Re: [asterisk-users] Redis in place of astdb
I’ll second that - for CDR you want the fastest sequential writing with possible batching of CDR records Best regards, Alex www.thirdlane.com > On Jul 9, 2020, at 1:37 AM, Antony Stone > wrote: > > On Thursday 09 July 2020 at 00:50:28, Jon Bonilla (Manwe) wrote: > >> DO you know odbc redis drivers? It would be nice to store cdrs ans other >> stuff in redis without patching asterisk > > A quick Google search turns up > https://www.cdata.com/kb/tech/redis-odbc-python-linux.rst > which I have no experience of and cannot comment on, but looks like what you > need. > > PS: I question the wisdom of storing CDRs in Redis - I think an RDBMS is the > correct tool for *that* job. I agree that Redis may be useful in other > areas, > though. > > > Antony. > > -- > All matter in the Universe can be placed into one of two categories: > > 1. Things which need to be fixed. > 2. Things which need to be fixed once you've had a few minutes to play with > them. > > Please reply to the list; > please *don't* CC me. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WebRTC as Softphone substitute ?
No, this will not be a problem. TURN relays media based on RTP headers which are unencrypted. SRTP encrypts only the payload of RTP packets. DTLS-SRTP is one of the things mandatory to implement and support things in WebRTC, and it will not speak RTP/AVP. Asterisk may have to unencrypt payload for media paths that require that, which TURN server is not able to do. Best regards, Alex Alex Epshteyn a...@thirdlane.com +1 (415) 261 6601 www.thirdlane.com > On Oct 2, 2018, at 7:39 PM, David P wrote: > > Thanks for sharing this, Alex. It sounds like TURN, as a media repeater, > wouldn't work if the media must be secured (via SRTP). Is that right? > > On Wed, 3 Oct 2018, 3:17 pm alex epshteyn, <mailto:a...@thirdlane.com>> wrote: > WebRTC requires a few specific things to be in place. We have blog posts that > talk about WebRTC based Thirdlane Connect, but most of the information > applies to WebRTC applications in general. > > https://www.thirdlane.com/blog/get-up-and-running-with-thirdlane-connect > <https://www.thirdlane.com/blog/get-up-and-running-with-thirdlane-connect> > > https://www.thirdlane.com/blog/nat-stun-turn-and-ice > <https://www.thirdlane.com/blog/nat-stun-turn-and-ice> > > Best regards, > Alex > > > Alex Epshteyn > a...@thirdlane.com <mailto:a...@thirdlane.com> > +1 (415) 261 6601 > www.thirdlane.com <http://www.thirdlane.com/> > > > >> On Oct 2, 2018, at 6:08 PM, Nasir Iqbal > <mailto:na...@ictinnovations.com>> wrote: >> >> @Olivior >> I agree that seting up WebRTC is hard, however when done it is smooth to >> use. For replication you can build RPMs with working configurations. >> >> Regarding stability, it is not being used widly, so can't say it is mature. >> However we have no complain so far regarding audio or connectivity. sometime >> we provide support for "allow media / mic" type issues, but you know it is >> security feature and not a bug. >> >> Regards >> >> On Tue, Oct 2, 2018, 13:03 Olivier > <mailto:oza.4...@gmail.com>> wrote: >> @Nasir: >> Thanks for replying here. >> >> Did you met in your deployments, the kind of stability issues Carlos >> reported earlier ? >> >> Le sam. 29 sept. 2018 à 13:32, Nasir Iqbal > <mailto:na...@ictinnovations.com>> a écrit : >> Hi Olivior, >> >> We have recently worked on a WebRTC based agent panel. As based on my >> experience I think that WebRTC based phones are far better and cheaper then >> those soft / sip phone. the big plus is that they are easy to customize and >> developer can use the power of browser and web to build / offer features >> which are not possible with regular phones. >> >> Regarding your concern about BLF or call history, for me as a being >> developer it is just a matter of customization. >> >> Regards >> >> Nasir Iqbal >> >> ICTBroadcast - an Auto Dialer software for ITSP >> <https://www.ictbroadcast.com/how-become-internet-telephony-service-provider-itsp-using-ictbroadcast-sp-edition> >> SMS, Fax and Voice broadcasting & Inbound / Outbound Campaigns >> http://www.ictbroadcast.com/ <http://www.ictbroadcast.com/> >> >> >> On Thu, Sep 27, 2018 at 1:06 AM Carlos Chavez > <mailto:cur...@telecomab.mx>> wrote: >> On 9/26/18 10:20 AM, Matthew Fredrickson wrote: >> >> > On Wed, Sep 26, 2018 at 9:40 AM Carlos Chavez > > <mailto:cur...@telecomab.mx>> wrote: >> >> On 9/26/2018 4:46 AM, Olivier wrote: >> >> >> >>> Hello, >> >>> >> >>> This morning, I asked myself if WebRTC could be a viable alternative >> >>> to softphone deployment. >> >>> >> >>> For me, main issue with Softphones is the amount of work needed for >> >>> installation and configuration. >> >>> Also, Softphones must be carefully choosen if Deskphone-like quality >> >>> is expected. >> >>> >> >>> Now that WebRTC becomes ubiquitous, it might make sense to trade >> >>> Softphone features (call history, BLF, ...) for WebRTC deployment >> >>> simplicity. >> >>> >> >>> What do you think of this ? >> >>> What kind of experience did you met with such WebRTC deployments ? >> >>> What about classic telephony features (CallTransfer) ? >> >>> Have you tried Cyber Maga Phone 2K ? >> >>> >> >> If you can get it to work WebRTC is a goo
Re: [asterisk-users] WebRTC as Softphone substitute ?
