Conference instead of MeetMe.
Thanks
Nikhil
On 12/08/2011 11:12 AM, Durgesh Mishra wrote:
Hi,
I am making confrence application.
In sip.conf
[phone1]
type=friend
host=dynamic
Takes an alphanumeric string.
context= employees
[phone2]
type=friend
host=dynamic
context= employees
[phone3]
type
.
Thanks
Nikhil
On 11/17/2011 08:43 PM, Danny Nicholas wrote:
In Asterisk 10.0 (you didn't state your version) you have sip qualify peer
and sip unregister peer
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nikhil
then also same result.
Is this feature available in asterisk ?
Thanks
Nikhil
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Does anyone face this issue.
Thanks
Nikhil
On 08/24/2011 10:10 AM, Nikhil wrote:
Hi
I am getting an issue when doing attended transfer from remote
server to asterisk.Asterisk is not sending BYE to replaced call once
it got invite with replaces from remote server.
scenario
handle_invite_replaces function,the sip_scheddestroy fun is
calling properly but still that dialog is not hangup up.
Asterisk version : 1.6.2.13
Note: Asterisk running in VOIP environment.
Please help on this.
Thanks
Nikhil
any answer on below..
On 08/18/2011 03:50 PM, Nikhil wrote:
Hi
Using AMI how can I get the presence feature.Below are the requirement.
-- List of all users in the PBX including analog and SIP
including registration status.
-- Status(BUSY or available ) of all users both analog
Hi
Using AMI how can I get the presence feature.Below are the requirement.
-- List of all users in the PBX including analog and SIP including
registration status.
-- Status(BUSY or available ) of all users both analog and SIP
Please help on this..
Thanks
Nikhil
Hi
How to send REFER with replaces from asterisk (Sending out) for
doing attended transfer.
Thanks
Nikhil
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Does anyone know about this...
On 06/20/2011 04:34 PM, Nikhil wrote:
Hi
In asterisk channel ,I so number of variable regarding the Codec ,Can
anyone explain what are those variable variable means.Below are the
variables
1. chan-readformat
2. chan-writeformat
3. chan
The rest of the logic happens in adhearsion.
--
Thanks,
Ishwar.
On Thu, Jul 28, 2011 at 6:33 PM, Nikhil d.nik...@cem-solutions.net
mailto:d.nik...@cem-solutions.net wrote:
Can you share the dialplan ,where SIP call is dialing...
Thanks
Nikhil
On 07/28/2011 06:15 PM, Ishwar
Hi
Any company proving radius based billing for asterisk only for
accounting ,not authenication and atherization.Please provide some links
Thanks
Nikhil
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Can you share the dialplan ,where SIP call is dialing...
Thanks
Nikhil
On 07/28/2011 06:15 PM, Ishwar Sridharan wrote:
Hello everybody,
We have an asterisk 1.8.4.1 setup, connected to a PRI line.
We're currently facing an issue where asterisk does not recognise the
event when the called
,and using CLI
command I can make calls outside and once call connected I can hear and
talk from my Headphone.
I planing to enhance chan_alsa module to get the features same as in
SIP client.
Thanks
Nikhil
On 07/26/2011 12:57 AM, Duncan Turnbull wrote:
Asterisk can run operator phones
. Is any other application available in asterisk to do this .
Thanks
Nikhil
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Hi
how to get all the users list that available in asterisk,
analog,sip,iax etc.Any cli command or AMI actions available to get this.
thanks
Nikhil
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in the SIP CC Transfer draft. (Transfer without
a GRUU)
In asterisk comment is written correct but it is not working.
Thanks
Nikhil
On 07/05/2011 09:44 PM, Kevin P. Fleming wrote:
On 07/05/2011 01:54 AM, Olivier wrote:
2011/7/5 Nikhil d.nik...@cem-solutions.net
mailto:d.nik...@cem
).
Does anyone knows about this?
Thanks
Nikhil
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Thanks for the helps
I use channel originate command to achieve this.
Command:
asteriskCLI channel originate SIP/201 application ConfBrigde 1234
This will make a call to the 201 user and when connected,it will be
routed to conference room .
Thanks
NIkhil
On 06/15/2011 02:17 PM, virendra
into the conference.I am using ConfBridge application for asterisk
version 1.6.2
Thanks
Nikhil
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any reply on below issue..
