Re: [asterisk-users] Looking for PRI Card with automatic fail over
Hi there, Using Sangoma Vega400 gateway you'll have what is called resilliency which is exactly what you are looking for. Regards. -- M. Shokuie Nia On Aug 3, 2015 18:51, Eric Klein eric.kl...@greenfieldtech.net wrote: Hi all, Strange request, I have a customer where we are putting an Asterisk PBX in front of a legacy (non-VoIP) PBX. One of the requirements it that the Asterisk PBX have 2 PRI ports (on towards the legacy PBX and one towards the carrier) with the ability to go to pass through should the Asterisk PBX (software or hardware level) fail. I did not see this feature in the Digium, Sangoma, Allo, or OpenVox cards. Does anyone know of a card that will do this? I know that Digium has an external box (the r850) that does something similar for 2 PBXs making them high availability, but in this case I only have the 1 Asterisk box acting as a gateway and passing some calls out over SIP and IAX2. Any suggestions would be appreciated. Thanks Eric -- _ -- 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] Help choosing the right card
Although it was better to ask it in Asterisk commercial list but you have different options like Digium, Sangoma or Openvox. TDM410P is the PCI one from Digium which suits your description. Just remember to buy two trunk (FXO) modules too and if you are looking for a best sound qulity get hardware echo canceller too. I just didnt get why your are going to set 5 ext on each IP Phone!!! -- M. Shokuie Nia On Sat, Jun 16, 2012 at 4:34 PM, Amit Patel pistolfir...@gmail.com wrote: I have been doing a lot of reading forums and elsewhere but am somehow unable to connect the dots. Here is what I am trying to accomplish initially and then wish for it to grow bigger from here on. I have two POTS (Analog) line that would connect to the Asterisk Box. I have, to begin with 5 IP phones (PoE), all connected to a switch. Asterisk Box with a LAN card also connects to the same switch. I wish to give out 5 ext to each IP Phone. Q) I am considering buying the TDM410p. Is this the right card for me ? T1/E1/Digital/Voip trunks atm are not available at my place. Q) Would Asterisk handle taking incoming calls via TDM410p and routing them via LAN card to the desired ext IP phone ? If yes, I would hit the 'Buy' button and start the journey. -- _ -- 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] Sangoma A400 background noise after a while
Dear folks, We have recently installed A400D card with 12 FXO modules, the serer is HP DL180 G6, cards works fine but after a while all the calls get an awful noise, you can not get what each side says. The noise cleares as soon as we restart wanrouter but not asterisk (i mean asterisk restart does not solve). We previsouly confronted this situation with PRI cards but not analogs, wanpipe version is 3.5.18 and zaptel 1.4.12 also tested with recent DAHDI with out any help. ifconfig doesnt show any overruns or errors. Once earlier we had the same problem and come to the conclusion to change the mainboard but this time i got mad as i couldnt change a 3000$ HP server that easy. Is there a way i could get if there is any problem of interrupts, when i check interrupts i could not see any shared interrupts for Snagoma card. Anyhelp would be highly appreciated. -- MSH -- _ -- 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 compatible cards?
Hi, The best cards I used are Sangoma's but if you cant afford them go for OpenVox, they are exactly the same with digiums and their 4 port ones doesnt need any driver installation even. Regards. -- M. Shokuie Nia. On Wed, Aug 11, 2010 at 8:55 AM, Faisal Hanif fai...@vopium.com wrote: Hi, We are using 4-PRI card from http://atcom.cn for our development LAB and we are satisfied with performance. It is also cheaper then other products. They also have analog. Regards, Faisal Hanif *VoIP Manager ***Vopium A/S On 8/10/2010 6:40 PM, Jeremy Betts wrote: I have always had very bad experiences with the x100p cards, they always have very bad echo. If you need decent call quality I would wait until you can afford a Digium card. On Tue, Aug 10, 2010 at 2:49 AM, Gordon Henderson gordon+aster...@drogon.net gordon%2baster...@drogon.net wrote: On Tue, 10 Aug 2010, Daniel Petre wrote: hello all, for my home purpose, i found nothing in my budget in romania, very limited options, so i turned to ebay where i found two cards in the 25 usd price range: a voxzone x100p and a infomatrix x100p, their descriptions from ebay: Never used them myself - always used Digium or openvox cards. I'd be surprised if there was any difference between them though. Google for x100p images and compare the chips... Gordon voxzone x100p: * Universal Voxzone X100P FXO PCI for DigiumTM Asterisk * Works with Official Asterisk Zaptel Driver * Connects Asterisk Box to PSTN * No Echo issues compared to MD3200 * Support via www.voxzone.com/forum/ * Original DAA Chipsets with Caller ID * Low profile bracket is available on request infomatrix x100p (a2 and b2) : # InfoMatrix ? X100P(A2), Intel Chipset, 2nd Generation, PCI, Single