WebRTC requires a few specific things to be in place. We have blog posts that talk about WebRTC based Thirdlane Connect, but most of the information applies to WebRTC applications in general. https://www.thirdlane.com/blog/get-up-and-running-with-thirdlane-connect https://www.thirdlane.com/blog/nat-stun-turn-and-ice Best regards, Alex Alex Epshteyn a...@thirdlane.com +1 (415) 261 6601 www.thirdlane.com > On Oct 2, 2018, at 6:08 PM, Nasir Iqbal wrote: > > @Olivior > I agree that seting up WebRTC is hard, however when done it is smooth to use. > For replication you can build RPMs with working configurations. > > Regarding stability, it is not being used widly, so can't say it is mature. > However we have no complain so far regarding audio or connectivity. sometime > we provide support for "allow media / mic" type issues, but you know it is > security feature and not a bug. > > Regards > > On Tue, Oct 2, 2018, 13:03 Olivier <mailto:oza.4...@gmail.com>> wrote: > @Nasir: > Thanks for replying here. > > Did you met in your deployments, the kind of stability issues Carlos reported > earlier ? > > Le sam. 29 sept. 2018 à 13:32, Nasir Iqbal <mailto:na...@ictinnovations.com>> a écrit : > Hi Olivior, > > We have recently worked on a WebRTC based agent panel. As based on my > experience I think that WebRTC based phones are far better and cheaper then > those soft / sip phone. the big plus is that they are easy to customize and > developer can use the power of browser and web to build / offer features > which are not possible with regular phones. > > Regarding your concern about BLF or call history, for me as a being developer > it is just a matter of customization. > > Regards > > Nasir Iqbal > > ICTBroadcast - an Auto Dialer software for ITSP > <https://www.ictbroadcast.com/how-become-internet-telephony-service-provider-itsp-using-ictbroadcast-sp-edition> > SMS, Fax and Voice broadcasting & Inbound / Outbound Campaigns > http://www.ictbroadcast.com/ <http://www.ictbroadcast.com/> > > > On Thu, Sep 27, 2018 at 1:06 AM Carlos Chavez <mailto:cur...@telecomab.mx>> wrote: > On 9/26/18 10:20 AM, Matthew Fredrickson wrote: > > > On Wed, Sep 26, 2018 at 9:40 AM Carlos Chavez > <mailto:cur...@telecomab.mx>> wrote: > >> On 9/26/2018 4:46 AM, Olivier wrote: > >> > >>> Hello, > >>> > >>> This morning, I asked myself if WebRTC could be a viable alternative > >>> to softphone deployment. > >>> > >>> For me, main issue with Softphones is the amount of work needed for > >>> installation and configuration. > >>> Also, Softphones must be carefully choosen if Deskphone-like quality > >>> is expected. > >>> > >>> Now that WebRTC becomes ubiquitous, it might make sense to trade > >>> Softphone features (call history, BLF, ...) for WebRTC deployment > >>> simplicity. > >>> > >>> What do you think of this ? > >>> What kind of experience did you met with such WebRTC deployments ? > >>> What about classic telephony features (CallTransfer) ? > >>> Have you tried Cyber Maga Phone 2K ? > >>> > >> If you can get it to work WebRTC is a good option. The problem is > >> that any changes in your network may disrupt it and even trying to > >> replicate your installation is difficult. I have it working fine on my > >> website so customers can call us directly from our web page but I never > >> could get Cyber Mega Phone 2K to work on the same server. We used JSSIP > >> to create the webrtc phone on our website. > > We just updated the documentation for how to get CMP2K working on the > > wiki [1]. We'd love some feedback if you still have issues getting it > > setup so that we can improve the docs. > > > > [1] > > https://wiki.asterisk.org/wiki/display/AST/Installing+and+Configuring+CyberMegaPhone > > > > <https://wiki.asterisk.org/wiki/display/AST/Installing+and+Configuring+CyberMegaPhone> > > > > Best wishes, > > Matthew Fredrickson > > > I followed the procedure indicated in the link but I cannot get > remote video. I can only see my own feed. We do have audio for a > little while. For some reason the users get disconnected after a few > minutes even though you can still see your video feed on screen. This > was done with Asterisk 15.6.0 > > -- > Telecomunicaciones Abiertas de México S.A. de C.V. > Carlos Chávez > +52 (55)8116-9161 > > > -- > ___
Re: [asterisk-users] softphone instead of desktop phones
