n packet).
SIP1 receive a redirect from SIP2 with SIP3 IP.
SIP1 makes a call to SIP3.
SIP3 finally helps in landing a call to “B”
All SIP are asterisk servers.
Please help in configuring asterisk to send 302 request back to the
server SIP1.
We are not able to get anywhere.
Regards,
A
Hi,
There isn't an astirectory driver for Asterisk version 1.4. So I guess
you'll have to use the asterisk realtime (res_config_ldap) driver.
cheers
Abhishek
On 9/5/07, Alessandro Russo [EMAIL PROTECTED] wrote:
Hi to all
I'm using Asterisk 1.4 and I'd like to use LDAP instead of sip.conf
Dear Mr Gavin,
Thank you once again. Will have to talk it over with my prof before
upgrading to Asterisk 1.4. The productive system is currently running on
1.2.6.
Thanks
Abhishek
On 8/28/07, Gavin Henry [EMAIL PROTECTED] wrote:
On 27/08/07, Abhishek M S [EMAIL PROTECTED] wrote:
Dear Mr Gavin
it be sufficiant if I were to copy the makefile and res_config_ldap.c
to the res/ directory of my running Asterisk and do make; make install? or
do I have to do LIBS=-lldap export LIBS ./configure before that? My asterisk
version is 1.2.6.
Thanks in advance,
Abhishek
*
*
* *
On 8/27/07, Gavin Henry
in the
extensions.conf file. Are there custom context modules for Asterisk
1.2.6version as well? If not, I'd really appreciate any suggestions or
help in
this regard.
thanks,
Abhishek
On 8/27/07, Seysan [EMAIL PROTECTED] wrote:
Thank you.
Now that the conexts are different can all the extension
/view.php?id=5768
Thank you for your time and patience,
Abhishek
On 8/27/07, Gavin Henry [EMAIL PROTECTED] wrote:
On 27/08/07, Abhishek M S [EMAIL PROTECTED] wrote:
Dear Mr Galvin,
Gavin! ;-)
As of today I am using the res_config_ldap of Astirectory in my test
Asterisk system
was the fact that there is no mapping required for SIP users and peers.
Regards
Abhishek
On 8/24/07, Gavin Henry [EMAIL PROTECTED] wrote:
Please see the official tracker in the Digium buglist:
http://bugs.digium.com/view.php?id=5768
Here are the schemas we did for OpenLDAP:
http
$ astRegseconds $ astname $ astLanguage ) )
Best Regards
Abhishek
On 8/16/07, Anthony Francis [EMAIL PROTECTED] wrote:
You will need to extend your schema to include all of the attributes
that can be used in sip.conf plus the extra ones that allow realtime to
store connection information. Please
user. I'd be extremely grateful for any
help or suggestion in this connection.
Thanks in advance,
Abhishek
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Dear Mr.Anthony,
Thank you so much for your responce.. Did go through the link as I'd done in
the past. Would really appreciate if you could give me more specific links
that could show a schema examples listing SIP attributes for LDAP.
Appreciate your time and patience
thanx
Abhishek
On 8/16/07
Title: Zorpia: Invitation Email
abhishek has sent you an invitation.
Hi ,
Your friend abhishek from
Sir
I want to set up asterisk server for only voip. I have installed it
but unable to configure the extensions and other things. Pl.
help.
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Hi
ryan ,
The header you are suspecting does not contains the
registration info. , it is actually the return path for the ACK which will get
generated in response to this packet.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Ryan
1 10:06:21 NOTICE[6019]: pbx_spool.c:266 attempt_thread: Call failed to
go through, reason 8
even when i can dial out manually through the same context(sip_proxy-out)
in sip.conf.
Can somebody help me out of this problem.
Thanks
Abhishek
of
my extension.
Please help me out . Thanks in advance.
Regards
Abhishek
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Hi,
This is test mail.
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Hi
terry ,
I am
an indian. Can u entertain my request , if so , i can send you my
resume
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I am facing problem in playing a wav or gsm file on asterisk. The error i
get whenever i tried is
*CLI -- Executing BackGround(SIP/1235-98f6,
/etc/asterisk-1.2.0/sounds/vm-goodbye.gsm) in new stack
Jan 13 20:08:57 WARNING[6181]: file.c:508 ast_openstream_full: File
: Friday, January 13, 2006 6:35 AM
Subject: Re: [Asterisk-Users] Please help
look in /etc/asterisk-1.2.0/sounds/ and see if you have sounds in that
directory.
On 1/14/06, Abhishek [EMAIL PROTECTED] wrote:
I am facing problem in playing a wav or gsm file on asterisk. The error i
get whenever i tried
help
just remember that in Background() Playback() and friends, you must
specify the file name without extension. Also, all sound should be in
the directory specified in asterisk.conf
Regards
On 1/14/06, Abhishek [EMAIL PROTECTED] wrote:
Hi tom,
Thanks for a very quick reply. But iam sure i am
I have tried Zyxel P2000W , it works very fine.
- Original Message -
From: Asterisk-User [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, January 10, 2006 3:23 AM
Subject: RE: [Asterisk-Users] Recommendations on
/0/0)
== Auto fallthrough, channel 'SIP/1235-e9b6' status is 'BUSY'
Configuration in sip.conf for 1234 is as
[1234]
type=friend
host=dynamic
context=default
username=1234
secret=1234
;regexten=1234
Abhishek
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exten= 1234,1,Dial(SIP/1234)
My clients are on Xlite softphone.
Can anybody help out ?/
Abhishek
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/${EXTEN:[EMAIL PROTECTED],30,r)
exten= 1234,1,Dial(SIP/1234)
My clients are on Xlite softphone.
Can anybody help out ?/
Abhishek
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= 1235,1,Dial(SIP/1235)
exten= _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
exten= 1234,1,Dial(SIP/1234)
Abhishek
- Original Message -
From: [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, December 21
Sir
I am a novice user and want to set up the asterix for only Voip as a project in my final yr. computer engineeering.
Pl. help me to do so . I will be highly thankful
Abhishek Gangal
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mine, on the stars of saturn
options:
Dione, Rhea, Titan, Mimas, Enceladus, Tethys, Hyperion, Iapetus, and Phoebe
Abhishek
--
Drishti-Soft Solutions Pvt Ltd
http://www.drishti-soft.com
On 5/12/05, Christopher Stephens [EMAIL PROTECTED] wrote:
Mine is called 'blacksun', as that's where it's
soft hangup channel name
-Abhishek
Drishti-Soft Solutions Pvt Ltd
http://www.drishti-soft.com
On 5/2/05, Asterisk guy [EMAIL PROTECTED] wrote:
1 after giving command oh323 show channels,
i want to disconnect a call, is there any command to disconnect a call?
2 how asterisk kill a hung
in pbx.c
-Abhishek
Drishti Soft
www.drishti-soft.com
On Sun, 13 Mar 2005 10:01:27 +, Umar Sear [EMAIL PROTECTED] wrote:
Checkout
http://www.voip-info.org/wiki-Asterisk+variables
I believe that should have the answer for you.
furthermore assuming that your number is always going
it working, what particular configuration used
on mediant (isdn signalling, framing, coding etc ??) and/or what configuration
on asterisk side ?? if anyone has it working, please send me the ini file for
the mediant and the zaptel.conf etc. would be extremely thankful.
Regards
Abhishek
Hi,
I am Abhishek from India.
I am have studying Cisco VOIP since a couple of months.Searching for Soft
PBX somenthing like (Cisco Callmanager) i came accros this Asterisk.
I have to provide a a solution to a clinet where he wants a connectivity
between his 3 offices across the WAN with a very
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