Displaying 20 results from an estimated 1000 matches similar to: "how to start callerid for india"
2004 Sep 30
0
UK Caller ID - todays CVS update knocks out a channel
I've updated to the latest CVS as of today (and rebuilt RedHat 9).
My setup is as follows:
Wildcard X101P - channel 1
TDM400P - channels 2-3 - fxs cards with fxo signalling, channels 4-5 - fxo cards with fxs signalling
I got CallerID to work on channel 4 with an old CVS, despite the usual "Didn't finish CallerID spill" message.
However, as soon as I insert the following
2010 Jan 05
5
CallerID on Indian PSTN is not working.
Hi,
I am using asterisknow 1.5.0 and Wildcard TDM410P card. Everything is
working fine except the caller ID of incoming call from PSTN line. The phone
display is showing "Unknown" when there is an incoming call. I think the
same problem listed here: https://issues.asterisk.org/view.php?id=6683
There is one patch on this link but i don't know how to apply patch on
asterisknow.
2012 Sep 14
2
Digium AEX410, MTNL Mumbai Caller-ID problems
Hi,
Continuing with the saga of Digium vs MTNL Mumbai, looking for
suggestions on handling incoming Caller-ID issues. The card manages to
grab a couple of (random) digits of the incoming CID, but they're more
or less useless. Is there any way to fix this?
Asterisk 1.8.13, Dahdi 2.5.0.1 on Debian Testing (Wheezy), MTNL Mumbai.
Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express)
2010 Apr 30
0
Caller ID on Asterisk and Astribank
Hi all...
I have a problem with caller id on my asterisk server.
here is my configuration :
centos-5, asterisk 1.6.2.1, dahdi-linux-complete-2.2.1, libpri-1.4.10.2
ibm X-3200 series, xorcom astribank (16fxo, 8fxs), 16 line telco (hunting)
everything fine until I try to feed my app with caller id.
My extensions.conf :
[incoming1]
exten =>
2007 Dec 27
2
No SMDI interfaces are available
Hi,
I'm a brand newbie to asterisk trying to set it up for the first time
and I can't get a softphone to connect, the connection times out.
I had a trixbox pro install working, but I need more control and would
like to learn to do it with asterisk.
In /var/log/asterisk/messages I see:
WARNING[17401] res_smdi.c: No SMDI interfaces are available to listen
on, not starting SMDI
2004 Sep 28
7
UK (British Telecom) Caller ID again
I've followed the recent thread on caller id with UK British Telecom
networks (where the caller id data is delivered before the first ring).
My understanding is that if I use a recent CVS head (e.g.
CVS-HEAD-09/18/04-17:45:52) and a TDM400 with FXO modules, all I need to
do is include the line:
usecallerid=uk
In my zapata.conf (in the [channels] section)
I've done this, but I get:
Sep
2004 Mar 31
3
SMDI support in Asterisk ?
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Hello,
Is there any work in progress for supporting SMDI in Asterisk ? if Not, could anyone tell how to get started implementing it for Asterisk.
Regards,
Tony
2006 Feb 10
0
TDM - Analog Trunk - CallerID question
Hello list.
I have a question about how to read the incoming calls' callerid on an
FXO interface of a TDM 400 analog card; (it's one of those RED modules).
Now -may this is the complexity adding step..- I have a GSM gateway
attached to this FXO thing; incoming calls are processed as they should.
But both when peeking on the CLI, as well as in the phone display I do
not see the caller id.
2008 Mar 05
1
Asterisk 1.6.0-beta5 Now Available
Greetings,
The Asterisk.org development team has released Asterisk 1.6.0-beta5. As of this
beta of 1.6.0, 1.6.0 is now feature frozen. In addition to a number of bug
fixes, the following new features have been added since beta4:
* The SMDI interface in Asterisk has been reworked to fix a number of
issues as well as add some new features. SMDI message information
is now accessed in the
2008 Mar 05
1
Asterisk 1.6.0-beta5 Now Available
Greetings,
The Asterisk.org development team has released Asterisk 1.6.0-beta5. As of this
beta of 1.6.0, 1.6.0 is now feature frozen. In addition to a number of bug
fixes, the following new features have been added since beta4:
* The SMDI interface in Asterisk has been reworked to fix a number of
issues as well as add some new features. SMDI message information
is now accessed in the
2004 Sep 28
2
SMDI Bounty - where?
