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

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

I need to use app_macro, but it seems to be absent from asterisk 16.30.1 Is there a workaround? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at:

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

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

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 busy. The problem is Avaya basically put the call on hold so audio

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?

is strictly prohibited. > Date: Thu, 11 Oct 2018 17:18:24 + (UTC) > From: Ivan Demkovitch > To: "asterisk-users@lists.digium.com" > > Subject: [asterisk-users] Is there any way to pass caller id to cell > phone? > Message-ID: <1490413779.83

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

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

We have following problem. On some of the extentions I call cell phone after 10 seconds or so.Or, like this one below- we call cell and office phone at the same time ;Eric on extension 105 exten => 105,1,Dial(${ERIC_CELL}&${ERIC_OFFICE},30)     same => n,VoiceMail(105@default,u) Where

[asterisk-users] What's the best way of extracting call data which has been written to flat files?

I'm dealing with a blind charity phone information system which writes its logs to two flat csv files (Although the log COULD actually now be written to dynamoDB or sqlite3, too if needed). The first file contains basic call information, one line per call and a unique call ID (distinct from

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,

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

Hello, Is there a way to let someone access to Asterisk CLI and type whatever command (s)he likes but the shell command (the ones started by !) ? Ideally, it could be an argument to rasterisk: rasterisk --no-shell When done, a session could be like this: > pjsip show endpoints ... > core

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

[asterisk-users] double NAT - one way audio

Hi all, I have a setup which is not working right now: Provider - DSL-Router (192.168.2.1) - Bintec-Router (10.17.46.66) - Asterisk (10.17.46.99) My issue: Everything works, but RTP is only going from my Asterisk towards the provider. Asterisk is configured to use SIP-ports 55060 and

[asterisk-users] First call - one way audio

Hello folks, I hope this is simple issue because it seems like something with registration expiration, etc. We use Asterisk (plain setup) with Cisco SPA phones (also nothing changed on settings, just proxy/UN/Password Everything on same LAN So, what we observe is when call coming in - we

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

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

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 asterisk 11: -- Blind transferring OOH323/7272-6385 to '0' (context fromtransfer) priority 1 -- Executing [0@fromtransfer:1]

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

[asterisk-users] Is there a way to escape text passwords in pjsip.conf?

Hello. I have plain text password for endpoint with outbound registration (someone else's server). My aim is to write it in pjsip.conf. md5 means that I know realm. I do not always know it. Is where any way? Dmitriy Serov. --

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,

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

Background: I dabbled with asterisk years ago, and more recently have more-or-less functioning IncrediblePBX systems for experimenting, but I want to understand more so I'm now working with distro packages (Ubuntu) and hand edited configurations files. I have three SIP trunks, each providing me

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

[asterisk-users] SOLVED: Re: Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available

El 02/05/14 11:41, Alex Villací­s Lasso escribió: El 02/05/14 10:49, Alex Villací­s Lasso escribió: El 27/04/14 07:47, Barry Flanagan escribió: On 26 April 2014 00:29, Alex Villací­s Lasso a_villa...@palosanto.com mailto:a_villa...@palosanto.com wrote: I am currently preparing a

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

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

Hello, My health care organization is looking for a way to do appointment reminders. We currently have staff members who spend part of each day manually calling patients to remind them of their upcoming appointments, and we would like to automate this process. Our electronic health record

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

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

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 145: [peer146] host=192.168.0.146 type=friend context=from-trunk [to-146] type=peer host=192.168.0.146

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

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

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 Dial function. The protocol is SIP only,

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?

...@lists.digium.com] On Behalf Of CDR Sent: Friday, 17 February 2012 10:00 p.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] Is there any way to make call fail after # of rings? My customer needs to set a forwarding based on number of rings,i.e., if the phone rings 5 times (user

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

[asterisk-users] Is there any way to terminate async origination initialized by AMY?

Hi I have an application. It connects to Asterisk via AMI, and when user decides it begins asynchronous origination to some device. But very often user decides to break origination and make another call. How can I achieve that? As much as I see, Asterisk doesn't returns any ID of dial process and

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

: 25a6e3604896da0e5482a7565560c...@79.80.154.99 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 Date: Wed, 28 Jul 2010 09:36:51 -0700 From: Jim Dickenson dicken...@cfmc.com Subject: Re: [asterisk-users] Nat issue one way audio on IP dial To: Asterisk Users Mailing

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

hi there, i have posted earlier on the list but got no satisfying answer. the problem is not big. I have asterisk server directly connected with internet (79.80.x.x) and clients are behind router. clients/users are registered with asterisk and are using sipura and xlite softphone. Now problem

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

[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? /voipfc -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

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

[asterisk-users] IAX peers one way voice

Hi all, i've 2 asterisk box with dahdi (server A ver. 1.4.29 and server B ver. 1.4.26) connected with IAX channel using gsm codec. - Calling from A to B the call has no problem: ring , answer a speak without problem. - Calling from B to A : B phone always listen ring also when A phone answer.

[asterisk-users] IP Kall One-Way Audio

I've scoured the web for hints, and find a lot of chatter about one-way audio with IP Kall, but no definitive explanation. I have the default range (5000-31000) of UDP RTP ports forwarded right to my asterisk box, and have no other difficulties with one-way audio on any other peers. Does anyone

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?

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

1. When i connected my analog phone to fxs card, i cannot get dial tone what could be the problem? I am using elastix 1.5.2 based on centos 5.2 Final. 2. On my 2 sip softphones using x-lite linux versions, i get one way audio how do i solve this?. This problem is also present when i use a

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

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

Hello, I'd like to implement some public sip uri's that poeple can call into and get an echo test. Is there a way to force a codec so that users can test various codecs? Something like: echo-t...@example.com (negotiates whatever codec, is there a way to figure out what codec was

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

[asterisk-users] Is there a way to get info who disconnected the call into CDR?

Hei! Here's my problem. I have an Asterisk with SS7 and SIP trunks. Asterisk version is 1.6. I'm setting up a custom CDR fields and I was wondering is there a way to know who initiated a hangup? Asterisk must be aware of that info somehow, cause in queue_log, that info is present

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

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

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

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 that whenever a call arrives on the Asterisk server the application will display the

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

: Thursday, September 10, 2009 12:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Looking for a way to show caller id information onthe desktop Hi there. My problem, I can't figure out how to ask this question. So, hopefully someone out here can point me

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 =

[asterisk-users] TDMoE in any way related to I-TDM

Hello all, recently i stumbled upon the I-TDM standard, e.g. see here http://www.picmg.org/v2internal/news2005.htm SFP.1, also known as I-TDM (Internal TDM), is a companion protocol specification to SFP.0 that is optimized for TDM traffic over high-speed fabrics such as 1 and 10 Gigabit

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

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

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 We want [EMAIL PROTECTED], and we can't have an entry in sip.conf

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