Thomas was asking how to save money and I was just offering an option. I am sorry if my post was inappropriate. That said, Thirdlane Connect itself is free, and we do offer a free version for companies with up to 10 users. -- Alex Epshteyn email: a...@thirdlane.com web: www.thirdlane.com phone +1 415.261.6601 - Original Message - > From: "Barry Flanagan" <barryf-li...@flanagan.ie> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Sent: Sunday, April 30, 2017 11:20:25 AM > Subject: Re: [asterisk-users] softphone instead of desktop phones > > > > > > > On 30 April 2017 at 16:54, Tech Support < aster...@voipbusiness.us > > wrote: > > > > I thought this was a non-commercial list. > > > > > Yeah, I wouldn't mind so much if it had actually answered the > original poster's query. "Switch to our proprietary solution and we > can offer you this proprietary solution" isn't a contribution, it's > an ad. > > > -Barry > > > > -----Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto: asterisk-users-boun...@lists.digium.com ] On Behalf Of Alex > Epshteyn > Sent: Saturday, April 29, 2017 08:59 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] softphone instead of desktop phones > > Thirdlane Connect can be used as a softphone. It works in modern > browsers > (no installation is required), on Mac, Windows and Linux desktops, > and on > mobile phones. > > Besides basic softphone functionality, it provides instant messaging, > group > chat (channels), voice and video conferencing, and screen sharing. It > integrates with a variety of applications and CRMs such as > Salesforce, Zoho, > Zendesk, Redmine, etc. > > Try it out! > > > -- > > Alex Epshteyn > web: www.thirdlane.com > > > - Original Message - > > From: "Amit Patkar" < a...@avhan.com > > > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > < asterisk-users@lists.digium.com > > > Sent: Saturday, April 29, 2017 9:16:05 AM > > Subject: Re: [asterisk-users] softphone instead of desktop phones > > > > > > Linphone is available for all major OS platforms. > > Then there is PortGo as well > > Regards, > > Amit Patkar > > > > > > On April 29, 2017 9:05:22 PM GMT+05:30, Thomas < > > thomasit...@gmail.com > > > wrote: > > > > Hello, > > Iam lookong for an Softphone for iPhor oder Android smartphone > > using > > togehter with an headset. > > I tried Zoiper and CSipSimple but quality was bad compared to an > > desktop SIP phone. > > > > Is there an better softphone? > > > > Or are there softphone solutions for PC desktop MAC or Android with > > an > > headset? > > I want to save cost for desktop phones. > > > > thanks Thomas > > > > > > -- > > _ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com > > -- > > > > Check out the new Asterisk community forum at: > > https://community.asterisk.org/ > > > > New to Asterisk? Start here: > > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > > > 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 -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _ &g
Re: [asterisk-users] softphone instead of desktop phones
Thirdlane Connect can be used as a softphone. It works in modern browsers (no installation is required), on Mac, Windows and Linux desktops, and on mobile phones. Besides basic softphone functionality, it provides instant messaging, group chat (channels), voice and video conferencing, and screen sharing. It integrates with a variety of applications and CRMs such as Salesforce, Zoho, Zendesk, Redmine, etc. Try it out! -- Alex Epshteyn web: www.thirdlane.com - Original Message - > From: "Amit Patkar" <a...@avhan.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Sent: Saturday, April 29, 2017 9:16:05 AM > Subject: Re: [asterisk-users] softphone instead of desktop phones > > > Linphone is available for all major OS platforms. > Then there is PortGo as well > Regards, > Amit Patkar > > > On April 29, 2017 9:05:22 PM GMT+05:30, Thomas > <thomasit...@gmail.com> wrote: > > Hello, > Iam lookong for an Softphone for iPhor oder Android smartphone using > togehter > with an headset. > I tried Zoiper and CSipSimple but quality was bad compared to an > desktop SIP > phone. > > Is there an better softphone? > > Or are there softphone solutions for PC desktop MAC or Android with > an > headset? > I want to save cost for desktop phones. > > thanks Thomas > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PBX selection
The solution you choose should be based on many factors which should include your business requirements, team's experience, your budget, growth expectations and more. You can choose Asterisk or Freeswitch as a platform and start building on that - but it is not simple and being new to VoIP you are likely to make mistakes. The "do-it-yourself" approach will some money initially, but will be the most expensive option long term - as you will be denying the economy of scale. Bringing a "smart programmer" won't help much as you will also create a "lock-in". In fact, this could be worse than a dependency created when you use a commercial or a known open source solution as while you would still be able to get help from the community for the "base" part of your pbx, your custom part will be much harder to deal with. Our company started building Asterisk based PBX in 2002 and Multi Tenant PBX in 2005 - we do this as our core business and are still finding areas for improvement :). As your experience with VoIP is minimal I would side with your CTO - you should find a solution high enough in the stack to avoid the complexity of building it all yourself. Good luck, Alex -- Alex Epshteyn email: a...