similar to: anyone using SIP trunks from Time Warner Telecom?

What is the deal with "CSR's"? TWTelecom is telling me that I can't port a number to their service without a Customer Service Record. Apparently this is easy with Verizon, and not so easy with some other companies. Basically I'm at a brick wall with a couple of ports because TWTelecom is telling me I HAVE to get a CSR and certain other providers (Time Warner Cable for

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

(Cross posting on purpose) What is the common wisdom on the list... find a telco that offers SIP termination or wait for Digium's DS3 card? Who are the telcos that offer SIP termination? Thanks,

Hello. I need some help. I was running uw-imap on my IMAP server (so I am using mbox email files), but I was having trouble with Outlook 2013, so I decided to move to dovecot. At first things were looking much better, except that most of my folders, other than the Inbox, were not showing up. I started changing both the dovecot configuration and the folder structure of my mail files,

Shortly, I'll be purchasing a Sangoma A102. I'm wondering if I should spring for the hardware echo cancellation circuit or not. Upon initial implementation, the 2 T1 Ports will be used as a passthrough as we slowly transition off of a legacy PBX. Eventually, we'll only be using one of the ports, and will be providing VoIP service to a bunch of SIP deskphones. So - with that usage

All - this is probably a simple problem, but I've been pulling my hair out trying to figure out what I'm doing wrong. I'm building a *simple* IVR menu. Here it is: [main-menu] exten => s,1,Answer exten => s,2,SetMusicOnHold(default) exten => s,3,DigitTimeout(5) exten => s,4,ResponseTimeout(30) exten => s,5,Background(logic-main) exten =>

Hello, I stumbled on this problem while troubleshooting a time synchronization problem. The Windows commands "w32tm /monitor" and "nltest /dclist:domain" appear to both use the same call to query the domain for a list of domain controllers. When the DC is Samba4 (2003 domain & forest level) these commands complete successfully. After raising the levels to 2008 or 2008_R2

Hello all - I'm about to deploy an asterisk server here at work. Before deploying, I'd like to do an extended load test on the system. I currently have T1 crossover cables connecting ports 1->2 and 3->4. Would there be an easy way to script generating a bunch of calls across these spans? I envision generating 23 calls over the 1->2 span and 23 over the 3->4 span. I'd

Following the successful deployment of asterisk servers at several of our branch offices, in the near future, I'll likely be implmenting an asterisk server at our HQ. We currently have a T1 PRI terminated on a legacy PBX. I'll be doing a stepped deployment in which, via a dual T1 linecard, the asterisk server will initially pass all incoming/outgoing calls directly through to the PBX.

I've been working with a telco for the past two days trying to get a PRI span up and running. This is a small-ish telco and I get the feeling they don't do this very often. Anyway, they specified a pretty standard setup: ni2 switchtype, esf framing, b8zs coding, etc. All of my b-channels are up, but we're having a heck of a time getting the d-channel to come up. He finds out that

Hello - I have (finally) obtained a fair amount of success in connecting my Intertel Axxess PBX to an asterisk box via a T1 PRI. I can place calls from the Intertel side through the T1, out to an IAX2 softphone and the calls get routed correctly and all of the CID information stays intact. However, when I call from the IAX side to an extension which should route back through to the Intertel

Hi, Is it possible to configure asterisk so it doesn't play MoH from certain phones? Regards, Jan

Hi folks, How I can change default level in asterisk from 3 level to 7level, using the script /etc/init.d/asterisk

I am having a weird issue with setting the recording file for the Page app. Here is some quick background info I have a macro that pages all my phones: [macro-pageall] ; Context for paging all devices. ; This will search the sip table in the realtime database ; for all phones that start with a number. That number is ; passed to this macro as ${ARG1}. ; ; ARG1 = The

At work, I have 4 branch offices at which I've deployed asterisk. Call termination/origination at each branch office is handled either through a frac PRI or 3rd party SIP provider. Soon, I'll be replacing the legacy PBX at our HQ with asterisk. Each branch office has between 3 and 20 employees, each with their own extension and DID, and at headquarters, we have about 70 people, again

I was trying to get my polycom phone to auto answer. I added this to the dialplan. Used a different phone to call "22" and the phone rang it did not auto answer. Did I miss something? exten => 22,1,SipAddHeader(Call-Info:=\;answer-after=0) exten => 22,n,SipAddHeader(Alert-Info: Ring Answer) exten => 22,n,Set(__SIPADDHEADER=Call-Info:\;answer-after=0) exten =>

All - I've been trying to pick out a bluetooth conference phone that I could use with a softphone along with my asterisk server. I've been looking at the TrendNet TVP-SP4BK. Have any of you used this device or any other bluetooth conference phone? How have your experiences been? Thanks! -Erik -- Erik Anderson http://andersonfam.org

Hello Everyone! According to voip-info.org the correcy syntax for the realtime function is: REALTIME(family|fieldmatch[|value[|delim1[|delim2]]]) on read REALTIME(family|fieldmatch|value|field) on write It seems from the syntax that it is only possible to retrieve a full row according to the value of only of column. This translates in SQL language as Select * from family where fieldmath =

Hi, I'm trying to add mysql CDR onto a vanilla Asterisk 1.2 install. The add-ons pack has been installed for a while, so now I'm trying to add the Mysql config. I've created a mysql database, added the grants for a user acces, and can run a mysql -u asteriskcdruser -p and can connect to the database. I've been using this as a guide:

I've been going back and forth with my telco for several days, trying different configurations to get a new PRI to come up. The bchannels are all up and the T1 is not in alarm status. The dchannel refuses to come up however. We've tried ni2, qsig, and now dms100 for the switchtype. The telco tech I've been working with says that he's been sending "reset all channels"