass the Zap channel that the original call came from to my macro embedded in the Dial command.

[Asterisk-Users] Dial command

HI I have 4 asterisk boxes both loaded from the CVS within a couple hours of each other. The machine that was loaded last does not have the dial command in the CLI. Was this taken out of the latest release? If so why? This is a good trouble shooting tool Ive gotten used to for trouble

[Asterisk-Users] Dial command r option

1. Should the r option of the Dial command always generate a ringing until the called party answers. I have such a scenario but the r option is not generating a ringing, when I use the m option however I do hear music. This does not seem correct. 2. Having read the docs etc is it correct that

[Asterisk-Users] Dial command problems

I am still testing Asterisk, but I am running in to a lot of problems. I set up numerous extensions, but Asterisk is not performing to tasks correctly. Here is an example. exten = 231,1,Dial(Zap/g1/231|3) exten = 231,2,Voicemail(u231) exten = 231,3,Hangup When I call in and enter extension 231,

Re: [Asterisk-Users] Dial command problems

Shouldn't that read: exten = 231,1,Dial(Zap/g1/231,30) Now it should be 30 seconds, which should be 30/6 or about 5 rings. Lyle - Original Message - From: Michael Little [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 13, 2004 12:15 PM Subject: [Asterisk-Users] Dial

Re: [Asterisk-Users] Dial command problems

I'm very new to Asterisk, but try: exten = 231,1,Dial(Zap/g1/231,3,r) exten = 231,2,Voicemail(u231) exten = 231,3,Hangup greeting Siam Am Fri, 13 Aug 2004 13:15:23 -0400 schrieb Michael Little [EMAIL PROTECTED]: I am still testing Asterisk, but I am running in to a lot of problems. I set up

RE: [Asterisk-Users] Dial command problems

Lyle, I tried your suggestion, but it still does not work. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lyle Giese Sent: Friday, August 13, 2004 2:19 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Dial command problems Shouldn't that read

RE: [Asterisk-Users] Dial command problems

I changed the Dial entry, but it still continues to ring the extension without going to voicemail. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of nospams Sent: Friday, August 13, 2004 2:23 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Dial

Re: [Asterisk-Users] Dial command problems

to voicemail and if I have my SIP phone off, it goes directly to voicemail. Lyle - Original Message - From: Michael Little [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 13, 2004 3:11 PM Subject: RE: [Asterisk-Users] Dial command problems Lyle, I tried your suggestion

Re: [Asterisk-Users] Dial command problems

Michael Little wrote: exten = 231,1,Dial(Zap/g1/231|3) exten = 231,2,Voicemail(u231) exten = 231,3,Hangup When I call in and enter extension 231, my call is routed to the correct extension, but it just keeps ringing On the * console, set verbose 100, and post here the output when you try dialing