Re: [asterisk-users] Dial() after the h extension has been invoked?

On Friday 12 November 2021 at 23:34:25, Steve Edwards wrote: > On Fri, 12 Nov 2021, Antony Stone wrote: > > I've never used AGI, so what would your suggested solution involve? > > If all you need is to update/insert/delete some rows in a database, ODBC > could be a solution. I already use ODBC

Re: [asterisk-users] Dial() after the h extension has been invoked?

On Fri, 12 Nov 2021, Steve Edwards wrote: I prefer to do database work in an AGI. I find quoting within the database to be obtuse and fragile. s/database/dialplan/g -- Thanks in advance, - Steve Edwards

Re: [asterisk-users] Dial() after the h extension has been invoked?

On Fri, 12 Nov 2021, Antony Stone wrote: I've never used AGI, so what would your suggested solution involve? If all you need is to update/insert/delete some rows in a database, ODBC could be a solution. I prefer to do database work in an AGI. I find quoting within the database to be

Re: [asterisk-users] Dial() after the h extension has been invoked?

On Friday 12 November 2021 at 18:08:11, aster...@phreaknet.org wrote: > On 11/12/2021 12:39 PM, Antony Stone wrote: > > On Friday 12 November 2021 at 17:36:07, Eric Wieling wrote: > >> Create a spool file from the 'h' extension to generate the call. > > > > Yes, I thought of that, but it somehow

Re: [asterisk-users] Dial() after the h extension has been invoked?

On Friday 12 November 2021 at 18:18:03, Eric Wieling wrote: > On 11/12/21 12:39, Antony Stone wrote: > > On Friday 12 November 2021 at 17:36:07, Eric Wieling wrote: > >> Create a spool file from the 'h' extension to generate the call. > > > > Yes, I thought of that, but it somehow feels a bit

Re: [asterisk-users] Dial() after the h extension has been invoked?

On Fri, 12 Nov 2021, Antony Stone wrote: Can anyone suggest how I might be able to do this? I need to perform a Dial() command after an inbound channel has hung up. I do not expect the Dial() to bridge to anything (the context being dialled simply does some database manipulation and then

Re: [asterisk-users] Dial() after the h extension has been invoked?

On Friday 12 November 2021 at 17:36:07, Eric Wieling wrote: > Create a spool file from the 'h' extension to generate the call. Yes, I thought of that, but it somehow feels a bit clunky, and was hoping for a neater solution :) Antony. -- Software development can be quick, high quality, or

Re: [asterisk-users] Dial() after the h extension has been invoked?

Create a spool file from the 'h' extension to generate the call. On 11/12/21 11:56, Antony Stone wrote: Hi. I have a setup which comprises some "front-end" Asterisk servers which have SIP trunks to external providers, and very simple dial plans, and some "back- end" servers which only talk to

Re: [asterisk-users] Dial() after the h extension has been invoked?

On Friday 12 November 2021 at 17:20:39, Frank Vanoni wrote: > On Fri, 2021-11-12 at 16:56 +, Antony Stone wrote: > > I use Dial() commands with custom SIP headers to pass information > > (eg: about the current state of a call) between the front-end and back-end > > machines, and this works

Re: [asterisk-users] Dial() after the h extension has been invoked?

On Fri, 2021-11-12 at 16:56 +, Antony Stone wrote: > I use Dial() commands with custom SIP headers to pass information > (eg: about > the current state of a call) between the front-end and back-end > machines, and > this works very well. > > I need to perform a Dial() > command after an

[asterisk-users] Dial() after the h extension has been invoked?

