Hi all,
I think i understand the problem, actually I have two asterisk server. In
the extension.conf file on one server I have added
exten => 3923903,1,GOTO(s,1,3923903.conf)
which reads the corresponding conf file when ever the extension no. through
PSTN is called and learns the location of inbound.php which contains the IVR
script to be executed.
Now what i want is that through this inbound.php , i should be able to call
another asterisk server, where I have also configured twinkle as a
softphone.
The problems:
--I am not able to register this softphone on the previous asterisk server
as user 2001, though i modified the server's extension and sip file to
include the user 2001 under [phones] context.
---cli>> chan_sip.c:15839 handle_request_register: Registration from
'"user1" <sip:2001 at 172.26.48.208 <sip%3A2001 at
172.26.48.208>>' failed for
'172.26.48.62' - No matching peer found
shows this error upon registration..
-->at my server it shows 3 unmonitored peers, but the previous server
doesn't show any peers on sip show peers..though i have added all three
users in sip file, and yes reloaded the dial plan.
WARNING[9041]: chan_sip.c:2984 create_addr: No such host: 2001
[Jun 10 12:26:46] WARNING[9041]: app_dial.c:1237 dial_exec_full: Unable to
create channel of type 'SIP' (cause 20 - Unknown)
is the error when i do not give ip..assuming 2001 to be registered at the
server.
when i give the ip of my server..
chan_sip.c:20039 handle_request_invite: Call from '' to extension
'2001'
rejected because extension not found.
is the error..call actually lands up on asterisk server but it shows the
above error and ofcourse can not be recieved with softphone.
Please help me out in this regard. Though above details may be confusing..I
have tried to briefly write in case any more 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
contact at 9793905858
email: rit2007033 at iiita.ac.in
niksinghania at gmail.com
http://profile.iiita.ac.in/RIT2007033/
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---------- Forwarded message ----------
From: nikhil singhania <niksinghania at gmail.com>
Date: 10 June 2010 14:08
Subject: asterisk registration
To: asterisk-users at lists.digium.com
Cc: Ma Hu Ma <anshumishra6827 at gmail.com>
Hi all,
I think i understand the problem, actually I have two asterisk server. In
the extension.conf file on one server I have added
exten => 3923903,1,GOTO(s,1,3923903.conf)
which reads the corresponding conf file when ever the extension no. through
PSTN is called and learns the location of inbound.php which contains the IVR
script to be executed.
Now what i want is that through this inbound.php , i should be able to call
another asterisk server, where I have also configured twinkle as a
softphone.
The problems:
--I am not able to register this softphone on the previous asterisk server
as user 2001, though i modified the server's extension and sip file to
include the user 2001 under [phones] context.
---cli>> chan_sip.c:15839 handle_request_register: Registration from
'"user1" <sip:2001 at 172.26.48.208 <sip%3A2001 at
172.26.48.208>>' failed for
'172.26.48.62' - No matching peer found
shows this error upon registration..
-->at my server it shows 3 unmonitored peers, but the previous server
doesn't show any peers on sip show peers..though i have added all three
users in sip file, and yes reloaded the dial plan.
WARNING[9041]: chan_sip.c:2984 create_addr: No such host: 2001
[Jun 10 12:26:46] WARNING[9041]: app_dial.c:1237 dial_exec_full: Unable to
create channel of type 'SIP' (cause 20 - Unknown)
is the error when i do not give ip..assuming 2001 to be registered at the
server.
when i give the ip of my server..
chan_sip.c:20039 handle_request_invite: Call from '' to extension
'2001'
rejected because extension not found.
is the error..call actually lands up on asterisk server but it shows the
above error and ofcourse can not be recieved with softphone.
Please help me out in this regard. Though above details may be confusing..I
have tried to briefly write in case any more 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
contact at 9793905858
email: rit2007033 at iiita.ac.in
niksinghania at gmail.com
http://profile.iiita.ac.in/RIT2007033/
--
Nikhil Kumar
summer intern:simmortel voice technologies
rit2007033
b.tech IT 6th sem
IIIT Allahabad
contact at 9793905858
email: rit2007033 at iiita.ac.in
niksinghania at gmail.com
http://profile.iiita.ac.in/RIT2007033/
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