Re: [asterisk-users] Is there a way to compile app_macro in 16.30.1

On 7/10/2023 8:55 PM, Federico wrote: I need to use app_macro, but it seems to be absent from asterisk 16.30.1 Is there a workaround? It's disabled (not built) by default. You'll need to enable it using menuselect[1], and load it in modules.conf Note that app_macro has been removed now and

Re: [asterisk-users] Is there a way to make asterisk send a INVITE in-dialog to re-establish the audio

Thank you Joshua -Original Message- From: asterisk-users On Behalf Of Joshua C. Colp Sent: Friday, May 24, 2019 9:53 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Is there a way to make asterisk send a INVITE in-dialog to re-establish the audio On Fri, May 24

Re: [asterisk-users] Is there a way to make asterisk send a INVITE in-dialog to re-establish the audio

On Fri, May 24, 2019, at 9:47 AM, Dan Cropp wrote: > > We are working with an Avaya switch. > > > We send them a REFER. If the transfer is successful, everything is > great. If it fails (busy), they send an INVITE in-dialog with a media > attribute of inactive. After that, they send a 486

Re: [asterisk-users] Is there any way to pass caller id to

to figure out when it's a call from office :))) Thank you,Ivan Message: 2 Date: Mon, 15 Oct 2018 23:39:31 +0200 From: Daniel Tryba To: Asterisk Users Mailing List - Non-Commercial Discussion     Subject: Re: [asterisk-users] Is there any way to pass caller id to     cell phone? M

Re: [asterisk-users] Is there any way to pass caller id to

On 10/16/18 1:42 PM, Antony Stone wrote: On Tuesday 16 October 2018 at 19:04:42, Ivan Demkovitch wrote: Thanks all, I did contact Callcentric about it and their tech support helped meget those headers established. They even helped to troubleshoot Asterisk dialplan. A the end all works as it

Re: [asterisk-users] Is there any way to pass caller id to

On Tuesday 16 October 2018 at 19:04:42, Ivan Demkovitch wrote: > Thanks all, > I did contact Callcentric about it and their tech support helped meget > those headers established. They even helped to troubleshoot Asterisk > dialplan. A the end all works as it should. For the benefit of others who

Re: [asterisk-users] Is there any way to pass caller id to

To: Asterisk Users Mailing List - Non-Commercial Discussion     Subject: Re: [asterisk-users] Is there any way to pass caller id to     cell phone? Message-ID: <20181015213930.2a4uulq2z6xbfjcb@bogus> Content-Type: text/plain; charset=us-ascii On Thu, Oct 11, 2018 at 05:18:24PM +, Ivan Demkovitch

Re: [asterisk-users] Is there any way to pass caller id to cell phone

-- Message: 2 Date: Mon, 15 Oct 2018 11:12:09 +0300 From: Eric Klein To: asterisk-users Subject: Re: [asterisk-users] Is there any way to pass caller id to cellphone? Message-ID: Content-Type: text/plain; charset="utf-8" Ivan, Be aware that what you a

Re: [asterisk-users] Is there any way to pass caller id to cell phone?

On Thu, Oct 11, 2018 at 05:18:24PM +, Ivan Demkovitch wrote: > Where problem comes in - if person not at the desk - his cell phone shows > call from OFFICE number and there is no way to tell who is really calling. > We use Callcentric as a trunk if it makes any difference. > I'd like to add

Re: [asterisk-users] Is there any way to pass caller id to cell phone?

Ivan, Be aware that what you are asking may cause problems with making the call to the cell phone. Think of it this way, you are taking an inbound call and then sending it out over your regular operator. They may object to accepting a call with a CLID that does not match your account and could

Re: [asterisk-users] Is there any way to pass caller id to cell phone?

On Thursday 11 October 2018 at 22:11:10, Ivan Demkovitch wrote: > Abdul, > Added code like you proposed, I see it in logs but still don't see caller > ID coming in: > -- Goto (internal,101,1) > -- Executing [101@internal:1] NoOp("SIP/callcentric13-06d1", "Call > ID: "DEMKOVITCH,IVAN"

Re: [asterisk-users] Is there any way to pass caller id to cell phone?