I am the one that placed the bounty. After it being there for 2 months
and getting no takers (and very few if any people asking about it), we
are almost finished writing it in house. I'll keep the bounty up untill
we do finish our product so if anyone beats us to getting it working
they'll get paid...
W. Kevin Hunt
CCIE #11841
MCSE, Linux+ SME
www.huntbrothers.com
>
2006 Apr 20
3
Asterisk Won't start after SVN Trunk Update
Hi:
I deleted old modules in /usr/lib/asterisk/modules
before make install. I built zaptel and libpri before
asterisk. Modprobe zaptel and modprobe -v wctdm
executed witiout complaint. Starting asterisk
produced the output below with several warnings and a
failure. Can someone help, please. I double-spaced
the warnings in the text below. The first warning is
about music on hold because it
2013 Nov 25
1
Asterisk 11.6.0 not starting up
Hello Friends:
I've just installed Asterisk 11 on my Linux (debian) server but it is not
starting up when trying with "asterisk -vvvvvvvvvvc" and "service asterisk
start". Starting process just stop and shows: "Illegal instruction" as
final output.
Looking at logs I fouind at /var/log/asterisk/messages :
[Nov 25 11:09:26] Asterisk 11.6.0 built by root @
2008 Aug 01
3
Asterisk Queues problem
Hi,
I have Asterisk 1.4.18 and I have been running call center queues on it.
Today it suddenly stopped adding inbound calls to queues. I am facing
with following error: app_queue.c:3939 queue_exec:
unable to join queue "myqueue"
In extension file:
Queue(myqueue|t|||120)
And my agents are joining in following
2006 Oct 30
0
Problem with incomming calls
<font size="2">I've got an odd situation where callerid is only picked up every other call. Is there anything I can do so callerid works on all calls?<br><br>I'm seeing the channel hangs up during a call<br><br> == Starting post polarity CID detection on channel 4<br> -- Starting simple switch
2010 Jul 31
0
Disconnect supervision tone detection working for india
Hi ,
Thanks danny nicholas. Finally we get the things done with following.
If i specify busypatten=500,500 then asterisk does not recognize hang up
signal. After removing it only all are working fine.
I choosed 2nd option as per your suggestions.
working chan-dahdi.conf:
====================
signalling = fxs_ks
busycount = 3
busydetect = yes
callprogress = yes
progzone=in
usecallerid=yes
2013 Jan 10
1
Segmentation fault after upgrading from asterisk-10.5.0 to asterisk-11.1.2
After upgrading from asterisk-10.5.0 to asterisk-11.1.2, I am getting a Segmentation fault.
[root at localhost asterisk-11.1.2]# asterisk -vvvvvvc
Asterisk 11.1.2, Copyright (C) 1999 - 2012 Digium, Inc. and others.
Created by Mark Spencer <markster at digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components
2005 Aug 30
1
X100P and UK CallerID
Hi,
I'm currently running asterisk 1.0.9-r1 and zaptel 1.0.9_p1-r1 (the
current gentoo ~x86 versions), with the UK CallerID patches from
http://www.lusyn.com/asterisk/patches.html applied.
The Zap interface itself seems to work fairly well - although it's a
little quiet, there is no echo. Unfortunately, there's also no
CallerID.
My zapata.conf is as follows:
[channels]
2009 May 03
2
Asterisk not starting up due to database problems
When I try and start asterisk I get the following, however I have commented out the data the connections in res_mysql.conf and res_pgsql.conf. I am not sure therefore why I am getting these errors. Do I have to change something else to turn this off?
Thanks
Asterisk 1.4.21.2~dfsg-3, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <markster at digium.com>
Asterisk
2007 Sep 11
1
TDM400P not answering or making calls
Hello,
I have recently purchased a TDM400P card with one FXO expansion card,
and I'm having problems.
The card does not pick up incoming calls. Asterisk detects the ringing
line and rings various SIP phones as required. When a sip phone
answers, the sip user hears nothing and the PSTN user continues to
hear ringing. Here is the asterisk output for an incoming call: