Re: [asterisk-users] pbx testsuite
I'd like to see it. That definitely interests me. Thanks Glen On Jun 15, 2016 9:05 AM, "Marek Červenka" wrote: > hi, > > we have in house developed pbx testsuite based on > >- node.js >- selenium >- protractor >- gulp >- pjsip - pjsua python >- docker > > there are helpers for testing > >- sip >- web >- api > > you can create end-to-end scenarios like > - create 2 users via web > - call from first user to second > - check CDR result via API > > but > we have some problems in "burning" tests with frozen jasmine reporter, > with account management in pjsua python, ... > > my questions are: > is there some similiar testsuite based on node.js technology? (i know > about asterisk-testsuite and xivo-testsuite) > is there interest in publishing our testsuite on github? > > -- > --- > Marek Cervenka > === > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Looking for some guidance with the Asterisk 12 ARI/API
Hi - I figured this was probably the best place to ask this question Is there anyone that has done anything practical with the API and/or Real Time Database config? If so, I would like to pick your brains if I may. Thanks - G -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11 - looking for ideas and a possible solution
Okay - I thank you for your time. Much appreciated! Glen On Sat, Jan 25, 2014 at 10:15 AM, Joshua Colp wrote: > On 14-01-25 11:13 AM, James Wystead wrote: > > On that note, Joshua , while I have your attention. I wanted to ask this: > > > > Is there an available API or some other medium that can interface with > > Asterisk? Something that we can write, perhaps some PHP scripts and put > > and get commands to some API that can manage Asterisk in that way? > > That's not an area I work in so I don't know what others have come up with. > > > I also understand that there is a real time database module. > > You can store stuff in a database and have it used as a source for > configuration and such, yes. > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: www.digium.com & www.asterisk.org > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11 - looking for ideas and a possible solution
On that note, Joshua , while I have your attention. I wanted to ask this: Is there an available API or some other medium that can interface with Asterisk? Something that we can write, perhaps some PHP scripts and put and get commands to some API that can manage Asterisk in that way? I also understand that there is a real time database module. Thanks - G On Sat, Jan 25, 2014 at 9:47 AM, Joshua Colp wrote: > On 14-01-25 10:44 AM, James Wystead wrote: > > Oh - no kidding! > > > > What we want to do is be able to create users, voicemail accounts and > > some of the basic features. Nothing fancy, I'll create the dialplan by > > hand of course. > > The Asterisk REST interface does not currently provide this > functionality. It's not for management, it's for writing telephony > applications (such as a new app_queue or app_voicemail) outside of > Asterisk. > > > So, if I understand, I can use the chan_sip, using the old ways > > (Asterisk 11) and the Asterisk 11 CLI commands - that is what I'm > > familiar with. > > Yes. > > > I can use this and still use the new AMI/API features as well as the > > real time database config? > > Yes. > > Cheers, > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: www.digium.com & www.asterisk.org > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11 - looking for ideas and a possible solution
Oh - no kidding! What we want to do is be able to create users, voicemail accounts and some of the basic features. Nothing fancy, I'll create the dialplan by hand of course. So, if I understand, I can use the chan_sip, using the old ways (Asterisk 11) and the Asterisk 11 CLI commands - that is what I'm familiar with. I can use this and still use the new AMI/API features as well as the real time database config? I just want to be 100% clear - sometimes I need to hear an explanation a couple of times before it sinks in! Thanks On Sat, Jan 25, 2014 at 9:07 AM, Joshua Colp wrote: > > On 14-01-25 09:54 AM, James Wystead wrote: > > First, let me thank all of you for your input - I was looking for some > > sort of an API to interface with Asterisk via REST. Python, Ruby or > > something that we could "webify" using PHP. > > It all depends on what you want to do the with API as well... > > > Some of you suggested Asterisk 12 - I love the idea, but unfortunately, > > we have an Asterisk 11 install. It seems, and I could be wrong, that > > Asterisk 12 is a whole new 'ball game' so to speak. The big thing is > > pjsip - nice idea and I really like it. It looks like a bit of a > > learning curve and I don't want have to re-learn at this point - I will, > > but not right this second! However, I have questions: > > > > 1. Is it necessary to use pjsip for Asterisk 12 as the extension.conf, > > sip.conf has changed gears a little. Can I backtrack to the previous SIP > > stack, which segways into another question: > > The chan_sip module has not been removed and works perfectly fine in 12. > You can even use both, provided you bind each to different ports. > Dialplan is also the same as in the past. > > > 2. If this is possible (the above scenario) is it still possible to use > > these new cool features along with the previous SIP stack so that I can > > implement the API? > > Depends on what you mean by cool new features. The ARI (Asterisk REST > interface) work is agnostic of channel drivers, it has no specialized > logic for any and any present channel driver can be used with it. > > Cheers, > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: www.digium.com & www.asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 11 - looking for ideas and a possible solution