@thirdlane.com web: www.thirdlane.com phone +1 415.261.6601 - Original Message - > From: "J Montoya or A J Stiles" <asterisk_l...@earthshod.co.uk> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Sent: Tuesday, April 18, 2017 1:40:47 AM > Subject: Re: [asterisk-users] PBX selection > > On Monday 17 Apr 2017, Speed Boy wrote: > > Hi all, I'm new to VoIP, now we have a project that needs a > > PBX with client APPs. > > In our team we have argument for choosing PBX. By so far, we > > have following candidates: > > > > A: Open source > > > > 1) Asterisk PBX (http://www.asterisk.org) (with longest > > history that almost every one knows it, now the last version using > > the > > PJSIP stack) > > 2) FreeSwitch (http://www.freeswitch.org) (A lot people > > recommended it to us) > > > > > > B: Commercial > > > > 1) Vodia PBX (http://www.vodia.com). It comes from SNOM, now > > acquired by a HongKong company now > > 2) PortSIP PBX (http://www.portsip.com/portsip-pbx). It > > also includes VoIP SDK, WebRTC and offer rebranding app for free. > > > > My boss prefers the Open Source PBX since they are free, > > but our CTO prefers the commercial editions, according to > > whom the business PBX has better support, and the > > performance is good, and easy to use - considering our team > > all are new to VoIP/PBX. > > Proponents of proprietary solutions always like to say "If an Open > Source > solution breaks, who can you call?" The answer is, "Any > sufficiently-competent > programmer -- it may be broken, but we have all the pieces". Whereas > if you > spend money on proprietary software and it breaks, then there is only > *one* > place you can call -- and you'd better hope they are interested to > fix your > problem. > > On the other hand, if you could get full Source Code and Modification > Rights > (basically, "everything we could do with a GPL program except > distribute > copies"), a proprietary solution might not be so bad after all. But > since > the goal of most proprietary software vendors is to extract money > from you and > maintaining you in a state of perpetual helplessness is highly > desirable in > the course of this, do not expect to get such a deal in real life. > > > We have did some searching of Asterisk, here are my questions: > > > > 1. Does the last Asterisk using PJSIP stack ? > > Yes. > > > 2. Does there has the comparison of PJSIP and reSIProcate, > > sofia(using by > > FreeSwicth) ? > > Not sure about this. We're still using the original chan_sip driver. > > > 3. Is it easy to compile and setup Asterisk? > > It's about as easy as compiling anything from Source Code. Harder > than LAME > MP3 encoder, but easier than the Linux kernel. If you altered > `monop` from > the BSDgames package to make the streets match your local edition of > the game, > you will have no problem whatsoever with building Asterisk. > > If you understand the process of what you are doing -- basically, > setting up > an automated process that will examine your server hardware and > software > configuration (configure), choosing which parts of Asterisk you > want to > include (make menuselect), compiling the selected human-readable > Sourc
Re: [asterisk-users] Showing sip subscriptions in Manager
You can use Command command, and sip show subscriptions as a parameter -- Alex Epshteyn email: a...@thirdlane.com web: www.thirdlane.com phone +1 415.261.6601 - Original Message - From: Leandro Dardini ldard...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, January 15, 2015 3:00:30 PM Subject: [asterisk-users] Showing sip subscriptions in Manager Hello, almost any useful CLI command has an analogue on Asterisk Manager Interface, but I cannot find a way to get the list of subscriptions using AMI. Which is the command, if any? The CLI command is sip show subscriptions Leandro -- _ -- 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
Re: [asterisk-users] Druid
