similar to: New TDM400P no dialtone

Everyone, Just thought I would drop a line telling everyone here I have the software RxFAX/TxFAX up and running without any real problems. I did have to..... RH 9.0 1) Install an audio devel rpm 1) install libtiff from source, and copy over a bunch of include files to /usr/local/include 2) build/install spandsp 3) move app_rxfax.c and app_txfax.c to apps/ dir in asterisk source tree. 4)

Has anyone gotten the voicetronix boards to work with Asterisk, what would it take? Or does anyone know where I can get 4 ports or more fxs PCI cards that do work with asterisk? Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-798-9106 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017

Like most others on this list I also have some really annoying echo whenever a call goes out to the PSTN from a SIP phone... SNOM/Budgettone -> Asterisk -> X100P -> PSTN I have tried every echo canceler in the makefile and turned on and off aggressive suppressor etc. etc. etc. tried 32,16,128, and 256 bridgetaps and I can get it reduced to only a few seconds on the intro of the call and

Hi, We are looking at consolidating our lines with PRI. This will allow the elimination of many fax lines. Some of them will be replaced with this type of config ... PRI * IAX * Channel-Bank FAX We will have daggressor suppressor enabled. Is anyone doing this and should I expect smooth operation? John This e-mail was scanned and found clean by Monroe-Woodbury CSD Antivirus.

Hello, How would I go ahead a record all phone calls into and out of my asterisk server. I know the legality issues behind it, but I could always play a recording to let people know they will be recorded. Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-537-2817 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017

Hello, I resolved my echo issue using grandstream/estara etc etc sip phones and wcfxo interfaces from digium. I swapped out my via kt400 based msi kt4vl motherboard for an asus p4pe? i845? based motherboard and the echo has completly gone away along with aggressive suppressor option in the makefile. I hope this helps others. Brian J. Schrock Anistone Technologies, LLC 6926 Avery Rd. Dublin, OH

Hello, I have my sip stuff seemingly working fine as well as my zaptel stuff working great... But I have a problem with sip registration timers (I'm guessing here). In my extensions.conf file I have this... exten => 2244,1,Dial,Zap/2|25 exten => 2244,2,Dial,Sip/brian|25 exten => 2244,3,VoiceMail,u2244 But if I close my sip phone and a call goes through it will still wait the 25

I'm wondering if anyone knows of ISPs with service that has QoS features that would be good to use with VoIP stuff. Granted, the QoS would only be supported as long as you stayed within their network, but it might be better than nothing. --Eric

I have a simple problem with sialing a SIP device. I'm SURE it's a syntax problem, but I dunno what it might be. Here are the debug messages: == Accepting call on 'Zap/1-1' ("PENSACOLA, FL" <8503846785>) -- Executing Goto("Zap/1-1", "2111|1") in new stack -- Goto (default,2111,1) -- Executing Dial("Zap/1-1",

Hello, I have read the information on echo and SIP in the FAQ and I have scoured the mailing list for possible solutions, but as yet I have not been able to get rid of this echo. I have a Sip phone, Digium 4 Port FXS, and 3 Digium FXO cards installed into an asterisk server. If I call between the Sip Phone (Budgettone-100) and the 4 FXS ports everything sounds great. If I call out to the PSTN

Well, I think I discovered even further why there is no ringback tone available. The following message, is displayed on the console in asterisk. ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device Failed to register zone 'United States / North America': No data available Looking more into it, I found that it was related to loading tones for a particular zone. The message is printed

Hello, my English isn?t very good but I try to explain my problem. Hopefully you can understand me. :-) I have a Linux-Server with a Digium Wildcard TE110P. I install and configure zaptel. But I have an error when I execute ztcfg ?vv: 31 channels configured. ioctl(ZT_LOADZONE) failed: Invalid argument Notice: Configuration file is /etc/zaptel.conf line 288: Unable to register tone zone

HI, basic question. I've got a TE110P card and I'm trying to set it up with ztcfg with polish zone. ioctl(ZT_LOADZONE) failed: Invalid argument Notice: Configuration file is /etc/zaptel.conf line 206: Unable to register tone zone 'pl' I've got loadzone and defaultzone set to pl, and there is a definition of that zone in zonedata.c but it doesn't work. Any hints? tia

Hi All, I just installed Asterisk 1.0.5, and the installation went fine (I ran modprobe zaptel and modprobe wcfxo). However, when I ran ztcfg I get the following: ioctl(ZT_LOADZONE) failed: Invalid argument Notice: Configuration file is /etc/zaptel.conf line 135: Unable to register tone zone 'us' After that I ran Asterisk and it seem to started ok, except that it won't pick up

I have an ATA-186 in a SIP configuration (following Shawn Djernes how-to), but I get the following error at the asterisk console when I try to call the phone connected to the ATA: ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device Failed to register zone 'United States / North America': No data available Everything works if I remove indications.conf from /etc/asterisk -

Hello all, I recently noticed the realTime effort and must say it is a nice idea! I would appreciate any help to get it running .. I downloaded the code & patches and succefully patched my asterisk (CVS-HEAD-11/29/04-12). - created a DB called asterisk, and a table sip using the schema supplied at http://bugs.digium.com/bug_view_page.php?bug_id=0002613. - entered an entry: insert into

Hello, I'm running * on a very basic configuration. I have a Wildcard TE405P with the first T1 connected to a PRI line and the remaining three to Adtran TA750 channel banks with FXS modules. I successfully configured everything to work with a couple of Swissvoice IP10S handsets (MGCP) and analog extensions connected to the channel banks. The problem I'm having is that when I pick up any

Hi all, I am having a problem on to different boxes and to different channel banks. I can't get dial tone out of either one. I can still send and receive calls but no DT. This is the error that I get: Jun 8 15:23:54 WARNING[5021]: chan_zap.c:6283 handle_init_event: Unable to play dialtone on channel 95 asterisk-1.2.9.1 zaptel-1.2.6 zapata.conf: signalling=fxo_ls context=gene

Hi everyone, Sorry for forwarding and top-posting this email again but its as if my TDM40b has keeled over yesterday. After a few hours last night and swapping the card to another asterisk server (with exactly the same result) I needed to have the FXS ports working ASAP this morning so I have repaced the functionality of the TDM40b with some Grandstream handytones which I already had in

Hi list, I experiencing a strange behaviour when transferring a call. The use case is like this: - Incoming call from Zap/1-1 - Routed to SIP phone SIP/1001 - The called user (SIP/1001) wants to redirect the call and presses "#" - IVR (default setup) says "Transfer" and user gets dial tone - User dials 1002 - IVR says "No such extension - please try again" ??? It