similar to: fxstest cant ring phone, but asterisk can !

Hi all, I have been looking at the code for both the zaptel driver (wctdm.c/wcfxs.c) and the asterisk channel driver (chan_zap.c) trying to figure out how much of this that has been implemented. So far I can see that the current stable 1.0.9.1 zaptel driver don't have the SETPOLARITY ioctl that would be required to properly signal the Swedish/Dutch CLIP, but the 1.2 beta1 has this

Greetings all- During some recent testing and debugging, I wanted to use the 'fxstest' application. However, I found it hasn't been built when doing the standard 'make, make install' shtick with dahdi-linux-complete-2.3.0.1+2.3.0... Can anyone tell me how to build fxstest? Thanks! --Tim

Hi All, I am wondering if I will be breaking the GPL, if I write for example, a channel driver or make some modifications to the astrisk source code, to interface at RUN TIME, through sockets, with a proprietary system. Eg. 1. I write chan_xxx + modify asterisk source (make changes + new code publicly available) 2. chan_xxx supports hardware by XXX Corp. 3, XXX Corps interface is

Hi Guys, An 8 channel Astribank is connected to Trixbox 2.8 and I ran freepbx-module-zapauto but I get the following when running these commands and can't make calls out: [Trixbox]# dahdi_genconf xpporder /usr/sbin/dahdi_genconf: warning - OLD DRIVER: missing '/sys/bus/xpds/devices/00:0:0/timing_priority'. Fall back to /proc pbx*CLI> dahdi show channels Chan Extension Context

Hello , I'm a student in Shahed university in IRAN . Now I'm become a member of our uni IT-Center . We are going to have an organized network with a en-bloc user authentication and proxy server. In comparison between MS AD and Samba , i want to choose Samba server ( cause of my belief in Open Source ) . Our plan have a Forest(root : shahed.ac.ir) and trees for faculties (like :

Ouch, part reset, quickly restoring reality (0) Power alarm on module 1, resetting! I have looked though a lot of email on this issue, and no one seems to have the answer. How many people are seeing this on a TDM400 card? Does anyone have a "REAL" answer to it. Yes, I do have the power connected to it! Regards Garry Taylor

>> I have more FXO ports on TDM400's than I have PSTN lines available for >> testing. When all the lines were used up (the FXO ports are all in >> zap group 2), and I did a Dial(Zap/G2/${phone},5) it thinks the Dial >> succeeded even though there is neither line voltage nor dial tone. >> Can at least the lack of voltage be detected? It would be good in

This is really cool! PREMIUM CHANNELS........Descrambled! EASY to assemble plans for only $7.00 ! YOU WILL BE WATCHING all your FAVORITE PAY STATIONS featuring MOVIES, SPORTS. Adult entertainment, and any other scrambled signal NEXT WEEK! You can EASILY assemble a cable descrambler in less

Hi Eric, > On 12 avr. 2016, at 17:48, Eric Cooper <ecc at cmu.edu> wrote: > > On Tue, Apr 12, 2016 at 04:36:58PM +0200, Bertrand LUPART - Linkeo.com wrote: >> I just made a asterisk / dahdi fresh install on Debian 8.4, and ended up with the following packages : >> [...] >> However, i can't find any dahdi startup script, neither init.d neither systemd

Hi, I am interesting how can I use the capabilities of ProSLIC to measure the following PSTN parameters: - Voltage (V) & Polarity (+-); - Current (A); - Frequency (Hz). Are there any ready for use tools? If there aren't ready for use tools how can I do the above measuring? Best Regards, Miroslav Nachev

(REPOST, sorry if you get this more than once.) Hello all, (Not sure if this is more appropriate for user or dev list) Does asterisk have any sort of "standards based" api that can enable an application to do call control on the switch ? For example, if I am developing a call center application using asterisk, I would like to be notified of inbound calls and then be able to route

Just for grins, pull off the cover and look at the electrolytic capacitors on the motherboard. Look for ones with the tops pushed up instead of being flat. This can cause all sorts of odd behavior. Often the machine with a blown capacitor will continue to run till it is powered off. Mostly I see this on Dell machines but have seen it on other machines as well. They are pretty easy to

Hello All, I am new to *, and this is my first post on the user list. I have had success with making / receiving calls to a SIP hardware Phone and the Console Channel Driver. My SIP phone *requires* that I register with a SIP server. For this, I used the NIST sip presence server (a version that I downloaded almost a year ago). I have a problem with getting asterisk to register properly,

Hello I have played about with a TDM400 card and plugged in some standard analog phones. I am using the card in FXS mode - for analog extensions. I did notice that one of my phones did not ring and I wondered why. I later read in Paul Mahler's book VoIP Telephony with Asterisk that in his section on the TDM400 on page 127 he says "In the UK, you may need an adapter that provides a

Hi All, Is it correct to say that by design, asterisk wont make use of any cards hardware dsp capabilities ? I don't think that any of the hardware cards currently supported have any dsp capabilities, but I wanted to know if for example, in the future a driver was written for a card that did have dsp capabilities, would asterisk be able to make any use of it ? I am only just starting out

Three months ago, I was experiencing all sorts of issues with my Asterisk box maintaining a connection to multiple trunks, etc. I also experienced various timing issues as well. In addition, Asterisk would sometimes take almost a minute to fully load and register its SIP and IAX trunks. Puzzled, I recompiled several times. No result. I checked my hardware. Didn't find anything. However, I did

Styma, Robert (Nokia - US) wrote: > > Just for grins, pull off the cover and look at the electrolytic capacitors > on the motherboard. > Look for ones with the tops pushed up instead of being flat. This can > cause all sorts of odd behavior. Often the machine with a blown capacitor > will continue to run till it is powered off. Mostly I see this on Dell > machines but

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 This seems to be an old problem, but I haven't found much about it. SuSE mentions "hwprobe=-pci" but, as far as I know, CentOS doesn't use it. This does stop the machine from booting (mounting /) quite a few times. Logs: SCSI subsystem initialized sym0: <895> rev 0x2 at pci 0000:00:13.0 irq 177 sym0: Symbios NVRAM, ID 7,

HiI don't have any digium cards and only want to install ztdummy with all asterisk functions. What I will need with ztdummy and what I can disable?Thanks a lotDmitri************************************ Zaptel Module Selection ************************************* Press 'h' for help.

Hi All, The link http://www.voip-info.org/wiki-Asterisk+Data+Configuration talks about how a Digium PRI card can be setup for data, and also be used by asterisk to handle voice. Is the same or similar possible with BRI ? There are a few links on the Wiki to BRI cards, and that BRI can be supported through CAPI compatible hardware. That will take care of voice. What I want to do is use a BRI