similar to: Help with Cisco PRI

* --> SIP --> CISCO --> PRI --> PSTN The PSTN sees no callerid. *---> PRI[zaptel]--> PSTN Callerid is there... which makes me think it's the cisco, not the PRI/PSTN/telco. CISCO PRI--> * PRI [zaptel] Callerid IS there... which makes me shake my head in disbelief, because * can see clid from the cisco pri, but pstn doesn't... but when * sends info on that pri,

Is it possible to send CLID NAME on a PRI? The numbers we send out are being received by telco and propagated, but the names we send out are not showing up. Is this a feature in PRI? Do we need to set PRI_NET instead of PRI_CPE? Is this just not possible? Is this a telco config issue? Thanks for your help... I've read voip-info, and various other sources, and search engines, and google...

Good morning, I have a Bell Canada PRI here (switchtype=national) and I am trying to perform a call-forward-unconditional on one of the DIDs. The idea is that when DID 5551234 receives a call, Asterisk redirects it back out the same PRI to some external number. This is simple enough to do with something along these lines: [PRI] exten => 5551234,1,Set(CALLERID(RDNIS)=${EXTEN}) exten =>

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 have a PRI comming into each of 2 buildings. How do I redirect an incomming call on PRI_A of particular DIDs to arrive at PRI_B instead? Thanks, John

I am ordering T1-PRI service from local service provider and have a few questions. Is there framing and coding considerations (or is it all one standard), if so what is best? How are calls routed based on DIDs - are these just dtmf tones passed after the call is picked up and treated as normal exten=> definitions? John This e-mail was scanned and found clean by Monroe-Woodbury CSD

Hello All, I am getting my first PRI installed in a couple of weeks and I wanted to ask for a little advice. I have a single span Digium card I will be using for the install. Id there a benefit to which protocol I use? When asked, I told them to set it up as NI2. The PRI is through MCI and will be used for local and long distance with DIDs and features like CallerID, etc. Any advice would be

I've tested it on MSVC, gcc, clang and icc. The solution in clang (in Decl.h) is not ideal (as you have said yourself). The solution I offer, is using a union of fields of class BitField (this is a new class that implements a bitfield of a specific type requested). With this, the definition, of the required bitfields in the subclass data, remains in the hands of the actual class needing them.

Hi, all I want to configure a few FXS ports in an Antribank-16 to be able to receive faxes sent throught a PRI: E1 ==>Zap * ==>FXS * ==>Fax machine My asterisk box has a Digium TE120P (for the PRI). Versions are *=> 1.4.17 | Zaptel=>1.4.8 | libpri=>1.4.5 The Astribank is not configured yet, because I am a little bit confused about how to do it. Let's say I configure

On Mon, Apr 27, 2020 at 6:42 AM Robinson, Paul via llvm-dev < llvm-dev at lists.llvm.org> wrote: > Older releases are still available for download at releases.llvm.org; I > believe the 3.0 release was supposed to be able to read 2.x bitcode, so you > should be able to upgrade the bitcode with 3.0 tools and proceed from > there. I **think** everything since 3.0 is still readable

Hi all, I was looking to build a SIP-to-PRI gateway using Asterisk (as in my other post), but there is also an alternative of using a Cisco router with something like an NM-HDV module with a T1 VIC module and DSP channel banks. The question is, would it be more reliable to offload all dahdi/zaptel/libpri type stuff to a dedicated gateway device (Asterisk or Cisco) and have the Asterisk PBX only

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of Adams, Gavin Is there any additional information I could provide to start tracking this down? I was thinking about looking into the various applications source to see how they access the data elements for callerid. I know where the values are pulled

Hey, all. I love having a PRI to play with -- lets me do all sorts of things with DIDs, fax-to-e-mail, etc. But for a small shop, a T1 is pretty pricey. Is there any reason that a BRI can't do exactly the same stuff, but on 2B+D instead of 23B+D? "Enquiring minds," etc. -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to

Hello Everyone, I am looking to getting converged with the local ILEC here in Canada (Bell or Telus), and was wondering if I can get some more information about typical setups. DIDs and channel offerings from third party clecs does not fit our business model and that's why we are looking to purchasing phone numbers directly from the ILEC, and manging our own PRIs for a convergence of sorts.

I admit I didn't know about that, but that is because I am handling a lot of *old* bugs, older than LLVM version 3.0. Version 3.0 has been released in Dec 1st 2011, and there are still many bugs open from before that point. The current BC Reader can't parse files produced by LLVM version 2.9 and below (I've checked). So, I wonder if there is anyone in favor, against (, or just

Hi all I wonder if anyone can give a little insight into this; asterisk@home 2.2, HP Proliant gl3, sangoma A101,Cable and Wireless, ISDN30. 10 zactive channels Incoming calls work fine no problems tested 43 DIDs all working. zaptel.conf # Global data loadzone = uk defaultzone = uk span=1,1,0,ccs,hdb3,crc4 bchan=1-10 dchan=16 zapata.conf [channels] language=en context=from-pstn

Hi everyone, In the reviewing system, after a patch is approved, what should be the following step? I assume I'll have to make sure the patch hasn't broken anything, before pushing it, so I'll wan't to run a full build+tests, probably in a remote sterile environment (like a jenkins server). Do we have an integration system like this? Or should I just trust the build+tests on my

This depends on what you mean by ?not involving the service provider.? If you are literally forwarding calls that come in on the PRI back out on the PRI, the most efficient way is with Two B-Channel Transfer (TBCT). Check it out in the wiki. You need to make sure your carrier supports the feature. When you want to do a ?transfer,? you have an incoming call alerting or answered, you

I currently have: Telco-PRI ---- Panasonic DBS576 PBX ---- E&M wink T1 ---- Asterisk. I have configured the Panasonic to forward my Asterisk DIDs to the Asterisk extensions over the T1. I do not get DID nor CID on the Asterisk, so I want to use PRI between the PBXs. I do not want to pay for another PRI card for the Panasonic. (T1 and PRI are different cards) I see this as my least

Quite often I get to work on an old bug, where an old ll/bc file is attached as a testcase. These files, in most cases (if not all), need to be converted somehow to the latest format, for the trunk version to be able to parse it without an error. So a few questions arise: 1. Is there a standard way to convert an old ll/bc to the latest? If not, what is the common approach for these cases? 2.