This is Kaleb, I have ABSOLUTELY no ties whatsoever with any VOIP service or product, I am just an end-user. I currently use sixTel, I love it. I have tried others and had very bad experiences, sellvoip.net being the absolute worst from my experience. I haven't ever tried any of the "unlimited" services like broadvoice, I only have tried the pay-as-you-go providers that support the IAX protocol. sixTel has been really good to me, they are a tad slow on email support, but of you catch the through MSN messenger ([EMAIL PROTECTED]) they usually are pretty good. I haven't dealt with their service department for about over a month, I have had an account with them for 2 months and had only one problem with inbound calls, their DIDs stopped working for a small spell due to a problem with their carrier, beyond their control (at least that is what they told me). When configuring their service it was 100% required that in iax.conf the context was "sixTel" with the capital T or inbound wouldn't work, can't say that I have ever noticed a problem with outbound. I will send a copy of my sterilized sixTel config to anyone that would like.
No Dan, I do not have a "tie" with them, I am just a happy customer. I especially like the fact that they email me automatically if they aren't able to reach my server when someone calls. (has happened a time or two, was MY fault). They also allow you to set up a fail-back number (can be pstn or cellular or whatever) if your server is unreachable by them. They win my business; If you don't like them, that is your call. Kaleb > I guess Kaleb has a tie up with them.. > > > >I could never get my sixtel try call out except once. They customer service >sucks as usual.... > > > >I regret paying to them...Their sample config are all done from my side my >my calls comes out with 'NO ANSWER" > > > >Dan > > > >On 25/01/06, Dovid Bender <[EMAIL PROTECTED]> wrote: > >Let me guess you have no affiliation with them what so >ever ? no commision on accounts either ? >--- "Kaleb L. Kunzler" ><[EMAIL PROTECTED]> wrote: > > I use iax.cc and find their service to be superior > to ANY other VOIP > provider I have tried. Their prices are > competitive, My calls always go > through, I always get my calls, I couldn't think of > a better provider. They > are "picky" with the context that you use in your > IAX.cc, but as long as you > use the sample config that they provide it works > beautifully. You only need > $5 to open the account, that really isn't bad at > all, others like > sellvoip.net <http://sellvoip.net> (bad) require $25 to open an account. > If you need help getting > their service to work for you, please contact me off > list. > > > >On 1/25/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> > wrote: > > > > You can try with www,iax.cc too but i guesst not > luck with a test > > account.. > > > > Dan > > > > > > On 25/01/06, Nilesh Londhe <[EMAIL PROTECTED]> > wrote: > > > > > > I use www.voipjet.com and find it OK. > > > > > > On 1/24/06, Roberto Pereyra < > [EMAIL PROTECTED]> wrote: > > > > Hi > > > > > > > > > I looking a good IAX service for a emerging > voip provider. > > > > > > > > Better with a test account to try. > > > > > > > > Thanks in advance. > > > > > > > > roberto > > _______________________________________________ > --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 > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060126/d65904 6f/attachment-0001.htm ------------------------------ Message: 10 Date: Thu, 26 Jan 2006 14:55:22 +0000 From: bails <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] TDM400 pinout To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=us-ascii; format=flowed BJ Weschke wrote: > On 1/26/06, bails <[EMAIL PROTECTED]> wrote: > >>Chris Bagnall wrote: >> >>>>Hi I'm looking for a pinout for the above. Note this has >>>>what i'd call >>>>RJ45 sockets (or someone smart can correct me). I need to >>>>plug into BT (rj13?). >>> >>> >>>Are you sure the TDM400 has RJ45 sockets? The pair I've got here have RJ12 >>>sockets. >>> >>>I assume with the mention of BT, you're in the UK. The line is on pins 2+5 >>>of the BT connector, which'd usually translate to the 2 inner pins of an >>>RJ11 connector (pins 2+3). You should find an old modem cable will do the >>>job fine. >>> >>>If your TDM400 really does have RJ45 sockets, then you'd expect the line to >>>be on the middle pins (pins 4+5), similar to a modtap used in structured >>>cabling environments. >>> >>>Regards, >>> >>>Chris >> >>Thanks, yes they are rj45, we have had rj12 in he past I look at the above. >> >>Like I said though, pity Digium dont supply the information on there >>site or with the cards, its a bit like everything in life today. We are >>only the customer, but we're expected to do the running around. >> > > > Earlier versions of the TDM400 I believe were RJ45. They were changed > to RJ11 I think I had heard at one point for compliance with some > telco standards outside the US. But, in either case, yes, the middle > pair is the "active" pair for your FXO/FXS ports on these cards > whether RJ11 or RJ45. > > -- > Bird's The Word Technologies, Inc. > http://www.btwtech.