Hi Dean, I would be happy to get your feedback and feature suggestions regarding the user portal. I don't dislike what we have now, but there is no doubt that it can be improved. You can see a demo on our website - http://www.thirdlane.com/products/pbxmanager-be/demo - you have to login as end-user for one of the existing extensions or create your own. Feel free to contact me off list, or call me at 1 415 721 7717 (I am in California). Best regards, Alex Alex Epshteyn Third Lane Technologies, LLC http://www.thirdlane.com _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Thursday, November 01, 2007 2:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Druid Nice - I like where it's heading, finally someone came up with a half decent looking UI for a User Portal (though not perfect) I've consulted to 3 different companies about User Portals and not a single one has implemented anything near what it should look like. I just don't get why Asterisk product designers don't talk hire proper UI consultants to build the right solution. That's what you get for being cheap, all the pieces available and still no clue. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Alan Lord Sent: Thursday, 1 November 2007 5:07 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Druid Alex Epshteyn wrote: I came across this one the other day: http://www.voiceone.it/index.php?synSiteLang=2 It's GPL (Important to me!) and it looks very slick. It also has a user interface as well as an admin interface. Although quite young from what I have seen of it - it looks like the dogs b's. Al -- The way out is open! http://www.theopensourcerer.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Druid
Dean, If you are looking for a non-restricting and extensible Asterisk GUI please look at Thirdlane http://www.thirdlane.com http://www.thirdlane.com/ . If you are comfortable installing OS, Webmin and Asterisk, I would suggest installing PBX Manager GUI (packaged as a Webmin module), otherwise Thirdlane Advantage (CentOS based ISO) may be a good option. Best regards, Alex _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Wednesday, October 31, 2007 3:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Druid Is anyone out there using Druid? After the switchbox announcement today I've been looking into some other gui's and as I'll probably do a trial install this weekend of the free switchvox iso but I thought I'd ask is there any other guis I should be burning trial ISO's of as well? Regards, Dean Collins Cognation Pty Ltd mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What web GUI are people happy with?
I hope no one will frown on my post here as our product is commercial. I just wanted to let you know that you can use Thirdlane PBX Manager to create complex dialplans. The way it works is that you can create scripts - equivalent of Asterisk macros with some extras and no limitation on what asterisk dialplan code you put there, which plug into the GUI, so other people (who may not be able to write or understand your scripts) may still use them through the gui by attaching them to user or special extensions. Thanks, Alex -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Wednesday, October 17, 2007 6:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What web GUI are people happy with? On Wed, 2007-10-17 at 14:53 -0700, shadowym wrote: Ok Thanks, I guess I'll have to give it a shot. I just assumed it would be more work than 30minutes (after the initial learning curve) for a moderately complex dialplan.. The other issue that arrives is that a complex dialplan can't be created/managed via a gui... So for an easy dialplan - both a gui and conf files are going to work out fairly quickly. With something more complicated, you end up having to edit the files. So it's worth knowing how to do that. PaulH ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi tenant
Hi Mujtaba, We have a multi-tenant version of our Asterisk based management and end-user software called Thirdlane PBX Manager. You can see a demo of a single-tenant version on our web site http://www.thirdlane.com/pbxmanager.htm the multi-tenant adds tenant and DID management, and allows to partition Asterisk to manage independent tenants with their own administrators, extensions, routes, queues, etc Please contact me off list for more information. Best regards, Alex Alex Epshteyn Third Lane Technologies, LLC http://www.thirdlane.com _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mujtaba Mahmood Sent: Thursday, October 04, 2007 2:14 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Multi tenant Hi all, i just wanted to know if any one has done any multi-tenant version of the asterisk. thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hints / State change on outgoing calls
Hi, I am trying to set BLF on SNOM phones. With call-limit=4 in sip.conf and hints in the extensions.conf a call to the extension correctly shows state as InUse (show hints) and BLF works. When call is originated from the extension the associated state remains Idle, so no notification and no BLF. Is there something else that has to be set for state to change (and watchers notified) on the outgoing calls? Also, Asterisk restart results in all the watchers being lost. Is there a way to force the phone to subscribe to notifications after restart (short of rebooting it) and is it phone specific? This is Asterisk 1.4.11. Thanks, Alex Alex Epshteyn Third Lane Technologies, LLC http://www.thirdlane.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RAW asterisk!