  == Spawn extension (internal, 101, 3) exited non-zero on 'SIP/callcentric13-06d1' From: Abdul Basit To: idemkovi...@yahoo.com; Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, October 11, 2018 12:42 PM Subject: Re: [asterisk-users] Is there any way to

Re: [asterisk-users] Is there any way to pass caller id to cell phone?

on-Commercial Discussion Sent: Thursday, October 11, 2018 12:42 PM Subject: Re: [asterisk-users] Is there any way to pass caller id to cell phone? Hi Ivan, Check whats CallerID you are getting before initiating dial command. ;Eric on extension 105 exten => 105,1,NoOp( Call ID: ${CALLER

Re: [asterisk-users] Is there any way to pass caller id to cell phone?

Hi Ivan, Check whats CallerID you are getting before initiating dial command. ;Eric on extension 105 exten => 105,1,NoOp( Call ID: ${CALLERID(all)} ) exten => 105,n,Dial(${ERIC_CELL}&${ERIC_OFFICE},30) same => n,VoiceMail(105@default,u) Also what Caller ID is set on outgoing trunk? Is that

Re: [asterisk-users] Is there a way to remove launching shell command from Asterisk CLI

Yes: I never thought of using sudo to also forbid access some apps. Using it for that is very smart ! Thank you for sharing it here. I'll experiment with this and report here my findings. Thanks again 2018-08-14 19:50 GMT+02:00 John Kiniston : > I use sudo to limit this. > > Cmnd_Alias

Re: [asterisk-users] Is there a way to remove launching shell command from Asterisk CLI

I use sudo to limit this. Cmnd_Alias CAPTAGENT = /sbin/service captagent stop, /sbin/service captagent start, /sbin/service captagent restart Cmnd_Alias ASTERISK = /sbin/service asterisk stop, /sbin/service asterisk start, /sbin/service asterisk restart, /usr/sbin/rasterisk, /usr/sbin/asterisk,

Re: [asterisk-users] double NAT - one way audio

> Can you get your own modem? (double) NAT is ugly hack. Unfortunately not. The provider is only supporting this hardware. > Not sure what is VoIP in the router here, but looks like some sort of SIP ALG > or VoIP passthrough - disable it! It rewrites ip addresses inside of the > packets ang it

Re: [asterisk-users] double NAT - one way audio

On Wednesday 15 of March 2017 07:55:09 Andre Gronwald wrote: > ISP won't change, but will check. > in the hidden menus it isn't changeable either. Can you get your own modem? (double) NAT is ugly hack. > However, it is working after i deactivated VoIP in the router. And even > after reenabling

Re: [asterisk-users] double NAT - one way audio

ISP won't change, but will check. in the hidden menus it isn't changeable either. However, it is working after i deactivated VoIP in the router. And even after reenabling VoIP it is still working. I don't understand why... However, it works. :-D thanks a lot. regards, andre -- Andre Gronwald

Re: [asterisk-users] double NAT - one way audio

Hi Andre, On your comment "unfortunately there is no bridge mode or any comparable mode available", sometimes the carrier (if it's a carrier supplied DSL router) will have these settings hidden from standard user's eyes. You may need to call your ISP and request them to place your DSL router

Re: [asterisk-users] double NAT - one way audio

Hi Glenn, unfortunately there is no bridge mode or any comparable mode available. I am using the same router (but another type) on my private homenetwork with another router at the back (=> same architecture as in this failing scenario), but everything works fine. There are only two differences:

Re: [asterisk-users] double NAT - one way audio

Hi Andre, Some routers just simply won't support this double-nat scenario you describe. Othera will... And without any special forwarding. Is it possible to put the first router into "bridge" mode, and use the second router as the actual NAT router? This may be the quickest solution to your

Re: [asterisk-users] asterisk 13 n-way call problem

On Tue, Dec 22, 2015 at 1:47 AM, Dmitry Melekhov wrote: > I spent some time reading docs and such change is not documented, so this > is bug. > I'll open issue... > > Not necessarily. Certain aspects of features was definitely changed in 13, and may require the use of a pre-dial

