similar to: SpanDSP and analog Digium channels (TDM400P)

Word. I'm doing a dupe of my production server this week as a CYA. Guess what: FC2. Once I yum update to the current kernel, no more yum. There's no reason to. You may have your own reasons (publicly avaliable server, for example) but why add uncertainty to an, at best, quite uncertain process (that of creating a stable Asterisk install given random hardware, network conditions, PSTN

Hi ! Does anyone managed to send multipage faxes (in single TIFF file) with app_txfax from spandsp package (i'm using 0.0.2pre18, libtiff 3.7.1)? If so, I'm interested in format of TIFF file that has been sent sent succesfully (tiffinfo <fax-filename>). I'm having problems with app_txfax, sending first page successfuly 99 % of the time, but never managed to send second or

I just tried, on a whim, to use an OpenSSH client vs SSH 1.2.26 client, and I can connect with it... Something drastic change between pre23 and pre24? Then again, I can't get pre23 to work anymore either, after further investigation :( OpenSSH clients connect no problem, just not the old SSH 1.2.26 clients ... Marc G. Fournier marc.fournier at acadiau.ca

<div><BR>Hello,</div> <div>&nbsp;I bought a TDM400P, and intended to use it with my analog phone, which is RJ11 ofcourse. So, the question now, how do I plug in my RJ11 phone to the TDM400P card, which has an RJ45 interface? Also, since it's an 11B card, I also intend to bring in an analog line into the RJ45, so i am still left with the same question....how do I go

Greetings, I have been using isolinux with Microsoft's cdimage.exe for my Utility / Preinstallation cds. I just upgraded from 3.36 to 3.55 and found it to be non functional. I didn't use the -debug version, it just hangs at the version banner. I went through several previous versions, and it appears that 3.50-pre23 is the last working version. I didn't see anything between that

Hi Guys, Looked at lotsa places on the Web/archives already. Does anyone have a Makefile for Asterisk 1.4 that integrates spandsp, app_rxfax, & app_txfax? This patch sure doesn't work with the Asterisk 1.4 Makefile: <http://soft-switch.org/downloads/spandsp/spandsp-0.0.2pre26/asterisk-1.2.x/apps_Makefile.patch> ==============================================================

Hi, I'm using spandsp-0.0.3 [http://www.soft-switch.org/downloads/snapshots/spandsp/ spandsp-20061116.tar.gz] on a bristuffed asterisk (1.2.13) [http://212.91.251.199/~junghanns.net/downloads/bristuff-0.3.0- PRE-1v.tar.gz] libtiff is at version 3.6.0 Running on: Linux router2 2.6.17-2-686 #1 SMP Wed Sep 13 16:34:10 UTC 2006 i686 GNU/Linux Debian testing distro. I've tried many

Hi, I saw your posting on Hipath and Asterisk.I have some doubts on the same.it would be really nice of you if you can help me out.My Doubt is as follows Currently I am using Hipath HG1500 V3.0 with Opticlient4.0. But i am not satisfied with the performance of Opticlient. I wanted to use SJPhone. Regarding this i had a talk with Seimens guys out here but they talk something ilogical. They told

Hi Stephen, You said that PRI works great. We are using HiPath 3550 and Siemens digital phone which using *11, *97 etc for function keys. However Asterisk uses the the * key plus one or two digits for function keys as well(it is common key combination for functions). So is it any way to disable *11, *97 keys in HiPath system and pass this keys to Asterisk? Thanks and regards, Isaac >Hi

Hi Group, I am currently facing a dead end and any help will be much appreciated. I have an a104d installed in an asterisk box, two of which is configured on ISDN pri. One is facing pstn and the other one is facing a hipath 300e Siemens. I am getting one way audio when a local on the hipath tries to make a pstn call but no issue on incoming calls from pstn going to the hipath locals. local

Hi, I will like to know if anyone would guide me about how I can to interconnect one SIEMENS HiPATH 3700 with Asterisk. HiPATH have VoIP card and my idea is to do one un IP trunk between them so we would to transfer calls and services (voicemail, IVR,..) between both. We havent PRI ports unused in HiPATH so cheapest method of interconnection is one IP trunk. Any help or comment about will be

Hi all, i have asterisk 1.2.17 with sip tcp support and i am trying to connect asterisk with HiPath 4000 V.3.0 using SIP. I can see the registration from the HG3540. But when i try to place a call from Asterisk to HiPath, the call fails with SIP/2.0 603 Declined. The strange thing is that the first INVITE uses tcp and the response is a 100 TRYING, the next 7 INVITE are using udp and the respose

Hi, I am currently using asterisk to record all incoming calls. My setup is as follows, the asterisk server has a two TE120P cards one of which sends/receives calls from the carrier and the other is connected to a Siemens HiPath 3000. All calls that come into asterisk use MixMonitor to record calls and this works fine, but if a call gets transferred the transfer information is not sent back to my

One our servers at work, I use ssh/rdist to sync binaries on various machines...one master server, several client machines. This requires the ability to ssh to the remote host without a password, so I have a .rhosts setup on the remote host, trusting the 'central server'... With pre25, I can't seem to get this to work anymore... I have PermitRootLogin yes IgnoreRhosts no set ...

Hi, I 'm using spandsp-0.0.2pre26<http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.2pre26/>, and thereis a bug adding headers: LOCALHEADERINFO and LOCALSTATIONID (I can't see them ). But faxes goes using rxfax and txfax fine. I also have tried development versions, the bug is fixed, but I get bad faxes (I get one page, but my tiff consists of three pages, and I get just

I am connecting to a siemens hipath 4000 with dahdi 2.1.0.4 and asterisk 1.4.23 using a Te210P card. the phone guy is saying that the lines are reporting always BUSY. however on my end the status shows OK. Anyone seen this? Is there something different about connecting PRI to siemens hipath? system.conf shows: loadzone=us defaultzone=us span=1,1,6,esf,b8zs bchan=1-5 dchan=24

hi all, we do have some troubles with zaptel timing source - we have a setup with 3 telco PRI lines, connected to asterisk digium card 0 - asterisk does some handling - calls are leaving on digium card 1 - going to a siemens hipath - there is some call handling - some of the calls then are going from the hipath over a qsig line to a bosch integral PBX - handling the rest of the calls. To be able

Hello, the bench that gives me a problem is the following: asterisk1 + spandsp ===>===PSTN===>=== asterisk2 + spandsp 1) i put a .call file inside Asterisk spool directory 2) the fax is sent successfully, I got this message in the sender's log: .... [Oct 4 17:57:02] DEBUG[17610] app_txfax.c: FLOW FAX Set tx type 8 [Oct 4 17:57:03] DEBUG[17610] app_txfax.c: FLOW FAX Set rx type 0

I am connecting 1.4.22 and dahdi 2.1.0.3+2.1.0.2 to a siemens hipath 300 and siemens hipath 4000. (2 channels to each switch) with a TE210p card setup as T1 with em_w. When the call is initiated to either switch the phone rings, when its answered then nothing... I hear no audio etc... After the timeout period the call is hung up. The phone switch 300 needs the T1 reset as the channel is not

Hi all, I have a connect between a siemens hipath & Asterisk system over PRI The connection works perfectly I can call from the Hipath to an Asterisk Extension. I want to allow the hipath extensions to dial out over a SIP trunk on asterisk but I keep getting "The number you have dialed is not in service" In this CLI I'm dialing from hipath extension 9 (prefix for sip trunk)