Displaying 20 results from an estimated 1000 matches similar to: "SIP Diversion RDNIS - how to get reason parameter?"
2017 Nov 21
2
How to correctly set REDIRECTING to indicate diversion reason
Hi Richard
Thank you
> You need to set more redirecting information [1].
>
> In sip.conf send_diversion=yes needs to be in effect. You also need
> to setup
> the from party id information (at least the from number) to indicate
> where you
> are redirecting from. You should also increment the redirecting
> count.
>
> Richard
>
> [1]
>
2017 Nov 20
2
How to correctly set REDIRECTING to indicate diversion reason
Hello List
Next question where google did not spit out an unsable answer.
When redirecting a call with Transfer, I would like to correctly
indicate the reason.
I did try this:
exten => XX,1,NoOp(Call to ${EXTEN} from ${CALLERID(all)})
exten => XX,n,Dial(SIP/ZZ)
exten => XX,n,set(REDIRECTING(reason)=cfb)
exten => XX,n,Transfer(SIP/YY)
I did try with 'reason'
2019 Nov 19
2
Global number rewriting rules affecting ALL headers?
Hi Joshua
I had a shot at your suggestion, bug still no success.
I fear the 181 is sent before the macro is called.
I want to change the Diversion Header in the 181 message sent back to
the caller to put the number it contains in the correct e164 format
(stripping the 0 and adding +41 for Switzerland) but just any 'dialplan
set' value would do for an example :-)
Could you please make
2010 Sep 24
1
RDNIS not passed from one box to another with BRI access
Hi,
I've configured a new Asterisk 1.6.1 box to replace an old Bristuffed 1.2
Asterisk.
Since then, it happens that forwarded calls are not presented the way they
used to be.
It seems that now, some endpoints are displaying the original caller id
(that's what I'm trying to achive), while some are displaying the
redirecting number :
so if A calls B, B forwards to C
depending on where
2010 Aug 11
1
Youmail RDNIS
Does anyone know the mechanism by which companies like YouMail (and MNO's
using their own voicemail system) are able to redirect ALL calls from a ALL
subscribers to *just one* voicemail DID, yet determine WHICH subscriber did
the redirection?
I had always assumed this was simply done using RDNIS. In other words, the
original calling party's CallerID is passed with the redirected
2006 Dec 30
0
Theory behind RDNIS and does it work or not?
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Hello everybody,<br>
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currently I'm implementing redirection
2014 Aug 28
1
RDNIS with tel: vs. sip: header
Has anyone had success patching chan_sip.c so that Asterisk will recognize
the tel: header for RDNIS information?
exten = get_in_brackets(tmp);
if (!strncasecmp(exten, "sip:", 4)) {
exten += 4;
} else if (!strncasecmp(exten, "sips:", 5)) {
exten += 5;
} else {
ast_log(LOG_WARNING, "Huh? Not an
2010 Jun 19
0
OT - Explain RDNIS
Hi,
1. Can you explain what RDNIS is when it should used with Asterisk ?
I've read this http://www.voip-info.org/wiki/view/RDNIS but I'm still
wondering if I should use it.
My understanding is:
when dialing out through an ISDN line, Asterisk is sending two numbers
(using signalling channel) :
- one is CALLERID(num) which is supposed to be whatever the Asterisk admin
whishes to be,
- one
2006 Apr 28
2
Asterisk DNID/RDNIS with Dial iax2
Dear Asterisk-Users:
Question:
========
How do I get asterisk to pass DNID/RDNIS information between
asterisk machines using iax2, in a Dial(IAX2...) command ?
Setup:
=====
I have two asterisk boxes, MASTER and SLAVE. MASTER is running
1.2.0 and SLAVE is running 1.2.1. The main box handles incoming calls
on a multiple lines (both via hardware connection to our internal PBX
and calls
2004 Oct 26
2
RDNIS
I'm trying to use RDNIS with asterisk, and I don't appear to be
receiving any information (the value is blank). The upstream who
provides the PRI says they are passing all the info through, I don't see
this value coming across. I've tried it with a Verizon call forward, as
well as a Nextel with the same results for both. I'm trying to use this
for Voicemail. I'm using
2009 Mar 24
0
Issue with RDNIS
Hello,
Does anyone know why I am unable to retrieve the "Redirecting Number"?
I've done a "pri debug span 1/1" and can see the number being passed
correctly to Asterisk:
< Redirecting Number (len=15) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
< Ext: 0 Presentation: Presentation allowed
of
2006 Dec 09
2
RDNIS question
Perhaps I've got the whole concept wrong, but here goes:
Using 1.4, when someone from the outside dials my direct line (123456),
I want it to call my extension at work (SIP/456), my extension in my
home office (vpn connection to corporate lan, SIP/678) and my mobile
(654321). So my dialplan is thus:
exten => 123456,1,Dial(SIP/456&SIP/678&Zap/G3c/07803654321,30)
exten =>
2005 May 20
1
RDNIS (DNID) Call Routing
I haven't been able to find much support for the RDNIS or DNID variables
online.
I am trying to prove a concept of call routing before we move towards
development of a production system. I need to have calls routed coming into
a call center based on DNIS. What type of syntax is needed in the
extensions.conf file and how can I test it with a softphone (ie: can I
emulate the DNIS from xlite)?
2010 Oct 26
2
Trim the RDNIS
What I am needing to do is to trim the 1 from beginning of the RDNIS and
I have tried using the CUT function but cannot seem to make it work for
me. What we have is a phone number like this, 18881232342 and want to
make it like this 8881232342. I appreciate any help that you guys can
give. Thanks!
--
*Chris Ramirez*
TELE-ONE COMMUNICATIONS, INC.
cramirez at tele-onecom.com
903-531-0777
2019 Nov 19
2
Global number rewriting rules affecting ALL headers?
Hi List
One more Problem I stumbled upon.
Using Asterisk in a TSP environement.
Incomming IC Calls are e164 and have a NPRN (Routing Number) prefixed.
Example: +4198055615995555
+41 country prefix
98055 Routing Prefix
615995555 effective phone number
Calls routed to Customers need to be put in the 'local' format.
0615995555
This is also the format of the From / To / Invite header
2009 Mar 27
2
SIP Diversion header
Hi,
Is anyone aware of SIP Diversion header ?
It seems currently supported by Comverse (formely NetCentrex) softswitch and
some hardphones (Thomson ST2030).
An old draft (draft-levy-sip-diversion-08.txt) mentions this header.
ha
I'm wondering if this could be used
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2008 Dec 16
4
RDNIS and asterisk
I have a couple of numbers that are diverted to a number that is
conected to an isdn30 card, running asterisk 1.4.
eg.
123456 => 22334455
654321 => 22334455
What I would like to know is the number of the orginal number dialled
(123456 or 654321). I thought that RDNIS was the answer, but it is
always coming up blank.
When I did a debug on the pri span, I saw the following message
2019 Nov 15
2
pre-dial handler, how to access variables from calling channel?
Hi List
Implementing screening and routing I have stumbled over this issue:
[pbx-router]
exten => s,1,NoOp(ROUTER FROM: ${CALLERID(Number)} TO: ${DESTINATION})
same => n,Set(SOURCE=${CHANNEL(name)})
same => n,Set(PAI=${PJSIP_HEADER(read,P-Asserted-Identity)})
same => n,Set(FROM=${CALLERID(Number)})
same => n,Set(TO=${DESTINATION})
same
2017 Nov 27
2
pjsip Transfer 'Failed to parse destination uri'
Hi Richard
> That could be possible and would be a bug in chan_sip.
Ok, so I switched to PJSIP to see if this behaves differently
So ip do a
Transfer(PJSIP/${DESTNUMBER}@trunk)
And this results in:
Failed to parse destination URI '[destnumber scrubber]' for channel
PJSIP/trunk-00000011
Do I have to specify the destination number differently when using
Transfer with pjsip that I
2006 Mar 24
3
* Meetme Freeze patch found
Hi all
Apparently there is a patch for those 1.2.4/5 MeetMe Freezes:
http://bugs.digium.com/view.php?id=5884
Haven't tried it out yet.
Benoit Panizzon
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