Bill, Please take a look at Thirdlane PBX Manager. It gives you both management and end-user GUI, and stores data in text configuration files. You can also extend it using what we call Scripts (basically GUI integrated self-documented Asterisk Macros), this way you can still use your Asterisk dialplan coding skills when required and hardly ever need the RAW mode. Best regards, Alex Alex Epshteyn Third Lane Technologies, LLC http://www.thirdlane.com -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Bill Andersen Sent: Thursday, August 16, 2007 11:38 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] RAW asterisk! I'm a network admin that maintains 3 commercial Asterisk servers for my employer. I am wanting to move away from the pre-packaged commercial PBXs to a more pure asterisk setup. The systems I have utilize a nice web GUI to make changes, but it really limits what I can do beyond what they have programmed into their GUI. Would I be better off starting with: a) Plain old asterisk from asterisk.org? (tutorial suggestions?) b) AsteriskNow c) Trixbox (not Pro) d) other suggestions. Thanks Bill ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: why there havn'tapp_meetme.sofileaboutasterisk1.4.0?
This would do it, but a better way would be to specify --with-zaptel=PATH (PATH is the directory of zaptel sources) when running configure. If you already did a build you probably want to run make dist-clean before running configure again. Best regards, Alex Alex Epshteyn Third Lane Technologies, LLC http://www.thirdlane.com -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Bill Gibbs Sent: Thursday, February 01, 2007 6:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Re: why there havn'tapp_meetme.sofileaboutasterisk1.4.0? Removed the DEPENDS_app_meetme=ZAPTEL from the file menuselect.makedeps And in menuselect.makeopts I removed the DEPSFAILED line that had meetme in it. It then compiled. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of ?? Sent: Thursday, February 01, 2007 9:01 AM To: Asterisk Users Mailing List - No Subject: Re: [asterisk-users] Re: why there havn't app_meetme.sofileaboutasterisk1.4.0? Steven,hello! Thank you so much, but I have installed Zaptel before Asterisk. You have to compile and install Zaptel first, for asterisk to build meetme. -- -- Steven http://www.glimasoutheast.org 李君 [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] asterisk-users@lists.digium.com hi, I install asterisk1.4.0 , when I use the meetme application. The console show that WARNING[9872]: pbx.c:1755 pbx_extension_helper: No application 'Meetme' for extension . I found that there havn't app_meetme.so in the directory of moudles. Then I complied the asterisk1.4.0 again , there is no app_meetme.so also. How to overcome this problem? Thanks, Amy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = = = = = = = = = = = = = = = = = = = = 致 礼！ 李君 [EMAIL PROTECTED] 2007-02-01 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Running Multiple Instances of Asterisk
Thirdlane PBX Manager multi-instance can be used to manage/configure multiple instances of Asterisk. If you have any questions please contact me at [EMAIL PROTECTED] Best regards, Alex Alex Epshteyn Third Lane Technologies, LLC http://www.thirdlane.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Monday, September 25, 2006 2:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Running Multiple Instances of Asterisk I don't see a problem here. Using includes you dedicate every company their own directory of configs. Macros are eithere system wide, or each comapny can create their own. I don't see why this is any harder than mutilple instances of asterisk. On 9/25/06, Douglas Garstang [EMAIL PROTECTED] wrote: -Original Message- From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED] Sent: Monday, September 25, 2006 11:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Running Multiple Instances of Asterisk Asterisk does not support this, as it already has features for multi-client configuration within a single Asterisk installation/process. Douglas Garstang wrote: I'd like to know if anyone has sucessfully managed to run multiple instances of Asterisk on the same system. - Did you run each instance as a separate user? - Did you have any install or config problems? - It looks like the G729 codec registration utility doesn't work when files aren't installed in standard places. Did you have this problem? - How many instances could be run on a single Asterisk box? What do you mean 'does not support'? How easy do you think the management of the configuration files is going to be if your trying to host several dozen companies on the one Asterisk instance? Sure, you can split things into contexts, but just try and imagine how complex the management is going to become when several companies comprise the same file space. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Running Multiple Instances of Asterisk