First, let me thank all of you for your input - I was looking for some sort of an API to interface with Asterisk via REST. Python, Ruby or something that we could "webify" using PHP. Some of you suggested Asterisk 12 - I love the idea, but unfortunately, we have an Asterisk 11 install. It seems, and I could be wrong, that Asterisk 12 is a whole new 'ball game' so to speak. The big thing is pjsip - nice idea and I really like it. It looks like a bit of a learning curve and I don't want have to re-learn at this point - I will, but not right this second! However, I have questions: 1. Is it necessary to use pjsip for Asterisk 12 as the extension.conf, sip.conf has changed gears a little. Can I backtrack to the previous SIP stack, which segways into another question: 2. If this is possible (the above scenario) is it still possible to use these new cool features along with the previous SIP stack so that I can implement the API? Thanks to anyone who can give me insight. I don't mean to sound lazy, but v12 seems to be so new that the information is not as plentiful as previous versions. I figured there has to be someone that was 'on the ball' with this and would be happy to share their knowledge! Thanks again! G -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question about Asterisk 12
Okay - maybe I'm just suffering from a moment of horrible ADD - but, I'm a little lost. I see that Asterisk 12 has a nice REST API - very nice - something I can use. However, and this is gonna sound dumb - but all the CLI commands are different now. What did I miss? Can anyone, please, anyone point me to a good, simple to understand tutorial on the new CLI? I am so, so freaking lost! I'm not looking for hand-holding, I just want to understand. Something that will show me how to: - create users - configure SIP trunks - configure basic dialplan I'm lost - anyone point me to a resource that is easy to follow? Once I get the jist, I think I'll be fine. I looked on http://www.voip-info.org - maybe I missed it? The Digium/Asterisk site - I see all sorts of cool things about the REST API, but CLI - maybe I missed it!!?? - again, I could be looking in the wrong place? Overwhelming - sigh. Thank much - any help would be appreciated - next time you are in Manchester NH - I'll make you my fave Tequila Sour drink! G -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk API
Good Day, Ishfaq; This may be a much better idea than the REST API. Correct me if I'm wrong, but the concept is this: You write to the database, and this gives the same result as perhaps modifying the dialplan, sip, voicemail, etc *without* having to physically modify the extensions.conf, sip.conf, voicemail.conf? Am I on the right track? Thanks! On Mon, Jan 13, 2014 at 4:16 AM, Ishfaq Malik wrote: > > On 10 January 2014 17:12, James Wystead wrote: > >> Hello Folks; >> >> I have an Asterisk server >> Asterisk 11.7.0 built by root @xxx on a x86_64 running Linux on >> 2013-12-27 18:47:44 UTC >> >> No FreePBX, no AsteriskNOW, no Elastix. Just Asterisk. >> >> Is there an API out there that anyone knows of that I can pass commands, >> etc to Asterisk? Creating Extensions, adding voicemail users, setting up >> voicemail, etc? >> >> I'm kind of clueless. Is there something available? >> >> Thanks - Glen >> >> >> > You could use asterisk realtime architecture and use your favourite > database to hold peer/voicemail/dialplan configuration. > > https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration > > > > > -- > > Ishfaq Malik > Department: VOIP Support > Company: Packnet Limited > t: +44 (0)845 004 4994 > f: +44 (0)161 660 9825 > e: i...@pack-net.co.uk > w: http://www.pack-net.co.uk > > Registered Address: PACKNET LIMITED, Duplex 2, Ducie House > 37 Ducie Street > Manchester, M1 2JW > COMPANY REG NO. 04920552 > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk API
Hello Folks; I have an Asterisk server Asterisk 11.7.0 built by root @xxx on a x86_64 running Linux on 2013-12-27 18:47:44 UTC No FreePBX, no AsteriskNOW, no Elastix. Just Asterisk. Is there an API out there that anyone knows of that I can pass commands, etc to Asterisk? Creating Extensions, adding voicemail users, setting up voicemail, etc? I'm kind of clueless. Is there something available? Thanks - Glen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Installing on an OpenVZ instance
Hello All; I'm attempting to build the dahdi on an OpenVZ instance: Linux serverx 2.6.18-274.7.1.el5.028stab095.1 #1 SMP Mon Oct 24 20:49:24 MSD 2011 x86_64 x86_64 x86_64 GNU/Linux Now, the kernel says that I have the proper one installed, as you can see from above. However, when I run the make all, this is what I see: You do not appear to have the sources for the 2.6.18-274.7.1.el5.028stab095.1 kernel installed So, my question is this - what is the best way to fix this? Feel free to ask anything you want as I really want to get this working. Thanks - Glen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gateway?
