similar to: Setting Caller-ID / Point Codes

Hey folks, I have a T100P card with a PRI; when doing outbound dialing over the PRI, I can use SetCIDNum(2024561414|a) to force caller ID to display as "The White House" on a land line. This is apparently done as a reverse lookup by Verizon, as I do not hand the PRI the words "The White House" -- 202-456-1414 is an actual White House number. Using SetCIDName("Flying

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

Hi, Did anyone hear about a recent ruling which makes it illegal to have caller-id set to anything except what is on the account of the user? IE... If your name is "Joe Smith" you can't have "Mary Smith" set as the caller-id name, unless mary smith is also on your account.

Alle, I'm following the instructions in section 19.3.3.2 of the docs @ http://www.centos.org/docs/5/html/5.1/Deployment_Guide/s2-nfs-config-autofs-LDAP.html, but I cannot add the following entry in LDAP: dn: automountMapName=auto.home,dc=subaru,dc=nao,dc=ac,dc=jp objectClass: top objectClass: automountMap automountMapName: auto.home After looking at the schemas in /etc/openldap/schema,

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.

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

As an employee in the technical operations of a CLEC this information is easily obtainable by anyone that has access to the Class 5 switch servicing that PRI... A Q.931 trace in the Class 5 Switch will tell the whole story.... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Rich Adamson Sent: Tuesday, June

Hi all, I am trying to setup a ZAP interface to do MF signaling for a handoff to a 911 tandem. The signaling I need to perform on the T1 is this: 9-1-1 Tandem: Wink CLEC end office: KP (Keypulse) NPA ST (Start) 9-1-1 Tandem: Wink CLEC end office: KP I (Info Digit) NXX XXXX ST As I'm not as familiar with the Zaptel configurabliity, I'm not really sure how to do this. Do I dial twice

I need some help understanding how on of the example data sets is formatted in the basic R installation. If I load the Mona Loa CO2 data, with the command: > data(co2) I can view the data with: > co2 And the data are in the form of 11 rows labeled as years (1994-2004) and 12 columns labeled (Jan - Dec). This structure appears to be a dataframe, however, if I type the command

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

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

My CLEC just called and asked if we will support the "MI2" protocol on our proposed T1 circuit. I think this is for CallerID name. Will the T100P support this? Thanks, Mike

Central business location has a PRI with a CLEC. Remote offices access the PRI for all voice traffic via VoIP. How does one get the telco to report the address of a remote office to the 911 call center when the call is made from that respective location?

David, I guess I'm a little confused here. Are you asking me to provide a "pencil copy" of an LOA for your review? I don't understand why you need an LOA from us. We need an LOA from you to order circuits that will be billed to us that will be attached to your CFA. It was also my understanding that you had an LOA ready to be given to us, which had already been reviewed by

Hi Folks, having an issue with outbound calls through a VOIP provider. Calls get sent out with the CallerID(number), but where does callerID(name) come from? Apparently not from provider, as we are seeing different (sometime missing) names on inbound calls, different than what we have configured. Apparently this comes from some telco database somewhere? Numbers were ported from a wired-telco.

I want to test asterisk with its DID functions on a smaller scale without the expense of a PRI. I am in the US, and want the most reliable/stable, and feature rich solution that I can have. What ISDN card would be reccommended? My clec switch operator says he can/will send DID info to my BRI's if I wish. Can Asterisk, or one of its addons support this? What BRI solution would be

http://www.boingboing.net/2006/06/05/play_zork_by_phone.html Let me preface this idea with one comment: I don't have the time to do this - I don't even have time to eat these days. But someone out there has the cycles to do this... and it would be very cool. OK, so now Zork is attached to Asterisk, but using the less-than-clear Festival engine. There are beta tests of the LumenVox

Hi all, Newbie quest here. I have 3 PSTN lines (2430-2432) setup by the CLEC in a hunt group coming in to a TDM04B and 3 Grandstream gxp-2000's (say a,b and c) and Asteriskathome installed. I want all phones (a,b,c) to be able to take calls from the hunt group. Does this mean I must make 9 SIP extensions on * ? 3 for each IP phone? and group them in trtiplets (ring groups?)

dear R users, I am using sweave to generate report for my data analysis. I recently updated R ro 2.8.0, and now I have the following results when compile the the tex file generated from R. This is pdfTeXk, Version 3.141592-1.40.3 (Web2C 7.5.6) %&-line parsing enabled. entering extended mode (./Lajos.tex LaTeX2e <2005/12/01> Babel <v3.8h> and hyphenation patterns for english,

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