Re: [asterisk-users] New VirtualBox Beta Has PCI Pass-Through Support
On Fri, Jul 08, 2011 at 07:28:39PM -0400, Doug Lytle wrote: Can you say a Virtualized Asterisk with a PRI card! http://www.phoronix.com/scan.php?page=news_itempx=OTY0OQ I'm not sure this is relevant to your everyday PRI card. Simple pass-through of PCI is rather simple to implement. It has been implemented for quite some time in e.g. qemu, IIRC. But there's one important feature that is tricky to implement: DMA. DMA allows a PCI card to basically free access to the memory of the system, without the CPU getting in the middle. Emulating this with the host CPU in the middle will work, but is exepnsive. But the host does not want to give the guest free access to the host's memory. The solution: IOMMU: http://en.wikipedia.org/wiki/Iommu . The CPU of the system has a Memory Management Unit (MMU) that maps virtual address spaces to processes. Likewise we know prevent the IO card from seeing physical addresses. Rather, it sees virtual addresses mapped by the IOMMU. Just like the operating system maps addresses for processes, the hypervisor maps address ranger to IO cards. See also the link in that Phoronix article: http://www.ibm.com/developerworks/linux/library/l-pci-passthrough/ I'm not well familiar with the relevant hardware, but I believe that not only not every CPU supports this, but many (most?) PCI cards don't support IOMMU. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] OT: Google Plus
On Sat, Jul 09, 2011 at 01:34:03PM +0200, randulo wrote: Go ahead and lambast me for this post, it isn't specific to Asterisk, but: G+ has only been open at all for a week and I already am chatting with over 200 people who are into VoIP, Asterisk and all the rest of the stuff we here care about. If you don't care or are anti-social, fine. But you owe it to yourself to check it, because a lot of cool VoIP people are there and after all, Google themselves are doing some great stuff with VoIP, XMPP and video, and steadily moving towards open source. Come drink the Kool-Aid! I'll believe it when I see it. Google Talk uses an existing federated protocol. It does use many extensions by Google (and many deviations of Google from the reference implementation it published, but never mind). Buzz was (technically: is) a federated protocol. Google Plus seems to be a walled garden. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] Thomson ST022 - External Call problems
Hy all of you, I've successfully installed a freepbx solution with 10 extensions : - 5 on Linksys SPA922 - 1 on Linksys SPA942 - 1 on Thomson ST022 Everything seems to work fine with all the hardphones excepts last week. The thomson has a strange behaviour. It can reach french mobile cell phones but when it reaches fix phones, the correspondant can't hear the caller. What is very stange is that for incoming calls for this extension, everything works fine. Thanks for your help, Regards -- _ -- 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] OT: Google Plus
On Saturday, 9 July 2011, Gordon Henderson gordon+aster...@drogon.net wrote: On Sat, 9 Jul 2011, Steve Davies wrote: On 9 July 2011 12:34, randulo rand...@randulo.com wrote: Go ahead and lambast me for this post, it isn't specific to Asterisk, but: G+ has only been open at all for a week and I already am chatting with over 200 people who are into VoIP, Asterisk and all the rest of the stuff we here care about. If you don't care or are anti-social, fine. But you owe it to yourself to check it, because a lot of cool VoIP people are there and after all, Google themselves are doing some great stuff with VoIP, XMPP and video, and steadily moving towards open source. Come drink the Kool-Aid! Can you suggest a good way of finding/following appropriate VoIP/Asterisk people once on Google+? How do you then group them? Just in a Circle, or some other mechanism? I've just created a VoIPy circle - So I can then invite people I know into the circle by email address, and/or looking at someone else's circles and seing if they have something relevant in their summary tag and adding them into your own circle... (Or using their people search - e.g. for 'randulo' :) You can have people in more than one circle. Right now, it's a bit like a media-rich version of twitter with excellent filtering (the circles). I don't have camera/microphone/speakers on my PC, (got real desk SIP phones!) so haven't tried the audio/video chat yet, but the typing instant messaging type chat works just fine. I think Google are still slowly gating people into + though. I did have some invites, but seem to have used them all up now (google didn't tell me how many, the invite button just went away after a while!) I'd love to see SIP integration into it, so I can use my existing SIP toys with it. Gordon Thanks for that Gordon. What appears to be missing at the moment is the ability to interface or collaborate with a group of 'strangers'. It would be good if there were a way to broadcast a 'we're here, come join us' to bring a group of VoIP people together, a bit like an IRC channel name can do, or a Facebook fan page. I thought that sparks might cover that, but I'm not entirely sure how sparks work yet. I agree that SIP integration would be great. I think it'll be a while yet but if anyone will allow it, it'll be Google. Cheers, Steve -- _ -- 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] OT: Google Plus