Guys and gals - these are all excellent answers - I am not being clear, I think. Let me see if I can illustrate it. If you cannot see my diagramme, let me know and I will make a word-type chart. So, the Ip device at the top is a SIP phone Asterisk Server Gateway /IP - This gateway is where the SIP Trunk is - so, a provider like Packet 8 or Broadcomm would have this - this connects directly to the public telephone system (somehow) - a Digium card would not work for me as I am not looking to connect to a dial tone. - Does this make sense? So, the Gateway/IP based - what the hell is that called? I am sure there is such an animal as most of us have configured SIP trunks on Asterisk - so, I'm thinking that this thing that connect to the public phone system is what we see as a SIP trunk - right? So, how the hell do I do that? Probably not that simple. Thanks! Glen [image: Inline image 1] On Tue, Apr 30, 2013 at 9:11 AM, Eric Wieling wrote: > On Monday 29 April 2013, James Wystead wrote: > > This is going to sound like a dumb-ass question: > > > > The device that allows you to bridge Asterisk (or any other PBX) into > > the pstn.. What is that called? > > For 1 - 2 ports they are usually called an ATA (Analog Terminal Adapter). > For more than 2 ports they are usually called Media Gateways. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > <>-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Gateway?
This is going to sound like a dumb-ass question: The device that allows you to bridge Asterisk (or any other PBX) into the pstn.. What is that called? So, I guess, not a SIP trunk, but the device that actually IS the SIP trunk. Am I making sense? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question for the group
Yes, I like the look of that. Researching it too - the commercial one looks nice too, but I don't know if there is a budget. G On Fri, Feb 10, 2012 at 11:52, Terry Brummell wrote: > I assume that solution was A2Billing? > > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson > Sent: Friday, February 10, 2012 11:41 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Question for the group > > - Original Message - > > Hello Folks; > > > > I know this is a non-commercial discussion group, but I am looking for > > some open-source software suggestions > > > > > > We are going to be setting up a prepaid PBX service with the following > > features: > > > > > > • Email to Fax and Fax to Email > > • Inward DID local and 800 services > > • Calling card SIP based and ANI authenticated > > > > > > I see there are many types of software that can be addons/installs/etc > > to Asterisk. > > > > So, the question that I ask is which one would be best suited for > > these needs? Of course, it needs to be scalable and work well (most > > opensource software does) > > > > So, any thoughts? > > > > You just posted this to the asterisk-biz list under a different name/email > address. The one response you received was immediately brushed off because > you apparently cannot read: "Thanks for this - but I am looking really for > a software type solution". The product offered *IS A SOFTWARE SOLUTION* > that would run on your hardware. The posted option is more than suitable to > your needs, and offered by folks with a highly deserved great reputation. > > Good luck to you. > > --tim > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New > to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question for the group
Hello Folks; I know this is a non-commercial discussion group, but I am looking for some open-source software suggestions We are going to be setting up a prepaid PBX service with the following features: * > >- Email to Fax and Fax to Email >- Inward DID local and 800 services >- Calling card SIP based and ANI authenticated > > * > I see there are many types of software that can be addons/installs/etc to Asterisk. So, the question that I ask is which one would be best suited for these needs? Of course, it needs to be scalable and work well (most opensource software does) So, any thoughts? Thanks G -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