We offer a management GUI for both options - multi-tenant (multiple companies within the same instance of Asterisk) or multi-instance (multiple instances of Asterisk). Best regards, Alex Alex Epshteyn Third Lane Technologies, LLC http://www.thirdlane.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann Boon Sent: Monday, September 25, 2006 5:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Running Multiple Instances of Asterisk Douglas Garstang wrote: -Original Message- From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED] Sent: Monday, September 25, 2006 11:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Running Multiple Instances of Asterisk Asterisk does not support this, as it already has features for multi-client configuration within a single Asterisk installation/process. Douglas Garstang wrote: I'd like to know if anyone has sucessfully managed to run multiple instances of Asterisk on the same system. - Did you run each instance as a separate user? - Did you have any install or config problems? - It looks like the G729 codec registration utility doesn't work when files aren't installed in standard places. Did you have this problem? - How many instances could be run on a single Asterisk box? What do you mean 'does not support'? How easy do you think the management of the configuration files is going to be if your trying to host several dozen companies on the one Asterisk instance? Sure, you can split things into contexts, but just try and imagine how complex the management is going to become when several companies comprise the same file space. Have you tried running asterisk in a chroot environment? It can do what you want. The only catch you'll have to specify the bindaddr for SIP. And, it works with IP aliases, so you can host multiple sessions on one NIC. Cheers. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Plan to free myself from AAH
Please take a look at PBX Manager - you may find it flexible and easy to extend. http://www.thirdlane.com Alex -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Strom Carlson Sent: Wednesday, May 17, 2006 10:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Plan to free myself from AAH On 5/17/06, David K Parker [EMAIL PROTECTED] wrote: I wouldn't knock the third party friendly interfaces to Asterisk too hard. They will evolve and improve over time. The adoption of Asterisk as a mainstream PBX is dependent upon a user friendly interface. Well, as soon as a GUI shows up that doesn't make configuring Asterisk like trying to sew with boxing gloves on, I'll give it a good, hard, unbiased look. For now, though, the available interfaces are really just not there yet - they don't allow enough flexibility and they are very easy to outgrow. -- Strom Carlson http://www.stromcarlson.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Re: Voicemail error
The problem is that the Voicemail application has no way to tell the difference between s in stephani and the s option, the same will happen to bob's b or ursula's u. If you absolutely need to use you could probably check for s, u, and b and uppercase them, but then messages will be left in a mailbox called Stephani, plus you will be in trouble again with VoiceMailMain where s is for skip password. Why not just use numbers? Alex -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of David L. West Sent: Saturday, May 06, 2006 2:36 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Re: Voicemail error Do you have lowercase on sip.conf and uppercase on voicemail.conf? The conf files are all lowercase. I didn't want to change those for fear of finding other case-related issues in the extension handling logic, so for this test I just hard-set ARG1 to STEPHANY.TOMAN in the stdexten macro. Further fooling around with case shows that some parts of Asterisk care, some don't. I'm therefore inclined to keep everything lowercase, and just convert to uppercase inside whatever function exhibits problems. Only trouble with that is it seems nobody has written a good uppercase/lowercase function for Asterisk yet. Amazing. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Web interface
I don't think I received the whole thread, but I just wanted to mention that the language selection has been added to the preferences page of PBX Manager. As Stefan mentioned, it is normally done in Webmin, but since some users may not be allowed to change anything outside of the module we added it there as well. I hope I am not jumping in the conversation about the different/wrong GUI :-). Alex Epshteyn Third Lane Technologies, LLC http://www.thirdlane.com -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Stefan-Michael. Guenther (in-put GbR) Sent: Monday, January 30, 2006 11:45 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Web interface Hello Zac, There is 1 problem.. I only took 1 semester of German 15 years ago. Looked all over the page for the English button, but I could not find one. I did wake up 10 minutes ago, so I could still be blind. the language of the module is influenced by the language you choosed for webmin. That's why there is no button in the PBX Manager to choose the language. I will rephrase the statement.. AMP hands down is STILL the best FREE asterisk manager... ACK ;-)) Bye, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Lösungen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music-on-Hold problem