Re: [asterisk-users] asterisk 13 n-way call problem

I spent some time reading docs and such change is not documented, so this is bug. I'll open issue... 22.12.2015 10:53, Dmitry Melekhov пишет: Hello! I need to use n-way call as it described here: http://habrahabr.ru/sandbox/52259/ It is in russian, but dial plan is quite clear. It works in

Re: [asterisk-users] Understanding the right way to get started with multiple trunks/extensions

In the 'home-number' example that was provided the caller ID was being replaced with the string 'Home' It's easy to prepend the caller ID instead however. Set(CALLERID(name)=Home-${CALLERID(name)}) You could even get fancy and set it based on what number was called, This would prepend the

Re: [asterisk-users] Understanding the right way to get started with multiple trunks/extensions

For some reason I didn't see David's reply by email, and have copy/pasted the following from the list archives to make my reply, sorry if that messes up anyone's threading. On 4 March 2015 at 12:15, David Duffett wrote: If you would like to set things up via the GUI on your incredible PBX,

Re: [asterisk-users] Understanding the right way to get started with multiple trunks/extensions

If you would like to set things up via the GUI on your incredible PBX, you could use queues instead of call groups (making your SIP clients agents of the appropriate queues), and in the queues configuration page there is an CID Name Prefix option, which allows you to add a label that will show up

Re: [asterisk-users] looking for a way to do appointment reminders

We did something like that - see http://blog.wombatdialer.com/post/24187267017/drstrangelove You can use the free version of the dialer if you have low traffic or just want to run a test. l. 2013/4/26 Ron Wheeler rwhee...@artifact-software.com Good comment. Another feature suggestion You

Re: [asterisk-users] looking for a way to do appointment reminders

Thanks very much to everyone for their ideas for my original posting. You've all given me much to consider and think about. Thanks again, Brandon On 5/2/2013 8:54 AM, Lenz Emilitri wrote: We did something like that - see http://blog.wombatdialer.com/post/24187267017/drstrangelove You can use

Re: [asterisk-users] looking for a way to do appointment reminders

Would love too hear more about this, as we are looking for a solution too. Good comment. Another feature suggestion You might to ask the person to press 1 to confirm or 2 to leave a message if the appointment is not going to be kept or 0 to reach the receptionist to reschedule the

Re: [asterisk-users] looking for a way to do appointment reminders

Hi Brandon, as you are asking for professional help for a commercial project, I would recommend you to place a bounty. You can contact me directly if you want my professional help... I have developed exactly what you´re looking for and this solution is running in a high-call-volume

Re: [asterisk-users] looking for a way to do appointment reminders

Hi Brandon! I have a wakeup call system based on call files that are generated by an external C program. The call files can be triggered by dialing a phone number (e.g. for waking up the hotel guest in room 333 at 6:15 am: *77*3330615) or from outside via a web interface, or whatever. It

Re: [asterisk-users] h323-sip: one way connection

oh yes, i'm using h323 not openh323 On Fri, Apr 26, 2013 at 1:16 AM, Asghar Mohammad asghar...@gmail.comwrote: nuFone h323 or openh323? On Thu, Apr 25, 2013 at 9:33 PM, s m sam.gh1...@gmail.com wrote: flavor? i do not understand what you mean. please explain more. thanks On Wed, Apr

Re: [asterisk-users] looking for a way to do appointment reminders

-Original Message- From: jg webaccou...@jgoettgens.de Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] looking

Re: [asterisk-users] looking for a way to do appointment reminders

Hans, they are currently calling patients. I think these calls apply only to a certain fraction of the patients, who are difficult to contact by other methods. jg -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] looking for a way to do appointment reminders

On 26/4/13 10:38 am, jg wrote: they are currently calling patients. I think these calls apply only to a certain fraction of the patients, who are difficult to contact by other methods. I suspect there will be different requirements depending on how 'helpful' to patients you wish to be. At the

Re: [asterisk-users] looking for a way to do appointment reminders

On 26/4/13 10:14 am, Hans Witvliet wrote: Only reasonable option is to send them an SMS. Given the likelihood that a sizeable percentage of people attending a medical establishment are going to be at the upper end of the age scale, it's possible they may not have mobile phones, and even if

Re: [asterisk-users] looking for a way to do appointment reminders

Chris! Brandon should probably be more specific about what he wants to achieve. It might even be preferable to have a semi-automated system that originates the calls based on a list of callees, available callers, and some timing heuristics. This way the callees would always talk to a human

