I am using dahdi in asterisk to manage all analog calls.
My scenario is
analog phone----> pbx---------> asterisk-------------> jitsi
my analog extension no is 77 and jitsi user number is 2000
Now when i am calling from 77 to 2000. I am getting the call. Its working fine.
But how to get callerid number that is 77 on jitsi.
Any command in asterisk to get callerid from dahdi.
Thanks a lot…
configs/chan_dahid.conf.sample:
;
; caller ID can be set to "asreceived" or a specific number if you want to
; override it. Note that "asreceived" only applies to trunk interfaces.
; fullname sets just the
;
; fullname: sets just the name part.
; cid_number: sets just the number part:
;
;callerid = 123456
;
;callerid = My Name <2564286000>
; Which can also be written as:
;cid_number = 2564286000
;fullname = My Name
;
;callerid = asreceived
;
; should we use the caller ID from incoming call on DAHDI transfer?
;
i already wrote in chan_dahdi.conf file
; General options
[channels]
usecallerid=yes
hidecallerid=no
callwaiting=yes
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
cidsignalling=dtmf
cidstart=polarity
callerid=asreceived
rxgain=0.0
txgain=0.0
;FXO Modules
group=1
echocancel=yes
signalling=fxs_ks
context=default
channel=1-20
#include dahdi-channels.conf
what should we write in extension.conf file. is there any command to get incoming caller id number.
I attached 77 number with dahdi. so that after calling on message IVR can be listen and send it to respective number. Actually user can call from any number such as 78 to 99. now i want to show that number from 78 to 99. From which user called on jitsi side user…
Thanks a lot
Your original description suggested that the analogue phone was on an FXS port. If you are actually talking about an exchange line on an FXO port, whether CLID is possible at all and how to configure it depends on the PSTN network operator and the service product that you have bought.
However, 77 still doesn’t sound like a PSTN number. It still sounds like a simple phone and simple phones don’t originate caller-ID; you must set an explicit value for the caller ID in dahdi.conf.
Transfer, etc., are not things that one would normally enable on an exchange line.
What dahdi hardware are you using, and is it a currently supported product?
my pc has TDM2405P card. from that my intercom is connected with extension no 10 to 99.
and my intercom is originate caller id. i am able to see on my called id phone.
How to read that caller no and ID from dahdi hardware. So i can forward it to my sip phone.
Thanks a lot for you reply.
How does it send caller ID?
My guess is that it is an after first ring system, not a line reversal system.
Does it use DTMF. Bell standards, or V.23, etc.?
Assuming it uses a supported standard, you need to set these two parameters to match what the PBX actualy uses:
cidsignalling=dtmf
cidstart=polarity
[code]; Type of caller ID signalling in use
; bell = bell202 as used in US (default)
; v23 = v23 as used in the UK
; v23_jp = v23 as used in Japan
; dtmf = DTMF as used in Denmark, Sweden and Netherlands
; smdi = Use SMDI for caller ID. Requires SMDI to be enabled (usesmdi).
;
;cidsignalling=v23
;
; What signals the start of caller ID
; ring = a ring signals the start (default)
; polarity = polarity reversal signals the start
; polarity_IN = polarity reversal signals the start, for India,
; for dtmf dialtone detection; using DTMF.
; (see doc/India-CID.txt)
;
;cidstart=polarity
[/code]
If you are in the USA, my best guess is that you want the defaults, but that is only completely reliable when dealing with a real PSTN line.
Although it is fairly unlikely that anyone knows your intercom, providing the make and model would do no harm and would help if they do know it. It shouldn’t have been necessary to drag out of you that were using this with a PBX, not with a phone.
i already mention
cidsignalling=dtmf
cidstart=polarity
in my chan_dahdi.conf
When i am calling from 99 extension to other number(which has callerid), it shows number 99 on caller id.
Like that way when i call on 77 number(which is connected to asterisk) from 99. 99 Number should display to my sip client.
so how to read is from dahdi and send to sip client. Is there any cmd to read incoming callerid.
Right now i am getting “asterisk” callerid on sip phone
Provided both sides are capable of caller-ID, caller-ID is passed through unless you actively do something to prevent it. If it is not going through, the most likely reason is that it is not being detected.
I am rather surprised that an “intercom” device generates line reversals, but you haven’t told us what it is, so I can’t check.
Given that most phones will recognize the US standards, even if you are in Denmark, Sweden or the Netherlands, it is more likely that it uses the US standard than that it uses DTMF (and line reversals).
The ${CALLERID(num)} function can be used to read the value and actively manipulate it.