FXO port , # Guarantied Caller ID/Redirection, Call transfer, Ring and Remote hang up detected # Support and work with Mutiple Protocols: SIP, IAX, H.323, MGCP, Skinny/SCCP... # Support Global codecs for all countries to use: G711, G726, G723.1, G729A, GSM # Fully tested to support the Asterisk and its appliance, all PC systems(old, new) # 100% compatible with Digium WildCard X100P, X101P card and more features. anyone has any idea which one should i buy? i intend to use it in a SFF dell computer with one pci port (its pretty tight to the PSU wall but i hope it will fit.. any interference because of that?) thanks! On Mon, 2 Aug 2010, Daniel Petre wrote: hello, i just subscribed to this list, i discovered asterisk and i would like to try it at home on my personal pc. the computer is a p4 at 3 ghz with 2 gb ram and 80 gb hdd, a 1 Mbit guarranted connection and runs a gentoo linux. i search about digium products but i can't find them in my area on any shops, i was wondering if good people here could recommend some PCI or PCIex cards for a beginner to play with one telefonic line (which i will install it soon via provider) If you really can't get digium cards, then look on ebay for x100p cards - you might get lucky... Failing that, OpenVox have some compatable cards - you might find an importer locally who deals in them. Gordon -- _ -- 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
Re: [asterisk-users] IAX Hardphones.
Hi Dude, I've used both AT530 and AT530P ip phones, they have good voice quality and somehow resistent to harsh environment like offices ;) , except the keypad which after a year or more just sometimes types two digits with one push on a button, the AG188N ATA is also a good choise and we have some installation which work without any problem yet. Regards. -- M. Shokuie Nia. On Fri, Oct 23, 2009 at 9:48 AM, Andrew Higgs andrew.m.hi...@gmail.comwrote: Hi Albert, We have also had success with the 530P phones. I think for their price they are very nice phones. Regards Andrew Higgs On Thu, Oct 22, 2009 at 6:33 PM, Albert Culleton a...@icmunicomp.ie wrote: Hi there, Has anyone Used ATCOM IAX Hard phones with any success? ___ -- 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
[asterisk-users] Sangoma A200 and battery removal detection ??!!!
Dear Folks, Anyone knows if Sangoma supports or going to provide support for battery removal detection on FXO lines?? As Tzafrir said earlier DAHDI supports it, which is a very nice feature but what about Sangoma? Regards. -- M. Shokuie Nia. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZAP and line disconnection detection
Hi Tzafrir, Thanks for the hint, I'll check it to see if Sangoma supports this or not. Regards. -- M. Shokuie Nia. On Thu, Sep 17, 2009 at 10:27 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Thu, Sep 17, 2009 at 09:34:56AM +0330, M Shokuie wrote: Dear Folks, Im looking for a way to detect if an analog line is connected to card or not (Im using Sangoma A200). Im using the dialtone detection when dialing but need a way to detect the disconnection of the line when it actually happens. I have no idea about the Sangoma drivers, but reecnt in-tree DAHDI drivers report this by raising a RED channel alarm if there's nothing connected. This means that Asterisk won't try dialing through it. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ZAP and line disconnection detection
Dear Folks, Im looking for a way to detect if an analog line is connected to card or not (Im using Sangoma A200). Im using the dialtone detection when dialing but need a way to detect the disconnection of the line when it actually happens. Anyone have any hints or tricks for this? Regards. -- Mohammad Sh. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Default login information for a ArtDio IPF-2600
Hi There, Im not sure about IPF-2600 but on IPF-2200L, it's 12345678 for web access and 1234 on the phone itself. Give it a try it might be the same for the your model too. I just want to know if you are satisfied with the phone or not IPF-2200L is unsatisfactory in different aspects. First is the very low mic volume, the other party can hardly hear you even with the highest volume possible, Second is the sip log in process which sometime might take 10 minutes, third is the poor echo cancellation. I'd be glad if you can inform me about the quality of your model too. Regards. --- M. Shokuie Nia. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of George Patterson Sent: 2006/10/26 05:13 ق.ظ To: asterisk-users@lists.digium.com Subject: [asterisk-users] Default login information for a ArtDio IPF-2600 Hello, I recently purchased a ArtDio IPF-2600 phone from voipsupply.com, but they did not include a manual. Does anyone know the default login information? I have tried all of the common ones that I can think of. If anyone knows, it would be greatly appreciated. Thanks! _ Stay in touch with old friends and meet new ones with Windows Live Spaces http://clk.atdmt.com/MSN/go/msnnkwsp007001msn/direct/01/?href=http://spa ces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends.aspxmkt=en-us ___ --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] Quintum DX as gateway to PSTN for Asterisk