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 > > Thanks I can confirm that this is indeed correct Bails ------------------------------ Message: 11 Date: Thu, 26 Jan 2006 07:56:16 -0700 From: Colin Anderson <[EMAIL PROTECTED]> Subject: RE: [Asterisk-Users] Bootable CD? To: 'Asterisk Users Mailing List - Non-Commercial Discussion' <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1" To clarify: You have to write it as a DISK IMAGE. If you simply drag the ISO file to your Nero project and write it, you will get a CD with a single file on it - the ISO image - and not the CONTENTS of the ISO Image. 1. Run Nero 2. In the New Compilation dialog click Cancel 3. Click File > Burn Image, select" All Files" under Files of Type and pick your ISO. 4. Click OK, then click Write hth -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Thursday, January 26, 2006 1:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Bootable CD? yup... its a bootable image.. go ahead and just write it directly... Dan On 26/01/06, Sohail Arham < [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> > wrote: ahan...then it mean it doesnt need to uncompress it..juss write on cd by nero burning software...?? _______________________________________________ --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> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060126/d19c42 eb/attachment-0001.htm ------------------------------ Message: 12 Date: Thu, 26 Jan 2006 09:58:07 -0500 From: Bill Michaelson <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] * point to point t1 solution? / alternatives To: [email protected] Cc: [EMAIL PROTECTED] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1; format=flowed This has been an interesting discussion for me (except for the sniping). The last post led me, out of curiosity, to this wiki entry: http://www.voip-info.org/wiki-Asterisk+TDMoE I was unaware of this feature, and it looks pretty good. I've been pondering replacing some T1's by leveraging IP capacity but of course have run up against the QoS issue. My idea was different... I don't have production experience with precisely this type of application, but I ask for validation from this list. Pardon me for stating what is undoubtedly obvious to many... The key to assuring adequate performance in replacing a TDM link with IP is to assure that adequate idle time is reserved for voice on the IP segment(s) involved in the route. In this way, latency can be stabilized, and if maintained below a certain (arbitrary) threshold, performance can be deemed acceptable. The first step, of course, is to assure that the virtual TDM allocation does not exceed the available IP bandwidth (so leave a margin, which is huge in the example given). The next step is to use routers which respect the TOS field (however it is used; diffserv/whatever), and finally, to assure that no non-VoIP traffic can be injected into the path with higher routing priority. On a point-to-point link, a pair of typical Linux boxes can do all this. Given the original problem, I would place Asterisk boxes at either end of the link, and have them blend the ordinary traffic with the VoIP traffic (which would probably use IAX to relay calls between the T1s), while assuring (enforcing) that VoIP packets are marked as highest priority. There are varied ways of accomplishing this, and a good reference which I've used in the past can be found at: http://www.lartc.org/lartc.html Additionally, I think one could use the tunneling techniques described in that guide to encapsulate the non-VoIP traffic such that its packets' originally marked TOS values are preserved for transit outside the segment used for TDM emulation. In this way, that part of the segment bandwidth not required for VoIP would function as a dedicated link, allowing other prioritization of traffic such as interactive vs. bulk (or even other voice!), with the added advantage that it could use the reserved VoIP bandwidth when it is otherwise not required (albeit in the case of a T-1 over 10Mb, that's insignificant). Is this easier or harder than TDMoE as described? Does the TDMoE shared idle bandwidth? What about stability (I'm thinking of SW releases)? What other drawbacks or advantages are there? >Date: Wed, 25 Jan 2006 23:53:59 -0700 >From: "Damon Estep" <[EMAIL PROTECTED]> >Subject: [Asterisk-Users] * point to point t1 solution? >To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[email protected]> > >Can anyone point me to a reference or sample config for bypassing a >nailed up (point to point) t1 between two PBXs with asterisk and a pair >of t1 cards? > >Right now I have 2 Nortel norstars connected to each other via a leased >line t1. I also have a solid 10mbps low latency microwave link between >the 2 sites. > >My goal is to run an asterisk box at each end with a t1 card and >Ethernet card to act as a TDM<>SIP gateway to bypass the nailed T1 in a >relatively dumb configuration, with the goal of migrating off of the >norstars eventually. > > In past situations I would have done this with a pair of Cisco routers >with T1 interfaces in