Nikhil
On 06/06/2011 07:23 PM, Nikhil wrote:
Hi
when I Upload MOH file from Asterisk GUI ,it is getting success and
even not getting any error,But if check the destination path the file
is not showing , even the source file and destination path and formate
in google that there is some known issue in asterisk 1.4 regarding
this ,I suspect this als o same .Can anyone explain what is expecting to
happen in back end when upload MOH file from asterisk GUI ,then I am
will able to debug more on this .
Thanks
Nikhil
On 06/03/2011 07:25 PM, Steve
Hi am
I am using 1.4 asterisk and asterisk GUI. If I do moh upload its is
not working .help me on this
Thanks
Nikhil
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I read about asterisk 1.10 in website https://wiki.asterisk.org. but
didnt find this release from asterisk community.
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Hi
How to set a threeway conference in asterisk only for VOIP (I am
using only SIP channel).
Thanks
Nikhil
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Hi
I would like to use asterisk as a SIP client(IP phohone ) with
multiple user,multiple line support . Using existing chan_alsa driver I
am not able to achieve my requirement . Please give some hint to do this .
Thanks
Nikhil
Hi all,
Does anyone compiled asterisk using NKD build in android. Please
give some suggestions.
Thanks
Nikhil
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Hi all
I am planning to use asterisk as a IP phone(Porting asterisk into a
hardware).Is there any limitations if I use asterisk as a SIP
client?,and asterisk has any advantages if use like this?
Thanks
Nikhil
document which explain how CDR works in asterisk and what are its
limitations.
Thanks
Nikhil
On 03/22/2011 11:35 AM, Tilghman Lesher wrote:
On Tuesday 22 March 2011 00:56:05 Nikhil wrote:
Hi all
In asterisk source code we can see lots of places
AST_CDR_FLAG_LOCKED flags is used.This
Thanks .. which version of asterisk have the CEL completely. Now I am
using 1.6.2 .
Thanks
Nikhil
On 03/22/2011 03:31 PM, Tilghman Lesher wrote:
On Tuesday 22 March 2011 02:20:24 Nikhil wrote:
Thanks for reply. I am trying to understand how CDR in asterisk is
working(Code wise),because some
Hi all
In asterisk source code we can see lots of places
AST_CDR_FLAG_LOCKED flags is used.This is for CDR purpose. Does anyone
what is exact usage of this lock in CDR.If I remove this flags where it
will impact,any data overwrite will happen..?
Thanks
Nikhil
Hi all
how to send SIP HOLD Invite from asterisk to other sip client/server.?
Thanks
Nikhil
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ok..that means I have to modify chan_sip . I wondering why this is not
available in asterisk.
Thanks
Nikhil.
On 03/16/2011 04:39 AM, Kevin P. Fleming wrote:
On 03/15/2011 04:18 AM, Nikhil wrote:
how to send SIP HOLD Invite from asterisk to other sip client/server.?
Asterisk's chan_sip
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
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New
Any reply..
On 03/01/2011 02:50 PM, Nikhil wrote:
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
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sory satish...my thunderbrid was not load.
Thanks for reply...
On 03/02/2011 09:59 AM, Nikhil wrote:
Any reply..
On 03/01/2011 02:50 PM, Nikhil wrote:
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
)
-- Executing [#@default:1] Playback(ALSA/hw:0,0, demo-thanks)
in new stack
-- ALSA/hw:0,0 Playing 'demo-thanks' (language 'en')
-- Executing [#@default:2] Hangup(ALSA/hw:0,0, ) in new stack
== Spawn extension (default, #, 2) exited non-zero on 'ALSA/hw:0,0'
Thanks
Nikhil
On 03/01/2011 07:55
Do I need to modify chan_phone application to make it works or it is
available in net.
Thanks
Nikhil
On 02/17/2011 12:52 PM, Khaled W. Chehab wrote:
Install asterisknow and begin from there.
http://www.asterisk.org/asterisknow/
and don’t miss to read the documentation
https
Hi
I wanted to use asterisk as SIP client in my centOS box.I should
able to make calls and receive calls.and should able to talk and listen
from the headset that I connected to my CentOS box.
I need a direction to start on this.
Thanks
Nikhil
waiting for replys..
On 02/11/2011 02:20 PM, Nikhil wrote:
Thanks for reply. Any other suggestions .
On 12/20/2010 05:52 PM, Service clients - VDI CENTER wrote:
i believe there is a way to do it using asterisk and flashphoner
++
2010/12/20 Gilles codecompl...@free.fr
Thanks for reply. Any other suggestions .