Alex, thanks so much, that was it - I don't know how I missed it. I guess I was looking for more complicated reasons :-). Cheers, Alex -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Alexander O. Lopez Sent: Tuesday, November 08, 2005 7:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Music-on-Hold problem Have you tried adding an answer before playing MOH??? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Epshteyn Sent: Tuesday, November 08, 2005 11:10 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Music-on-Hold problem Hi, We are experiencing a strange problem playing music-on-hold - or perhaps it is a problem with the configuration of a Zap channel. When a call comes in from PSTN (FXO card) and MusicOnHold application is executed, the music on hold starts (Asterisk reports that the moh has started - and you can see that the mpg123 process is running) but the caller continues hearing ringing and no moh. Also, and possibly related, Zap channel stays in Offhook state after the caller hangs up. We tried a variety of options to make Asterisk detect hangup (busydetect, callprogress, etc) with no success. Could it be a hardware problem? Does anyone know of any bugs/issues/configuration errors that are likely to cause this? It appears that somehow the music being played is not delivered to the channel (could it be device configuration?). We are running Red Hat with 2.6 kernel, with udev configured as specified in README.udev, mpg123-0.59r. I apologize for not describing the whole environment, software versions, etc - I am not sure what info would be relevant. Help would be very much appreciated. Thanks, Alex ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Music-on-Hold problem
Hi, We are experiencing a strange problem playing music-on-hold - or perhaps it is a problem with the configuration of a Zap channel. When a call comes in from PSTN (FXO card) and MusicOnHold application is executed, the music on hold starts (Asterisk reports that the moh has started - and you can see that the mpg123 process is running) but the caller continues hearing ringing and no moh. Also, and possibly related, Zap channel stays in Offhook state after the caller hangs up. We tried a variety of options to make Asterisk detect hangup (busydetect, callprogress, etc) with no success. Could it be a hardware problem? Does anyone know of any bugs/issues/configuration errors that are likely to cause this? It appears that somehow the music being played is not delivered to the channel (could it be device configuration?). We are running Red Hat with 2.6 kernel, with udev configured as specified in README.udev, mpg123-0.59r. I apologize for not describing the whole environment, software versions, etc - I am not sure what info would be relevant. Help would be very much appreciated. Thanks, Alex ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to configure Asterisk through webmin
Hi Seshu, I would be happy to walk you (or anyone else who may be interested) through the Thirdlane PBX Manager features, to explain that while it wont magically configure Asterisk for you, it does help quite a bit. It is all really about the expectations and the target audience what is a good tool for some is too limiting for the others, and whatever is not limiting may appear too complex and not immediately useful. Please contact me off list at [EMAIL PROTECTED], or even better, we could spend a half an hour on the phone that may change your opinion. Best regards, Alex From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT) Sent: Friday, November 04, 2005 6:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] How to configure Asterisk through webmin The Thirdlane PBX Manager solution is just a few perl scripts. This is no better than what you can do by directly modifying the Asterisk Config files or many Open Source GUIs like Phonecall etc you have out there. Infact Areski's A2Billing has a good extension configurator in the solution. So that may be something you can consider. Seshu Kanuri From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pikoro Sent: Thursday, November 03, 2005 7:09 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to configure Asterisk through webmin I tried the third lane asterisk manager thingy for webmin and let me tell you, it did not work. Only made things harder and i had to result to making the configuration by hand in order to get asterisk to work. Going to email them today and ask for a refund. That webmin module by third lane looks like a good solution, but the thing i noticed by reading the manual was that there are quite a few references to you'll have to change that in the config file type lines. Basically, it's good for creating extensions, but nothing more. Aaron Stefan-Michael. Guenther (in-put GbR) wrote: On Thu, November 3, 2005 17:46, nr k said: Hi allI configured asterisk and webmin.i dont know how tointegrate webmin with asterisk and how to accessasteriskthrough webmin.pls do the needful.regardsramakrishnan.n Asterisk is not managed through webmin. Webmin is a tool to helpadminister the rest of the server. and Asterisk, too:Have a look at THIRD LANE ASTERISK PBX MANAGERhttp://www.thirdlane.com/opensource.htm#managerStefan NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
