similar to: asterisk-providers mailing list?

Hi, Inspired by a recent rant about one particular provider, I am getting very curious about something I've never mastered. I'd like someone to explain this here or at least post a link or two that can educate me and probably countless others who have no knowledge in this area. I'm sure there are several of you reading this that know all about the subject. What are the various

Hello Everyone, We are currently having talks with various service providers, and trying to determine what the best way is to interconnect in order to have access to the PSTN network. As you know there are two ways of doing this: Traditional PRI: Have trunks grouped into a transport layer such as OC3/12. With DIDs attached to the group. As you many know, this approach would also require a POP

Hi all, With the recent 's' and 'n' priorities, as well as the advantage of labels, dialplan management has become *much* simpler IMHO. However, I have one suggestion for possible improvement. In any of the Goto[If|IfTime] statements, the ability to do 's' + a number or label + a number would be _nice_. Example extensions.conf: exten => 1,1,NoOp(Start) exten

If you are a Canadian VoIP provider or CLEC, help make a difference by joining the CAVP: www.cavp.ca >OMFG, >I thought April Fools day was over. This is hard to believe. If true it >tells me that Bell has not changed at all. They are still trying to >manipulate and take advantage of the parts of the market they have absolute >control over. The CRTC was right to continue to

I've got the XTEN Lite soft phone mostly working with * but it's dropping out like a very bad cell phone call. The GSM codec is worst (unusable), G711u and G711a are best but not good enough to use. I don't think it's a lack of bandwidth. What tuning options or approaches should I be investigating to make this work. Also, what's the best soft phone(s) for Windows XP?

OK.. I'm up to my eyes in LD BS! I can't for the life of me understand how any carrier, either VoIP or traditional service provider can make heads or tails of how to hand off an * based call to an LD provider. Every provider I talk to, says I have to have a traditional T1 put in to their respective networks. I don't want to do this. I want a LD provider that can take a IP, SIP, IAX

Hi All, I am looking for a provider/s of inbound DID - IAX numbers, for UK, USA, and Australia. Preferably free or low cost :-) Can anyone make a good reference? Many thanks Chris PS: I appreciate this is perhaps a little OT, please feel free to reply off list. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Greetings folks, I'm currently dealing with a company to let me set Caller-ID-Name on outbound calls. So far pretty happy with their services. The basic service works like this: * CLEC sets Point Code to point to this company * CLEC has to sign LOA saying they give me permission to set the Caller-ID-Name through this company. * I go into web interface and set name. However, the CLEC is

Using a SIP connection with a CLEC, the downstream (received) audio is perfect when the mute button is activated on the phone. However, when there is upstream audio (i.e., talking or even breathing into the microphone), the downstream audio is cut off. It's kinda like having a half-duplex audio connection. When I divert outgoing calls to another provider, these calls are fine.

Nah, this is the first time I've posted here in years. Gennaly things just tick over with the application moving along in a nice straightforward manner, improvment upon improvement. Rsync is one of the mose important and usefu apps available for Linux and OS X (powered by APPLE!!!!); followed by ffmpeg and vlc (and Apache, I guess, but I don't do much of that anymore). The docs are very

I receieved my first PAP2-NA yesterday from our distributor(Tech Data). It installed pretty easily and has worked great so I went to order some more of these units today. When I logged into Tech Data this morning, the PAP2-NA was now marked as discontinued and no longer available and only the PAP2 version was available which is the Vonage branded version. :( I saw someone on the list say that

Pardon my ignorance, but I was hoping someone could clear up something for me. - For a few POTS lines, digium has a single port card for that, or a T1 card to a channel bank. - For 10 or more lines, digium has a T1 or E1 card for that too based on PRI channels - For 100's to 1000's of lines, I suspect a soft-switch is in order??? A traditional phone company will sell: - POTS lines for

I would rather the maintainers would spend their time improving the application, and not worrying aobut the OP's hurt little open-sourced religious feelings. I use rsync to transfer pretty massive (wait for it) ADOBE PHOTOSHOP files around my LAN. Neener neener fucken neener. On Wed, Jul 27, 2016 at 11:37 AM, Karl O. Pinc <kop at meme.com> wrote: > On Wed, 27 Jul 2016 14:56:07

Checking the earlier mails, it stated that the phones were $75 (100) & $85 (102) ref :- http://lists.digium.com/pipermail/asterisk-users/2003-June/013483.html Well, I just called Ovislink/dgtimes and was quoted $90 & $100 and the person said there was no price change. Anyone on this list actually bought them at the $75 & $85 rate ??? Regards...Martin -- Too much is just enough.

I have the following configuration: CLEC -> T-1 -> Asterisk -> Adtran Channel Bank -> (analog) -> Nortel Don't complain that it's ugly. I've already done plenty of that. The CLEC manages their Adtran remotely and needs to be able to continue to do so. I assume they use FDL to do the management. We are using the Zaptel DACS/DACSRBS to cross connect some of the

Arrrrr, I think I've found a bug in the behavior of the stats::cor function when NAs are present, but in case I'm missing something, could you look over this example and let me know what you think: > a = c(1,3,NA,1,2) > b = c(1,2,1,1,4) > cor(a,b,method="spearman", use="complete.obs") [1] 0.8164966 > cor(a,b,method="spearman",

I'm in the process of turning up a PRI in one of my markets and have run into a problem I have never seen before. I am unable to place a local outgoing call. Long Distance over the same PRI works fine. When I attempt to place a local call using the PRI I see Asterisk attempt to dial, and am greeted with a busy signal. This signal appears to originate on the telco's switch. I have had

Hi Masters! Soy nuevo en esto y por favor necesito su ayuda.... Tengo 2 enlaces de internet, en un router linux, necesito saber como se puede hacer que si se cae un enlace, automaticamente con ip route configure el Gateway y la ip del otro enlace para que los usuarios salgan a internet. En si es hacer una contingencia de los dos proveedores cosa que si cae un enlace levante el otro y que despues

Hauptman-Woodward Medical Research Institute (Buffalo,NY) is an independent, non-profit research facility that is committed to improving human health through biomedical research. In addition to that, it's a really cool place to work. HWI is a casual, low stress environment. Each of us is very busy with our work, but it is work that we enjoy. Even though our retirement plans are very good,

[crossposted to isp-clec and asterisk-users] As part of several larger projects, the question of rate importation from termination carriers has come up. If a firm has four different LD partners, as an example, then it is obvious that the firm needs to determine at the origination of a call where that call would be best sent on a cost basis. I'll ignore things like quality of service and