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Hi, I just got my TDM400P card (2 modules) and i installed it no probs. The card is detected fine, but for some reason when I add the card to zaptel.conf i get the following error: --snip-- ZT_CHANCONFIG failed on channel 2: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? --snip-- My /etc/zaptel.conf looks

Does anyone know a place in Australia that sells a power supply suitable for the IAXy? I haven't had any luck tracking one down. -- jeremy bogan [ jeremy@segpub.com.au ] segment publishing - design.develop.host

hello all, I am using the msql cdr module to store cdr in db, I realised that it does't capture the start and end time af a particular call record. Therefore I dive into the source code to add the start and end time into the query (add something like cdr->start, cdr->end), but end up getting segfault. the original version of cdr_mysql.so works fine but I need the start time and end

Hello I Installed Asterisk on RedHat 9. I am currently try to configure minimum with two softphone talking to each other over the LAN. I am using X-Lite softphones from xten.com site. I defined 3 phones in sip.conf and also specifies in extensions.conf file. I am able to ring other computer but there is no voice exchange ( i can't hear any think except ring). Here is the portion of sip.conf

I have problem to change from handsfree mode to handset mode. When I switch from handset to handsfree while waiting for connection I press the green speakerphone button once. It is all well. Once it is connected I don't want to give the called party too much echo and I want to switch it back to handset. If I press the green button again I lose the call. Anyone knows whether it is possible

Hello I am trying to setup Asterisk on 2 servers PBX300 and PBX200. PBX300 has X100P card with 1 telephone line. PBX200 don't have any Zap device. Softphone from PBX200 can talk to softphone on PBX300 but no outgoing call from PBX200. I can call from PBX300 outside but I am unable to configure soft Phone defined in PBX200 to dial out side using PBX300 Zap devices. I am geting error

For some unknown reason, two of my IAX peers started registering on strange ports. Nothing has changed in the config, but they cannot make calls to me, however I can still make calls to them. In my IAX2 peers, the following is showing: user1/user1 203.XXX.XXX.XXX (D) 255.255.255.255 4585 OK (21 ms) user2/user2 203.XXX.XXX.XXX (D) 255.255.255.255 11280 OK (37 ms) I've

I have two grandstream budtetone-100 and cisco 7960g phones. When I talk via speaker phone on either of the phones I get a lot of echo. Any suggestions? Also how do I turn on the mark echo canceller. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Tuesday, April 13,

Hi All, Something really weird is going on. I just stopped and started my asterisk server, and it just decided to stop starting. I only made one change to the config file for the musiconhold.conf, and now for some strange unknown reason, i'm getting the errors: [chan_modem_i4l.so]Apr 28 15:26:42 WARNING[16384]: loader.c:240 ast_load_resource:

What happens when you do "stop now" like the error states? Sean -----Original Message----- From: Ryan Parlee [mailto:listbox@jesca.com] Sent: Saturday, April 03, 2004 9:56 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Unabled to exit console No matter what I try, Asterisk won't let me out of the console. If I CTRL+C, of course, the process will terminate. I

When I try to start up zaptel, whilst running ztcfg, I get the following error: Jan 1 10:48:18 bu ztcfg: ZT_CHANCONFIG failed on channel 2: No such device or address (6) My /etc/zaptel.conf is: fxsks=1 fxoks=2 loadzone = au defaultzone=au Channel 1 is a X101P card connected to the PSTN and channel 2 is a S100U box driving an analogue phone. The zaptel kernel module gets loaded OK as does the

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

I have looked though the archives on this issue and I can not find a final answar on it. So I will bring it up again. I have an x100p card.....if Asterisk picks up a call, and it see if there are fax tones on it, and if so transfer it somewhere. What is a user has picked it up. Can it still transfer it, or would the user have to? Thanks again, Michael

Does anyone know what VoIP phones are available in australia for 'decent' prices (ie, decent for me is under $250 - $300 max) So far I've found the following: D-Link DPH-100S $620 Various Cisco phones at > $500 each ZyXEL Prestige 2000W $380 Haven't been able to find much else around. I would like to get at least one grandstream but the delivery charges will probably add up

Hey guys, Cal someone help me. I'm register voiceplus DID i try to config fllow example but not work. When i test call to number and debug iax2 in my asterisk not found packet. My iax.conf -------- register => in-xxx:yyy@gw5.voicepulse.com [voicepulse] context = voicepulse-incoming secret=yyy auth=md5 type=friend host=gw5.voicepulse.com ------ extention.conf ---- [voicepulse-incoming]

Hi, I have two E100P boards connected to my PC. I wish to setup two E1-ISDN-PRI lines, the first SPAN with 15 channels and the second one with 30 channels. When I try to load zaptel modules, I get an error message: "Loading zaptel framework: Loading zaptel hardware modules: wct1xxp wcusb Running ztcfg: ZT_CHANCONFIG failed on channel 63: No such device" Follows my zaptel.conf:

Hi there, I've got an installation where there's 12 POTS line incoming into *, and am trying to get some insight as to which VoIP hard phone would be most suitable for this scenario. Most of the VoIP phones I've looked at only have 4-6 line presentations; is anyone aware of one that has more? I tried to get some info about Snom's Keypad 220 since it has loads of programmable

Hi, I'm cutting my teeth setting up a home asterisk server, but having some trouble getting Zaptel to recognize a X100P clone PCI (Motorola SM56 which should work according to various wiki's). What struck me was that the "/proc/interrupts" do not include IRQ 11 as the "lpcpi -v" is reporting. Am I missing something fundamental here? Any hint is greatly

Hi, I have been trying to figure out why the S100U is not performing very well on RH9.. Here is my thinking..( may be totally wide of the mark but here goes anyway) I remember reading somwhere that the sound system used by RH has changed... Does the S100U not depend on the sound subsystem?? So what I think is that the sound subsystem in RH9 and the S100U are not happy working together.. Does

Does anybody developed Predictive Dialer using Asterisk/Digium PBX? Another question: does anybody developed an Dialer using the X100P board? Julio -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030718/be051d49/attachment.htm