Hi,
I'm trying to connect two asterisk instances using the method described
here..
http://ofps.oreilly.com/titles/9781449332426/asterisk-OutsideConn.html
under the section
Connecting two Asterisk systems together with SIP
I have an user named venu in serverA and vijay in serverB
the serverA
@Alec,
Thanks.. That was the error.. got it working now.. :)
On Sun, May 5, 2013 at 2:34 PM, Alec Davis siva...@paradise.net.nz wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Sandeep Raju
, Sandeep Raju sandeepr...@practo.com wrote:
@Alec,
Thanks.. That was the error.. got it working now.. :)
On Sun, May 5, 2013 at 2:34 PM, Alec Davis siva...@paradise.net.nzwrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun
Hi,
I'm using Cicso - Linksys SPA3102 to connect to asterisk. I have followed
the official user manual and the blog post here
http://www.skelleton.net/2012/08/02/linksys-spa-3102/
When I call an extension say 225 from the analog phone, I can get the IVR I
have setup in my dialplan. But when I
it be distro-related?
I have various versions of asterisk (from 1.4 upto 11.3) running
paravirtualized or HW-virtualized with XEN.
Normally i use the pre-build packages from suse, only when i want to try
a release-candidates i need them myself.
hw
-Original Message-
From: Sandeep
@Hans,
Now I feel its distro related as I am getting the same error when I try to
compile and run asterisk 1.8.. what distro are you using? I think I need to
change the distro I'm running on..
On Tue, Apr 23, 2013 at 1:44 PM, Sandeep Raju sandeepr...@practo.comwrote:
Hi Hans,
If we use
@Hans, I just tried installing from pre-built packages (which has asterisk
1.8). Its working fine! :) only the compiled installed versions were
giving me the error!..
PS: sorry for spamming with multiple mails..
On Tue, Apr 23, 2013 at 2:10 PM, Sandeep Raju sandeepr...@practo.comwrote:
@Hans
1.8.22 and again
the problem was back..
so, i think the problem is same as the one in the issue...
On Tue, Apr 23, 2013 at 6:07 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Tue, Apr 23, 2013 at 02:17:47PM +0530, Sandeep Raju wrote:
@Hans, I just tried installing from pre-built packages
my gcc version is as follows
gcc (Ubuntu/Linaro 4.6.3-1ubuntu5) 4.6.3
On Tue, Apr 23, 2013 at 6:12 PM, Sandeep Raju sandeepr...@practo.comwrote:
@Tzafrir,
I uninstalled the version 11.2 and compiled the version 1.8.12.2 as
mentioned in that page... its working fine now.. as my virtual
Hi,
I'm trying to install Asterisk 11.2 on a virtual machine in my private
opestack cloud.. When I compile Asterisk 11.2 from source (./configure,
make, make install) as specified in the Asterisk book and run it, it gives
me the error: Illegal instruction (core dumped).
Any ideas how I can solve
@Doug, Yes.. I can compile other applications.
I discussed this issue on the #asterisk irc and they pointed me to this,
https://issues.asterisk.org/jira/browse/ASTERISK-20128
I think the issue is with my asterisk version (which is 11.2)... not sure
though!
Any help would be grateful :)
On
, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Mon, Apr 22, 2013 at 03:44:45PM +0530, Sandeep Raju wrote:
Hi,
I'm trying to install Asterisk 11.2 on a virtual machine in my private
opestack cloud.. When I compile Asterisk 11.2 from source (./configure,
make, make install) as specified
Hi All,
Can any body tell how to enable call forward facility in INDAI
for an asterisk IPPBX.
Regards,
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Hi All,
Can any body tell how to enable call forward facility in INDIA
for an asterisk IPPBX.
Regards,
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of calle's extension.
thanks
sandeep.___
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hi all,
how to establish a call between two asterisk servers for the sip users
registered for the servers.
- Original Message -
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, February 10, 2008 11:30 PM
Subject: asterisk-users Digest, Vol 43, Issue 30
Send
Hi all,
I am not getting the dial tone when i dial the zero digit.
And i am using analog card,for my operator phone caller id is not displaying on
the phone.I am in india.
In india is it possible to get the caller id for analog cards.
Can any body help me.
Please reply.
ThanksRegards,
hi all,
how to set the caller id facility for
the TDM400p card in INDIA.
thanks
sandeep.s
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hi all,
how to set the caller id facility for
the TDM400p card.
Please help me
thanks,
sandeep.s
- Original Message -
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, January 15, 2008 3:09 PM
Subject: asterisk-users Digest, Vol 42, Issue 51
Send
,
-Sandeep A
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Hi ,
Has anyone earlier tried integrating asterisk with LDAP.
I am interested to integrate LDAP for authentication purpose for any SIP
Incoming calls..
Pl. suggest pointers.
Thanks and Regards
--Sandeep Kalra
Ph: +91-120-4342000-X-2966
is immaterial.
Thanks and Regards
--Sandeep Kalra
Ph: +91-120-4342000-X-2966
: +91-120-4342966 (direct)
www.globallogic.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brent Torrenga
Sent: Monday, December 18
Try
Exten = _351217588XXX, 1, Dial ( ... )
Thanks and Regards
--Sandeep Kalra
Ph: +91-120-4342000-X-2966
: +91-120-4342966 (direct)
M- 9810683168
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joao Pereira
Sent
with
signelling fxsks
Is there any solution or work around for it .
I am ready to give more details or experimentation.
Thanks
-Sandeep
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In india no distributer for digium cards
If any body is going to us u can ask them to bring it.
I got in that way
-sandeep
Ankit wrote:
where did u purchase ur card frm, im not able to find ne distributor
of digium cards in india, and if i order it frm their site it will
have to pay arnd
of 10 hunks FAILED -- saving rejects to file apps/app_meetme.c.rej
my asterisk version is :
#asterisk -V gives
Asterisk CVS-HEAD-04/12/05-18:15:04
Pl suggest me what went wrong.
Thanks
Sandeep
Peter Svensson wrote:
On Thu, 2 Jun 2005, Mohamed A. Gombolaty wrote:
I was trying to make
For bridging VOIP with PSTN Lines
Which one is giving better performance SIP or IAX ?
I am looking at a result without NAT in picture ?
Can some body give details from experiance ?
Also with single SIP/IAX channel can I use more than one call at a time ?
Thanks
Sandeep
(SIP/abc2,10,t)
exten = 2158,1, Dial(SIP/xyz2,10,t)
exten = _1XXX,1, Dial(SIP/pbx/${EXTEN})
Thanks
Sandeep
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systems
[pbx]
type=friend
username=pbx
secret=pbx
host=192.168.X.Y
dtmfmode=info
insecure=very
qualify=no
disallow=all
allow=ulaw
Do you want any more details ?
thanks
-Sandeep
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phones in India that I got desperate and didnt know which list to post
this on.
--
Sandeep A.S [EMAIL PROTECTED]
Netcontinuum Pvt Ltd
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