Hi. I have a setup which comprises some "front-end" Asterisk servers which have SIP trunks to external providers, and very simple dial plans, and some "back- end" servers which only talk to the front-end machines, and have the majority of my dialplan logic on them. I use Dial() commands with

Re: [asterisk-users] Dial(PJSIP/xx) - finding the IP address it connected to

Hi, pls try: exten => s,n,Set(pjsiprip=${CHANNEL(pjsip,remote_addr)}) https://wiki.asterisk.org/wiki/display/AST/Asterisk+19+ManagerAction_Getvar On Thu, 4 Nov 2021 at 15:50, Kingsley Tart wrote: > On Thu, 2021-11-04 at 10:10 -0300, Joshua C. Colp wrote: > > > Do you know whether it is

Re: [asterisk-users] Dial(PJSIP/xx) - finding the IP address it connected to

On Thu, 2021-11-04 at 10:10 -0300, Joshua C. Colp wrote: > > Do you know whether it is possible to get the remote_addr from the > > AMI? > > I don't know off the top of my head. AMI actions and events are > documented on the wiki[1], so you could look there and see. > > [1]

Re: [asterisk-users] Dial(PJSIP/xx) - finding the IP address it connected to

On Thu, Nov 4, 2021 at 10:07 AM Kingsley Tart wrote: > On Thu, 2021-11-04 at 09:45 -0300, Joshua C. Colp wrote: > > The information may not yet be available. Why that would be, I do not > > know. > > Right OK, a bit of a mystery then. > > I have tried to figure out whether this information is

Re: [asterisk-users] Dial(PJSIP/xx) - finding the IP address it connected to

On Thu, 2021-11-04 at 09:45 -0300, Joshua C. Colp wrote: > The information may not yet be available. Why that would be, I do not > know. Right OK, a bit of a mystery then. I have tried to figure out whether this information is available via the AMI but I haven't been able to find anything. Do

Re: [asterisk-users] Dial(PJSIP/xx) - finding the IP address it connected to

On Thu, Nov 4, 2021 at 9:39 AM Kingsley Tart - Barritel Ltd < kingsley.t...@barritel.com> wrote: > On Thu, 2021-11-04 at 08:52 -0300, Joshua C. Colp wrote: > > Thanks, that looks perfect. What is the syntax? I have tried a few things > but none work: > > > ${CHANNEL(pjsip,remote_addr)} > > > Hmm,

Re: [asterisk-users] Dial(PJSIP/xx) - finding the IP address it connected to

On Thu, 2021-11-04 at 08:52 -0300, Joshua C. Colp wrote: > > Thanks, that looks perfect. What is the syntax? I have tried a few > > things but none work: > > > > ${CHANNEL(pjsip,remote_addr)} Hmm, I can't get this to work. This dialplan code: exten => s,n,NoOp(### state=${CHANNEL(state)} ##)

Re: [asterisk-users] Dial(PJSIP/xx) - finding the IP address it connected to

On Thu, Nov 4, 2021 at 8:43 AM Kingsley Tart wrote: > On Wed, 2021-11-03 at 16:25 -0300, Joshua C. Colp wrote: > > On Wed, Nov 3, 2021 at 3:31 PM Kingsley Tart wrote: > > Hi, > > When dialling a remote SIP host with PJSIP, is it possible either > within the dialplan or via the AMI to find out

Re: [asterisk-users] Dial(PJSIP/xx) - finding the IP address it connected to

On Wed, 2021-11-03 at 16:25 -0300, Joshua C. Colp wrote: > On Wed, Nov 3, 2021 at 3:31 PM Kingsley Tart > wrote: > > Hi, > > > > When dialling a remote SIP host with PJSIP, is it possible either > > within the dialplan or via the AMI to find out the IP address of > > the > > remote host? > >

Re: [asterisk-users] Dial(PJSIP/xx) - finding the IP address it connected to

On Wed, Nov 3, 2021 at 3:31 PM Kingsley Tart wrote: > Hi, > > When dialling a remote SIP host with PJSIP, is it possible either > within the dialplan or via the AMI to find out the IP address of the > remote host? > The CHANNEL dialplan function[1] provides functionality for pjsip to get the

[asterisk-users] Dial(PJSIP/xx) - finding the IP address it connected to

Hi, When dialling a remote SIP host with PJSIP, is it possible either within the dialplan or via the AMI to find out the IP address of the remote host? If for example a remote host has multiple A records, I would like to know which one Asterisk has connected to. We have an issue with some

Re: [asterisk-users] Dial(${PJSIP_DIAL_CONTACTS(Alice)} & ${PJSIP_DIAL_CONTACTS(Bob)}) how not to fail if one endpoint has no registered AOR?

Le 10/06/2019 à 10:53, Benoit Panizzon a écrit : What about to put eveything in a variable and the remove the last character if it equal & Yes, I considered this... What if you dial three endpoints and the middle one (or last one) is empty? You would also need to remove the first & and any

Re: [asterisk-users] Dial(${PJSIP_DIAL_CONTACTS(Alice)} & ${PJSIP_DIAL_CONTACTS(Bob)}) how not to fail if one endpoint has no registered AOR?

> What about to put eveything in a variable and the remove the last > character if it equal & Yes, I considered this... What if you dial three endpoints and the middle one (or last one) is empty? You would also need to remove the first & and any double & within that string. Is it faisable with

Re: [asterisk-users] Dial(${PJSIP_DIAL_CONTACTS(Alice)} & ${PJSIP_DIAL_CONTACTS(Bob)}) how not to fail if one endpoint has no registered AOR?

Le 09/06/2019 à 13:19, Benoit Panizzon a écrit : Dear List Hello It's probably been more than a year now I switched from chan_sip to pjsip. pjsip works much cleaner than chan_sip. But! I have come across a Problem I was not able to solve with Asterisk Dialplan Logic. With pjsip an

[asterisk-users] Dial(${PJSIP_DIAL_CONTACTS(Alice)} & ${PJSIP_DIAL_CONTACTS(Bob)}) how not to fail if one endpoint has no registered AOR?

Dear List It's probably been more than a year now I switched from chan_sip to pjsip. pjsip works much cleaner than chan_sip. But! I have come across a Problem I was not able to solve with Asterisk Dialplan Logic. With pjsip an endpoint can have multiple AOR, so you need to expand them with

Re: [asterisk-users] Dial to FastAGI application appears as 1-second CDR - how do I fix?

In article <7d8dc02f-0fce-4d47-72d9-604994c33...@palosanto.com>, Alex Villací­s Lasso wrote: > El 29/05/18 a las 05:24, Tony Mountifield escribió: > > In article <3a005ff6-19a4-215b-4751-bee616ec7...@palosanto.com>, > > Alex Villací­s Lasso wrote: > >> In my application, I am using AMI

Re: [asterisk-users] Dial to FastAGI application appears as 1-second CDR - how do I fix?

El 29/05/18 a las 05:24, Tony Mountifield escribió: In article <3a005ff6-19a4-215b-4751-bee616ec7...@palosanto.com>, Alex Villací­s Lasso wrote: In my application, I am using AMI to run an Originate command between a channel and a dialplan application (NOT a context). In my case, the

Re: [asterisk-users] Dial to FastAGI application appears as 1-second CDR - how do I fix?

In article <3a005ff6-19a4-215b-4751-bee616ec7...@palosanto.com>, Alex Villací­s Lasso wrote: > In my application, I am using AMI to run an Originate command between a > channel and a dialplan application (NOT a > context). In my case, the application I want to invoke is FastAGI. The >

[asterisk-users] Dial to FastAGI application appears as 1-second CDR - how do I fix?

In my application, I am using AMI to run an Originate command between a channel and a dialplan application (NOT a context). In my case, the application I want to invoke is FastAGI. The Originate AMI command works correctly, but Asterisk generates a very short (0-1s) duration for the CDR that

Re: [asterisk-users] Dial an extension to modify dialplan

On Wed, 2017-05-10 at 12:56 +0200, Frank Vanoni wrote: > exten => 2001,1,Dial(SIP/Dial(SIP/deviceA/deviceB/deviceC) > > exten => 2002,1,Dial(SIP/Dial(SIP/deviceA/deviceB) Whoops... sorry for the typo (in the hurry of copy & paste)! exten => 2001,1,Dial(SIP/deviceA/deviceB/deviceC) exten =>

Re: [asterisk-users] Dial an extension to modify dialplan

Dear Digium List First of all, I thank all of you for all the replies and the interesting suggestions. I thank you very much. I can only learn from people like you. :-) I will remember all the different solutions for a future use in other scenarios. On Mon, 2017-05-08 at 16:35 +0200, Frank

