similar to: received wrong state events for originate command

I was cleaning and working on laptops most of the day. Check my logs, I did plenty of work. -----Original Message----- From: "asterisk-users-request at lists.digium.com" <asterisk-users-request at lists.digium.com> To: "asterisk-users at lists.digium.com" <asterisk-users at lists.digium.com> Sent: 11/29/2008 1:13 PM Subject: asterisk-users Digest, Vol 52, Issue 81

Hi, I have the setup of my xp100 plugging into my dlink gateway that i use with a voip provider. I notice that when someone calls my pstn # that goes to the asterisk box it works but when they hang up asterisk does not recognize the hangup. What needs to be done to make it work with a dlink gateway. when i plug the xp100 into my land line jack everything works fine and the xp100 reconizes

Wanting to connect my asterisk box off of 2 unused analog extensions on the non* PBX system. Can I bring those lines in to * on XP100 FXO cards? Any special wiring/open loop issues to watch for? The desired config is inbound PRI -> non* PBX, attendant picks up and if caller wants services on * box then xfers call to one of the analog extensions where * picks up and handles call from that

Has any one found any issues (bug?) with ver 1.2.5 as CallerID generated on an outgoing FXS port (to the handset) fails when UK tones are used, with a message 'Didn't finish Caller-ID spill. Cancelling.' Any tips on getting this running ? Looked at Mantis, but only known bugs seem to relate to XP100 cards - not the TDM card. Thanks

Has any messed with getting Asterisk to work using the Voice Pulse Open Access plan? I currently have 2 numbers with Voice Pulse, 1 is a number that is assigned to their hardware device (Sipura SPA-2000), the other is a Open Access number that uses SIP from any device (you must authenticate with them). I want to be able to use the Open Access number on my Asterisk server here at home with no FXO

Hi John, I've been using Packet8 via a physical ATA and XP100 card for some time. As far as I know it is not possible to connect to the Packet8 service without the ATA. If this is not the case I would be very interested to hear this. In addition since moving to the USA I now only have a single packet8 line into my asterisk box (I used to have this and a 2nd regular pstn line) I used to be

> Hello, > > I would like to implement a home GSM gateway using asterisk. What > would you recommend me as a low-cost hardware for creating a gsm > channel? I found 2n gsm gateway, that supports sip and chan_blue for > bluetooth connections. Any recommendations? > > Basically, I want to end calls to some GSM number in my sip > telephone and for some prefixes dial

Hello all We succesfully added a H323 Gateway to our CallManager 3.0 that resides in Mexico and were/are able to make calls from CallManager SCCP phones to the Asterisk Server phones in the U.S.; however, we have not been able to call from Asterisk server in U.S. to CallManager phones in Mexico Here is what we tried: 1. Adding a Gatekeeper into CallManager and then have Asterisk (and also

Hi, I'm getting errors while originating a call through AMI. [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() returned error: Broken pipe [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() returned error: Broken pipe [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() returned error: Broken pipe Asterisk version 11.0.1

Hi Guys, I am trying to make a web form where a person is allowed to put in $phoneNumber, $dialNumber, and $spoofNumber to make a call with spoof caller ID. There are a few problems that I am facing with Asterisk AMI Originate command. The reason why I want to use the darn AMI Originate is because I am sending calls to mobile phones and I want to have some accountability and to know if a call was

Hello Everyone, I am making a simple index.php file which will allow a web user to enter his $phoneNumb, $dialNumb, and callerID ($spoofNumb) and get the call bridged. Following is the index.php and the contents of extensions_custom.conf. When I submit the form nothing happens. I don't even see Manager Connected msg. Your input will be much appreciated. I am thinking I have some syntax

I found this tracing a bug when experimenting with a new sort() function using match.arg(). It was triggered because mosaicplot.default(.) has an argument called `sort' and calls itself the sort() function in which I was using match.arg()... Here is (input for) a small clean example : #### bad match.arg() // sys.function() bug : #### MM, 22.3.2002 callme <- function(a = 1, mm =

I downloaded R v2.1.1 earlier this morning to compile under Fedora Core 4. It compiled without incident, but 'make check' failed. Below is the relevant part of its report. Is this a known problem? I used a locally compiled version of GCC v4.0.0 that reports [kent at d89h102 R-2.1.1]$ gcc -v Using built-in specs. Target: i686-pc-linux-gnu Configured with: ../gcc-4.0.0/configure

Sorry for this long message. The two first questions asks for general coding standard when creating packages, the third one asks if the .R files are concatenated in lexical order or not, the fourth question is a "how-to" question. All questions are related. If there is a reference where I can read about this please tell me, because then I could repost a more restricted set of questions.

I downloaded R v2.1.1 earlier this morning to compile under Fedora Core 4. It compiled without incident, but 'make check' failed. Below is the relevant part of its report. Is this a known problem? I used a locally compiled version of GCC v4.0.0 that reports [kent at d89h102 R-2.1.1]$ gcc -v Using built-in specs. Target: i686-pc-linux-gnu Configured with: ../gcc-4.0.0/configure

Hi, I am having trouble passing variables via the call files, here is my call file via the php: fputs($oSocket, "Action: login\r\n"); fputs($oSocket, "Events: off\r\n"); fputs($oSocket, "Username: $strUser\r\n"); fputs($oSocket, "Secret: $strSecret\r\n\r\n"); fputs($oSocket, "Action: originate\r\n"); fputs($oSocket,

Hi, I have cvs updated all my modules (zapata, libpri, zaptel and asterisk). I have also read in the archives & seems that no-one has run into this problem. What I'm trying to do is simple. Just make and outbound call using the /var/spool/asterisk/outgoing directory. I copied /usr/src/asterisk/sample.call and only changed the context & extension. I configured my Zap1 to the same

*Good morning! * *I verified that the trixbox version Trixbox 2.6.1.13 has support for OpenR2, I verified in the repository that has to libraries of the project openR2, but I don't manage to do to work in the trixbox, when I type the command (it colors show channeltypes)ele no demostra the support to MFC+R2, they could help finding out which package is necessary of the trixbox and which the

Hi this is my code for * manager: $oSocket = fsockopen($strHost, 5038, $errnum, $errdesc) or die("Connection to host failed"); fputs($oSocket, "Action: login\r\n"); fputs($oSocket, "Username: $strUser\r\n"); fputs($oSocket, "Secret:

> -----Original Message----- > From: Ronald [mailto:asterisk107@gmail.com] > Sent: 26 May 2005 10:47 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Little Php question > > > Hi > I'm trying to make a call from a local webpagee through my > xlite softphone > (xlite1) > BTW when I'm trying to do it through