login | register
Sat 05 of Jul, 2008 [03:14 UTC]

voip-info.org

Search with Google
Search this site with Google. Results may not include recent changes.
 
Google Ads
Shoutbox
  • Samuel, Thu 03 of Jul, 2008 [13:41 UTC]: ok thank you
  • Mats Karlsson, Thu 03 of Jul, 2008 [13:37 UTC]: Nice Samuel, will look forward to rad it.
  • bwl_fernstudent, Thu 03 of Jul, 2008 [09:08 UTC]: Your blog shows some usefull code
  • Samuel, Thu 03 of Jul, 2008 [08:04 UTC]: I'll translate it, for sure
  • Mats Karlsson, Wed 02 of Jul, 2008 [20:46 UTC]: LOL, in french! Translate it to English and I will read it.
  • Samuel, Wed 02 of Jul, 2008 [08:07 UTC]: Hello, i wrote a blog about Asterisk, speaking about installation,programming and more http://sambranche.blogspot.com/
  • Nick Barnes, Tue 01 of Jul, 2008 [17:46 UTC]: Steve - Asterisk doesn't 'fit into linux' - it's an application which runs on top of Linux.
  • Steve, Mon 30 of Jun, 2008 [18:07 UTC]: anyone know where I can find a block diagram of how asterisk fits into linux. my f'ing bosses want me to draw something up.. ugh.
  • akbar, Fri 27 of Jun, 2008 [10:37 UTC]: marley_boyz@yahoo.com how to configure call forward, call back, call pick up using TDM and asterisk 1.2.13... please help me.. thx...
  • Matthew Williams, Tue 24 of Jun, 2008 [22:37 UTC]: We are looking for Tier II VoIP Support Technicians in St Louis. Send resumes to mwilliams AT voxitas DOT com.
Server Stats
  • Execution time: 0.39s
  • Memory usage: 2.59MB
  • Database queries: 33
  • GZIP: Disabled
  • Server load: 0.55

DNID

DNID sometimes contains the number that was dialed.

From asterisk-1.4.20.1.tar.gz; docs/channelvariables.txt:
${DNID} - Dialed Number Identifier (Deprecated; use ${CALLERID(dnid)})

Asterisk implementation comment:
I do not know when it works or not nor when it is supposed to work or not.
See http://bugs.digium.com/view.php?id=2590

See also


Created by fredo, Last modification by timbo on Tue 03 of Jun, 2008 [18:40 UTC]

Comments Filter

DNID for SIP incoming calls from VoIP provider

by Vladimir Latyshev on Monday 07 of April, 2008 [09:30:35 UTC]
Thanks Paul, it works for SIP :)
Full decision in dialplan (when "to" headers look like <sip:xxxxxxx@sipnet.ru>):
exten => s,1,Set(FOO=${SIP_HEADER(TO):5})
exten => s,n,Set(DNID=${CUT(FOO,@,1)})
exten => s,n,Goto(${DNID},1)

Started a page that may be useful

by michigantelephone on Monday 23 of October, 2006 [08:02:43 UTC]
Regarding the Anonymous comment, I started a page entitled "Routing calls using a free international calling service" that gives a couple of solutions (one lets you use a custom extension to "speed dial" individual numbers, and the other lets you create a trunk for such calls, though neither is a "perfect" solution, and they are very much geared toward FreePBX/Trixbox users). The page is here:
http://www.voip-info.org/wiki/view/Routing+calls+using+a+free+international+calling+service

DNID relating to Asterisk

by Paul Durcek on Saturday 04 of March, 2006 [19:41:54 UTC]
I am not sure if it works in all situations, but I was able to get the actual dialed number information from the SIP header using the following command:

exten => s,1,NoOp(${SIP_HEADER(TO)})
Edit

Could prove very useful

by Anonymous on Sunday 23 of January, 2005 [17:50:49 UTC]
Hi all,
This variable could prove very useful. for instance, one could incorporate it into an international callback, without prior pick-up (i.e one stage dialing). for example, a user would dial the * phone number immediately followed by the ultimate destination he/she wishes to reach (should work for calls coming over zap channels at least) . * authentifies the user based on CID, then connects through a conference room the CID and DNID numbers (after proper CDR recording). If anyone interested in co-developing (I could pay for this) let me know at o (underscore) olivier (at) hotmail (dot) com

Please update this page with new information, just login and click on the "Edit" or "Add Comment" button above. Get a free login here: Register Thanks! - support@voip-info.org

Page Changes | Comments

Sponsored by:

Terms of Service Privacy Policy
© 2003-2008 VOIP-Info.org LLC

Powered by bitweaver