similar to: design check

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

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, I have configured ssh for port forwarding local 8080 to out company web proxy server. # ssh -L 8080:proxy:8080 myname at ssh-host I've tried the configuration by using # telnet localhost 8080 and it works fine. But whenever I try to use it with any http browser, it just fails. I investigated the problem with Ethereal and I've noticed that packets travel on loopback interface,

Hello, I've been trying to setup a Fax2Email mecanism on my Asterisk box. I have been using the following: 1) An incoming IAX line on Unlimitel (Im not even sure if it's worth mentionning the provider, but I do just in case) 2) NVBackGroundDetect from Newman Telecom 3) The following extension to test: exten => fax,1,SetVar(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif) exten

James, I'm a systems and telecom professional with experience going back more than thirty years, to the days of teletype, current loop, POTS (2600hz signalling anyone?) and echo cancellation via analog level control and hybrid balance. Your read of the intent of the VOIP/SIP design correctly. The intent WAS to make making connections between endpoints as easy as using a browser.

Hi All, I am seeking input from anyone who may have seen a similar configuration and dealt with similar issues to what I'm experiencing. Configuration: - 2 sites (site A and B) - Asterisk 1.2.23 on each site (Trixbox) - Internet 512/512 symmetric at each site, dedicated to VOIP calls only. - IAX trunk between the sites, with data travelling across the 512/512 Symmetric link - PSTN

Hi friends, Is their any possibility to setup our own Toll-Free Number in Asterisk using some PCI or FXO Card? I have one number from my local Telecom called 123XXXXXXXXXXXX and i would like to setup this number in my asterisk if some one called this number from his mobile or land line he should not be charged when the call will come i can route to my SIP or IAX in asterisk internally. In this

Hi all! I am looking for some software that would work as a proxy between a SIP device (SIP phones and ATA boxes) and the Asterisk system running IAX. The reason is that I can only get IAX trow the firewalls, and can't change the settings. One solution I am using are getting several Asterisk system communicate trow IAX bout in this case would I rater have every persons computer act as a proxy

Hi, I?ve been reading up on SSO-based logins for the last couple of weeks. I?ve found a lot of information about it, but nothing that matches my situation. Here?s the gist of my situation... - I have a Samba 3 PDC in our corporate office as well as three remote offices. - Each remote office is in a different physical building and connected to the Corporate office either via Point-to-Point T-1

Hi to ALL my name is Dimitri and im a CEO of startup Company in Italy focused on Internation call traffic i usualy use Asterisk (very good app :-) ) for switching call. I ask now to Asterisk User of Telecom Company if is possible to cooperate in creation of network International POP call Termination through Voip Tunnel from us. What we think about? Thanks to all Dimitri Bellini

Hey guys, I'm actively trying to get the "big" picture on how all this works and relates to each other. I've gone through some basic examples from the book and from the sample files just fine. Now, I've setup an account with a VOIP provider which does IAX termination (exgn.net) After getting an account and following their steps, I can make calls out using my IAX (cubix) and

On Tue, 2 Jun 1998, Luke Kenneth Casson Leighton wrote: > On Tue, 2 Jun 1998, Robert Vasvari wrote to samba-ntdom: > > > > > Hi All, > > > > I'm planning to run my own SMB server on some box, > > binding to a user level port (>1024). Problem is, > > WINDOWS (both Nt and 95) only connects to an SMB > > server on ports 137-139. So, the

http://www.smh.com.au/articles/2004/04/05/1081017104255.html SingTel ready to break into web telephony April 6, 2004 Singapore Telecommunications is teaming up with US internet phone start-up SIPphone to offer low cost, and in some cases free, phone services over the web. The deal, expected to be announced today, will allow SIPphone - started by MP3.com founder

Well I know from personal experience that NuFone is working on a solution for its customers as fast as it can. I know they found an alternate termination provider and are working to have a solution for the TF and Local DID's he currently has on his platform. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of

Put your love of puppet to work with us at Speakeasy. We need Engineers who want to use puppet in conjunction with re-architecting our infrastructure. We are based in downtown Seattle, overlooking Elliot Bay. Please send your resume or any questions directly to me and off-list. Thanks, -g Company summary: Speakeasy, one of the nation’s leading broadband voice (VoIP), data and IT service

Hello list, I am currently doing a job for a summer camp. They would like to have several phones around the camp from which people can call in to the main office. It is an older campus and it is comprised of mostly old nungalow type housing. I need to install these phones several hundred yards from where the Asterisk server will be. The way thier telecom wires are currently set up is that they

Hello, In setting up * for my company's office and remote employees, I have a question about how to log one username into * as either a SIP account, or IAX account. For example, we will be using SIP phones in the office locally to the * server, however some employees travel, and want to use IAX (as it's much friendlier with firewall/proxy setups than SIP) clients on their notebooks.

I currently have a single server with a few SIP and IAX upstreams for origination and termination with IAX clients. I am adding a second server that will have a much higher capacity and will be handling a larger call volume. However, this second server is not going to be geographically near the first. It will largely share the same upstreams. I would like for this to be an integrated system

What does "t" mean in a CDR entry? This is in place of where the number that was dialed normally goes. For one IAX termination provider it always has a t instead of the number dialed. Also, we always see the word "hunguup" in the same record entry. This is the provider we have set to our secondary not primary. Is it transfer of some sort? I don't think there was a

To whom it may concern, I recently touched base with Chris Lattner who recommended I should email the LLVMDEV mailing list for help. I am currently looking for Compiler Front End Software Engineers, £60,000K + 10% bonus + shares + benefits + relocation package. Are you interested or are you able to recommend anyone? Please let me know your thoughts either way. Job Purpose and