On 12/20/2010 05:52 PM, Service clients - VDI CENTER wrote:
i believe there is a way to do it using asterisk and flashphoner
++
2010/12/20 Gilles codecompl...@free.fr mailto:codecompl...@free.fr
On Fri, 17 Dec 2010 15:51:33 +0530, Nikhil
Hi everyone
Any one used Radius based accounting for asterisk.Please give me details.
Thanks
Nikhil
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Any suggestions ..
On 02/02/2011 09:04 AM, Nikhil wrote:
Hi everyone
How can I get the current calls details in asterisk.if I use cli
commad core show channels,there is two channels of each call.But the
requirement is, need to get caller ,calee,starttime ,duration of the
current
Hi Thorsten
Thanks of reply. The command core show verbose is working. but the
problem is, for one call we can see 2 results,there is no common field
on these two.How I parse these result to get the proper
values.(Caller,callee,starttime,duration).
Thanks
nikhil
On 02/02/2011 03:11 PM
.
Thanks
Nikhil
On 02/02/2011 03:58 PM, Ishfaq Malik wrote:
On Wed, 2011-02-02 at 09:04 +0530, Nikhil wrote:
Hi everyone
How can I get the current calls details in asterisk.if I use cli
commad core show channels,there is two channels of each call.But the
requirement is, need to get caller
,and scenarios.Please help me on this.
Thanks
Nikhil
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this is using..?
Thanks
Nikhil
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HI
In My asterisk OPTIONS packet is retransmitting continuously ,any one
know the reason for this.I am using asterisk 1.6.1.1.
Thanks
Nikhil
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New
SIPp is a good option.
Thanks
Nikhil
On 12/27/2010 11:38 PM, Bruce B wrote:
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
Hi
Enable debug level to more than 1 ,you may get something.
Thanks
Nikhil
On 12/22/2010 11:26 AM, Michael wrote:
Spawn extension (incoming-private, , 3) exited non-zero on
'SIP/Proxy-0031'
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On 12/17/2010 03:51 PM, Nikhil wrote:
Hi
Does anyone ported Asterisk to Android OS .please give details
Thanks
Nikhil
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reply please
On 12/17/2010 10:03 AM, Nikhil wrote:
Hi
Does anyone knows how to find out a call in a asterisk is
external incoming ,external out going or internal
Thanks
Nikhil
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Hi
Does anyone ported Asterisk to Android OS .please give details
Thanks
Nikhil
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Hi
Does anyone knows how to find out a call in a asterisk is external
incoming ,external out going or internal
Thanks
Nikhil
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New to Asterisk
(means copy paste the logic).Does this works or I need to do anything
else.waiting for reply to start change the code.
Thanks
Nikhil
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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
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of call forward it will show
properly.Please correct me I am wrong
Thanks
NIkhil
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anyone have a idea on this..
On 11/22/2010 10:50 AM, Nikhil wrote:
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
Thanks,Now I understand the problem,Now I am trying to change CDR to fix
these issues.
Thanks
Nikhil
On 12/01/2010 08:31 PM, Steve Murphy wrote:
On Wed, Dec 1, 2010 at 5:56 AM, Nikhil d.nik...@cem-solutions.net
mailto:d.nik...@cem-solutions.net wrote:
anyone have a idea
Hi everyone
Does anyone know how to check the TRANSFERRED Target Number is a
local extension or a PSTN number.
Thanks
Nikhil
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please reply on this if u know
On 11/18/2010 09:24 AM, Nikhil wrote:
Hi everyone
Anyone please explain me How Account code is use for billing.,
Thanks
Nikhil
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Check X-lite sending register request or not to asterisk buy checking
the asterisk console,if not there would some problem in X-lite
configuration settings,if sending check the console and see what error
logs you are getting..
Thanks
Nikhil
On 11/18/2010 04:06 PM, Phuong Hoang wrote:
Hi
on this.
Thanks
Nikhil
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Hi everyone
Anyone please explain me How Account code is use for billing.,
Thanks
Nikhil
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parking , Anyone have a solution on this. And anyone knows how to use
accoutcode and amaflags to solve this problem.
Thanks
Nikhil.
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Hi
Will radius client in asterisk can use with third party radius
servers instead of freeradius ?,if supports how do I configure asterisk
to make it work.
Thanks
Nikhil
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/sbin/asterisk -g for first asterisk
$ /home/asterisk2/usr/sbin/asterisk -g for second asterisk
Thanks
Nikhil
On 10/05/2010 03:42 PM, bilal ghayyad wrote:
Hi All;
Did anyone try to implement (installation and configuration and running) for
more than one asterisk instance (two or three instances
Hi
Anyone knows how to do cross compile asterisk 1.6.2.13 using
mipsel linux.?
