similar to: gap in priorities - what happens

The FXO lines on my TDM400 are connected to PSTN lines in Israel. As far as I know, the lines around here have disconnect supervision (I've seen some other Israelis on this list, anyone know for sure?), because it's worked on Dialogic cards, which reported hangup, not busy detect (while when I connect a Dialogic card to a PBX, I have to measure the busy signal's frequency/cadence or

I have a Linksys PAP2. Identical setups for the two channels in both the unit and in Asterisk. In particular, both channels enable g729 and set it as the preferred codec, and have disallow=all and allow=g729 in sip.conf. If we make a call on one channel, it works (and uses g729), but if we make a call on the other channel when the first one is still connected, it fails. We have three g729

Hi all, we have a problem with a reinvite sent by our SIP provider to change audio codec due to the recognition of a fax tone. After that the SIP call session has been established (INVITE and 200 OK) we have the following codec situation: UAC ASTERISK UAS | ASTERISK UAC PROVIDER g711 <----------------------> g711

> -----Original Message----- > From: Jacqueline Lee [mailto:jlee@isdomaininc.com] > Sent: Friday, September 23, 2005 11:46 AM > To: asterisk-users@lists.digium.com > Subject: channel offhook state > > > We are using a digium card (TDM400) with asterisk for our access to the > PSTN. Initially when the server starts, all the zap channels on the card > are in the

I'm not having any trouble with interrupts, but here's my /proc/interrupts on Fedora Core 2 on a hyper-threading CPU and using the SMP kernel (2.6.5-1.138). I don't think I need to worry about uhci_hcd, nothing is plugged into USB, but libata is the disk driver. How do I get libata and wctdm to use different interrupts? $ cat /proc/interrupts CPU0 CPU1 0:

I have a PRI that if you dial a number that is busy, the channel does not hang up, it then sends "h|1"to the phone company which will then plays back to the end sip user "You don't need to dial a one or zero" I am running stable CVS-v1-0-01/20/05-02:45:17. I have placed the important bit from the extension and sip configs below. Simplest possible example that will show

[cc'ing back to r-help] On Fri, Nov 18, 2011 at 4:39 PM, matteo dossena <matteo.dossena at gmail.com> wrote: > Thanks a lot, > > just to make sure i got it right, > > if (using the real dataset) from the LogLikelihood ratio test model1 isn't "better" than model, > means that temporal auto correlation isn't seriously affecting the model? yes. (or

I have: RedHat 9.0 TDM40B asterisk-0.9.0 compiled from sources zaptel-0.9.1 likewise /etc/zaptel.conf contains fxoks=1-4 loadzone = us defaultzone=us loaded modules zaptel and wcfxs /etc/askterisk/zapata.conf contains [channels] language = en signalling = fxo_ks context = phones channel => 1-4 /etc/askterisk/extensions.conf contains [general]

I have an interesting problem. I am attempting to install a payphone utilizing a Cisco ATA-188. The payphone actually works, but there are some timing issues. What happens is you pick up the payphone and the ATA grabs a line and goes offhook. While you monkey with putting money in and dialing the number, you are eating up the time before you get the offhook reorder tones (or howler tones

I would like the ability for a sipura sip device to instantly connect to an asterisk server as soon as the sipura sip device goes offhook and before any digits are pressed. This way asterisk can provide the dialtone and the dialplan. This also allows me to play a greeting to the phone before giving them a dialtone. Is there any way to do this, like possibly having the sipura device dial a

I've noticed that some callers listen to our main menu and don't press any keys. I have it set up to restart the menu a few times and eventually hang up. I'm wondering if these are wrong numbers (in that case, why don't they hang up) or they really want to speak to someone here but don't understand the menu (what's so hard about "for the operator, press

I am trying to put together a list of 'departments' to request as voice prompts. I have the biggies (sales, accounting, shipping, etc...) but I want to make sure I do not miss any. If anyone anyone has some suggestions (Ha... that is like going to an NRA meeting ans asking if anybody has a gun :-) ) please forward them to me (and / or post here although, with the volume of this

Happens quite often. X100P FXO card puts the PSTN line offhook, so that no calls go out or come in. The outside callers get a busy siganl while inside callers cant dial PSTN. Its a DELL optiplex P3 128MB ram 500MHz processor. Here is some more info: (see the zapata.conf in the end) Please direct me where to look for problem. Thanks!!! ======================================== pbx1*CLI> zap

im getting: Fatal error: Unknown function: mssql_get_last_message() in /var/www/html/tikiwiki-1.8.2/lib/adodb/drivers/adodb-mssql.inc.php on line 415 to the wiki.. Jason

I have an Asterisk@Home 1.3 server (Asterisk 1.0.9) and recently added a TDM400P with (1) FXO card on port 4. Inbound calls are always successful but outbound calls fail 75% of the time with intercept messages from my dial tone provider that include "we're sorry, your call did not go through", and "we're sorry, when placing a local call it is now necessary to dial an area

>Asterisk and it works fine untill the following >situation: > >- one of the telco lines occasionally becomes mute after call is completed, would not provide dial tone, (not sure about ringing on that >line) - both via old and new PBX. >- zap show channel <n> would show that line as 'Offhook', though no telephone is off hook. > >If physical line would be

Does anybody know if there are any external telephone-keypads for sale anywhere? (containing the keys 0-9, *, # and onhook/offhook would do) I am looking for a keypad to control a softphone and would prefer the controls to be in the physical world instead of as a window. Sincerely, Markus Hakansson

I have a TDM400p with 3 FXS mods and 1 FXO mod. I have all set up with what seems to be correct settings (according to digium and asterisk wiki). As soon as I plug in my POTS line into FXO mod the line goes into offhook state (whether I have power to the card or not). Should this happen? When I try to call * box all I get is busy signal. I've installed stable version, cvs version, change

I am searching for a way to add a 2 second delay before calling out with Dial(). Sometimes I get the message "you must first dial a 1 to place this call". I presume the phone company is missing the first digit pulsed out sometimes. How do I put a 2 second delay after coming offhook and before dialing the digits? Thanks, jerry

I've been having trouble with making outbound calls to my TELCO from a TDM400 card (FXS KS signalling) after upgrading from 1.6-beta9 to 1.6.0. The problem is completely intermittent. When it fails, I get this message: [Nov 9 19:12:26] WARNING[18916] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) At some point, it starts working, but I don't know what