On Sun, Jul 10, 2011 at 11:07 AM, Steve Davies davies...@gmail.com wrote: Can you suggest a good way of finding/following appropriate VoIP/Asterisk people once on Google+? How do you then group them? Just in a Circle, or some other mechanism? It's word mouth now, but I think there will be discovery mechanism soon. I've just created a VoIPy circle - So I can then invite people I know into the circle by email address, and/or looking at someone else's circles and seing if they have something relevant in their summary tag and adding them into your own circle... (Or using their people search - e.g. for 'randulo' :) Once you found me, you should have been able to find the post where I've put names of most of the VoIP USers COnference people. More then added their own. https://plus.google.com/104027218792812194992/posts/Xvnbp1YWf9K You can have people in more than one circle. Right now, it's a bit like a media-rich version of twitter with excellent filtering (the circles). I don't have camera/microphone/speakers on my PC, (got real desk SIP phones!) so haven't tried the audio/video chat yet, but the typing instant messaging type chat works just fine. You have to try the Hangou because that's an amazing feature and it's the one I want to see with SIP interface so we can bridge to a SIP conference. I think Google are still slowly gating people into + though. I did have some invites, but seem to have used them all up now (google didn't tell me how many, the invite button just went away after a while!) I'd love to see SIP integration into it, so I can use my existing SIP toys with it. That would be my wish, too. In the end, it is a process of finding the right people. You can see all public posts in the stream. However if there were 20 people say, from this list in my Asterisk Circle talking aout SIP integration, we'd keep it private, NOT to hide, but to not bore our other friends in Basket weaving Circle. I encourage anyone who's the + and interested to look me up. I can easily blast out more names as suggestions. :r -- _ -- 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] OT: Google Plus
On Sun, Jul 10, 2011 at 11:07 AM, Steve Davies davies...@gmail.com wrote: Thanks for that Gordon. What appears to be missing at the moment is the ability to interface or collaborate with a group of 'strangers'. You watch the stream to discover people but obviously that's a long process. It will be even longer as adoption grows. It would be good if there were a way to broadcast a 'we're here, come join us' to bring a group of VoIP people together, a bit like an IRC channel name can do, or a Facebook fan page. Hangouts are broadcast to the public (everyone's stream) unless you state otherwise. Nothing stops anyone from blasting out names. If you can find my post about VoIP people, you can add your name in the comments or asl me and I will blast it out. I thought that sparks might cover that, but I'm not entirely sure how sparks work yet. Sparks is currently just a topic search and it's pretty lame in everyone's opinion. I agree that SIP integration would be great. I think it'll be a while yet but if anyone will allow it, it'll be Google. :r -- _ -- 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] References customers
Hi All; How can I find a references customers that used Asterisk as IP Telephony or Call Center or IVR? In which link they are mentioned? Regards Bilal -- _ -- 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] OT: Google Plus
On Sun, Jul 10, 2011 at 10:16 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: Google Plus seems to be a walled garden. Wait for the API. :r -- _ -- 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] OT: Google Plus
On Sun, Jul 10, 2011 at 12:17:52PM +0200, randulo wrote: On Sun, Jul 10, 2011 at 10:16 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: Google Plus seems to be a walled garden. Wait for the API. Will that API allow me to run a separte (compatible) server? API normally implies that you connect as a special client to a server. If I want to set up my own independent service that is allowed to chat with people using Google Plus (federated, as in as in XMPP, SIP and SMTP), I guess I will not be able to. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] OT: Google Plus
On Sun, Jul 10, 2011 at 12:25 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Sun, Jul 10, 2011 at 12:17:52PM +0200, randulo wrote: On Sun, Jul 10, 2011 at 10:16 AM, Tzafrir Cohen I don't see you on G+, are you there? :r -- _ -- 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] OT: Google Plus
On Sun, Jul 10, 2011 at 12:26:45PM +0200, randulo wrote: On Sun, Jul 10, 2011 at 12:25 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Sun, Jul 10, 2011 at 12:17:52PM +0200, randulo wrote: On Sun, Jul 10, 2011 at 10:16 AM, Tzafrir Cohen I don't see you on G+, are you there? Me? You may see me there if it proves to be a federated service. I'm likewise not on Twitter and prefer Status.Net (ATM I actually have an account on http://identi.ca , but I may set up my own server). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] Queue Issue : Duration between 2 agents call
Hy, I'm currently working with one queue and whatever I change in the config, it stills a gap of 6 seconds during which no agents are ringing for this queue. Is ther any parameter to configure there? regards -- _ -- 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] References customers
