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Hi, does anyone have any recommended (read tried and tested) way of making asterisk be able to handle incoming faxes. I've a PC running asterisk with a digium E1 card in and simply want to be able to route a call to some application which will take a fax call and save the fax as an image. I guess the bits I'm unsure of are how to terminate a fax call without a modem (if this is

Hi, I am struggling to get rid of a conflict on DELL 2800 : PCI Parity error (EB113 on the display) I am learning linux and asterisk as I go along, there might be obvious things I should know, but bear with me. >From demsg below my 2 digium cards installed are listed (no config or connections done to digium cards yet), the conflict is with the TDM400P card, without that card, in any slot, no

mdew:~# tc qdisc add dev imq0 handle 1: root htb default 12 r2q 1 Cannot find device "imq0" mdew:~# lsmod Module Size Used by Not tainted ipt_REDIRECT 728 0 (autoclean) ipt_MARK 728 2 (autoclean) iptable_mangle 2100 1 (autoclean) ipt_REJECT 2712 4 (autoclean) iptable_filter 1672 1 (autoclean)

Any thoughts on the following? I am running asterisk from CVS (downloaded yesterday's version, just to be sure) on a test system with no digium cards in it, so I have installed ztdummy (see logs and screenshots below) as a timing source. When I call the music on hold extension from a Sipura Sip connected analog phone, I hear nothing and start getting Warning[98310]: chan_sip.c:674

zaprtc 'load' is failing on my machine (the make was fine, same output as other posts to this list) [xxx@xxx zaptelrtc]# make load sync modprobe zaptel insmod ./zaprtc.o ./zaprtc.o: init_module: Input/output error Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or

Hello, I am posting this after spending hours digging through the list archives. Problem : When asteirsk plays a voice prompt, the voice clip is really choppy. I figure that this is something to with the sound card, the timing of playback etc. But cannot seems to find an answer. Here is the Notice which appear when voice prompt is played. NOTICE[1217602880]: File sched.c, Line 209

I have had a web server listining sql-1433, www 80, ftp-21 using proxy arp with sub-netting in a three interface DMZ. All these ports are in the rules file as ACCEPT. With one exeception that 1433 allows a few host from the net. 21 and 80 allow all net to dmz connections. The policy is DMZ to net ACCEPT This has been working great for about a month or more until I rebooted the

Hello everybody, After going through multiple posts on Internet and trying different things I can't seems get zaptel wcfxo loaded on the server correctly. I run dual P3 500 Mhz box. After reading README.udev I put a file 60-zaptel.rules /etc/udev/rules.d # Section for zaptel device KERNEL="zapctl", NAME="zap/ctl" KERNEL="zaptimer",

I have 1 * box having x100p installed and the other has no zaptel card at all. both of the * box has compiled in ztdummy module and both have been activated by modprobe ztdummy. When using trunk to connect the 2 *box. The one without zaptel card complaint about unable to accept trunked packet: no matching peer. On the one has zaptel card I have tried to remove the ztdummy module and connect

Installing DevkitLite hardware (Very similar to John Lange's post on Tue Oct 08 2002) I cannot get anything to work on the phone connected to the s100u. I dont know what to do. Can someone please help me? I used the sample configuration files from digium documentaion that was supposed to be "sane" defaults for the kit. Very similar to John Lange's post on Tue Oct 08 2002 Here

Has anyone ever setup Asterisk to pass Feature Group D access while using a CIC code for outbound calls? If so can you please email the configuration you have done? I have tried to get this up and running but with no luck. I have also contacted support and I cant seem to get this going. Thanks in Advance, Jason Miller

Hello all, I hope this is not off-topic, if it is please let me know. I'm currently playing with an Asterisk at home, in order to get to know it's ins and outs. Very very impressive indeed. I've got it hooked up to my home phone line via a Wildcard clone board (Intel modem with Ambient chipset), and it works like a charm. Zaptel picks up the card as a generic clone, and works with it

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 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

On my debian woody (kernel 2.4.31) the tcpdump doesn''t work with imq0 devices. If I try to tcpdump imq devices there is no packet seen: [...] rt1:~# tcpdump -n -i imq0 Warning: arptype 65535 not supported by libpcap - falling back to cooked socket tcpdump: WARNING: imq0: no IPv4 address assigned tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on

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

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

Does anyone keep a known telemarketer caller id database? If not has anyone proposed an Asterisk community project to share this information? Sort of a nation wide blacklist so Asterisk'ers can cut down on the garbage calls... -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

https://bugzilla.netfilter.org/cgi-bin/bugzilla/show_bug.cgi?id=68 ------- Additional Comments From laforge@netfilter.org 2003-03-20 10:55 ------- This looks strange. The BUG in slab.c tells us that there is a GFP_ATOMIC missing. This means that we are allocating kernel memory from softirq context with only GFP_KERNEL. If I understand your backtrace correctly, what happens is: - you are