Re: [asterisk-users] Dial an extension to modify dialplan

Greetings, I think this is a better solution: I've created a simular solution for our main incoming line. Extentions can add/remove themselfs from the distrubuting extention. I used the DATABASE functions of Asterisk to accomplish this following. my example: first create a few

Re: [asterisk-users] Dial an extension to modify dialplan

I would use a Queue with RingAll strategy. Then, I would Pause/Unpause Agents. A "Paused" agent would not receive calls from the Queue, but can still receive direct calls. You can set an extension to Pause the member and another to Unpause it, using the applications PauseQueueMember and

Re: [asterisk-users] Dial an extension to modify dialplan

> Hello > I have the following scenario: > [mynicecontext] > exten => 2000,1,Dial(SIP/deviceA/deviceB/deviceC) > As expected, by dialing 2000, all three devices will ring. And that's fine. > However, there are situations where I only want "deviceA" and "deviceB" to ring. I would like to have an

Re: [asterisk-users] Dial an extension to modify dialplan

https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerAction_DialplanExtensionRemove Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 8 May 2017 at

Re: [asterisk-users] Dial an extension to modify dialplan

On Monday 08 May 2017 at 15:44:47, Marcelo Terres wrote: > https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+ManagerAction_Dialpl > anExtensionAdd > > Is it enough? Is there a similar call to delete an extension, or to modify an existing one? On the basis that the OP already has extension

Re: [asterisk-users] Dial an extension to modify dialplan

You could use the DIALGROUP function for this and not need to shell out. https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Function_DIALGROUP On Mon, May 8, 2017 at 7:35 AM, Frank Vanoni wrote: > Hello > > I have the following scenario: > > [mynicecontext] >

Re: [asterisk-users] Dial an extension to modify dialplan

On Monday 08 May 2017, Frank Vanoni wrote: > By dialing 4000 or 4001, the dialplan is modified and reloaded > accordingly. > > Is there a better solution? That's an . interesting . way of doing things! We would be thinking in terms of using a GLOBAL variable, or an ASTDB entry, to

Re: [asterisk-users] Dial an extension to modify dialplan

https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+ManagerAction_DialplanExtensionAdd Is it enough? Regards, Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres

[asterisk-users] Dial an extension to modify dialplan

Hello I have the following scenario: [mynicecontext] exten => 2000,1,Dial(SIP/deviceA/deviceB/deviceC) As expected, by dialing 2000, all three devices will ring. And that's fine. However, there are situations where I only want "deviceA" and "deviceB" to ring. I would like to have an extension

Re: [asterisk-users] Dial() using full SIP account details

> Incidentally, I do know I can put a Register statement into sip.conf, and > then be able to use the Dial() application just using the username (and > this works), however I need a solution which can support two or more > accounts at different remote providers having the same username. You can

[asterisk-users] Dial() using full SIP account details

Hi. I'm having problems with the Dial() application when I use full SIP account details in it. I'm looking at the O'Reilly book https://www.amazon.co.uk/dp/1449332420 on page 135, where it says "The Dial() application also allows you to connect to a remote VoIP endpoint not previously defined

Re: [asterisk-users] Dial() from the console?

On Wed, Jan 11, 2017 at 07:31:31AM -0500, Doug Lytle wrote: > >>> On Jan 11, 2017, at 7:20 AM, Thufir Hawat hawat.thu...@gmail.com wrote: > > >>> Can I dial directly from the asterisk console with the Dial() application? > > > console dial number@context Note, however, that it is a different

Re: [asterisk-users] Dial() from the console?

On Wed, 11 Jan 2017, Doug Lytle wrote: On Jan 11, 2017, at 7:20 AM, Thufir Hawat hawat.thu...@gmail.com wrote: Can I dial directly from the asterisk console with the Dial() application? console dial number@context Thanks, that's much more intuitive :) -Thufir --

Re: [asterisk-users] Dial() from the console?

>>> On Jan 11, 2017, at 7:20 AM, Thufir Hawat hawat.thu...@gmail.com wrote: >>> Can I dial directly from the asterisk console with the Dial() application? console dial number@context Doug -- _ -- Bandwidth and Colocation

[asterisk-users] Dial() from the console?

