similar to: Voicetronix

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, Does anyone know what may be causing this? Asterisk was built from cvs tonight. Ztcfg also says us is an invalid tone zone. Anyone got some information on what this is, why is it happening, and possibly some solutions? [root@anistonetech zapata]# asterisk -f -d DEBUG[1024]: File config.c, Line 653 (__ast_load): Parsing /etc/asterisk/asterisk.conf DEBUG[1024]: File config.c, Line 653

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)

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

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

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",

Hi All, Just a quick question, but I could really use some help on this one. I've got the CVS-Head of * installed and running, and am using a VoiceTronix OpenSwitch12 to connect to 12 analouge lines. I've got callerid activated by the Telco, and can get callerid using a std phone. However, using *, I always get an error 'Cannot decode callerid'. Does anyone know if I need to

Hi. I need help with my brand new Voicetronix OpenLine4 board that I installed into Asterisk. After building the Linux device driver and inserting the module, I modified the /usr/src/asterisk/channels/chan_vpb.c file to uncomment the US settings and comment out the Austrailian ones. I made the appropriate entries for routing in vpb.conf and extensions.conf.... All appears to be well, except

Hi we've got Asterisk CVS-HEAD 18-Aug-04 (modified by Voicetronix as available on their site for use with the vpb driver) and an OpenLine4 (4xFXO). The same server also has two X100P. Calls on the Voicetronix card drop instantly when the called party picks up. The vpb driver reports that it detected a hangup (loop drop) yet there is no hangup when connecting the X100Ps or analog phones to

hi there, i've been able to successfully run asterisk with the Voicetronix OpenLine4 card, it can accept calls and function normally. The only problem I'm experiencing so far is getting the card to outdial to a third party. What I'm trying to achieve is basically call bridging, where the caller dials in to asterisk, some IVR plays and then attempts to perform a "transfer"

dear Hackers, i have a voicetronix Openswitch card, and i have been finding it very dificult to get it to work with asterisk. i intend to connect 8 ports to the PSTN and 4 as station ports. problem 1: On running asterisk all i get at first i get : event[9=>[11] station OFF hook] on vpb/1-12 even [12=>[11] loop drop on vpb/1-12 event [12=>[11] Tone detect:GRUNT event [2=>[11] Dial

Hi all, Trying to get fax reception going using a voicetronix openline4 card, however, there are two issues (as far as I can see) (1) Currently the voicetronix world (cards, firmware, drivers, channel prog etc.) does not do fax tone detection. I have spoken to the voicetronix guys , understand what has to happen there and may do some hacking if I can resolve the second issue. Meanwhile, I

Hi all, I used to have an OpenLine4 card, but decided against using it due to some problems with hangup detect. Does anyone on the list actively use Voicetronix's OpenSwitch12? What are your opinions on the card? Cheers, Flynn

Hi there, Besides the problem of Voicetronix dialing too early before the carrier gives a dial tone, there also appears to be issues with the audio quality when more than 1 channel is utilized.

Hi there, I am trying to delay sending out DTMF from Voicetronix OpenLine4 to the CO line. The reason being is that Voicetronix sends out the DTMF too fast even before the line is fully established with the carrier. Usually when dialing an 8 digit number, only 7 digits are actually successfully heard by the carrier. Currently, my dial plan is: exten => _9.,1,Dial(vpb/1-1/${EXTEN:1}) Daniel

Since updating * via CVS earlier this week, I've been having problems with cdr_mysql. Prior to that time my queries and cdr all worked fine. Now, even though my queries still work, I get the messages similar to this: ERROR[1211374384]: cdr_addon_mysql.c:203 mysql_log: Failed to insert into databas -- Hungup 'Zap/74-1' Anyone have any suggestions what went wrong?

Any guidance on how to get my openline4 to get recognized by asterisk@home? I've got my vpb drivers running, but not sure how to add it as a trunk, should it be via zap? or is there another way? Thanks, Chuck

Hello, Hacker I install VoiceTronix OpenSwitch 12 port PCI Telephone Card, and setting vpb.conf, extensions.conf following My problem is: When i dial to fxo(channel 9-12), it is ok, but when i continue press exten 102, the channel crach with error messages following exception caught: VPBAPI_DIAL_INVALID_LEVEL, file: vpbdial.cpp line:872 Do i ignore some setting for VoiceTronix OpenSwitch12