similar to: Why using usecallerid=no?

Hi, I have these two cards, the Sangoma has 4 fxo interfaces and the digium has 1 fxo and 1 fxs. After install the sangoma card, my zaptel.conf was configured for that card. I'm trying to configure the Digium one together thinking that the Digium ports should be 5 and 8 but it doesn't works. Someone has some example about this? Thanks in advance Pau?p

Ottawa, Canada ? February 15, 2006 - PIKA Technologies Inc. today announced that they have integrated PIKA?s high-density analog computer plug-in boards with the open source Asterisk PBX, with the introduction of PIKA Connect for Asterisk. PIKA Connect for Asterisk is a software layer, available free of charge and distributed under the GNU Public License (GPL), which allows interoperability

Hi, I'm currently having problems detecting FSK BT (UK) caller id in our API (Pika boards). I have a recording to test on but it is giving me checksum errors. I'm wondering if someone from UK using BT lines could send me a recording with the FSK signal so I can have more data to work on? If you have something, please send it to paulo.astuser at gmail.com. I appreciate all help about

Every week we try to get guests with ideas, products and services you haven't had time to check out to come and talk about what they're doing. Tomorrow, Pika Technologies will be with us. Friday, March 7that 12:00 PM (Eastern US) 9AM PST, 5PM GMT *** Call (724) 444-7444 or SIP:123 at 66.212.134.192 *** After the call connects, enter the conf: 22622# and your_PIN# (or 1# if you

hi, i have problem with pass-through faxing with this scenario hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.2.X(xen virtual) - linksys ATA i can fax to fax2mail on hylafax but after upgrade asterisk2 to 1.4 faxing is not working hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.4.X(xen virtual) - linksys ATA configuration is same do you hava any idea what is

Hello, I've spent a couple days searching and posted into the forum with no luck, apologies to anyone who reads the Digium forums for the cross-post. I'm having a problem with an asterisk set up where I have a TDM402B connected to a POTS line. Also connected to the POTS line are plain telephones, non SIP, just plain old telephones. When one of the normal handsets goes off-hook,

Hi I am mouving from PC-NFS to SAMBA and thare are some thinks that I would like to see in SAMBA. In NFS, when I do not want a share to be accessible, I just use the command: unshare /shared/directory This make the /shared/directory imediataly not accessible. Even by some one already using it! I?m using SAMBA 1.9.18p10 in a SUN Solaris 2.5 and i try to change smb.comf to disable one share,

Judging from log analysis, there has been in the past some interest in the Asterisk example files that I have made available from my experimental platform (home office) so I've put my newest versions up on my site. Feel free to dig through these and cut pieces up for your own use, or just marvel at the bad logic flows. :) http://www.loligo.com/asterisk/current/ JT

Please, I would like to know how to generate a truncated multivariate normal distribution k - dimensional, X ~ NT(mu, Sigma), where the elements of X to be non-negative (except the first), and the first dimension is strictly larger than zero. Example: X ~ NT_2(mu, Sigma), where mu=c(0.5, 0.5) and Sigma=c([120, 191], [191,154]), with X_1>0 and X_2>=0 Could anybody help

Okay, I can't find what might be causing this. Here is what I got: Asterisk server hooked up to a digital T1 line (full 24-channel) via a Digium card. Verizon has turned on caller ID on the first line (I can guarantee it is on as I can hear the FSK tones on this line but not the others). Using zttool an ZapScan() I have determined the following: 1) The RxB/RxD bits toggle from 1 to 0

Hi all, the company I work for has an aging Digital PBX attached to an E1. This PBX has a few analogue lines, one of which we use a 'traditional' fax machine on. I want to upgrade our PBX and Asterisk is almost a perfect fit. The only problem I can't seem to find a working solution for is Faxing. I don't want to use Hylafax or other similar methodologies. I believe there

Hi, I would like to understand how those features (subject) work on fxs ports. Unfortunately I don't have a digium card with this kind of port, then any help will be appreciated. I tried to gather some information from google and this list history, but I still need some help. 3-way-call - As I could understand, when you are talking to A-person, you can press *flash*, call to B-person and

Hello! I'd like to use PriorityInheritance option, despite it is still experimental. (Why it could be experimental, when it looks quite simple feature, no?) But this option doesn't change the TOS field of outgoing UDP packet. I just do tcpdump on outgoing interface from vty1: $ sudo tcpdump -vni eth0 udp dst port 655 And from vty2 I do ping with setting of tos field to EF(0xb8). $

Hi All, I have the following scenario, it may already have been answered elsewhere, but I cant find the solution. I already have a PBX and would like to start implementing asterisk. I have ordered a 4 port card from the asterisk store (2 port FXS and 2 port FXO) and am waiting for it to arrive. I do not want to plug my incoming lines into my FXO ports yet as not all the desks have IP phones

I would recommend that you look at the Pika Technologies Daytona MM board. It has onboard DSP and onboard analog bridging taking up much less horsepower. Please contact me off-list if you would like more information. Bill Hunt Stroudwater Contact Point 207 347 8080 x219 877 870 1234 Toll Free www.stroudwater.com "Realize the Value of Customer Contact!"TM This e-mail is intended

On 2016-08-29 12:28, Eric Klein wrote: > Hi Hooman, > > What you probably want is a PRI PBX running Asterisk. > > You should either plan to build your own (with the cards you need) or get one of the low cost options: > > * Allo.com has their Mega PBX with 1 PPR port (http://allo.com/megapbx-line.html) > * Pika Tech has the Warp PBX with BRI

Hello, The company I work for recently purchased 2 Rhino CB24s and a Rhino PCI-E R4T1. The channel banks are plugged into the R4T1, as well as 2 PRIs from our telco. The CB24s are for all internal analog phones. Most of the phones are setup in "batphone mode", which is "immediate=on" in the DAHDI config. They are set up this way because we are an outgoing call

Hello For those SOHO customers (ie. at most, a couple of POTS/ISDN connections and simultaneous SIP calls) who'd rather not use a big, noisy PC to run Asterisk, I'd like to offer an alternative that has the following features: - not old hardware sold on eBay, ie. it must be up-to-date hardware sold by a company currently in business - compact, silent - has room for a 2.5" hard-disk,

Hello all. I just received a TE110P and am trying to hook it to my Mitel 200SX has anyone successfully done this? My configuration is as follows. Asterisk -> TE110P ->Kentrox (csu/dsu) -> Mitel T1 Card. All I get is a blinking yellow on my TE110P card and an alarm on my Mitel. T1 card. Any advice would be great. Zaptel.conf span=1,0,1,d4,ami e&m=1-23 dchan=24

Hi, Built a new server at the weekend and install Asterisk 1.6.0.9 and IAX and SIP work great :) The one problem I am having is getting the OpenVox (TDM400 type card) to work. It is successfully identified using WCTDM kernel module and dahdi_scan picks it up just fine. The issue is when I try and setup dahdi_channel.conf as it fails everytime. When running asterisk -rvvvv I see the port pick