similar to: Incoming phone dialing / IXJ

Hi all, Currently trying to get asterisk to dial out with an Internet Line Jack card, however, it does not use the pots line, only on the line it dials out of. This is similar to the previous thread/posting "Asterisk won't answer pstn ring", but I didn't find any follow up to get it working. My asterisk setup is like this: iptelephony:/etc/asterisk# cat phone.conf | grep -v

My thanks for previous help on fixing IXJ issues in 1.2.40; I now have problems with a just-built 1.4.35 on the same host: [Aug 18 17:26:48] WARNING[27209]: app_dial.c:1298 dial_exec_full: Unable to create channel of type 'Phone' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) O/S: Linux 2.4.27; IXJ driver for Linux Rev. 3.5, gcc 3.01 I applied the patch for

Greetings: We have been running a CVS HEAD version of asterisk from Mar 10, 2005 on ix86 (PIII-600) Linux 2.4.27 with ixj (chan_phone) hardware. In a hope of getting better 'chan_skinny' support (to attempt using a Cisco 7920 IP phone) I built asterisk 1.2.40 on this box. Initial tests verify that our previous dialplan is working (iax2 trunks, register sip phones, registering withour SER

Hello All... I have searched in the archive and also followed Zara's instruction on getting incoming calls to work with Asterisk...but I still can't get Asterisk to answer incoming call on Linejack's LINE port. I attached a phone set to the PHONE port, and telco line to the LINE port on the Linejack(ISA) card. I have downloaded, compiled and installed the newest driver for

G'day, First off, I'm not subscribed to the list, so if there are any responses that should be directed to me, feel free to CC me in :) The below url is an updated patch of Professor Tom's earlier SRP patches for SSH. The only things changed was so that it would compile on a newer openssh version. For more information regarding SRP, see http://srp.stanford.edu This isn't

Hello, I'm trying to figure out how to get Asterisk to dial an extension when a call comes from the outside and contains the extension already. (Somebody wants to call a user of Asterisk with extension "111" from the outside) For example: I've hooked Asterisk to sipgate.de and received a landline phone number (say 0781205237). Now if you dial 0781205237 and and an extension

Thanks a lot for all the suggestions! Unfortunately, it still gives problems. Most common error message is "ast_realaudio_callback Failed to write frame" after "paying the beep". Then it says "User disconnected". Also, it doesn't react to any extension entered and doesn't do any forwarding (as it should in "exten =>

Hello list: If I am not mistaken the module 'ixj' is for the cards 'Quicknet LineJack' is possible not to load it when starting asterisk ?. How call asterisk this module ?. -- Jose M? Guisasola Consultor T?cnico CMSI 2002 S.L.

Hi, I have a problem with TDM2400P card. The card is detected ok, I can make a call but only with pulse dialing (not tone dialing) without hear sounds from the headset. When I receive a call, I can to establish a communication, but without hear sounds from the headset. When I dial any phone key, I can hear dtmf tone. I'm using Elastix 1.5.2. These are my configuration files:

I am trying to get a way to have * forward calls that are dialed to an extension, to end up at an extension on my old analog phone system. I will have 7 lines coming into * using the new Digium cards via PSTN, and then lines coming from * into the PSTN lines on the analog system. So that if for example someone dials extension 110: The system will call the analog system, the system will assume

On Tue, 2005-01-18 at 09:49 -0600, asterisk-dev-request@lists.digium.com wrote: > > Hi, > > I am Arnaud F?vrier, I teach in a technical university in Marseille. > > I'd like to know if is is possible to connect a very old phone to > asterisk and dial pulses with it? > > Are digium cards pulse dial compatible? > > Is there any specific configuration

Dear listers, I am trying to fit a model using nlsList() using alternately a SSfol() selfstart function or its developped equivalent formulae. This preliminary trial works well mydata<-groupedData(Conc~Tps|Organ,data=mydata) mymod1<-nls(Conc~SSfol(Dose,Tps,lKe,lKa,lCl),data=mydata) as well as a developped form: mymod2<-nls(Conc~Dose * exp(lKe+lKa-lCl) *

Hi list, I have some problems to get the QuickNet Internet PhoneJack working. What .conf files do I have to edit to get a dialtone for the first test with the standard configs from asterisk? I have the ixj driver running and a cat /proc/ixj after asterisk start tells me one reader and one writer. But if I pick up the receiver I don't get a dialtone nor I'm able to dial a number.

I have been wanting something similar. I paid some money for a busy detect routine from newman telecom, but it is not yet done. We'll see what happens. Greg -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Min Hwan Chang Sent: Tuesday, August 09, 2005 6:57 PM To: Asterisk-Users@lists.digium.com Subject:

I'm having this problem where if the phone is ringing from IncomingCall #1, IC#2 will be immediately sent to VM. Is there somethign wrong with my dial plan? I currently have 4 incoming lines going into a TDM400 with the group set to g0. Could it be that the way I've set this up, if any of the phones are busy, it goes immediately to VM? exten => s,1,Answer() exten => s,2,Wait(1)

Hi list! My skinny phone can make outgoing calls but incoming calls just keep ringing for the calling end but the phone that actually should ring doesn't ring at all. I guess I have something messed up with the dial command. I have this in skinny.conf: [z4040] device=SEP000000000000 (actual number removed) nat=0 callerid="z4040" <105> callwaiting=0 mailbox=199 transfer=1

I sent the following message a few days ago, but never received a reply, so I thought I'd ask again.. Can anyone tell me how me to get asterisk to dial out a phone number using BTP when a bluetooth device is not detected? I can get BTP to dial to a SIP phone, but I can't get it to dial through a POTS phone line using the Zap interface.. I've tried putting the following under the

Can anyone tell me how me to get asterisk to dial out a phone number when a bluetooth device is not detected? I've tried putting the following under the clients section in /etc/asterisk/btp.conf: client =>user,00:12:34:56:78:90,Zap/4/1234567890 and in extensions.conf: exten => 222,1,Playback(pls-hold-while-try) exten => 222,2,Dial(BTP/user,60,m) exten => 222,3,Hangup but

Hi, I'm facing something strange but maybe I haven't the right solution. What I want ot do is : Someone from outside call my phone number, I check some informations using an IVR script and then I want to transfer the call to an external phone number. The point is that when I'm doing that using the Dial command, I've got a message telling me no is able to answer the call. But

A couple of weeks ago there was a question regarding apparent convergence in nls when using the SSfol selfStart model for fitting a first-order pharmacokinetic model. I can't manage to find the original message either in my archive or in the list archives but the data were time conc dose 0.50 5.40 1 0.75 11.10 1 1.00 8.40 1 1.25 13.80 1 1.50 15.50 1