similar to: Caller ID problem

I have a new asterisk installation (1.4.2) that is working fine with SIP. Now I'm trying to add 2 cisco ip phones (7960) running SCCP (latest chan_sccp). I have the phones booted, and the tftp directory all setup, etc. But the phones do not quite work right. When I lift the handset I only get a dial-tone 1 out of 5 or so times I try, though hitting the speaker button works. I can dial

I have looked all over the site(s) for help. But heres the problem. Im missing something. In coming works fine from FreeWorld via IAX. But when Dialing out i get: May 13 13:42:01 WARNING[1150495040]: chan_iax2.c:5256 socket_read: I don't know how to authenticate iaxtel to 65.39.205.121 my IAX.conf if as follows [general] port=5036 register => ######:xxxxxxxxxxxxx@iax2.fwdnet.net

Hi, I am posting my question to both storage-discuss and zfs-discuss as I am not quite sure what is causing the messages I am receiving. I have recently migrated my zfs volume from b104 to b134 and upgraded it from zfs version 14 to 22. It consist of two zvol''s ''vol01/zvol01'' and ''vol01/zvol02''. During zpool import I am getting a non-zero exit code,

We have antiquated caller ID schemes here in Australia. We barely support numbers from other local carriers, let alone OS ones. Certainly no names either.

I got a call from an overseas call center telling me about the problems with the Windows machine I was using. They wanted to remote in and fix things for me ... (Ignore the fact I use a MacBook Pro or an ASUS laptop with Debian). What I found curious was the caller's name was Asterisk, and the caller's number was Asterisk at 10 or or Astrk at 10 similar. (I don't recall the exact

Caller*ID used to work as some point, but I can't seem to get it going these days. The card is a x101p. I've tried going up and down the rxgain scale. Can the txgain effect it at all? When I plug in a phone into the line with a splitter it can decode caller id with no problems. Reading through the mailing list archives hasn't given me any move clues. Any ideas?

How do I make Lithuanian characters display correctly in R graphics? Instead of the special characters for Lithuanian language I get question marks... I use Ubuntu Feisty, the locale is utf-8 ... Do I need to specify somewhere the locale for R, or - default font for the graphics? -- Donatas Glodenis http://dg.lapas.info

Since the X100P/X101P/Clone cards does not work in all countries that use DTMF based Caller-ID systems, I've developed a hardware that you connect to a serial port and the PSTN. You then run a perl script "cid_logger.pl" as a daemon, and modify extensions.conf to call an agi script whenever a call comes in, and if it's on the X100 card it will get the caller id information

dear all, i am looking to enable CALLERID on an Asterisk system comprising a X101P FXO interface connecting to BT PSTN in the uk seems this is supported by the interface but there seems to be varying information on how to enable it in zapata.conf 1. usecallerid=uk 2. ukcallerid=yes being two of the configuration statements offered TIA GT

Hello! My Dom0 config: Xubuntu 8.10 + "Xen-3.3.1 and kernel-2.6.18-xen.hg" I have Nexenta CP2 RC1 (Opensolaris b104) on ZFS file system. I installed this OS in HVM. For PV i used pygrub. My config file: ''ncp2.pygrub'' bootloader = "/usr/bin/pygrub" boot = "c" memory = 512 name = "NexentaPV" uuid =

Hi All, Having searched the archives, I can see there has been much discussion at various points regarding capture of caller id information from good old BT. If I understand correctly, it seems that not only do the drivers not currently support it, but my X101P possibly/probably can't do it anyway due to hardware? So, that leaves me with the modem route, which seems more and more unlikely,

Has anyone gotten CID from Bell Canada to work properly with *? We have our * box down at our datacentre in St Louis, and whenever we call it from a Bell Canada Telephone line, all we see is '' for the CID. I did some digging on google and the mailing lists and couldn't find anything pertaining directly to Bell-Canada and * CID, but didn't find much. I did however find :

Hello! Can i get uuid of DomU from this DomU ? What is needed? Thanks. _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

Hello all, I was wondering if anybody knows where one might obtain a PCI ISDN card supporting a single BRI for use with Asterisk in Australia (and using something like chan_capi). Because of the Isdn4Linux DTMF issue, I don't want one of those cards. I've already spent too much time messing about with my current card. I'm after something like the AVM Fritz! cards. I found one place

can somebody that has these hardwares(X101P and S400P) working on his asterisk system please assist.................. you can send the solution to austino@skannet.com..... the error message is what i have below. ---------- Forwarded message ---------- Date: Thu, 1 May 2003 21:01:21 +0100 (WAT) From: austino@skannet.com To: asterisk-user@lists.digium.com Subject: Error working with X101P and

Wow ! No reply... May be I must talk about X100P instead of X101P ? Is someone has yet encountered this kind of "no sound" problem when bridging two FXO lines like this (first as input, second as output) ? Any idea ? TIA. Best Regards, Francois BERGERET, France. ----- Original Message ----- From: "Francois BERGERET" <f6hqz-m@hamwlan.net> To: "Asterisk Users

Hello, I would to allow users to place calls overseas such as India and Malaysia but only with a security code. if they don't have a security code I want to be able to drop the calls. can someone point me to a right direction to achieve this goal? Thanks, Motty -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a friend traveling overseas. I want to allow him to call a number which will give him a busy signal (so no charge), but will then send me an email that he has called. I know how to use a call file to trigger a call (I created a callback system for myself when I traveled overseas a few months ago), but I don't know how to trigger an email. All I want is a simple message like

When x101p clone receive ring signal from phone line, my voip phone start ringing. But, if caller hang-up at some time, phone continues to ringing 10 second more. How can I shorten that time? Pause betwen incoming rings on my phone line is 4s, so when x101p clone (wcfxo driver) do not receive next ring signal after 4.5 sec, call should be consider as ended. What should I change to set that time

I'm looking to get an x100p off ebay and am not particularly familar with the "life cycle" of the card.. An "Authentic X100P" listing has a buy it now of $29.95 and says There are 3 types of cards Asterisk would recognize: *Screenshots from the official, original driver install Cheap "OEM X100P","Clones", "Compatibles", Knock-Offs