Re: [asterisk-users] looking for a way to do appointment reminders

On 26/4/13 12:24 pm, jg wrote: This way the callees would always talk to a human being If possible, this would definitely be a Good Thing. Many people (myself included) will disconnect a call as soon as they realise it's a recorded message. It also means the human caller can confirm they

Re: [asterisk-users] looking for a way to do appointment reminders

... Essentially, I suggested a predictive dialer (http://en.wikipedia.org/wiki/Predictive_dialer). In this case this could be a reasonable thing to do. jg -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] looking for a way to do appointment reminders

Good comment. Another feature suggestion You might to ask the person to press 1 to confirm or 2 to leave a message if the appointment is not going to be kept or 0 to reach the receptionist to reschedule the appointment. Ron On 26/04/2013 7:06 AM, Chris Bagnall wrote: On 26/4/13 10:38 am, jg

Re: [asterisk-users] h323-sip: one way connection

try UserByAlias=yes in general and type=user in user context. On Fri, Apr 26, 2013 at 9:48 AM, s m sam.gh1...@gmail.com wrote: oh yes, i'm using h323 not openh323 On Fri, Apr 26, 2013 at 1:16 AM, Asghar Mohammad asghar...@gmail.comwrote: nuFone h323 or openh323? On Thu, Apr 25, 2013 at

Re: [asterisk-users] h323-sip: one way connection

flavor? i do not understand what you mean. please explain more. thanks On Wed, Apr 24, 2013 at 8:16 PM, Asghar Mohammad asghar...@gmail.comwrote: what flavor of h323 you are using? On Wed, Apr 24, 2013 at 8:50 AM, s m sam.gh1...@gmail.com wrote: thanks Asghar, i do it, but no thing

Re: [asterisk-users] h323-sip: one way connection

nuFone h323 or openh323? On Thu, Apr 25, 2013 at 9:33 PM, s m sam.gh1...@gmail.com wrote: flavor? i do not understand what you mean. please explain more. thanks On Wed, Apr 24, 2013 at 8:16 PM, Asghar Mohammad asghar...@gmail.comwrote: what flavor of h323 you are using? On Wed, Apr

Re: [asterisk-users] h323-sip: one way connection

thanks Asghar, i do it, but no thing happened:( asterisk do not identify host line as ip address of the other end On Tue, Apr 23, 2013 at 9:15 PM, Asghar Mohammad asghar...@gmail.comwrote: try type=peer instead of friend. On Tue, Apr 23, 2013 at 10:04 AM, s m sam.gh1...@gmail.com wrote:

Re: [asterisk-users] h323-sip: one way connection

what flavor of h323 you are using? On Wed, Apr 24, 2013 at 8:50 AM, s m sam.gh1...@gmail.com wrote: thanks Asghar, i do it, but no thing happened:( asterisk do not identify host line as ip address of the other end On Tue, Apr 23, 2013 at 9:15 PM, Asghar Mohammad

Re: [asterisk-users] h323-sip: one way connection

i know what is the exactly problem. i enable debug for h323 and it says: could not find user by name 200 or address 192.168.0.146 when i change peer-146 to 200 every thing is ok and i can call from two side. but it is not good for me because 200 is the name of extension and when i config asterisk

Re: [asterisk-users] h323-sip: one way connection

try type=peer instead of friend. On Tue, Apr 23, 2013 at 10:04 AM, s m sam.gh1...@gmail.com wrote: i know what is the exactly problem. i enable debug for h323 and it says: could not find user by name 200 or address 192.168.0.146 when i change peer-146 to 200 every thing is ok and i can call

Re: [asterisk-users] h323-sip: one way connection

please post cli output for both calls. On Mon, Apr 22, 2013 at 11:32 AM, s m sam.gh1...@gmail.com wrote: hello everybody i want to have sip connection between two asterisk systems (145 and 146). connection from 145 to 146 is ok but i can not call from 146 to 145. this is h323.conf file in

Re: [asterisk-users] Is there any way to make call fail after # of rings?

On Fri, 2012-02-17 at 04:00 -0500, CDR wrote: My customer needs to set a forwarding based on number of rings,i.e., if the phone rings 5 times (user-selectable), then try another number. Is there a way to do such a thing with Asterisk? I could not find way to do it based on the documentation of

Re: [asterisk-users] Is there any way to make call fail after # of rings?

Simply, without checking for BUSY, DND or TIMEOUT I'm assuming each ring period is 3 seconds. exten = 8512,1,Dial(SIP/8512,15) exten = 8512,n,Dial(DAHDI/GO/101233456,15) Or another way. Maybe the FollowMe application, allow multiple numbers to be tried, each after a configured timeout. from

Re: [asterisk-users] Is there any way to make call fail after # of rings?

On Friday 17 February 2012, CDR wrote: My customer needs to set a forwarding based on number of rings,i.e., if the phone rings 5 times (user-selectable), then try another number. Is there a way to do such a thing with Asterisk? I could not find way to do it based on the documentation of the

Re: [asterisk-users] Is there any way to make call fail after # of rings?

Try this exten= yournumberhere,1,Dial(SIP/peern1,60) exten= yournumberhere,n,GotoIf($[${DIALSTATUS} != ANSWER]?4) exten= yournumberhere,n,Hangup exten= yournumberhere,n,Dial(SIP/peer2,60) exten= yournumberhere,n,GotoIf($[${DIALSTATUS} != ANSWER]?9) exten= yournumberhere,n,Hangup you can

Re: [asterisk-users] Nat issue one way audio on IP dial

Do you have your softphone setup to use a stun server so it can send it's public IP address in the SIP packets? I see in the SIP debug output a 192.168 address for the RTP packets to go to which of course will not work. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On

Re: [asterisk-users] Is there a way for a peer to clear its registration from a server?

Frank Church wrote: Is there a way for a client to tell a server where it is registered to remove the registration? Assuming you are talking about a SIP peer (since you didn't specify), yes, the SIP peer can cancel the registration by sending an update to the registration and setting the

Re: [asterisk-users] Is there a way for a peer to clear its registration from a server?

Is there a way for a client to tell a server where it is registered to remove the registration? Yes, it needs to send an UNREGISTER sip message. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] Is there a way for a peer to clear its registration from a server?

11 mar 2010 kl. 15.17 skrev Philipp von Klitzing: Is there a way for a client to tell a server where it is registered to remove the registration? Yes, it needs to send an UNREGISTER sip message. There's actually not an UNREGISTER method in SIP. As Kevin stated, you send a REGISTER with a

Re: [asterisk-users] Is there a way for a peer to clear its registration from a server?

Hi There's actually not an UNREGISTER method in SIP. As Kevin stated, you send a REGISTER with a zero expiry to cancel a current registration. Yes, of course you are right there, sorry for the confusion. I was thinking about the resulting Asterisk CLI message: Unregistered SIP 'peername'

Re: [asterisk-users] Is there a way for a peer to clear its registration from a server?

Thanks. Is there command is used for that? I have checked the help show and there is no command like sip register or sip unregister in the list. Is it available on version 1.4? On 11 March 2010 13:08, Kevin P. Fleming kpflem...@digium.com wrote: Frank Church wrote: Is there a way for a

Re: [asterisk-users] No tone, one way communcation.

ystdm8xx+e159:0001 Yeastar YSTDM8xx From: Tzafrir Cohen tzafrir.co...@xorcom.com To: asterisk-users@lists.digium.com Sent: Mon, October 26, 2009 4:05:16 PM Subject: Re: [asterisk-users] No tone, one way communcation. On Mon, Oct 26, 2009 at 05:02:18AM

Re: [asterisk-users] No tone, one way communcation.

On Mon, Oct 26, 2009 at 11:20:07PM -0700, PATRICK KANGETHE wrote: my lsdahdi output is; 1. [r...@elastix ~]# lsdahdi ### Span 1: WCTDM/8 YSTDM8xx REV E Board 9 (MASTER) 1 FXSFXOKS (In use) 2 FXSFXOKS (In use) 3 EMPTY 4 FXS

Re: [asterisk-users] No tone, one way communcation.

