First, here's what I want to do / what I have: X100P and a Quicknet PhoneJack. I want to be able to pick up the analog phone (connected to the phonejack) and dial another computer (with the same hardware) or just make a regular phone call which will be decided by asterisk depending on the phone number dialed. I know that this won't be taking full advantage of asterisk, but I'm just trying to get a connection before I go into deeper configuration. So far, I have asterisk running, and I get a dialtone when I pick up the phone ... however, when I dial a digit, I get a busy signal. I'm pretty sure I need to create an ID for the computer to connect to in iax.conf [remote] type=friend host=x.x.x.x context=default I have read the documentation on the dialplans and I am a little confused: I have a PhoneJack card connected to my analog phone, so should I use IAX channels in extensions.conf? How can I send a local call through the regular phone line, instead of through the internet? Sorry if this seems like I'm jumping all over the place? _________________________________________________________________ STOP MORE SPAM with the new MSN 8 and get 2 months FREE* http://join.msn.com/?page=features/junkmail
First make sure you can configure X100P. Look at www.digium.com Documentation->FAQ->Configuration. There is some guide to install one X100P in the system. Now to be able to use Phonejack you have to have quicknet drivers loaded (they come with the kernel ... they'll called ixj or something like that and when you compile your own kernel you'll see them in "Telephony"). Then you have to configure asterisk's phone.conf and use the interfaces somehow like this: exten => 1000,1,Dial,Phone/1 Martin On Fri, 25 Apr 2003, Derek Beaumont wrote:> First, here's what I want to do / what I have: > > X100P and a Quicknet PhoneJack. > I want to be able to pick up the analog phone (connected to the phonejack) > and dial another computer (with the same hardware) or just make a regular > phone call which will be decided by asterisk depending on the phone number > dialed. I know that this won't be taking full advantage of asterisk, but > I'm just trying to get a connection before I go into deeper configuration. > > So far, I have asterisk running, and I get a dialtone when I pick up the > phone ... however, when I dial a digit, I get a busy signal. > > I'm pretty sure I need to create an ID for the computer to connect to in > iax.conf > > [remote] > type=friend > host=x.x.x.x > context=default > > I have read the documentation on the dialplans and I am a little confused: > I have a PhoneJack card connected to my analog phone, so should I use IAX > channels in extensions.conf? > How can I send a local call through the regular phone line, instead of > through the internet? > > Sorry if this seems like I'm jumping all over the place? > > _________________________________________________________________ > STOP MORE SPAM with the new MSN 8 and get 2 months FREE* > http://join.msn.com/?page=features/junkmail > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
Does anyone know how to do the following: 1. Caller calls in 2. Asterisk answers. 3. Asterisk rings nominated extensions 4. Caller keys in certain digits while extensions are ringing 5. Caller is directed to another extension based on the digits keyed in I can achieve this if I have Asterisk play a background message after answering and before ringing the extensions (between steps 2 & 3). But I cannot get it to work if the extensions are rung straight away. Any help would be greatly appreciated. Simon Brown
Can anyone shed some light on this ??? Or is this not the "right" sort of question to ask? Simon Brown -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Simon Brown Sent: Wednesday, 4 August 2004 11:44 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Dialplan question Does anyone know how to do the following: 1. Caller calls in 2. Asterisk answers. 3. Asterisk rings nominated extensions 4. Caller keys in certain digits while extensions are ringing 5. Caller is directed to another extension based on the digits keyed in I can achieve this if I have Asterisk play a background message after answering and before ringing the extensions (between steps 2 & 3). But I cannot get it to work if the extensions are rung straight away. Any help would be greatly appreciated. Simon Brown _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Hello Asterisk,
is it possible to make an extensions that write a call file
(like a call back to the callerid) in
the outgoing directory WITHOUT using a perl AGI ?
--
Best regards,
Danny mailto:dannyz@belgonet.com
belGOnet.com a Euro-pictures division - internet solutions
place princesse elisabeth 9/11 - 1030 Brussels - Belgium
Tel : +32-(0)2-215.67.65 - Fax : +32-(0)2-215.66.65
domains - hosting - hardware - VoiP - consultancy - backuping
CISCO - HP/COMPAQ - SUN - EMC - JUNIPER - IBM - DELL - NORTEL
No legal consequences can be derived from the contents of the email
neither is belGOnet.com committed to them. The content of this email
is exclusively intended for adressee(s) and information purposes.
belGOnet.com accepts no liability for any damage resulting from the
use and/or acceptation of the content of this email.