What do you mean by customers? Are you looking for testimonials from satisfied users? -- On 7/10/11 11:53 AM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; How can I find a references customers that used Asterisk as IP Telephony or Call Center or IVR? In which link they are mentioned? Regards Bilal -- _ -- 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] Problem with setting up fresh 1.8.5 Asterisk
Is there some detailed documentation for 1.8.5? I am tryin to make Asterisk 1.8.5 with MySQL backend, TLS transport and SRTP encryption. For now with no success :-(-- _ -- 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] Problem with setting up fresh 1.8.5 Asterisk
On 07/10/2011 05:02 PM, Matiss Jekabsons wrote: Is there some detailed documentation for 1.8.5? I am tryin to make Asterisk 1.8.5 with MySQL backend, TLS transport and SRTP encryption. For now with no success :-( https://wiki.asterisk.org/wiki/display/AST/Asterisk+1.8+Documentation Regards, Patrick -- _ -- 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] Problem with setting up fresh 1.8.5 Asterisk
Cool thx :) dont know why i didnt found it myself :D Quoting Patrick Lists asterisk-l...@puzzled.xs4all.nl: On 07/10/2011 05:02 PM, Matiss Jekabsons wrote: Is there some detailed documentation for 1.8.5? I am tryin to make Asterisk 1.8.5 with MySQL backend, TLS transport and SRTP encryption. For now with no success :-( https://wiki.asterisk.org/wiki/display/AST/Asterisk+1.8+Documentation Regards, Patrick -- _ -- 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 -- -- Best regards Matiss Jekabsons Procerto Ltd. ICT project manager GSM: (+371) 22440298 E-Mail: mat...@procerto.lv Dzelzavas Str. 117. Riga, Latvia-- _ -- 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] OT: Google Plus
On Sun, Jul 10, 2011 at 12:39 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: I don't see you on G+, are you there? Me? You may see me there if it proves to be a federated service. Tzafrir, I know you so I know you won't take this as a personal insult. Why comment on something you aren't a part of? I can easily understand people not wanting to be on any of these networks, but I don't understand how they (not you in particular) can know what they're talking about if they haven't even seen it first hand. I guess it ends with the statement, not federated, not worth doing. That is a limitation I don't agree with, but we're not all the same. :r -- _ -- 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] OT: Google Plus
On Sun, Jul 10, 2011 at 07:08:55PM +0200, randulo wrote: On Sun, Jul 10, 2011 at 12:39 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: I don't see you on G+, are you there? Me? You may see me there if it proves to be a federated service. Tzafrir, I know you so I know you won't take this as a personal insult. Why comment on something you aren't a part of? I can easily understand people not wanting to be on any of these networks, but I don't understand how they (not you in particular) can know what they're talking about if they haven't even seen it first hand. I guess it ends with the statement, not federated, not worth doing. That is a limitation I don't agree with, but we're not all the same. I corrected a few factual errors on your part. Then I answered some direct questions by you. But if you only look for feedback from the believers, why do you bother asking here? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] Help anybody - how to manage SRTP with TLS trasport
Working on that about a week and not getting closer. Now upgrading to Asterisk 1.8.5 with a bit of hope that will work. TLS is working just fine, but not SRTP. Module is not loading and thats it. Asterisk 1.8.4.4 -- Best regards Matiss Jekabsons Procerto Ltd. -- _ -- 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] References customers
I mean: What are the customers (big customers I mean) that they installed Asterisk in their company to be as a reference? Example: Toyota, GM, Hilton, Shiraton hotel, ... etc An example of such companies, whom? Is there a link that mention them? Regards Bilal --- What do you mean by customers? Are you looking for testimonials from satisfied 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] References customers
On Sun, Jul 10, 2011 at 5:22 PM, bilal ghayyad bilmar...@yahoo.com wrote: I mean: What are the customers (big customers I mean) that they installed Asterisk in their company to be as a reference? Example: Toyota, GM, Hilton, Shiraton hotel, ... etc An example of such companies, whom? Is there a link that mention them? Regards Bilal --- What do you mean by customers? Are you looking for testimonials from satisfied users? http://www.digium.com/ scroll down to the bottom Google, Yahoo, US Army, IBM just a few small little businesses. -- ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- 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] What is the use for the agent password if login via exten?
Hi All; Why we use the agent password when we configure the agent in the agents.conf if the agent login by dialing the number configured in the extensions.conf? example: exten = 28, 1, AgentLogin(1001) I know that agent username is used to assign the agent to the queue, but when we use the agent password? Regards Bilal -- _ -- 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] How to logout!