Once the card was configured correctly, have you set on the GUI the correct port to your zap extension? On Mon, 26 Oct 2009 05:02:18 -0700 (PDT), PATRICK KANGETHE patricemb...@yahoo.com wrote: 1. When i connected my analog phone to fxs card, i cannot get dial tone what could be the problem?

Re: [asterisk-users] No tone, one way communcation.

On Mon, Oct 26, 2009 at 05:02:18AM -0700, PATRICK KANGETHE wrote: 1. When i connected my analog phone to fxs card, i cannot get dial tone what could be the problem? What is the output of: lsdahdi dahdi_hardware I am using elastix 1.5.2 based on centos 5.2 Final. Consider also asking

Re: [asterisk-users] Is there a way to force a codec on an incoming sip uri call?

long time ago I added the SIP_CODEC variable that you can set from within the dialplan, eg: exten = s,1,Set(SIP_CODEC=alaw) exten = s,n,Answer exten = s,n,whatever now if the remote side actually supports the chosen codec Asterisk will try to use that one ... there's no error reporting as far as

Re: [asterisk-users] Is there a way to get info who disconnected thecall into CDR?

Found it, I use the g flag in Dial command, that helps :) Rennes From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rennes Neps Sent: 1. oktoober 2009. a. 16:05 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Is there

Re: [asterisk-users] Looking for a way to show caller id information on the desktop

On Windows systems you can use Xtelsio http://www.xtelsio.com/ Jonathan Moore schreef: Hi there. My problem, I can't figure out how to ask this question. So, hopefully someone out here can point me to the FM on this. I would like to have either a web page or an application that I can view

Re: [asterisk-users] Looking for a way to show caller id information on the desktop

As an ultra cheap way of doing it, you could simply output the caller-id to a log file and display a tail 20 of it on a web page. Something like this: exten = s,1,System( echo${EPOCH}|${CALLERID(num)} /var/log/asterisk/incoming ) It should be trivial to display the last n lines of it on a web

Re: [asterisk-users] Looking for a way to show caller id information onthe desktop

The asterisk GUI offers this. To Roll your own, simply do an asterisk -rx core show channels and filter that output to your browser. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan Moore Sent:

Re: [asterisk-users] Looking for a way to show caller id information onthe desktop

Un-top-posting... [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan Moore My problem, I can't figure out how to ask this question. So, hopefully someone out here can point me to the FM on this. I would like to have either a web page or an application that I can view

Re: [asterisk-users] Looking for a way to show caller id information onthe desktop

-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Thursday, September 10, 2009 1:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Looking for a way to show caller id information onthe desktop

Re: [asterisk-users] Looking for a way to show caller id information on the desktop

On Thu, 2009-09-10 at 12:21 -0500, Jonathan Moore wrote: I would like to have either a web page or an application that I can view that whenever a call arrives on the Asterisk server the application will display the callerid information. A good friend of mine has Asterisk send a Jabber message

Re: [asterisk-users] Looking for a way to show caller id information onthe desktop

You can also use our jabber/xmpp integration and send an Instant message to the user/desktop before you place the call with dial(). Or do it in the dial() macro as soon as someone answers. /O ___ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Looking for a way to show caller id information onthe desktop

On 11/09/09 6:24 AM, Olle E. Johansson wrote: You can also use our jabber/xmpp integration and send an Instant message to the user/desktop before you place the call with dial(). Or do it in the dial() macro as soon as someone answers. Yep, that's what we do: exten =

Re: [asterisk-users] Is there a way to specify the fromdomain from the dialplan?

Eric Chamberlain schrieb: Is there a way to override the fromdomain specified in the sip.conf and instead set the value from the dialplan? If we use: Set(CALLERID(num)[EMAIL PROTECTED] The SIP From header turns into: [EMAIL PROTECTED]@10.10.10.10 Maybe you could abuse

Re: [asterisk-users] is there a way

Steve Totaro wrote: My only wish is that Linux had a facility like XP to bridge NICs without running all sorts of commands for brctl. Just a GUI like XP. Last time I setup a bridge in Linux, I had to change many kernel options and rebuild the entire kernel to get bridging working

Re: [asterisk-users] is there a way

Brent Davidson wrote: Babcock, Michael Alex wrote: hey; i'm at best western and am curious is there a way i could find out if our best western, with out asking, is using asterisk? oh and petsmart i think is using asterisk they have alason voice for there main voicem enu. mike thanks

Re: [asterisk-users] Is there a way to test SIP credentials withoutmaking a call?

