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Tue 09 of Feb, 2010 [20:17 UTC]

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DNID

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



Comments

Comments Filter
222

333DNID for SIP incoming calls from VoIP provider

by latv, 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)
222

333Started a page that may be useful

by michigantelephone, 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
222

333DNID relating to Asterisk

by _pd, 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)})
222

333Could prove very useful

by , 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