similar to: Re: Asterisk-Users Digest, Vol 13, Issue 123

Hello All I read in www.sineapps.com have Asterisk 2.0 rewritten C# and run on windows, any body could be mail or send to me URL to download. Thanks Tin Trung Nguyen Technical Team Mobile: 084-91.365.4857 website: www.daivietcontrol.net -------------- next part -------------- An HTML attachment was scrubbed... URL:

My TDM400P w/ 4 FXO cards seems to have trouble with onhook/offhook switching. It dials perfectly, but does not seem to be changing the onhook/offhook state appropriately. It changes sometimes, but it's not really reliable. For example: When I booted the machine, it started as onhook. It remained "onhook" through the entire first call (which was silent on both ends --

> -----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

>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

On Wed, Jul 23, 2008 at 11:24:49AM +0200, Karel Zak wrote: > On Wed, Jul 23, 2008 at 04:43:30AM -0400, Christoph Hellwig wrote: > > On Wed, Jul 23, 2008 at 10:39:38AM +0200, maximilian attems wrote: > > > klibc mount has only short options thus uses the following syntax > > Frankly, it seems like a klibc problem... well not directly, but right klibc-utils should have

Is it just my box, or is there something flaky in the implementation of "show manager commands"? Note: I'm using putty. About half way through this, I toggled my KVM over to the desktop and logged in to try and recreate it. The output was the same as the last two entries in this dump. bebop*CLI> show manager commands bebop*CLPing Ping bebop*CLLogoff Logoff Manager

Hello, Caller -> PRI -> SIP -> application. If application goes offhook right away, dtmf/audio works fine in both directions. If application, before going offhook (sending OK) plays a message and wants dtmf/voice from the caller, then caller can hear this message but his dtmf/voice don't reach application. Any way to configure it? Thank you. Alex Zarubin -------------- next

There isn't much documentation on adsi, but I called NETXUSA (the vendor of my 480e) and they helped me along. My experience: 1. I really had no experience with ADSI so I had (probably still have) some misconceptions on how the configuration is loaded onto the phone. 2. I set the following in my /etc/asterisk/asterisk.adsi (most of this is the stock asterisk.adsi script): ;

Hello All How to detect remote called offhook. i make a context as below i created call file. copy to /var/spool/asterisk/outgoing. Channel: vpb/g0/9050718 MaxRetries: 1 WaitTime: 10 Context: ext-callout Extension: s Priority: 1 then when i copy to /var/spool/asterisk/outgoing. the asterisk auto call and wait for 10 second, auto playing wave file in context ext-callout, exten s, without

Hi With an FXO module + Zaptel, I'd like to know if there are ways to know when the remote party has answered the phone, whether calling through a callfile or by sending DTMF's. I read about {CHANNEL(state), ChanIsAvail(), and ${DIALSTATUS}: Are those reliable ways to know when the channel is available for dialing out and the call has been answered?

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

Hello All ! i'm purchased sangoma card A-101. i connect to E1 with MF/R2 signalling. but card don't work. negotiation with E1 fail. please help me to correct it. i dont' know some parameters such as: MTU, BAUDRATE Thanks Tin Trung -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Thanks. I know that line quality is a factor, and I know I could get a 50$ fax with a PSTN line (that is what I have now). But I have my reasons to want to setup a fax over IP, and I want to keep going. Where do I find info on this debug mode? Is there a detaild log in Asterisk that show exactly what happens when the fax is trying to come in? Also, could this console output help? - Executing

klibc fails to compile on sparc64 since the kbuild switch: In file included from include/sys/mman.h:11, from klibc/malloc.c:8: linux/include/asm/page.h:18:2: error: #error No page size specified in kernel configuration the current dirty build hack is to define CONFIG_SPARC64_PAGE_SIZE_8KB in the sparc64 MCONFIG. belows patch removes the asm/page.h include. another way to fix

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

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

Hi, We are having a problem with asterisk detecting that an analog ext has been put down. This seems only to happen after a number of calls have been made. We have an FXO port (TDM400P with FXO module) connected to our PBX and are using this to test asterisk prior to rolling our for our small office. What happens is that we make a number of calls to this ext which 1st rings a phone (FXS)

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

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