On Oct 11, 2008, at 8:28 PM, Rob Hillis wrote: Eric Chamberlain wrote: Is there a particular reason you /can't/ register? It would seem that registration would provide the functionality you require, even if you're only making outbound calls. In the case of a server like Asterisk,

Re: [asterisk-users] Is there a way to test SIP credentials withoutmaking a call?

On Mon, Oct 13, 2008 at 12:37 PM, Eric Chamberlain [EMAIL PROTECTED] wrote: We're developing the client and don't have control over the server, which may or may not be Asterisk. Adding extra extensions isn't possible. Can OPTION packets be used to verify authentication? -- Eric

Re: [asterisk-users] Is there a way to test SIP credentials without making a call?

At 9:37 AM -0700 2008/10/13, Eric Chamberlain wrote: On Oct 11, 2008, at 8:28 PM, Rob Hillis wrote: Eric Chamberlain wrote: Is there a particular reason you /can't/ register? It would seem that registration would provide the functionality you require, even if you're only making

Re: [asterisk-users] is there a way

Steve Totaro wrote: My only wish is that Linux had a facility like XP to bridge NICs without running all sorts of commands for brctl. Just a GUI like XP. Last time I setup a bridge in Linux, I had to change many kernel options and rebuild the entire kernel to get bridging working properly.

Re: [asterisk-users] Is there a way to test SIP credentials withoutmaking a call?

hi, (i am no sur): the user credential is tested during SIP Registration Step thanks and tel me if this is a error - Original Message - From: Eric Chamberlain [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday,

Re: [asterisk-users] Is there a way to test SIP credentials withoutmaking a call?

I should have clarified, we're only making outbound calls, not inbound, so there is no registration. On Oct 11, 2008, at 9:27 AM, Meftah Tayeb wrote: hi, (i am no sur): the user credential is tested during SIP Registration Step thanks and tel me if this is a error - Original Message

Re: [asterisk-users] Is there a way to test SIP credentialswithoutmaking a call?

then this is a error from me, thanks - Original Message - From: Eric Chamberlain [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, October 11, 2008 6:03 PM Subject: Re: [asterisk-users] Is there a way to test SIP

Re: [asterisk-users] Is there a way to test SIP credentials withoutmaking a call?

Eric Chamberlain wrote: I should have clarified, we're only making outbound calls, not inbound, so there is no registration. Is there a particular reason you /can't/ register? It would seem that registration would provide the functionality you require, even if you're only making

Re: [asterisk-users] Is there a way to test SIP credentials withoutmaking a call?

On Oct 11, 2008, at 1:41 PM, Rob Hillis wrote: Eric Chamberlain wrote: I should have clarified, we're only making outbound calls, not inbound, so there is no registration. Is there a particular reason you /can't/ register? It would seem that registration would provide the functionality

Re: [asterisk-users] Is there a way to test SIP credentials withoutmaking a call?

Eric Chamberlain wrote: Is there a particular reason you /can't/ register? It would seem that registration would provide the functionality you require, even if you're only making outbound calls. In the case of a server like Asterisk, wouldn't sending a register disrupt the flow of

Re: [asterisk-users] is there a way

Babcock, Michael Alex wrote: hey; i'm at best western and am curious is there a way i could find out if our best western, with out asking, is using asterisk? oh and petsmart i think is using asterisk they have alason voice for there main voicem enu. mike thanks for reading Systems

Re: [asterisk-users] is there a way

On Fri, Oct 10, 2008 at 5:55 PM, Brent Davidson [EMAIL PROTECTED] wrote: Babcock, Michael Alex wrote: hey; i'm at best western and am curious is there a way i could find out if our best western, with out asking, is using asterisk? oh and petsmart i think is using asterisk they have

Re: [asterisk-users] is there a way

no i'm a guest at the bestwestern On Oct 10, 2008, at 1:55 PM, Brent Davidson wrote: Babcock, Michael Alex wrote: hey; i'm at best western and am curious is there a way i could find out if our best western, with out asking, is using asterisk? oh and petsmart i think is using asterisk they