Hi there, I had the same configuration and it nearly took me a week to solve the problem and atlast I'm not sure if what I've done is the right way. I need Phone-Ast-Quintum-PSTN, so i defined a trunk in quantum with proper fxo lines in it then a hop off in the quintum with proper extension that Ast sends to it and include this hup off in the trunk. This way calls with the defined prefix from Ast forwarded to PSTN in the quantum. Regards. --- M. Shokuie Nia. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of doki_cti Sent: 2006/10/25 03:57 ب.ظ To: asterisk-users@lists.digium.com Subject: [asterisk-users] Quintum DX as gateway to PSTN for Asterisk Hello, I try configuring Quintum DX gateway as link to PSTN for *. Now, I can dial number which is connect to Quintum, and call is diverted to *. I don't know what I should set, if I want call from SIP_phone registred in Asterisk to PSTN via Quitnum. I set in sip.conf account for Quintum [sip_proxy-out] type=peer outboundproxy=QUINTUM_IP , and changed extensions.conf. When I call from SIP Phone, I see in Quintum log, that call is received with good caller and called numbers, but I think that quintum don't how route this call (he diverte this call to asterisk). So, can you give me advice what I should set, when I want route all calls from IP to PSTN and from PSTN to * via IP? How set password and user for quitnum and calls from SIP? Is it posible on Quintum or I should use for this radius? Regards Doki ___ --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] VoiceOne 0.4.0 released: a new web-based and opensource GUI
Hi Alex and dev team, I've just checked the demo on your site and going to install it without any time waste. It's absolutely marvelous and there isn’t any free, save as to open source comparable project on the web. You would have plenty of users with no doubt in near future. Regards --- M. Shokuie Nia -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Sent: 2006/10/25 11:22 ق.ظ To: asterisk-users@lists.digium.com Subject: [asterisk-users] VoiceOne 0.4.0 released: a new web-based and opensource GUI Hi all! We've released VoiceOne 0.4.0, a web-based and open source solution which allows to fully manage an Asterisk service hosted on a LAMP server. We focused on an charming and overall user-friendly interface. Thanks to the authentication based on roles, once configured by a super user, the PBX may be easily maintained even by an Asterisk unskilled users. From a technical point of view, the application is made up of two modules: one for the client - i.e. the user interface - and the other for the server. Thanks to the web services provided by the server module and the use of a database, VoiceOne may be easily integrated with other applications (e.g. CRM software). The project has grown and has received positive response so far. Nowadays there's a little but enthusiastic community of developers, supporters and users. Translations in several languages (e.g. English, Spanish, Russian, etc.) are already available. On the project website at http://www.voiceone.it you'll find the online demo and the links to download the source files from Sourceforge, as well as a support forum. We would be pleased if you could give it a try and let us know your feedback, comments, ideas, or suggestions replying here or posting a message on our forum. Thanks for your kind attention. Regards, Alex ___ --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] rxfax problem
Dear folk, My problem solved after two day research and try and error method ;). It was related to rxgain of the board im using. I've set the rxgain to 12 and it seems made some problem. As far as I got the spandsp is so sensitive about noise on the line and because of that it couldn’t hand shake with other side well. HTH. M. Shokuie Nia. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim McIver Sent: 2006/10/19 06:17 ب.ظ To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] rxfax problem Did you ever get an answer to this problem ? I too am seeing this and it’s driving me mad !!! Jim ___ --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
[Asterisk-Users] Using * and other gateways together
Deaf folks, Actually this is my first post here, so sorry for any inconvenience. Im planning for a solution a bit larger in scale than ususal. I'm goin to use * as a PSTN gateway with E1 links and use two other 3rd party Gateways for FXO lines. I should be able to switch from every incoming channel to any outgoing one and also to some SIP softphones. I planned to use SER as a sip server but really dont know were I should enforce my call routing mechanisms. Is SER applicable of doing that or should i write any application on the SER to do so ro is there any need for a softswitch at all? Or as a more basical question is there any need for SER, Asterisk cant do it itself? Any help and hints would be highly appreciated, M. Shokuie Nia. ___ 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