Thanks
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Thanks
Nikhil
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On 09/16/2010 04:11 PM, A J Stiles wrote:
On Thursday 16 Sep 2010, Nikhil wrote:
Hi
I got the bellow error when I try to configure asterisk code.
$./configure --with-ssl=/usr/local/ssl
...
...
...
checking for mandatory modules: OPENSSL... fail
configure: ***
configure
Its working now when I installed openssl using yum.
yum install -y openssl-devel.
Thanks
Nikhil
On 09/16/2010 05:26 PM, Nikhil wrote:
On 09/16/2010 04:11 PM, A J Stiles wrote:
On Thursday 16 Sep 2010, Nikhil wrote:
Hi
I got the bellow error when I try to configure asterisk
Hi all,
I would like to install asterisk as my home pbx, Anyone can suggest
which sub version of 1.6 is stable?
Thanks
Nikhil
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New to Asterisk
to a halt, with massive packet loss to other applications. At that point,
not even (a sane amount of) money helps, as you can't buy a higher upload
rate (aside from regrading to ADSL2+, which I'm looking into now).
Thanks in advance,
Nikhil
packets, rather than the 30 or so I was expecting - and these
included about 4000 packets arriving from one host with SIP registration
attempts, fully 200 per second...
Best,
Nikhil.
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, what ever would be implemented easily.
--
Nikhil Kumar
summer intern:simmortel voice technologies
rit2007033
b.tech IT 6th sem
IIIT Allahabad
cont...@9793905858
email: rit2007...@iiita.ac.in
niksingha...@gmail.com
http://profile.iiita.ac.in/RIT2007033
HI list-users,
Greetings!!
I have been using call file, i playback my file using *
application:playback*
and once the playback is over the call is disconnected. Is there any way it
can wait and also record the dtmf inputs once the playback is over.
Thanks in advace
Nikhil Kumar
summer
functions which we can
use , to play some file or actually control this call. I have to play some
file and get the user response sitting on the server itself.
Please can anyone help!!!
--
Nikhil Kumar
summer intern:simmortel voice technologies
rit2007033
b.tech IT 6th sem
IIIT Allahabad
cont
the ringing tone in the
phone which called and not able to talk . What may be the problem.
--
Nikhil Kumar
summer intern:simmortel voice technologies
rit2007033
b.tech IT 6th sem
IIIT Allahabad
cont...@9793905858
email: rit2007...@iiita.ac.in
niksingha...@gmail.com
http://profile.iiita.ac.in
hello world echoed on output.
--
Nikhil Kumar
summer intern:simmortel voice technologies
rit2007033
b.tech IT 6th sem
IIIT Allahabad
cont...@9793905858
email: rit2007...@iiita.ac.in
niksingha...@gmail.com
http://profile.iiita.ac.in/RIT2007033
, Zeeshan Zakaria zisha...@gmail.com wrote:
you should post this to the list, not to my personal email.
Zeeshan A Zakaria
--
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On 2010-06-16 2:45 AM, nikhil singhania niksingha...@gmail.com wrote:
Here is my extensions.conf:
[general]
static=yes ; default
of the asterisk server
i.e. 172.26.48.208. In the softphone it shows registration successful.
Thanks in advance
--
Nikhil Kumar
summer intern:simmortel voice technologies
rit2007033
b.tech IT 6th sem
IIIT Allahabad
cont...@9793905858
email: rit2007...@iiita.ac.in
niksingha...@gmail.com
-- Forwarded message --
From: nikhil singhania niksingha...@gmail.com
Date: 16 June 2010 12:15
Subject: Re: [asterisk-users] can't seem to register, status unmonitored
To: Zeeshan Zakaria zisha...@gmail.com
Here is my extensions.conf:
[general]
static=yes ; default
Hi everybody,
I am trying to register my softphone(twinkle) on an asterisk server.
Everything seems to be fine.
Here is the output on show registrations in twinkle:
Tue 18:57:51
nikhil: you have the following registrations
sip:2...@172.26.48.208 sip%3a2...@172.26.48.208;expires=3013
208 is ip
-15 4:11 AM, nikhil singhania niksingha...@gmail.com wrote:
Hi everybody,
I am trying to register my softphone(twinkle) on an asterisk server.
Everything seems to be fine.
