similar to: [Fwd: Re: udev or not?]

Charles S. Antrim wrote: > I am using a card that has an fxo and fxs module. I am no where near an expert but I have my sip phone working through my pstn line and this is my config. /etc/asterisk/sip.conf [general] port = 5060 bindaddr = 192.168.69.1 context = sip disallow = gsm allow = alaw disallow = ulaw nat=disable srvlookup=no localnet=192.168.69.0/255.255.255.0 subscribecontext =

This may seem obvious or silly but have you tried using a different phone cord? A bad phone cord played havoc on a colleague of mine during initial config. Fro what you show as your output from ztcfg, you should have one channel configured successfully then. Your example shows you have the channel set to one so no problem there. If you run ZAP SHOW CHANNELS from the CLI what do you see? W

Juan, I dont see people here with "attitude" or any need to shut down this list as you suggested. Contrary to your experience, I find people to be quite helpful here. Steven's comment "Start here and then ask a real question" is definitely not arrogant IMHO. He is just trying to knock some etiquette on Luan. Besides, he did give him all the answer Luan was looking for.

Hey all, ? I'm trying to get an E400P and 2 X100Ps working together in the one box and don't seem to be having much luck. ? I can get the two different types of boards working separately, but not together.? I've made calls on both the X100P and have seen sync and correct signalling on the E400P.? But when I try to enable to configs together I get the following: ? modprobe wcfxo

I'm just getting started with asterisk on White Box Linux 3.0 I have a x100p card in the PC. If I reboot and immediately do a "ztcfg -vv" I get the well known "Unable to open master device '/dev/zap/ctl'. If I then do a "modprobe wcfxo" this error disappears and Asterisk runs fine. I've read the advice about modifying a file in /etc/udev. Well this system

Steven Critchfield <critch@basesys.com> wrote: __________ >On Wed, 2004-05-26 at 14:23, tpanton@attglobal.net wrote: >> On which subject, has anyone else >> got time to work with me on a chan_dialogicGC ? It looks do-able but I am ignorant of how asterisk does threading. > >Do you have GPL drivers for the dialogic card? Follow previous comments >about licensing.

I just updated to the new CVS and now I am getting the following error from chan_zap (modprobe wcfxo works fine): WARNING[16384]: File chan_zap.c, Line 577 (zt_open): Unable to specify channel 1: Device or resource busy ERROR[16384]: File chan_zap.c, Line 4781 (mkintf): Unable to open channel 1: Device or resource busy here = 0, tmp->channel = 0, channel = 1 ERROR[16384]: File

Thanks for the help. I was able to get my application to load with Asterisk, now I just need to get it to work. After reading your comment, I don't know that I fully understand what's going on as far as the channels and extensions. Are you saying that the MWI is tied to the channel? If that is the case, then for my SIP phones, would the channel be (SIP/XXXX)? If it should be, then

You should see something like this.... (I have 8 channels) tuxpbx*CLI> zap show channels Chan Extension Context Language MusicOnHold pseudo incoming en 1 incoming en 2 incoming en 3 incoming en 4 incoming en 5 incoming en 6

My thoughts on this after reading Steven's very politically worded reply is that IMO your best bet would be to go with RedHat, I am not going to go into details about the if's, when's, why's, and but's.. I am running Asterisk quite well on RedHat 9 and if you like I have created an install guide for setting up an Asterisk box on RedHat 9 which I can send to you if you are

Well I dont want it to operate as a group. I understand that it is *8 on most PBX systems but I would like it to work as dialing *8+extension-thats-ringing to have it go over to my extension. I am having a lot of trouble finding more information from the wiki and google. Thanks! -James -----Original Message----- From: asterisk-users-bounces@lists.digium.com

With the increased traffic as of late, I'm wondering if it is time to split the list again. Specifically I am wondering if it should be split along the various VoIP protocols and zap hardware, then leave a general list that does configuration other than VoIP related? The hope is that those asking SIP or H323 questions could get help from the various supporters while the main list can deal

Just some clarification here. On Monday, June 30, 2003 12:04 AM, Steven Critchfield [SMTP:critch@basesys.com] wrote: > > As for slips and bipolar violations... > T1s are just high speed serial lines. A sleep is when you loose sync > with the far side and when you see a 1 come across the line, you may > not > know which bit it was for. A slip is when one end of the circuit is

swar sir, can u please unsubscribe me for your list b.regards jihad chalhoub --- asterisk-users-request@lists.digium.com wrote: > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, > visit > > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message

At 11:08 AM -0500 7/12/04, Steven Critchfield wrote: >[snip] > >Would it be something people would like to be able to add filters to a >line? Consider normalization as a filter. Monitor could then be moved to >a filter as well. Echo cancel could be a filter. Set it up so multiple >filters could be added and chained together. This could help those with >echo chain a couple of

Hi, Every time I dial into my asterisk box i hear nothing but asterisk jittering. The following is an example of what I get on the asterisk CLI Thanks *CLI> DEBUG[81926]: File chan_sip.c, Line 3826 (check_user): Setting NAT on RTP to 0 DEBUG[81926]: File chan_sip.c, Line 4807 (handle_request): Check for res DEBUG[81926]: File chan_sip.c, Line 952 (find_user): Call from user

For reference... http://www.voip-info.org/wiki-Asterisk+zap+channels Not sure it is relevant but go ahead and remove the spacing on the channel line so it will read.... channel=>1 Here is the original incoming context you showed. [incoming] exten => s,1,Answer ; Answer the line exten => s,2,Playback,demo-thanks ;for playing a file The Playback looks malformed based upon the wiki

Hi All, Can we stop this thread pl. This lady has no intentions to learn asterisk. She is just a troll and wasting our time. With her corporate attitude, what she expects is support that available with paid commercial products. Her company has enough money to buy commercial products, let she go there. Hey lady, whoever u are, dont waste our time. this is not for u. Lets move on to something

Hey.. Your saying I can not use flash with ISDN ? What options to I have to transfer a call directly ? ( So I have a free line afterwords) >> What interface are you using? ZapBRI? if so you might be able to do the >> hairpinning if it is supported. Im not using any interface.. But if you know how to do that, let me know and I install that interface. Thx for your answer :) Gr?sse

Hi, I have already installed the X100P drivers but when I make to make call call, I always get No channel type registered for 'Zap'. Any idea how to solve this problem? It seems that the Zap module is not loaded but I have already load the driver by modprobe zaptel > modprobe wcfxo > ztcfg -vv thanks -------------- next part -------------- An HTML