Can I dial directly from the asterisk console with the Dial() application? or, is channel originate preferred: channel originate SIP/thufir extension 18003569377@outbound thanks, Thufir -- _ -- Bandwidth and Colocation

Re: [asterisk-users] Dial and start music on hold after timeout

On Wed, Aug 24, 2016 at 6:02 AM, Israel Gottlieb wrote: > Are you sending progress? > > > בתאריך 24 באוג׳ 2016 13:40,‏ "Saint Michael" כתב: >> >> I have the same exact issue. I cannot push any sounds or even Playtones to >> the caller, unless the channel is

Re: [asterisk-users] Dial and start music on hold after timeout

Are you sending progress? בתאריך 24 באוג׳ 2016 13:40,‏ "Saint Michael" כתב: > ​I have the same exact issue. I cannot push any sounds or even Playtones > to the caller, unless the channel is answered, which is not possible for > billing reasons. > I am also using the Local

[asterisk-users] Dial and start music on hold after timeout

​I have the same exact issue. I cannot push any sounds or even Playtones to the caller, unless the channel is answered, which is not possible for billing reasons. I am also using the Local channel & Dial(PJSIP/...). I think this is a bug in Asterisk 13. The Dial function has not answered yet, so

Re: [asterisk-users] Dial and start music on hold after timeout

Yes, sorry, my idea was too simplistic, as it did not take into account that the caller would already be hearing the ringing and therefore the announcement would need to be somehow mixed with that ringing... On 24 August 2016 at 08:27, Jean Aunis wrote: > Using Progress

Re: [asterisk-users] Dial and start music on hold after timeout

Using Progress didn't solve the problem. If I cannot find another way, I will use your solution of recording the ring tone. Le 23/08/2016 à 20:53, Israel Gottlieb a écrit : Maybe try progress() instead of answer () בתאריך 23 באוג׳ 2016 7:19 PM,‏ "Jean Aunis"

Re: [asterisk-users] Dial and start music on hold after timeout

Maybe try progress() instead of answer () בתאריך 23 באוג׳ 2016 7:19 PM,‏ "Jean Aunis" כתב: > Thank you, I just tried your suggestion. Strangely, the announcement is > played only if I try to dial a SIP peer which is not available (not > registered to be more precise). If

Re: [asterisk-users] Dial and start music on hold after timeout

Damn, I was going to suggest trying a Queue with a single member using the 'r' option to play ringing instead of MOH and using an announcement but the queue will stop ringing your agent while it plays the announcement. It'd go right back to ringing after the announcement however. On Mon, Aug 22,

Re: [asterisk-users] Dial and start music on hold after timeout

Thank you, I just tried your suggestion. Strangely, the announcement is played only if I try to dial a SIP peer which is not available (not registered to be more precise). If the SIP peer is available, I only get the ring tone, and never hear the announcement. Here is the dialplan (I had to

Re: [asterisk-users] Dial and start music on hold after timeout

How about: exten => s,1,Dial(SIP/alice/555@delayed-announce,40) [delayed-announce] exten => 555,1,Wait(20) same => n,Playback(myannouncement,noanswer) same => n,NoOP(Whatever else you want to do goes here) The 'noanswer' option on the Playback means that SIP/alice should continue to ring for

Re: [asterisk-users] Dial and start music on hold after timeout

You could m and make a moh file that has ringing the first 30 sec and then the anouncment בתאריך 22 באוג׳ 2016 7:19 PM,‏ "Jean Aunis" כתב: > Thank you for the idea. The problem with RetryDial, is that it will cancel > the first call, play the announce and then dial the

Re: [asterisk-users] Dial and start music on hold after timeout

Thank you for the idea. The problem with RetryDial, is that it will cancel the first call, play the announce and then dial the SIP peer once again, so the telephone will display a missed call. I would prefer to do everything in a single call. Le 22/08/2016 à 17:57, John Kiniston a écrit :

Re: [asterisk-users] Dial and start music on hold after timeout

You could try using RetryDial() instead of Dial, It supports playing an announcement. On Mon, Aug 22, 2016 at 8:45 AM, Jean Aunis wrote: > Sorry, I forgot to write that the SIP peer must keep ringing while the > announcement is being played. > > Le 22/08/2016 à 17:42,