Hi All; How agent logout if he logged in using AgentLogin? Also, there is not ready and not ready status? Or only agent login and logout? Appreciate ur kindly help. Regards Bilal -- _ -- 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] How to logout!
Dear
check it ,1 for log in .0 for log out.
exten = 1,1,AddQueueMember(team,SIP/${CALLERID(num)});
exten = 1,2,Playback(agent-loginok)
exten = 0,1,RemoveQueueMember(team,SIP/${CALLERID(num)});
exten = 0,2,Playback(agent-loggedoff)
On Mon, Jul 11, 2011 at 3:23 AM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi All;
How agent logout if he logged in using AgentLogin?
Also, there is not ready and not ready status? Or only agent login and
logout?
Appreciate ur kindly help.
Regards
Bilal
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Re: [asterisk-users] What is the use for the agent password if login via exten?
you have 2 options, add an agent to the queue or add a registered ip phone( or pstn line) to the queue. in first option, your operator must enter a password to identify as agent. but next option does not need password. On Mon, Jul 11, 2011 at 3:06 AM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; Why we use the agent password when we configure the agent in the agents.conf if the agent login by dialing the number configured in the extensions.conf? example: exten = 28, 1, AgentLogin(1001) I know that agent username is used to assign the agent to the queue, but when we use the agent password? Regards Bilal -- _ -- 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 -- Pezhman Lali -- _ -- 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] Thomson ST022 - External Call problems
check the codecs, it's too important On Sun, Jul 10, 2011 at 1:22 PM, Florent THOMAS mailingl...@tdeo.fr wrote: ** Hy all of you, I've successfully installed a freepbx solution with 10 extensions : - 5 on Linksys SPA922 - 1 on Linksys SPA942 - 1 on Thomson ST022 Everything seems to work fine with all the hardphones excepts last week. The thomson has a strange behaviour. It can reach french mobile cell phones but when it reaches fix phones, the correspondant can't hear the caller. What is very stange is that for incoming calls for this extension, everything works fine. Thanks for your help, Regards -- _ -- 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 -- Pezhman Lali -- _ -- 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] DB Driven IVR
Hi, You can use combination of dial-plan and AGI for making DB driven IVR. On Sat, Jul 9, 2011 at 5:50 PM, G M gm.cu...@gmail.com wrote: Anyone has Experience ? On Fri, Jul 8, 2011 at 2:18 PM, G M gm.cu...@gmail.com wrote: I am using Vicidial and I am looking for someone who can help with DB Driven IVR. -- _ -- 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] Problem in Detecting Dtmf on FXO line.
Hi, I tried with cid_rxgain,rxgain to put upto 5.0 and 10.0 values but not getting success. regards Dhaval On Fri, Jul 8, 2011 at 8:34 PM, Ruben Rögels ruben.roeg...@jumping-frog.org wrote: Am 08.07.2011 08:58, schrieb DHAVAL INDRODIYA: Hi All, I am having Problem in detecting DTMF on analog lines. basically are system is in india and telco provider is BSNL [Bharat sanchar Nigam LImited]. We have Purchased Analog card From chinaroby.com http://chinaroby.com which is X1600P 16 port FXO card. they also provide us wctdm.c file. card is detected successfully, incoming and outgoing calls scenario is also fine. we are unable to receive dtmf properly it means there is some digit are missing when we receive dtmf the ratio of sucess is about to 70% and 30% of calls are getting wrong dtmf . Dahdi version is 2.3.0.1 and asterisk version 1.6.0.24 I load module using modprobe wctdm opermod=INDIA cidbeforering=1 cidbuflen=1 fixedtimepolarity=16 here id chan_dahdi.conf. Hello, did you try plaing with rxgain and txgain? When I set up a TDM400, I had some issues with DTMF because the signals where overmodulated. Regards, Ruben -- _ -- 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] Queue Issue : Duration between 2 agents call
Check 'retry' in queues.conf [SATISH] Mumbai, India. On Sun, Jul 10, 2011 at 4:34 PM, Florent THOMAS mailingl...@tdeo.fr wrote: Hy, I'm currently working with one queue and whatever I change in the config, it stills a gap of 6 seconds during which no agents are ringing for this queue. Is ther any parameter to configure there? regards -- _ -- 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] OT: Google Plus
On Sun, Jul 10, 2011 at 7:48 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: I corrected a few factual errors on your part. Then I answered some direct questions by you. But if you only look for feedback from the believers, why do you bother asking here? My bad, in that case. Apologies! :r -- _ -- 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
