Displaying 20 results from an estimated 10000 matches similar to: "Asterisk as a Operator Phone"
2010 Dec 20
5
DIALSTATUS on CANCEL
Hello,
We have a strange situation (asterisk 1.6.2.14), where we get a result for
DIALSTATUS for BUSY and No-ANSWER, but nothing for CANCEL.
This is the (relevant) test dialplan:
--------------------------------
[incoming-private]
exten => _X., n, Dial(SIP/1001,30)
exten => _X., n, NoOp(${DIALSTATUS})
exten => _X., n, Gosub(incoming-status,s-${DIALSTATUS},1)
[incoming-status]
exten
2010 Nov 22
3
Is existing CDR in Asterisk is enough for complete billing
Hi everyone,
I am facing lots for problem with CDRs in 1.6 and above
versions,its shows wrong records when I do transfer(caller side and
calee side),callforward,call parking.Is the present CDRs in 1.6 is
enough for Complete billing.?What I need to do to make it proper.Please
help me on this.
Thanks
Nikhil
2010 Nov 18
3
usage of account code in CDR
Hi everyone
Anyone please explain me How Account code is use for billing.,
Thanks
Nikhil
2010 Oct 05
5
Implementing more than one asterisk instance in the same hardware machine?
Hi All;
Did anyone try to implement (installation and configuration and running) for more than one asterisk instance (two or three instances), where each asterisk instance to work on a difference IP than the other where the server already has more than one IP address.
We need to implement this situation because in case we need to do testing for any scenario of configuration, then other
2010 Dec 06
1
Callee side blind transfer is failing in 1.8
HI
callee side blind transfer is failed in 1.8 but caller side blind
transfer is succes,Transfer doing by refer method,please help me on this
Nikhil
2011 Jun 02
1
Three-way conference in Asterisk
Hi
How to set a threeway conference in asterisk only for VOIP (I am
using only SIP channel).
Thanks
Nikhil
2010 Dec 27
6
Using SIP stack within Asterisk to reboot phones - Possible?
Hi Everyone,
I use Asterisk for regularPBX use it's made for. But I want to take it a bit
further and use it at cmmand level to be able to send SIP notifies to
restart a phone or take advantage of a phone's UPnP capabilities. Is
Asterisk capable of that? If so, what is a simple SIP reboot message like
and how can I invoke it from a Asterisk CLI?
If Asterisk is not the best tool for this
2011 Oct 06
3
Digium FFA + Gafachi T38 outgoing issues
Hi, folks.
I'm having a heck of a time trying to get outgoing T38 faxing (I don't
need inbound right now) working with FFA and Gafachi. G711 faxing works
(as well as can be expected over the internet), but I want the higher
reliability of T38.
I'm running Asterisk 10-beta1.
When I drop my callfile in to make the call, I get this:
-- Attempting call on SIP/18884732963 at
2011 May 12
8
Light indicator managed by Asterisk
Hello,
is there some way to make Asterisk light up a certain light on an IP-phone ?
Like MWI, the message waiting indicator can light up if there is voicemail.
Could this light, or even other lights (like BLF-buttons) be used to
give a visual notification to the user ?
For example : if a certain value is set in the Mysql-DB and Asterisk
reads out this value, can Asterisk react upon it inside
2011 Mar 01
6
wav files are not playing asterisk
Hi
I try to play a wav file in asterisk ,but its accepting only gsm
files.Do u know where I need to change to make it works.
Thanks
Nikhil
2012 Feb 02
1
T38 faxing - UDPTL creation failed
Hello guys.
When I am trying to send fax through T38 to linksys SPA (properly
configured etc. - I have tried it with other systems), I'm getting error
and fax is not delivered.
I'm getting this errors in asterisk.log:
WARNING[687] udptl.c: No UDPTL ports remaining
ERROR[687] chan_sip.c: UDPTL creation failed
WARNING[687] udptl.c: No UDPTL ports remaining
then, couple lines down:
2013 Feb 04
1
CallerID external call after Attended Transfer
Hello,
using Asterisk 1.8.12.2
case :
I call with my cellphone to our public telephone number
Our receptionist answers the incoming call and does an attended transfer
to my colleague ( A )
Colleague answers and the receptionist tells him that I am on the other
side.
Receptionist transfers the call and I am connected to my colleague ( B )
My question is about the CallerID that the
2013 Nov 20
5
Movistar sip Mexico
Hello,
I have a problem with movistar in Mexico with a sip calls. Movistar send to
me T38 and G729 in the INVITE and they say that I have to ignore T38 and
use G729 in the voice call.
When a fax call is made Movistar send only T38 in the INVITE.
Invite example:
v=0
o=GDL-BMSW-12D 19913379 19899826 IN IP4 192.168.1.2
s=sip call
c=IN IP4 192.168.1.2
t=0 0
m=audio 6370 RTP/AVP 18 101
a=fmtp:18
2013 Apr 27
1
Radius Based Accounting for Asterisk
hi,
you still interesting in it?
that I made long time ago.
http://lists.digium.com/pipermail/asterisk-dev/2010-March/043096.html
also I keep another patches and things and I need dedicated ftp for
it. if you can give me such things I'll provide this patch to you.
On 3 February 2011 09:44, Nikhil <d.nikhil at cem-solutions.net> wrote:
> Hi everyone
> Any one used Radius based
2011 Apr 07
4
asterisk SIP MESSAGE method support
Hello List,
I have found that asterisk supports only forwards in-dialog MESSAGE method. That is, if the MESSAGE method is sent within an active call.
But according our requirement we need to send MESSAGE method to the other leg without being in a call (general stateless proxy forward). Is it possible to do this in asterisk using some tricks?
Regards,
Rajib Deka
SIEMENS Ltd.
Robert V Chandran
2011 Feb 25
5
[OT] Yealink IP Phones
Hello all,
After numerous issues with Snom phones (360/370/870) potentially looking to migrate too Yealink as their product range looks very promising indeed.
Are any of you using them with Asterisk ? How do they perform ? Do you use mass deployment at all ?
Would be very interested to hear from you.
--
Thanks, Phil
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2010 Dec 02
5
Push central phone book to phones
Hello,
I have Snom, Cisco, Grandstream & YeaLink phones.
Is there a way to push a centralized phone book to these phones ??
Kind regards,
Jonas.
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2010 Sep 22
1
Cross compile Asterisk for mipsel-linux
Hi
Anyone knows how to do cross compile asterisk 1.6.2.13 using
mipsel linux.?
Thanks
2012 Mar 05
1
Call notification on IP Telephone
Hi everybody,
I'm seeking information on how to report an IP phone
on a call that is occurring on another IP phone.
Example:
While the A phone is ringing, Asterisk sends a
notification to a phone B on the call that is going to A, but this
notification is displayed on the B phone display and the user does not
need to hit anything to view the information.
I'm
2011 Apr 13
4
[OT] Yealink Phones
I've just started deploying these (well the T28P model) after years of
Snom issues and they look pretty good (although the documentation is
execrable; if you thought the Snom stuff was obtuse Yealink have got
them knocked into a cocked hat!).
Anyway, for provisioning I use HTTP with a DHCP entry like:-
#
# Yealink Phones
#
group {
#