Re: [asterisk-users] Dial and start music on hold after timeout

Sorry, I forgot to write that the SIP peer must keep ringing while the announcement is being played. Le 22/08/2016 à 17:42, John Kiniston a écrit : This seems like the obvious answer but maybe I'm misunderstanding the question. exten => s,1,Dial(SIP/alice,20) same =>

Re: [asterisk-users] Dial and start music on hold after timeout

This seems like the obvious answer but maybe I'm misunderstanding the question. exten => s,1,Dial(SIP/alice,20) same => n,Playback(myannouncement) same => n,NoOP(Whatever else you want to do goes here) On Mon, Aug 22, 2016 at 8:36 AM, Jean Aunis wrote: > Hello, > >

[asterisk-users] Dial and start music on hold after timeout

Hello, I am searching a way to dial a SIP peer, and if it does not answer within 20 seconds, play an announcement to the caller. This means that the caller would hear a ring tone for 20 seconds, and only then hear the announcement if the callee did not answer. I know it is possible to do

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:

[asterisk-users] Dial()-Function

Hi all! :) I search a function or option for application Dail(). My situations: I have two or more Dial()s with multiple devices (Handgroups). Level1: Dial(SIP/device1,20) Level2: Dial(SIP/device1/device2,20) Level3: Dial(SIP/device1/device2/device3,20) When in level one, no one accept the

Re: [asterisk-users] Dial()-Function

Hi all! :) I search a function or option for application Dail(). My situations: I have two or more Dial()s with multiple devices (Handgroups). Level1: Dial(SIP/device1,20) Level2: Dial(SIP/device1/device2,20) Level3: Dial(SIP/device1/device2/device3,20) When in level one, no one accept

Re: [asterisk-users] Dial()-Function

No, i think unfortunately it is not easier. :/ I have a string from database (Macro/appdata) in the format: function|timeout|function|timeout|function|timeout| Up to seven value pairs. "function" can be "Queue" (Identified by: "qu"-string), "Voicemail" (Identified by: "vm"-string),

Re: [asterisk-users] Dial()-Function

hi jg, jg schrieb am Don, 17. Mär 14:05: > Wouldn't it be easier to use a local channel and do something like > is done in the "Delay Dialing Devices Example"? > > https://wiki.asterisk.org/wiki/display/AST/Delay+Dialing+Devices+Example No, i think unfortunately

Re: [asterisk-users] Dial your phone and contact phone from within outlook?

From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Thursday, March 03, 2016 10:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dial your phone and contact phone from within

Re: [asterisk-users] Dial your phone and contact phone from within outlook?

Is TAPI still available on Windows 10, for instance ? 2016-03-02 23:22 GMT+01:00 Neeraj Chand : > Hi Travis, > > Have a look at this: > > http://www.ipcom.at/en/telephony/siptapi/ > > I have used this in the past to do something similar, unless you have an > Exchange

Re: [asterisk-users] Dial your phone and contact phone from within outlook?

ailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dial your phone and contact phone from within outlook? Hi Travis, Have a look at this: http://www.ipcom.at/en/telephony/siptapi/ I have used this in the past to do something similar, unless you have an Ex

Re: [asterisk-users] Dial your phone and contact phone from within outlook?

On Wednesday 02 Mar 2016, Ryan, Travis wrote: > I am wondering what the best solution is for initiating a call from Outlook > Contacts. I imagine something that would start the call from the outlook > card (or similar) and then dial the user's extension and the contact's > phone number and place

Re: [asterisk-users] Dial your phone and contact phone from within outlook?

Hi Travis, Have a look at this: http://www.ipcom.at/en/telephony/siptapi/ I have used this in the past to do something similar, unless you have an Exchange Enterprise setup in which case I would suggest exploring unified messaging Thanks, Neeraj On Thu, Mar 3, 2016 at 8:22 AM, Ryan, Travis

[asterisk-users] Dial your phone and contact phone from within outlook?

I am wondering what the best solution is for initiating a call from Outlook Contacts. I imagine something that would start the call from the outlook card (or similar) and then dial the user's extension and the contact's phone number and place them in a bridge. Anyone use something like this?

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 L options and attended tranfer

Hello guys, there is something i'm not sure and would like to ask please : let's say i have B calling and talking to A.( A -- B ) Then B would like to attended transfer A to C. In the behind extensions.conf when B is calling C (first part of the attended transfer) the Dial

Re: [asterisk-users] dial out with channel variable; sub-string usage

BABY appears to be a global variable in your example. In your CLI output testcarrier is a peer, It's not a variable at all. The context field for your peer testcarrier is where incoming calls from testcarrirer will be routed to. Here is some example dialplan showing how you can use one context

Re: [asterisk-users] dial out with channel variable; sub-string usage

On 15-04-09 12:06 PM, Chad Wallace wrote: but don't know where to put those lines. I have BABY defined as channel variable: BABY = SIP/babytel_out but that seems circular, somehow. You put them in the context for your clients... From what you show below, I'd say they go in the local_200

Re: [asterisk-users] dial out with channel variable; sub-string usage

On Wed, 08 Apr 2015 16:10:30 -0700 thufir hawat.thu...@gmail.com wrote: I want to do something like: exten = _NXXXNxx,1,Dial(${BABY}/${EXTEN}) exten = _Nxx,1,Dial(${BABY}/${EXTEN}) exten = _1NXXNxx,1,Dial(${BABY}/${EXTEN}) exten = _011.,1,Dial(Dial({TOLL}/${EXTEN}) exten =

[asterisk-users] dial out with channel variable; sub-string usage

I want to do something like: exten = _NXXXNxx,1,Dial(${BABY}/${EXTEN}) exten = _Nxx,1,Dial(${BABY}/${EXTEN}) exten = _1NXXNxx,1,Dial(${BABY}/${EXTEN}) exten = _011.,1,Dial(Dial({TOLL}/${EXTEN}) exten = _9NXXXNxx,1,Dial(${BABY}/${EXTEN}) exten = _9Nxx,1,Dial(${BABY}/${EXTEN})

[asterisk-users] Dial to PJSIP Channel with Typo PJSIP// Causes Asterisk Shutdown

I found an issue with how PJSIP handles a typo in the Dial application. If the Channel is mistakenly typed with two slashes (i.e Dial(PJSIP//...), the Dial applications fails (obviously), but it also kills the server. I put some code in my pbx_config to check for that string and not let the

Re: [asterisk-users] Dial to PJSIP Channel with Typo PJSIP// Causes Asterisk Shutdown

On Thu, Mar 26, 2015 at 9:28 AM, Trey Hilyard kct...@gmail.com wrote: I found an issue with how PJSIP handles a typo in the Dial application. If the Channel is mistakenly typed with two slashes (i.e Dial(PJSIP//...), the Dial applications fails (obviously), but it also kills the server. I

[asterisk-users] Dial Plan Issue

I am trying to transition an application over from a FreePbx box to a Standard build Asterisk 11.6 box. I have a job that creates a call file and plays a sound file. If it detects a voicemail, then it plays it, waits 1 second and replays it. The FreePbx box works fine but the Standard Asterisk

Re: [asterisk-users] Dial Plan Issue

-users-boun...@lists.digium.com] On Behalf Of Haley,Scott A Sent: Tuesday, February 10, 2015 3:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dial Plan Issue I am trying to transition an application over from a FreePbx box to a Standard build Asterisk 11.6

[asterisk-users] Dial international number over dahdi trunk

Hi all, I am trying to perform the following outgoing call: exten = _49.,1,Log(NOTICE,Dialing German number: ${EXTEN}) same = n,Set(route=DAHDI/g1/00${EXTEN}) same = n,Dial(${route}) exten = _0049.,1,Goto(${EXTEN:2},1) exten = _01149.,1,Goto(${EXTEN:3},1) exten =

Re: [asterisk-users] Dial application b subroutine arguments not passing?

On Fri, Aug 2, 2013 at 3:05 PM, Mitch Claborn mitch...@claborn.net wrote: On 08/02/2013 01:28 PM, Matthew Jordan wrote: On Fri, Aug 2, 2013 at 12:57 PM, Mitch Claborn mitch...@claborn.net mailto:mitch...@claborn.net wrote: Asterisk 11.1.0 I'm trying to use the b subroutine of the

[asterisk-users] Dial application b subroutine arguments not passing?

