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. >

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

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

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 ___ --

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

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

Hi, With the g option, you just have to continue in the CALLER Dialplan, you have nothing to do, just continue your Dialplan i.e: exten= s,n,Dial(what you want) = and when the Called hangup you're goto the next line exten= s,n,Goto(where you want) or exten= s,n, 'DO WHAT YOU WANT: playback,

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

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

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

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

RE: [Asterisk-Users] Dial command

Exten = 222,1,Dial(SIP/polycom601||20) Exten = 222,2,Dail(Zap/2/ww09123456789# -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Wednesday, March 08, 2006 5:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion

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

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,

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,

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 =

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

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

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 to the

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

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