Here is the output on show registrations in twinkle:
Tue 18:57:51
nikhil: you have the following registrations
sip:2
written a
simple php script which utilises the exec_dial function inbuilt in
phpagi.php file.
I have tried diff ways but can't seem to get it work.
Can please some one suggest me anything in this regard.
--
Nikhil Kumar
summer intern:simmortel voice technologies
rit2007033
b.tech IT 6th sem
IIIT
information as the ip of the asterisk server
i.e. 172.26.48.208. In the softphone it shows registration successful.
Thanks in advance
Nikhil Kumar
summer intern:simmortel voice technologies
rit2007033
b.tech IT 6th sem
IIIT Allahabad
cont...@9793905858
email: rit2007...@iiita.ac.in
-- Forwarded message --
From: nikhil singhania niksingha...@gmail.com
Date: 10 June 2010 14:08
Subject: asterisk registration
To: asterisk-users@lists.digium.com
Cc: Ma Hu Ma anshumishra6...@gmail.com
Hi all,
I think i understand the problem, actually I have two asterisk server
explanation needed, please mail
me.I am stuck in this so please help.
Thanks in advance
Nikhil Kumar
summer intern:simmortel voice technologies
rit2007033
b.tech IT 6th sem
IIIT Allahabad
cont...@9793905858
email: rit2007...@iiita.ac.in
niksingha...@gmail.com
http://profile.iiita.ac.in
i dial through PSTN it gives beeps sound, but without this line
program runs smoothly.
Can someone help???
--
Nikhil Kumar
rit2007033
b.tech IT 6th sem
IIIT Allahabad
cont...@9793905858
email: rit2007...@iiita.ac.in
niksingha...@gmail.com
http://profile.iiita.ac.in/RIT2007033
scenarios (router or phone line down) the DNS requests weren't being
answered, so Asterisk was waiting, preventing it from doing other things
properly.
So, as suggested, I'll go ahead and install dnsmasq (or similar), and that
should fix things.
Cheers,
Nikhil
: no luck.
- Rebooted machine and tried again: still no luck.
Again, the logs indicate that Asterisk thinks the SIP phones are
unreachable.
Was there anything special I needed to do with the setup of dnsmasq, or
its interface with Asterisk? If not, I'm stuck again.
Thoughts?
Nikhil
that's
correct. eth0 has address 10.9.8.1, while eth1 has a global internet IP
address.
Cheers,
Nikhil.
-
Extract from sip.conf:
[general]
context=incoming
srvlookup=yes
realm=nikhil-nair.net
; Various register= statements, not relevant to the local phones
[101] ; Aastra 9112i at 10.9.8.101
On Thu, 4 Feb 2010, Joseph wrote:
[...]
Does your router runs DHCPD, assigning network addresses on on your LAN?
Nope, the DHCPD on the ADSL router is disabled, because I'm running DHCPD
on the Debian box. In any case, the ADSL router is not directly
accessible from the local net.
If
On Thu, 4 Feb 2010, Joseph wrote:
On 02/05/10 02:35, Nikhil Nair wrote:
On Thu, 4 Feb 2010, Joseph wrote:
[...]
Does your router runs DHCPD, assigning network addresses on on your LAN?
Nope, the DHCPD on the ADSL router is disabled, because I'm running DHCPD
on the Debian box. In any
On Thu, 4 Feb 2010, Joseph wrote:
On 02/05/10 02:05, Nikhil Nair wrote:
Extract from sip.conf:
[general]
context=incoming
srvlookup=yes
realm=nikhil-nair.net
Your resolve authentication to an outside server, isn't it?
No, that's just a Realm string which has to match when the Asterisk
, I'm blind, which partly limits my choice of applications (I've been
using console based ones, as I haven't worked out how to use X).
Best wishes,
Nikhil.
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to pay a modest amount for this.
Thanks in advance for any suggestions!
Best wishes,
Nikhil.
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, with these options, I should have ample opportunity to test
things properly.
BTW, Igor and Atis: sorry, I certainly didn't intend to start any sort of
an argument, however short! For what it's worth, I feel you've both been
generous, and I very much appreciate that.
Best regards,
Nikhil
much appreciated!
Best wishes,
Nikhil.
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, apparently). Maybe Linphone running on the
same box as Asterisk will be the solution, after all.
That is, if Linphonec will consent to run for me without giving all sorts
of errors and segfaulting... and I thought this release of Debian was
supposed to be stable! ;) Ah well.
Cheers,
Nikhil
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