them but in this case I want to get asterisk into >the picture as an eventual replacement for the norstars. > > > > ------------------------------ Message: 13 Date: Thu, 26 Jan 2006 10:08:44 -0500 From: Bill Michaelson <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] * point to point t1 solution? To: [email protected] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1; format=flowed You've clarified your requirements for me. Please indulge me - I really want to understand - what are the application implications of this? In other words, what system behavioral changes will your users experience in the various scenarios (pure circuit emulation vs. relay via IAX or similar)? Date: Thu, 26 Jan 2006 07:00:02 -0700 From: "Damon Estep" <[EMAIL PROTECTED]> Subject: RE: [Asterisk-Users] * point to point t1 solution? To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1" Lets put the TDMoE aside for a minute... The same trunking could be achieved with SIP or IAX, could it not (with higher latency)? The rest of the question remains - is there a way to get asterisk to output, bit for bit, on a t1 interface, the same data that is input on a remote asterisk box t1 interface - using any trunking protocol. This is what would be required to truly emulate a "signaling un-aware" point to point t1 like one that you would get from a telco if you ordered a point to point esf/b8zs t1 from A location to Z location. Pure circuit emulation - not ISDN/CAS/E&M signaled voice. Does that clarify the question at all? ------------------------------ Message: 14 Date: Thu, 26 Jan 2006 09:15:47 -0600 From: "Ross C" <[EMAIL PROTECTED]> Subject: RE: [Asterisk-Users] * point to point t1 solution? To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1" Uhhhhhh..maybe you should ask Jean-Michel for a refund. Wait, you havent paid a dime for this. Or Asterisk. Or most of the Asterisk add-ons. I always see people getting mad at other people for bad advice or bad answers to their questions; people seem to forget that all this stuff is FREE. If Jean-Michels advice isnt what youre looking for, say Thanks for the info, but Id really like to know.. (geez, I feel like someones mom). Hes taken time out of HIS day to try to help YOU for FREE. If a high level of support and definitive answers are a must for your situation, pay someone with experience, or see the following: <expensive IP telephony> http://www.cisco.com <http://www.cisco.com/> http://www.nortel.com <http://www.nortel.com/> http://www.inter-tel.com <http://www.inter-tel.com/> http://www.avaya.com <http://www.avaya.com/> http://www.3com.com <http://www.3com.com/> </expensive IP telephony> _____ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Thursday, January 26, 2006 3:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] * point to point t1 solution? Actually, it is a quite appropriate response to ANYONE that includes this type of comment in their reply You probably need a couple of T1 cards, and some paid consulting to get it working (I've never done it myself but that's how I would do it if I was in a hurry) Perhaps something like this would have been better received; I know it can (or cannot) be done, and here is the name of someone that might be willing to help you for a fee Look back though the archives and you will see that I have had some participation here myself in the past D _____ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Simon Woodhead Sent: Thursday, January 26, 2006 2:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] * point to point t1 solution? Bad day Damon? I think your comments are a little harsh towards someone who is an active and informed contributor to the list. Jean-Michel could have ignored you but he chose to share what he could. Maybe someone else will have the complete answer to your question. On 1/26/06, Damon Estep <[EMAIL PROTECTED]> wrote: Jean-Michel, You missed the entire point - the question is IS ASTERISK CAPABLE OF EMULATING A POINT TO POINT T1 BETWEEN 2 BOXES, AND IF SO ARE THERE ANY WEB BASED HINTS I MIGHT LOOK AT? Not WILL YOU DO IT FOR ME? Your response to this post was un-informative and quite frankly it is the type of useless response that most mailing lists and newsgroups could do without. Damon > -----Original Message----- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> ] On Behalf Of Jean-Michel Hiver > Sent: Thursday, January 26, 2006 1:36 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] * point to point t1 solution? > > Damon Estep a icrit : > > > Can anyone point me to a reference or sample config for bypassing a > > nailed up (point to point) t1 between two PBXs with asterisk and a > > pair of t1 cards? > > > > > > > > Right now I have 2 Nortel norstars connected to each other via a > > leased line t1. I also have a solid 10mbps low latency microwave link > > between the 2 sites. > > > You probably need a couple of T1 cards, and some paid consulting to get > it working (I've never done it myself but that's how I would do it if I > was in a hurry) > > > > My goal is to run an asterisk box at each end with a t1 card and > > Ethernet card to act as a