similar to: Enterprise quality SIP provider

BJ, >BJ Weschke <bweschke@gmail.com> >Subject: Re: [Asterisk-Users] Do Both! :) Re: Telecom >SIP termination vs. DS3 >To: Asterisk Users Mailing List - Non-Commercial >Discussion <asterisk-users@lists.digium.com> >Message-ID: <79cf63305051908056c284cc9@mail.gmail.com> >Content-Type: text/plain; charset=ISO-8859-1 >Did I miss pricing/availability

Thanks a lot for help in advance. I am switching from matlab to R and I guess I need some time to get rolling. I was wondering why this code : > fit.0 <- lm( Response ~ 1, data = ds3) > step(fit.0,scope=list(upper=~.,lower=~1),data=ds3) Start: AIC= -32.66 Response ~ 1 Call: lm(formula = Response ~ 1, data = ds3) Coefficients: (Intercept) 1.301 is not working

Greetings!! We're getting some strange results using call files.. no matter the technology, DAHDI, SIP, etc., we get a "Call failed to go through, reason (3) Remote end Ringing" message when attempting to originate a call from a call file. Numbers changed to protect the innocent.... using call file.... //------------CALL FILE------------// Channel: DAHDI/g1/918005551212

Message: 16 Date: Thu, 19 May 2005 00:16:34 -0600 Michael, Do both! As for Sip Termination: ----------------------- Contact Kristi Eggers @ Txlink.net for month to month Originating/Termination VoIP Toll Free or Local USA DID #s. Yes they do both Sip and IAX. You must have seperate accounts for either Sip or IAX and fund your account with a minimum of $100. This is what I did. Once I get

Hi All, We are trying to implement a DS3 capacity calls (672 concurrent calls) using asterisk server. I wanted to ask are there any compatible DS3 cards with asterisk? I tried searching a lot but could find DS3000P from digium but unable to get this product. Does anybody have any idea of having any DS3 card in asterisk box so as to handle around 600 calls? Thanks Sandesh -------------- next part

Hi, Not sure if this is a PBKAC or not:- root at ds3:/usr/share/postfixadmin# doveadm pw -s SHA512-CRYPT -p password {SHA512-CRYPT}$6$aUgGXP0UshkMj7hY$9JV4yMRsjIe/98CzmglYrMjf.9NJ.FXzxcLE9B0v3doCRUWo2wRncc6hg6VCs0DCUHQbeC/bRDZdGCge/nB/h/ root at ds3:/usr/share/postfixadmin# doveadm pw -t

Is anyone using a media gateway with a DS3 for TDM/PSTN access? I am trying to determine the best way to scale Asterisk beyond 4 PRI's. What about the TNT Max? Can it be configured with a DS3 PSTN interface, the appropriate software and hardware, and Ethernet ports to support SIP-PSTN/DS3 media gateway services for multiple * boxes? What about TNT - SER - * ? Is there anyone here with real

Hello all, We are looking for some hardware requirements/recommendations to be able to handle a full DS3's worth of TDM -> VoIP traffic. The DS3 would bring 24 calls per T1 x 28 T1s = 672 simultaneous calls. We would then need to convert those calls into G729 SIP VoIP calls to send to our asterisk box over ethernet. Since everything is going in/out of asterisk is 729, and no features

Do you remember when Digium annouced channelized DS3???...vapour. You watch, this CODEC board will go the way of the DS3 boards...vapour. Mark my words. --------------------------------- Yahoo! Messenger with Voice. PC-to-Phone calls for ridiculously low rates. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear R experts, Wish you all a HAPPY NEW YEAR! How do I go about plotting (using lattice) overlays of an ID (group=ID) observed, fitted data in each panel of a multiple panel plot (each panel identified by DS1 + DS2 + DS3)? "x" variable is X1 in the accompanying section of a dataset. each individual is identified by color and Y's are identified by "pch" or "lty".

Hello Everyone, We are looking to interconnect with a local ILEC over an OC-n transport layer. They basically gave us two options in terms of mapping the SONET to the DS3: * VT1.5s mapping * DS1s mapping The second option is quite clear. We would MUX the connection, and plug the lines into qaud t1 cads etc... The tech mentioned that with the second option we would also need a DACS to convert

Is anyone out there using Lucent brand equipment to handle an incomming DS3, converting all 672 calls to SIP (as G729) and sending those to Asterisk/SER over ethernet? If you are and are willing to speak to my boss about your experiences (over the phone) with it, please contact me off list. We have a possible contract with a local CLEC to handle their long distance, and they want to send to

Obviously, there are no DS3 TDM cards that are currently compatible with Zap channels. (or are there?) Does anyone know of an inexpensive DS3 card that could perhaps be used with Asterisk if one were to try to port the Zap drivers to such a card? PCI, of course, would be the bus of choice. I think there are quite a few discouraging comments to be made on that question. Firstly, most

I have a customer installation with an Adtran DS3 mux. The DS1's go into my Asterisk servers that run IVR/Call recorders. The DS3 provider is Qwest, and they tell me that they routinely drop the DS3 service to redundant back-up's and that this is a common practice that happens thousands of times to DS3 lines daily across the US without any service interruptions. They say that the

I have been using the inbound 800 services from vitelity. Slowly the usage has been rising and in the month of Jan the bill was for $650. I am currently on a 1.9 cents a minute plan. Am I paying too much ? Some suggestions my team generated to reduce the toll free incoming call bill were: 1. When people call in on the 800 number take the local number they are calling from and then call them back

I don't want to start the "How many calls can Asterisk handle?" discussion or "How many angels can stand on the point of a pin?" discussion either. But can anyone contribute some practical knowledge of systems that take in channel bank T1s or DS3s from "far away", and process the calls? I am looking for real world, been there, done that, or "check the

I called bandwidth.com to buy a sip line from them for $30 a month. But they said they will not sell me a sip line since the address on the account is Citrus Heights CA and they can not provide services in that area. On asking further the person clarified that there is no e911 service available in the 916 area code for bandwidth.com But other providers like www.broadvoice.com are able to provide

Hi all, I want to get results for cox proportional hazards on SNP data. I'm trying to get HRs, CI's, & p-values for each individual SNP - this is represented by cov[,i]. When I run my code, I get the following error: subscript out of bounds. I don't know why I am getting this error. I have looked through the R logs, but nothing that has been previously suggested has helped so

Follow this diagram: Many POTS lines <-> Many Channel Banks <-> Mux <-> DS3 <---cloud---> DS3 <-> demux into T1s <-> Many Asterisk's If someone calls into 1 of the asterisk boxes (via PRI or VoIP), and I send the call back down the line above to a POTS phone, who will provide caller id to the POTS line? I think its called FSK signaling. Anyone know

All, This may be a stupid question, but here it is... What interface gives the most density? Do I top out at T1's? For instance, 4 t1's to the Digium Quad span t1 card. Is there an interface available for T3 or DS3? Thanks, Chris