similar to: pridialplan/prilocaldialplan

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'm using the biult in feature attended transfer. If someone calls me, I hit the #, dial another extension and connect these two extensions. When hitting # and dialing the nr, asterisk only diales the new nr for 15 seconds. Is it possible to increase this time? I've only found the timeout for the digits, not for the call time. Anyone

Hi Chris, Yes the call was from PSTN and your solution worked great! I've read about SetCallerPres earlier but I didn't connect the dots this time. Thanks alot! :) Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r Christoph F?rstaller Skickat: den 29 mars 2007 15:29 Till: Asterisk Users

Hi I'm beginner in list. I have doubts about the options pridialplan and prilocaldiaplan in chan_dahdi.conf. I interconnect the Asterisk with a Siemens PBX, but i saw that the changes in the file do not take effect in debug of the span or calling/called number. How to use this options? In that cases to use? Ps.: sorry for the english, i'm brazilian. Thanks -- Att, Rafael Saraiva

Hi list! What exactly is the meaning / function of the pridialplan & prilocaldialplan? I've been trying to find out what the different possibilities for these settings are but couldn't find a clear answer. The possible parameters I could find are are : local,unknown,dynamic,national,international and maybe there are more? Thanks!

Hi all, I've been batting my head against a brick wall for the best part of the day and still haven't got any further (apart from getting a big headache, that is). I've searched the Wiki and Googled the hours away, but I still can't find supportive documentation. I've just replaced my ISDN Fritz!/chan_capi setup with a HFC/Zap configuration and had the following problems: 1

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I' try to implement an automatic callback mechanism, just for local SIP calls.. Callback on busy and on no answer. If the other party doen't answer, it should be possible to press 5 to place an callback. Here is my dial: exten => _X.,1,Set(EXITCONTEXT=callback) exten => _X.,n,Dial(${DIALNUM},${ARG2},dtT) And here the script for

Hi, This is really strange (but probably simple solution). The CALLERID(all) setting doesn't seem to work when the incomming callerid is 'unknown'. Dialplan looks like this: exten => _3072,1,Answer exten => _3072,n,Set(CALLERID(all)=DIRECT <0850553072>) exten => _3072,n,Dial(SIP/2001&SIP/2002&SIP/2003&SIP/2004&SIP/2201&SIP/2202&SIP/2

People, I have a Debian box with Asterisk and I've installed the Destar package in order to get web managing of my voip system. After I installed Destar, it runs on "localhost:8080", but my server does not have X-Window to access to it so I can engter the web interface.. So how can I change localhost:8080 to IP_ASTERISK:8080 in order to access Destar via web from another PC ???

Hi, I have FRITZ!Card PCI card. I have installed misdn-1.1.0 on stable debian 3.1 with 2.6.8. kernel. Then I reboot system, and it doesn't boot, it stops near "Apache2 starting...". I started my system with "recovery" kernel, and tun off misd, then my system works fine. I think it's problem with memory. Has anybody debian and misdn working fine? Maybe you can

Please remove this email from your mailing list. UNSUBSCRIBE Thank you. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Thursday, March 29, 2007 9:14 AM To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 32, Issue 118 Send asterisk-users

Does anyone have any recommendations for a phone that has easy to understand/implement central provisioning? I've used CISCO 79XX phones, and they're great (but too expensive). I like Grandstream phones, but their provisioning sucks. What is everybody else using in large environments where individual config is not an option? ---------------------------------------- Rod Bacon

Hi, Lately we've noticed that some Zap channels on one of our PRIs are unavailable. We have 2 PRI lines with 60 channels in total. On the first PRI there are currently 20 channels that are not being used for some reason. I tried googling around and found some similar problems but there really was no solution (?). I'm not sure if this problem has occured now because of more load on the

I have an x100p which suffers from echo (no surprise there apparantly :), and a few of the things I've read about tuning out echo say the first thing to get right is the tx and rx gain, and for that you need a few different types of test lines from the telco (Telstra, Optus, whatever). Does anyone have any numbers in Australia for such lines? I think they just need to send out a tone of a

Hi, I'm having strange warning from asterisk when I try to dial GSM Gateway: -- Executing [1011501522xxx at sm:1] NoCDR("SIP/ibm-b2c52848", "") in new stack -- Executing [1011501522xxx at gsm:2] Dial("SIP/ibm-b2c52848", "Zap/R3/501522xxx") in new stack -- Requested transfer capability: 0x00 - SPEECH [Jul 13 11:58:50] WARNING[18208]:

Asterisk 1.6.0.6 with dahdi 2.1.0.4 is showing a strange "Unrecognized prilocaldialplan" error with the SIP username when a SIP call is dialed to a PRI trunk. The error shows up like this: Unrecognized prilocaldialplan TON modifier: a Unrecognized prilocaldialplan TON modifier: b Unrecognized prilocaldialplan TON modifier: c Where abc is the SIP username. Is this a bug

Only a warning, and doesn't seem to do anything bad. But I can't seem to figure out what these warnings mean? -- Requested transfer capability: 0x00 - SPEECH [Feb 17 12:33:03] WARNING[10750]: chan_dahdi.c:3096 dahdi_call: Unrecognized prilocaldialplan NPI modifier: k [Feb 17 12:33:03] WARNING[10750]: chan_dahdi.c:3096 dahdi_call: Unrecognized prilocaldialplan NPI modifier: o [Feb 17

Hi, all. This e-mail is a follow-up to an exchange I had several weeks ago. I've got an Asterisk box with a dual-span T1 card. I want to place it between the PSTN and my company's legacy PBX. I actually did do that, but international calls from the legacy PBX were having the "011" stripped off *AT* the PBX -- and someone pointed out that the PBX was probably using the

For whom which may be interested: Here in Italy we have GSM #numbers without leading zero PSTN instead has prefix starting with '0' to have '0' recognized by * i need to insert nationalprefix=0 as Jason Williams suggested me in irc; now, you cannot have: pridialplan=natonal otherwise * will not be able to call GSM phones you need to setup: pridialplan=local

Hello, I have a little problem with signalling. An E100p is connected to an Alcatel PBX, wich has an E1 to the outside. Located in Germany. zapata.conf: switchtype=euroisdn pridialplan=local prilocaldialplan=local overlapdial=yes signalling=pri_cpe .... With asterisk 1.0.2 I can call from a SIP phone to a phone connected to the Alcatel and the SIP number is correctly displayed at the caller.

After setting the pridialplan=unknown I seeing the Called Number TON change to Unknown Number Type but not the Calling Number TON. Should both be following this parameter or not. If not is their another option to change the Calling Number TON? > Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) > Ext: 1 Progress