TDM<>SIP gateway to bypass the nailed T1 in > > a relatively dumb configuration, with the goal of migrating off of the > > norstars eventually. > > > If it's a point to point Asterisk <-> Asterisk configuration, why use > SIP rather than IAX? IAX configuration is very easy, so once you get the > norstar <-> asterisk link up it'll be a piece of cake. > > Cheers, > Jean-Michel. > > -- > Jean-Michel Hiver - http://ykoz.net/ > Dicouvrez la Riunion des Technologies IP & Telecom > TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE > > > _______________________________________________ > --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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060126/d0dddc 65/attachment-0001.htm ------------------------------ Message: 15 Date: Thu, 26 Jan 2006 15:16:25 -0000 From: "Steve Langstaff" <[EMAIL PROTECTED]> Subject: RE: [Asterisk-Users] * point to point t1 solution? / alternatives To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1" Remember, however that TDMoE is TDMoE, not TDMoIP - it's not routable (unless you encapsulate it somehow, I guess). -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bill Michaelson Sent: 26 January 2006 14:58 To: [email protected] Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] * point to point t1 solution? / alternatives This has been an interesting discussion for me (except for the sniping). The last post led me, out of curiosity, to this wiki entry: http://www.voip-info.org/wiki-Asterisk+TDMoE I was unaware of this feature, and it looks pretty good. I've been pondering replacing some T1's by leveraging IP capacity but of course have run up against the QoS issue. My idea was different... I don't have production experience with precisely this type of application, but I ask for validation from this list. Pardon me for stating what is undoubtedly obvious to many... The key to assuring adequate performance in replacing a TDM link with IP is to assure that adequate idle time is reserved for voice on the IP segment(s) involved in the route. In this way, latency can be stabilized, and if maintained below a certain (arbitrary) threshold, performance can be deemed acceptable. The first step, of course, is to assure that the virtual TDM allocation does not exceed the available IP bandwidth (so leave a margin, which is huge in the example given). The next step is to use routers which respect the TOS field (however it is used; diffserv/whatever), and finally, to assure that no non-VoIP traffic can be injected into the path with higher routing priority. On a point-to-point link, a pair of typical Linux boxes can do all this. Given the original problem, I would place Asterisk boxes at either end of the link, and have them blend the ordinary traffic with the VoIP traffic (which would probably use IAX to relay calls between the T1s), while assuring (enforcing) that VoIP packets are marked as highest priority. There are varied ways of accomplishing this, and a good reference which I've used in the past can be found at: http://www.lartc.org/lartc.html Additionally, I think one could use the tunneling techniques described in that guide to encapsulate the non-VoIP traffic such that its packets' originally marked TOS values are preserved for transit outside the segment used for TDM emulation. In this way, that part of the segment bandwidth not required for VoIP would function as a dedicated link, allowing other prioritization of traffic such as interactive vs. bulk (or even other voice!), with the added advantage that it could use the reserved VoIP bandwidth when it is otherwise not required (albeit in the case of a T-1 over 10Mb, that's insignificant). Is this easier or harder than TDMoE as described? Does the TDMoE shared idle bandwidth? What about stability (I'm thinking of SW releases)? What other drawbacks or advantages are there? >Date: Wed, 25 Jan 2006 23:53:59 -0700 >From: "Damon Estep" <[EMAIL PROTECTED]> >Subject: [Asterisk-Users] * point to point t1 solution? >To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[email protected]> > >Can anyone point me to a reference or sample config for bypassing a >nailed up (point to point) t1 between two PBXs with asterisk and a pair >of t1 cards? > >Right now I have 2 Nortel norstars connected to each other via a leased >line t1. I also have a solid 10mbps low latency microwave link between >the 2 sites. > >My goal is to run an asterisk box at each end with a t1 card and >Ethernet card to act as a TDM<>SIP gateway to bypass the nailed T1 in a >relatively dumb configuration, with the goal of migrating off of the >norstars eventually. > > In past situations I would have done this with a pair of Cisco routers >with T1 interfaces in them but in this case I want to get asterisk into >the picture as an eventual replacement for the norstars. > > > > _______________________________________________ --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 End of Asterisk-Users Digest, Vol 18, Issue 167 *********************************************** _______________________________________________ --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