Asterisk 11.1.0 I'm trying to use the b subroutine of the Dial application so that I can do some stuff with our internal applications that need to have access to the called channel information. I can see that the subroutine is being executed, but the arguments I pass don't see to make it to

Re: [asterisk-users] Dial application b subroutine arguments not passing?

On Fri, Aug 2, 2013 at 12:57 PM, Mitch Claborn mitch...@claborn.net wrote: Asterisk 11.1.0 I'm trying to use the b subroutine of the Dial application so that I can do some stuff with our internal applications that need to have access to the called channel information. I can see that the

Re: [asterisk-users] Dial application b subroutine arguments not passing?

On 08/02/2013 01:28 PM, Matthew Jordan wrote: On Fri, Aug 2, 2013 at 12:57 PM, Mitch Claborn mitch...@claborn.net mailto:mitch...@claborn.net wrote: Asterisk 11.1.0 I'm trying to use the b subroutine of the Dial application so that I can do some stuff with our internal

[asterisk-users] Dial plan flow control

Arch = x86_64 OS = CentOS-6.4 (freepbx) Asterisk = 11.4 FreePBX = 2.11.0.4 I am trying to understand flow control in Asterisk dial plans and not having very much luck. I have read the Asterisk book from O'Rielly, or at least those parts I believe might apply, but that has not helped me much on

[asterisk-users] Dial problem with Asterisk 1.8.4.4

One of my sites asked for a way to identify if the person they are calling on another extension is already on another call. To that end, I wrote a bit of code in the dialplan for my extensions that checks to see if the extension they are dialing has a device status that is anything other than

[asterisk-users] Dial-App / Feature Disconnect

Hi, I configured in features.conf, that the Dial-App may be cancelled by pressing the pound key. That works fine. The caller can cancel the bridged call. BUT can I configure it that way, that the dialing itself can NOT be cancelled? My dial should only be cancelled by the timeout or by the

Re: [asterisk-users] dial and bridge

...@lists.digium.com] *On Behalf Of *Lenz Emilitri *Sent:* Tuesday, May 14, 2013 11:16 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] dial and bridge ** ** Hi all, I need some advice - I have been working on originating multiple calls using

Re: [asterisk-users] dial and bridge

Hi Mitul, I agree that the dialplan way is easier, but it's a client requirement to avoid using it. I was wondering if there was a way to send a call directly to a parking slot right from the originate, because that is cheaper than running conferences, and then joining the second call right to the

Re: [asterisk-users] dial and bridge

Hi Warren, the problem is that all I have is two channels, so the specs might be join SIP/123 and SIP/345 not join SIP/123 to 456@from-internal. They might be Local channels, but this should be able handle the general case. The reason why I have channels and not ext@ctxt is that I read them live

Re: [asterisk-users] dial and bridge

The dial n bridge might work, but there ain't indefinite wait in that scenario. Direct calls to parking you might try Local(70X@from-internal) but I m not sure if this method works reliably. The method I mentioned is used by vicidial and it works flawlessly, yes it comes with some computing load,

Re: [asterisk-users] dial and bridge

I never actually used parking, but should it work if I call the Park application as the second leg of the Originate (w/o going through the dialplan)? I dont seem to be able to make it work. l. 2013/5/15 Mitul Limbani mi...@enterux.in The dial n bridge might work, but there ain't indefinite

Re: [asterisk-users] dial and bridge

I think you could use twice the Park action to park the channels - https://wiki.asterisk.org/wiki/display/AST/ManagerAction_Park In the end you will have to bridge the parked channels. HTH, Ioan On Wed, May 15, 2013 at 1:03 PM, Lenz Emilitri lenz.lo...@gmail.com wrote: I never actually used

Re: [asterisk-users] dial and bridge

BTW - what was exactly the problem when trying to bridge the two channels that you have sent to the wait application? On Wed, May 15, 2013 at 4:29 PM, Ioan Indreias indre...@gmail.com wrote: I think you could use twice the Park action to park the channels -

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