Re: [asterisk-users] Asterisk + E1 with MFC/R2 in Argentina?
Yes, Telefonica is able to do PRI but just in a very restrictive area. RR Libera Ilan Rabinovitch escribió: We briefly used it with iPlan, but found that there were some problems with the stock asterisk implementation and Argentina variation of R2. We ended up convincing iPlan to switch us to PRI. As soon as we switched to PRI all problems disappeared. Any idea if Telefonica will be able to do PRI instead of R2? On 10/11/06, R.R. Libera [EMAIL PROTECTED] wrote: Hello, Has somebody installed this configuration: Asterisk + E1 with MFC/R2 (Telefónica Argentina) in Argentina before? I need to know if it´s possible with MFC/R2 argentine variation. Thanks in advance. R.R. Libera ___ --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] Billing Solution ?
Try www.asterisk2billing.org Noc Phibee escribió: Hi what is the best billing solution for Asterisk ? With WWW manager interface for user can see the real invoice... Thanks bye ___ --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
[asterisk-users] hardware requirements..
Hello, I have Asterisk running on Debian Sarge. I use AGI to billing and now I´m planing to separate mysql from Asterisk Box, resulting: (Asterisk + AGI) - (Iptables + showrewall + Apache + PHP + MySQL) What hardware configuration do you recommend for the second box, in order to maintain the quality in calls? I´m planning to run upon 30 (E1) concurrent calls. Thanks in advance. R.R. Libera ___ --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] VoicePulse Connect 4 Channel Limit?
Good prices means (exactly) reasonable prices. I´m a newbie, so I´m asking for good experiences... Thanks in advance... R.R. Libera Lacy Moore - Aspendora escribió: So, What´s your recommendation for a production environment? I was looking for good prices, good voice quality for USA Origination and I´d like to hear about good experiences PSTN. Just can't beat the quality :-) Wait, you said good prices. Sorry. ___ --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] VoicePulse Connect 4 Channel Limit?
So, What´s your recommendation for a production environment? I was looking for good prices, good voice quality for USA Origination and I´d like to hear about good experiences thanks in advance.. R.R. Libera Andrew Joakimsen escribió: I would worry about using Voicepulse as your primary provider, even if they didn't impose their draconian policies. You could have 20 numbers paying $220/month in your account and you still get only four calls,. However if you were to open 20 voicepulse connect accounts and put one number on each, you would still pay the same $220/month however you could get up to EIGHTY -- thats 20 times more! -- for the SAME EXACT PRICES Also VoicePulse DOES NOT use Tier 1 providers. We notice every week or so during peak hours a very bad degrigation of the voice quality. If you have an IVR and call it from a landline, it will sound like crap. It's the quality of service you would expect from a free provider. Aggrivated to this, when you contact them they try to blame YOU for their issues. They told me I HAD to run PingPlotter (a WINDOWS program, besides the fact this is VoicePulse Connect for Asterisk and Asterisk is software for Linux) which was not possible on a co-located machine. Also we ported a bunch of phone numbers and the DTMF does not work. If you dial 5551212 VoicePulse might recognise and pass to us 55112 and again instead of trying to troubleshoot the issue (from the SAME phone it always produced CONSISTANT behavior -- the ported number does not accet DTMF correctly, assigned # work!) they blame us and the phones we use. I went as far as going to Sprint PCS store and EVERY CDMA phone in the store would produce the same result! In the end, don't bother with VoicePulse. The quality of the service and the support and just the treatment you get is not worth the price. For $11/month per number and their draconian channels and also billing policy (I wont even get into that) I expect a PREMIUM service and they deliver something about par for a free service. Here's some typical behavior from their servers: ug 24 14:55:48 NOTICE[13944]: chan_iax2.c:7813 iax2_poke_noanswer: Peer 'voicepulse01' is now UNREACHABLE! Time: 71 Aug 24 14:55:59 NOTICE[13944]: chan_iax2.c:7145 socket_read: Peer 'voicepulse01' is now REACHABLE! Time: 1059 Aug 24 15:11:05 NOTICE[13944]: chan_iax2.c:7813 iax2_poke_noanswer: Peer 'voicepulse01' is now UNREACHABLE! Time: 40 Aug 24 15:11:15 NOTICE[13944]: chan_iax2.c:7145 socket_read: Peer 'voicepulse01' is now REACHABLE! Time: 49 Aug 21 15:33:08 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 39 Aug 21 15:33:18 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 43 Aug 22 13:40:41 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 45 Aug 22 13:40:52 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 1064 Aug 22 16:57:15 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 41 Aug 22 16:57:25 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 43 Aug 23 11:02:36 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 43 Aug 23 11:03:49 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now UNREACHABLE! Time: 37 Aug 23 11:08:19 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now REACHABLE! Time: 39 Aug 23 11:08:26 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 39 Aug 23 11:16:31 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 46 Aug 23 12:10:25 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 1246 Aug 23 14:01:37 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 45 Aug 23 14:04:27 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 258 Aug 23 15:28:36 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 44 Aug 23 15:28:37 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now UNREACHABLE! Time: 39 Aug 23 15:31:27 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 41 Aug 23 15:31:27 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now REACHABLE! Time: 56 Aug 23 15:40:31 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now UNREACHABLE! Time: 40 Aug 23 15:43:31 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 43 Aug 23 15:49:42 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 42 Aug 23 15:57:22 NOTICE[13944] chan_iax2.c: Peer 'voicepulse02' is now REACHABLE! Time: 40 Aug 23 16:37:49 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 44 Aug 23 16:38:00 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 48 Aug 23 17:31:07 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 43 Aug 23 17:32:17 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now REACHABLE! Time: 42 Aug 23 17:34:21 NOTICE[13944] chan_iax2.c: Peer 'voicepulse01' is now UNREACHABLE! Time: 42
Re: [asterisk-users] considering purchasing a t1 card, any recommendations?
Wow, this is a completely neutral and very valuable review. Thanks a lot Zoa. I´m an * newbie; my little box will only needs 20 extensions to give termination to remote users and I´m about to buy a PRI interface; I decide to get Sangoma hardware.. a lot of people recommended it to me. In you review you said: The biggest choice you need to make is if you want onboard echo cancellation or not. How can I really know if I will need echo cancellation? I´m planning to get a single span card (which doesn´t include echo cancellation) but how can I know if I really need this feature?? Thanks in advance. R.R. Libera Zoa escribió: I think the recent Digium and Sangoma cards are quite similar. (and about the same price) I didn't try sangoma so far, never had any issues with the digium cards, I have no clue how the digium helpdesk is, i never needed to call them. (well not really correct i did call them once, years ago for a firmware problem with their first te410p revision, causing a crash once every few months they had the distributor send me replacement cards right away, before i returned the old ones, so that i could swap them without having to shut down the server for a week). Configuration and installation for the cards is pretty straightforward, all you need to do is compile the kernel modules for your kernel. I personally installed at least 20 digium pri cards, all on different hardware without problems related to the digium hardware. (sometimes i did have bad cables, bad pri's, oh and my embedded pc didn't provide enough power for FXO ports). You will probably find more people on the list with problems with digium than people with problems with sangoma. This might be because a lot more people seem to use the digium cards with asterisk than sangoma cards with asterisk. (Based on the people i speak to, i'd guess 1 to 5% use sangoma?). The biggest choice you need to make is if you want onboard echo cancellation or not, you might not need it and if you want it its going to cost you a lot more than without. (both for sangoma and digium hardware). - They both seem to use exactly the same Octasic echo cancellation module. If you need on board echo cancellation but don't need 4 ports, digium is the only choice with their 2 port card with Octasic echo cancellation module. (Afaik sangoma doesn't have such a 2 port board with on board E.C. but i could be wrong.) Btw, there are more options, dialogic has compatible cards and so does eicon. (you will need deeper pockets though, the eicon retails at +/- 12000 euro for a quad span i think - people who buy these for asterisk usually do so for hardware faxing or interconnection to different carriers at the same time.) Some people prefer digium over sangoma because they sponsor the asterisk development that way. I'm not one of them, i buy digium cards (or tell my customers to buy them) because i'm happy with their product. Dislaimer: I know some of the people within Digium quite well, so maybe i get exceptional support or they ship me handpicked gold plated, overclocked versions of their cards (not really since i just buy them from a reseller). Cheers, Zoa. Dovid B wrote: Can I now 5th it ? All this makes me wonder why Digium dosent work harder. I have mainly only seen others praise Sangoma over Digium. - Original Message - *From:* Tom Vile mailto:[EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com *Sent:* Wednesday, October 18, 2006 4:22 PM *Subject:* Re: [asterisk-users] considering purchasing a t1 card,any recommendations? I 4th it. On 10/18/06, *Matthew Thompson* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On 17 Oct 2006, at 22:09, Richard wrote: I would have to second the Sangoma buy. Their tech support is second to none and more then helpful. I've never had any problems with their products that wasn't my own fault. Thirded - I've just done another install with a Sangoma A102 - the setup guides you through all the way and takes no more than 30 minutes (Including recompiling zaptel, which it does for you) [EMAIL PROTECTED] :o) -- Matthew Thompson [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com http://www.baldwintechsolutions.com Phone: 518-631-2855
Re: [asterisk-users] considering purchasing a t1 card, any recommendations?
Thanks Steve, it was helpful to read your post. Neither Digium or Sangoma single span cards have built in E/C, I´m wrong? To get one with this feature enable is completely out of my budget; since it is no for commercial use. I hope I wont have echo problems or they can be solve by means of software parameters. Thanks a lot. R.R. Libera Steve Davies escribió: On 10/20/06, R.R. Libera [EMAIL PROTECTED] wrote: Wow, this is a completely neutral and very valuable review. Thanks a lot Zoa. I´m an * newbie; my little box will only needs 20 extensions to give termination to remote users and I´m about to buy a PRI interface; I decide to get Sangoma hardware.. a lot of people recommended it to me. In you review you said: The biggest choice you need to make is if you want onboard echo cancellation or not. How can I really know if I will need echo cancellation? I´m planning to get a single span card (which doesn´t include echo cancellation) but how can I know if I really need this feature?? (Hopefully this does not oversimplify) Fundamentally, since Digium updated the Hardware E/C on their recent boards, there is little difference between the PRI-side electronics of the Sangoma or the Digium cards (at least, not which would worry the end user) The remaining H/W difference is the PCI interface hardware. Sangoma have far more hardware development resource, to their PCI and now their PCI-e interfaces are more compatible with variations in motherboards. Digium are a software company, so the software support for their cards will be easier to get going as it is built in - On the other hand, Sangoma's build environment has come a long way, and their support IS good. As far as do I need echo cancellation is concerned. I would say that if you can afford it, and it is available, then buy it. It generally saves a lot of heartache in the longrun, regardless of the manufacturer. As you say, the Sangoma single-port cards do not have EC, so that decides the issue :) When using software EC, the MG2 echo canceller in the 1.2.x release is good. We have almost no problems on PRI circuits. I am not as confident of the 1.4.x version of the EC, but time will tell I imagine. Hope that helps. Steve ___ --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] DID is not working (call is not routing)
Hello Chandra, What about Teliax´s service? Is it recommended? How´s their call quality? Thanks in advance... On 10/10/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi William,My DID is working and am receiving calls. The problem is with Teliax settings from their end. Thank you for spending your valuable time for me. Regards,Chandra.William Piper [EMAIL PROTECTED] wrote: Your server seems to be doing exactly what you are telling it to do: -- Executing Playback(SIP/216.89.79.2-09e1d020, ss-noservice) in new stack-- Playing 'ss-noservice' (language 'en') Read the extensions.conf directions on the wiki site: http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf bp On 10/8/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi,I have created SIP extenstions and created Teliax Trunk using IAX2. I am making outgoing calls to USA successfully. When I am making a call to my DID number from outside, its telling that The number you have dialed is not inservice. Here I am giving the output from Asterisk server console: *CLI -- IAX2/teliax-2 answered SIP/350-09e3b540 -- Executing GotoIf(SIP/216.89.79.2 -09e1d020, 0?from-trunk||1) in new stack -- Executing Set(SIP/216.89.79.2-09e1d020, TIMEOUT(absolute)=15) in new stack -- Channel will hangup at 2006-10-06 11:27:55 UTC. -- Executing Answer(SIP/216.89.79.2-09e1d020, ) in new stack -- Executing Wait(SIP/216.89.79.2-09e1d020, 2) in new stack -- Executing Playback(SIP/216.89.79.2-09e1d020, ss-noservice) in new stack -- Playing 'ss-noservice' (language 'en') -- Executing Congestion(SIP/216.89.79.2-09e1d020, ) in new stack == Spawn extension (from-sip-external, s, 6) exited non-zero on 'SIP/216.89.79.2-09e1d020' -- Executing NoOp(SIP/216.89.79.2-09e1d020, Hangup) in new stack -- Executing Set(SIP/216.89.79.2-09e1d020, DID=s) in new stack -- Executing Goto(SIP/216.89.79.2-09e1d020, s|1) in new stack -- Goto (from-sip-external,s,1) -- Executing GotoIf(SIP/216.89.79.2-09e1d020, 0?from-trunk|s|1) in new stack -- Executing Set(SIP/216.89.79.2-09e1d020, TIMEOUT(absolute)=15) in new stack -- Channel will hangup at 2006-10-06 11:28:04 UTC. -- Executing Answer(SIP/216.89.79.2-09e1d020, ) in new stack == Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/216.89.79.2-09e1d020' When I am calling from outside phone, call is coming to my server and is not routing. I am making calls to USA and between SIP extensions successfully. Please tell me the solution. Looking forward to your response. Thank you. Regards,Chandra. Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Stay in the know. Pulse on the new Yahoo.com. Check it out. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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
[asterisk-users] Asterisk + E1 with MFC/R2 in Argentina?
Hello, Has somebody installed this configuration: Asterisk + E1 with MFC/R2 (Telefónica Argentina) in Argentina before? I need to know if it´s possible with MFC/R2 argentine variation. Thanks in advance. R.R. Libera ___ --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] Echo Cancel Cards
I´m about to acquire an E1 interface. I was reading about TE110P and hardware incompatibilities issues with some boards, servers and chipsets. I also read a lot of compliments about Sangoma Hardware (specially for E1/T1 interfaces) and I was wondering if A101 from Sagoma is a better choice (technically speaking) than Digium TE110P. I read now, on this post, an opinion about Sangoma interfaces and echo cancellation issues.. I have a PC with Asus P5LD2 board (Intel 945P chipset). I asked Digium support for how compatible is the TE110P with my box.. and they said that no incompatibility issues had been reported with the chipset I use.. BUT, they had no test TE110P with this chipset... I´m not a Sangoma or Digium fan... I´m just a newbie who don´t want to get the wrong piece of hardware. I really appreciate any advice from people with a lot of experience and skills on this topic. Thanks in advance R.R. Libera Dovid B escribió: I have never used T1 cards but as far as POTS line cards I would say that I like sangoma better. It is a little bit harder to set up but works wonders. - Original Message - From: Thomas Kenyon [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 10, 2006 12:31 PM Subject: Re: [asterisk-users] Echo Cancel Cards Joseph wrote: On Mon, 2006-10-09 at 20:41 -0400, Forrest Beck wrote: Anyone using the echo cancelation cards from digium? We are using the single span T1 card with out echo cancel and I was curious if it was worth the money. I'm running Asterisk 1.0.11 with few Sipura 3000/2000 units and have no echo whatsoever. I just tried new Asterisk 1.2.12.1 and the first thing I've noticed was terrible echo, not to mentioned that it keep crashing constantly to a point this that is not possible to use it. I've got an SPA-3000 at home that is constantly crashing, echoey and is almost unusable. (The CS4660-based ATA and PA1688-based handsets have otherwise been fine, as were the the Cisco 468 and Linysys PAP2 when they were in use). ___ --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
[asterisk-users] USA Origination recommended service?
Hello, I need an advice about USA Origination (only). Who you recommend to me for a production environment like a customer care support? I need: 1) Excellent call quality. 2) Stability. 3) Excellent support. What will be the best increment schema? 60/15, 60/6, 6/6 ?? Thanks in advance. R. R. Libera. ___ --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] Intel Chipset 945p compatible?
Hello, I had recently install an Asterisk PBX into a brand new PC: Intel Pentium D 3.4GHz Dual-Core + P5LD2 motherboard + SATA HDD. I´m planning to handle one E1 with a TE110P interface and I want to know the compatibility between TE110P and Intel 945P chipset. I already buy the hardware and the only thing I got into account was the compatibility between the hardware selected and Debian Sarge (the distro I selected). I´ll accept any suggestion, advice or comment. Thanks in advance. R.R. Libera ___ --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] Need USA DID + trunk provider
Hello, I need an USA DID + 15 b-channels. The only option I already have is OpenVox and I want to see some alternatives. Sound quality is my priority. Thanks in advance. R.R Libera ___ --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] Need USA DID + trunk provider
Sorry, when I said OpenVox I should say VoxBone. Regards, ___ --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] Digium TDM or SPA-3000?
Hello, Which analog hardware is the best in quality-of-voice terms? In a production environment, which one gives more reliability and stability?? Thanks in advance, R.R. Libera ___ --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] Digium Interfaces in Tampa?
Where can I buy Digium Hardware in Tampa? Not Ebay or e-commerce. Thanks a lot. R.R. Libera ___ --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
