similar to: tdm2400p and asterisk 1.2.1

Hi, I have a box with Debian Sarge, Asterisk 1.2.1 (and zaptel 1.2.1) and a TDM2400P with echo canceller. I installed the card but no echo cancellation is being made...seems like the echo canceller module does not work, infact the software cancellation is working. My zapata.conf has echocancel = 128 and echocancelwhenbridged = yes but no echotraining parameter which gives a warning. I found

Hi all, I'm confused, in digium website, it says: TDM2400P: It supports a combination of up to 6 FXS and/or FXO modules for a total of 24 lines. 6 plus 6 is 12, how come it's 24? if I have 24 PSTN lines, i'll be needing 24 FXOs. Pls. elaborate. thanks. Lito -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm installing an Asterisk box with a TDM2400P in Brazil. I can make analog phones work while lines are not working. Since I do not know anything about brazilian lines, is there anybody who can tell me what is wrong/missing in my conf files (below)? TIA Giorgio _zaptel.conf:_ fxoks=9-16 fxsks=17-24 defaultzone=br loadzone=br* * _zapata.conf:_ context = inbound_zap echocancel = 128

Hi, I cannot make my TDM2400P (with Echo Canceller module) detect faxes. I tried with a TDM400P and it worked at 80% (20% of faxes were lost). My test conf.files are: _zapata.conf_: context = inbound faxdetect = incoming language = it musiconhold = default signalling = fxs_ks callerid = "call_or_fax" channel => 1 extensions.conf: [inbound] exten => s,1,Answer exten =>

Hi all, since Asterisk 1.4 seems to have too many differences from previous versions, wouldn't be nice to have a new mailing list? Giorgio Incantalupo

Hi, I am trying to use zaptel module on an Ubuntu 5.10 distro (2.6.x kernel) using gcc 4.0.2. Compilation does not give me errors so after a 'make install' I try to load zaptel module with insmod but the following error arise: *insmod: error inserting '/lib/modules/2.6.12/misc/zaptel.ko': -1 Invalid module format* Is there anybody who can help me?? TIA Giorgio --

Hi, I'd like to build a personal PBX connecting 4 or 5 analogic phones with a ADSL line and I'd like to know what is the right card I need I visited digium site and I think TDM400 could be the right choice but I cannot understand how it works...I think it has 4 slots where 4 modules (FXS or FXO) can be inserted. How many cards do I need to connect my ADSL line to 5 phones? I think I

This is a function of the Phone itself. Asterisk has nothing to do with it as it does not know anything about the call until after the SIP device 'sends' it. To my knowledge it is not posible. I don't even think a SIP standard is available for this. This 'feature' along with changing CallerID Display after a call has been answered is something missing from the RFC. >

Hi, we have an asterisk 1.2.1 box and 2 polycom SIP phones. We'd like to show the called name on the calling polycom display instead of his /her extensions as I do with the caller name on the called polycom. Is it possible? If yes, how? TIA Giorgio Incantalupo

pbx18*CLI> module load codec_g729-ast14-gcc4-glibc-pentium3.so Unable to load module codec_g729-ast14-gcc4-glibc-pentium3.so Command 'module load codec_g729-ast14-gcc4-glibc-pentium3.so ' failed. [Dec 22 15:52:45] WARNING[4491]: loader.c:757 inspect_module: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' does not provide a license key. [Dec 22 15:52:45] WARNING[4491]:

Hi, is there anybody who knows what this warning means?? WARNING[27309]: chan_sip.c:8875 reload_config: Section '10' lacks type TIA Giorgio -- ____________________________________________________________________ GIORGIO INCANTALUPO Tel. +39 02 9350 4780 (104) FG&A Software 20017 Rho - Via Puccini, 8 E-Mail : gincantalupo@fgasoftware.com Internet: http://www.fgasoftware.com

Hi, every second I get on the console: Remote UNIX connection Remote UNIX disconnected which gives no problem but makes console unusable. Is there anybody who has encountered the same problem? How did you solve it? TIA Giorgio Incantalupo

Hi, is it possible to force FOP to reload its configuration files (op_buttons.cfg and op_style.cfg) while it is working? I tried to click on the refresh icon but nothing happens. TIA Giorgio Incantalupo

Hi, does anybody know what *zttranscode *module* *is for*?* Thanks!! Giorgio -- _________________________________________________ Giorgio Incantalupo, mailto:gincantalupo@fgasoftware.com FG&A srl - http://www.fgasoftware.com - Voice@Work - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172

Hi all, I need to connect my Asterisk 1.4.26 with a Sangoma PRI card (configures with signalling=pri_net)) to a Cisco 2800 PBX. After connecting the cables everything seems fine (ifconfig w2g1 is ok, wanpipemonitor gives no errros, the span is up and active, green light on the card) but when I make a test with my iax phone, there's no way to dial the PBX and I get this WARNING: [Jul 2

Hi, I'm using Asterisk 1.2.1 on Sarge. it seems like if I call a phone and nobody answers, asterisk does not jump to the next priority...it freezes. Take a look at this: exten => 777,1,NoOp(before) exten => 777,2,Dial(SIP/7|60|g) exten => 777,3,NoOp(after) priority 3 is never executed but this worked with Asterisk 1.0.7!!! TIA Giorgio Incantalupo

Hi, is it possible to remove the "no timeout" combo box in flash operator panel? How can I reduce the flash area? I set small buttons and half of the area is white and I want to resize it. TIA Giorgio Incantalupo

Hi, I live in Italy and I have an Asterisk 1.0.7 box with a HFC monoBRI card connected to my cisco router which is connected to FastWeb provider: does anybody knows why every time my cisco router turns off, my telephone connection to Fastweb drops (while internet connectior is ok)? Restarting Asterisk is worth nothing. TIA Giorgio --

Hi, I'm using Asterisk 1.2.1 on a Debian Sarge with a TDM400P and a monoBRI using chan-mISDN from beronet site. It seems to work all right except for autodial calls, monoBRI ISDN channel behaves differently waiting for the caller to answer and then continue. Asterisk console says: analog: -- Attempting call on Zap/2/3391818250 for 104@inbound_originate:1 (Retry 1) > Channel

... Activating "sip debug" shows the register packets but nothing in return. ... I think that this is a network related issue, but you have to solve it by using a Asterisk config file. Unfortunately I think that the faster way to solve your problem is trying to understand if sip messages are correctly sent to tnet. I strongly suggest to use http://www.wireshark.org/ previoulsly named