Re: [asterisk-users] is there a way

steve; thanks a lot mike On Oct 10, 2008, at 2:20 PM, Steve Totaro wrote: On Fri, Oct 10, 2008 at 5:55 PM, Brent Davidson [EMAIL PROTECTED] wrote: Babcock, Michael Alex wrote: hey; i'm at best western and am curious is there a way i could find out if our best western, with out

Re: [asterisk-users] is there a way

nmap for scanning and identification. cross platform and even a nice gui for windows. Eric On Fri, Oct 10, 2008 at 3:20 PM, Steve Totaro [EMAIL PROTECTED] wrote: On Fri, Oct 10, 2008 at 5:55 PM, Brent Davidson [EMAIL PROTECTED] wrote: Babcock, Michael Alex wrote: hey; i'm at best

Re: [asterisk-users] is there a way

I will look into that when I get my Acer Aspire One running FC8, it came with windows XP and I got the 1gig ram, 120gig HD. I am following threads on howto but nobody has a definitive guide yet, that allows the embedded webcam and the NIC to work properly. Maybe (probably) my USB Alpha AWUS036H

Re: [asterisk-users] is there a way

On Fri, Oct 10, 2008 at 07:33:45PM -0700, Eric Fort wrote: nmap for scanning and identification. cross platform and even a nice gui for windows. What nmap does is called fingerprinting. it mostly uses the fact that when faced with normal behaviours, most stacks behave the same. But when faced

Re: [asterisk-users] [FreeBSD 6.3] Right-way to recover Zaptel?

If Asterisk is running that will happen. Make sure to shutdown asterisk cleanly before doing that. Anthony Vincent wrote: Hello I'm running Asterisk 1.4.20.1 on a FreeBSD that I compiled from the Ports collection. It's the second time I'm having an issue with a FXO card and/or the

Re: [asterisk-users] [FreeBSD 6.3] Right-way to recover Zaptel?

On Sat, 06 Sep 2008 12:47:58 -0600, Anthony Francis [EMAIL PROTECTED] wrote: If Asterisk is running that will happen. Make sure to shutdown asterisk cleanly before doing that. Sorry, forgot to say that I couldn't restart or stop/start Asterisk: [Sep 6 19:06:17] WARNING[23110]: chan_zap.c:4157

Re: [asterisk-users] Is there a way to encrypt passwords stored in the realtime database?

Philippe Sultan [EMAIL PROTECTED] writes: Well, if someone steals the md5secret (HA1) for a given username and realm, he can use it to authenticate to the SIP proxy or B2BUA that serves the target user. This is unavoidable with password-based systems. Either you transfer the password

Re: [asterisk-users] Is there a way to encrypt passwords stored in the realtime database?

On 20 Aug 2008, at 18:00, Eric Chamberlain wrote: We are exploring using Asterisk for a project and we are looking for a way to encrypt/decrypt the peer passwords stored in the realtime database (postrges). Ideally, we want to use a public key to encrypt the passwords before they go into

Re: [asterisk-users] Is there a way to encrypt passwords stored in the realtime database?

On Wed, Aug 20, 2008 at 10:00:55AM -0700, Eric Chamberlain wrote: We are exploring using Asterisk for a project and we are looking for a way to encrypt/decrypt the peer passwords stored in the realtime database (postrges). Ideally, we want to use a public key to encrypt the passwords

Re: [asterisk-users] Is there a way to encrypt passwords stored in the realtime database?

Tzafrir Cohen wrote: On Wed, Aug 20, 2008 at 10:00:55AM -0700, Eric Chamberlain wrote: We are exploring using Asterisk for a project and we are looking for a way to encrypt/decrypt the peer passwords stored in the realtime database (postrges). Ideally, we want to use a public key to

Re: [asterisk-users] Is there a way to encrypt passwords stored in the realtime database?

I was thinking the same thing I believe Tzafrir just alluded to. If the passwords are encrypted in the DB with a public key then...asterisk needs to have the private key stored somewhere to be able to decrypt the values to authenticate the user. In this way there is nothing preventing whoever

  1   2   3   >