> -----Original Message----- > From: Danny Zak [mailto:dannyz@belgonet.com] > Sent: Sunday, September 26, 2004 5:29 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: [Asterisk-Users] Dialplan question > > > Hello Asterisk, > > is it possible to make an extensions that write a call file > (like a call back to the callerid) in > the outgoing directory WITHOUT using a perl AGI ? >The only way that I can think of would be to execute a system Shell script or something to that effect. To my knowledge there is no way to write a file directly from within the dialplan. Hope this helps, Robert Jackson
I forgot to add a link to the system command: http://www.voip-info.org/wiki-Asterisk+cmd+System> -----Original Message----- > From: Robert Jackson > Sent: Sunday, September 26, 2004 5:57 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Dialplan question > > > > > > -----Original Message----- > > From: Danny Zak [mailto:dannyz@belgonet.com] > > Sent: Sunday, September 26, 2004 5:29 PM > > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > > Subject: [Asterisk-Users] Dialplan question > > > > > > Hello Asterisk, > > > > is it possible to make an extensions that write a call file > > (like a call back to the callerid) in > > the outgoing directory WITHOUT using a perl AGI ? > > > The only way that I can think of would be to execute a system > Shell script or something to that effect. To my knowledge > there is no way to write a file directly from within the dialplan. > > Hope this helps, > > Robert Jackson > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/as> terisk-users > To > UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users
Hello, I have a dial plan that tries to place a call over several different
outbound gateways, like this:
exten => _1X.,1,Dial(SIP/${EXTEN}@proxy1)
exten => _1X.,2,Dial(SIP/${EXTEN}@proxy2)
exten => _1X.,3,Dial(SIP/${EXTEN}@proxy3)
exten => _1X.,4,Dial(SIP/${EXTEN}@proxy4)
exten => _1X.,5,Hangup
it works fine if one of the gateways is busy [rolls to the next dial
statement]. However, if the phone number itself on the proxyX gateway is
PSTN-busy, then it correctly returns 486 busy here, but execution continues,
which wastes trunks trying a busy number on each gateway.
What is the best way to handle this? Inserting +101 extensions with the
Hangup command ? Will that still properly signal 486 busy here back?
Should I be using Congestion instead of Hang up ?
Matthew Simpson wrote:> Hello, I have a dial plan that tries to place a call over several > different outbound gateways, like this: > > exten => _1X.,1,Dial(SIP/${EXTEN}@proxy1) > exten => _1X.,2,Dial(SIP/${EXTEN}@proxy2) > exten => _1X.,3,Dial(SIP/${EXTEN}@proxy3) > exten => _1X.,4,Dial(SIP/${EXTEN}@proxy4) > exten => _1X.,5,Hangup > > it works fine if one of the gateways is busy [rolls to the next dial > statement]. However, if the phone number itself on the proxyX gateway > is PSTN-busy, then it correctly returns 486 busy here, but execution > continues, which wastes trunks trying a busy number on each gateway. > > What is the best way to handle this? Inserting +101 extensions with the > Hangup command ? Will that still properly signal 486 busy here back? > Should I be using Congestion instead of Hang up ?The Dial command sets the DIALSTATUS variable as documented in "show application dial" and you can see usage examples in the stdexten macro in extensions.conf.sample in the asterisk/configs source code.
If someone has a minute, I would appreciate their help configuring my
dialplan. I am using 2 Sipura-2000s to connect to the CO ports on my
legacy PBX. I'm tyring to setup the dialplan so that when someone
enters an extension (1XX), it will determine which of the 4 sip
extensions are available, call it, and then dial the extension entered
original (since the auto attendant immediately picks up). However, it
looks as though ChanIsAvail doesn't work for SIP extensions. Here is
what I have:
exten =>
_1XX,1,ChanIsAvail(SIP/759011&SIP/759012&SIP/759021&SIP/759022);
exten => _1XX,2,Cut(theChannel=AVAILCHAN,,1); Remove any CallerID
exten => _1XX,3,SetCIDName(${CALLERIDNAME});
exten => _1XX,4,Dial(${theChannel},90,D(${EXTEN}));
exten => _1XX,102,Playback(all-circuits-busy-now);
exten => _1XX,103,Hangup;
It works great as long as the first line (759011) isn't in use.
However, if it is, it will not fail over to the second, third, or forth
lines. If anyone has any ideas, I would really appreciate it. Thanks
in advance.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20050623/88d51143/attachment.htm