similar to: ERROR: chan_dahdi.c: PRI Span: 3 PROBLEM: General: Badly Structured Component

Hi All, I have installed asteriskNow with Asterisk 1.6.2.11 and FreePBX 2.7.0.10. I have configured x100p fxo card in my asterisk box. But in my cli mode i am not getting the command *"dahdi show status"* Output of CLI : astrisks*CLI> *dahdi show status* No such command 'dahdi show status' (type 'core show help dahdi show' for other possible commands) I

Hi, Is it possible for a Manager/Supervisor to intercept and listen in on live calls for training and evaluation purposes? Thanks -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Afternoon, I was hoping someone could point me in the right direction. I have an asterisk PBX deployed in China using a TDM400P based card. The incoming calls are being picked up correctly, but are not being hung up. I suspect that this might be an issue with the signaling that has been selected. If anyone here has deployed asterisk in china using an analog card, it would be a great help

Hi, I'm making some tests with Tellfree brazilian provider. I'm using 2 users A and B, one for calling and the other to receive calls. When I make a call I can see (from the CLI console) user A is calling user B but user B does not answer (the phone continues to ring) even if the "sip show registry" command says user B is registered. In my sip.conf I have: register =>

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

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]:

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

Dear all I have asterisk 1.4.11 on centos 4.x i have installed 2 PRI on is asterisk and it is working fine but i got this DTMF error on asterisk CLI what is it ?? -- Zap/36-1 is ringing -- Zap/36-1 answered SIP/5406-9fa59770 -- Channel 0/1, span 2 got hangup request, cause 31 [Sep 13 22:10:29] WARNING[7191]: app_dial.c:741 wait_for_answer: Unable to forward voice or

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 --

Dear folks, I had to rewrite UPS::Nut because it was not developed since 2002 and it was based a very old version of upsd speaking a totally outdated protocol. If some of you interested in it you can get from my debian repository. package name is libups-nut-perl. Add this to your APT sources: deb http://debian6.ki.iif.hu/ lenny contrib deb-src http://debian6.ki.iif.hu/ lenny contrib Or simply

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'm testing attended transfer with 3 SIP phones. I noticed about 10% of my transfers make the call drop and I get this on my log: Jul 5 10:42:32 WARNING[23960]: file.c:592 ast_readaudio_callback: Failed to write frame -- Playing 'beep' (language 'it') Jul 5 10:42:32 WARNING[23960]: res_features.c:745 builtin_atxfer: Failed to play transfer sound! Moreover, every

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

Would someone be able to give an example of a working use of setvar from chan_dahdi.conf? I am trying to create a custom variable like I use in sip.conf but I have been completely unsuccessful getting any variable set using setvar to appear for a DAHDI channel. I am running 1.8.11-cert8 and am using the newer format (but I have tried using the older [channels] format). Here is an example:

Hi, I get a lot of these messages on my Asterisk CLI: "Failed to authenticate user 1000<sip:1000 at MY_OWN_IP_ADDRESS>;tag=03f82bb9" as if my PBX machine is trying to authenticate to itself. It seems someone is attacking my asterisk PBX. Is there a way to fix this problem? Thank you. Giorgio Incantalupo

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

Hi all, When I do a test call into the box (which is running latest version of Trixbox) it all goes fine. If i decide to hangup the cellphone (during the ivr playing options) the system does not recognize the hangup and the system continues through and ends up at the timeout option. What settings do I need to change to fix this. Is it the rxgain? If so is there something i can use to figure

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, emails to astricon.net seems to bounce (at least for me) I need information about proper & authorized Asterisk training in the Miami, FL area and the possibility of later DCAP testing. Thanks, -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama