30 Mar
2013
30 Mar
'13
10:32 p.m.
Hello,
I followed the step by step guide
(http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb) that
describe the realtime integration between Kamailio and Asterisk. I have no problem with
registration but when I try a call from 101 to 102 I get the followng error:
[Mar 31 01:18:44] NOTICE[32330][C-00000006]: chan_sip.c:25195 handle_request_invite: Call
from '101' (192.168.1.100:5060) to extension '103' rejected because
extension not found in context 'DEFAULT NULL'.
Kamailio and Asterisk are running in the same machine.
Any idea about the cause of this problem?
Best Regards,
Theo
31 Mar
31 Mar
9:35 p.m.
On 30 March 2013 21:32, aaaa aaaa <indefix1(a)yahoo.gr> wrote:
Hello,
I followed the step by step guide (
http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb)
that describe the realtime integration between Kamailio and Asterisk. I
have no problem with registration but when I try a call from 101 to 102 I
get the followng error:
[Mar 31 01:18:44] NOTICE[32330][C-00000006]: chan_sip.c:25195
handle_request_invite: Call from '101' (192.168.1.100:5060) to extension
'103' rejected because extension not found in context 'DEFAULT NULL'.
Looks like you have an issue with your database table. The context being
returned from the database appears to be "DEFAULT NULL" which is definitely
not what you want. I don't know how you ended up with that value in your db
by following the instructions.
Quick fix might be to change the value of the field sipusers.context for
all entries to something - say "mycontext" and then in your asterisk
extensions.conf make sure that you are defining your extensions in this
context, like so:
[mycontext]
; our phones use 3 digit extensions, starting with 1
exten => _1XX,1,Dial(SIP/${EXTEN})
exten => _1XX,n,Voicemail(${EXTEN},u)
exten => _1XX,n,Hangup
exten => _1XX,101,Voicemail(${EXTEN},b)
exten => _1XX,102,Hangup
(You will need to reload/restart Asterisk after making these changes)
The issue you are having is an Asterisk one, not Kamailio and would be
better asked in the Asterisk list if you dont get it working fr m the
above..
Hope this helps.
-Barry
9:57 p.m.
New subject: Σχετ: Problem with Realtime Kamailio-Asterisk integration
Barry,
It solved with your guidance.
Thanx!
________________________________
Απο: Barry Flanagan <barry(a)flanagan.ie>
Προς: aaaa aaaa <indefix1(a)yahoo.gr>gr>; Kamailio (SER) - Users Mailing List
<sr-users(a)lists.sip-router.org>
Στάλθηκε: 10:35 μ.μ. Κυριακή, 31 Μαρτίου 2013
Θέμα: Re: [SR-Users] Problem with Realtime Kamailio-Asterisk integration
On 30 March 2013 21:32, aaaa aaaa <indefix1(a)yahoo.gr> wrote:
Hello,
I followed the step by step guide
(http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb) that
describe the realtime integration between Kamailio and Asterisk. I have no problem with
registration but when I try a call from 101 to 102 I get the followng error:
[Mar 31 01:18:44] NOTICE[32330][C-00000006]: chan_sip.c:25195 handle_request_invite: Call
from '101' (192.168.1.100:5060) to extension '103' rejected because
extension not found in context 'DEFAULT NULL'.
Looks like you have an issue with your database table. The context being returned from the
database appears to be "DEFAULT NULL" which is definitely not what you want. I
don't know how you ended up with that value in your db by following the instructions.
Quick fix might be to change the value of the field sipusers.context for all entries to
something - say "mycontext" and then in your asterisk extensions.conf make sure
that you are defining your extensions in this context, like so:
[mycontext]
; our phones use 3 digit extensions, starting with 1
exten => _1XX,1,Dial(SIP/${EXTEN})
exten => _1XX,n,Voicemail(${EXTEN},u)
exten => _1XX,n,Hangup
exten => _1XX,101,Voicemail(${EXTEN},b)
exten => _1XX,102,Hangup
(You will need to reload/restart Asterisk after making these changes)
The issue you are having is an Asterisk one, not Kamailio and would be better asked in the
Asterisk list if you dont get it working fr m the above..
Hope this helps.
-Barry
7 May
7 May
11:27 a.m.
New subject: Σχετ: Problem with Realtime Kamailio-Asterisk integration
hello :
I followed the step by step guide
(http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb)
that describe the realtime integration between Kamailio and Asterisk.
I have same question:
"Asterisk listens on IP 192.168.178.25 port 5080"
which asterisk's configuration file is setting?
zhengyw
--
View this message in context:
http://sip-router.1086192.n5.nabble.com/Problem-with-Realtime-Kamailio-Aste…
Sent from the Users mailing list archive at Nabble.com.
8 May
8 May
9:21 a.m.
New subject: Σχετ: Problem with Realtime Kamailio-Asterisk integration
sip.conf
--
WBR, Victor
JID: coyote(a)bks.tv
JID: coyote(a)bryansktel.ru
I use FREE operation system: 3.8.4-calculate GNU/Linux
9:32 a.m.
New subject: Σχετ: Problem with Realtime Kamailio-Asterisk integration
when I try a call from 101 to 102 I get the followng error :
1. in the asterisk console: " Unresolvable destination (478/SL)"
2. in the kamailio log:
May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: <core>
[resolve.c:1540]: ERROR: sip_hostport2su: could not resolve hostname:
"(null)"
May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: tm [ut.h:327]:
failed to resolve "(null)"
May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: tm [t_fwd.c:1530]:
ERROR: t_forward_nonack: failure to add branches
May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: sl
[sl_funcs.c:371]: ERROR: sl_reply_error used: Unresolvable destination
(478/SL)
Any idea about the cause of this problem?
ps:kamailio and asterisk are running in the same machine.
--
View this message in context:
http://sip-router.1086192.n5.nabble.com/Problem-with-Realtime-Kamailio-Aste…
Sent from the Users mailing list archive at Nabble.com.
12:07 p.m.
New subject: [SR-Users] Σχετ: Problem with Realtime Kamailio-Asterisk integration
Hello,
The port is configured in the sip.conf file.
Gj
-----Original Message-----
From: sr-users-bounces(a)lists.sip-router.org
[mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of zhengyw
Sent: dinsdag 7 mei 2013 11:27
To: sr-users(a)lists.sip-router.org
Subject: Re: [SR-Users] Σχετ: Problem with Realtime Kamailio-Asterisk
integration
hello :
I followed the step by step guide
(http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb
)
that describe the realtime integration between Kamailio and Asterisk.
I have same question:
"Asterisk listens on IP 192.168.178.25 port 5080"
which asterisk's configuration file is setting?
zhengyw
--
View this message in context:
http://sip-router.1086192.n5.nabble.com/Problem-with-Realtime-Kamailio-Aster
isk-integration-tp116944p118180.html
Sent from the Users mailing list archive at Nabble.com.
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6 comments
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participants (5)
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aaaa aaaa -
Barry Flanagan -
Gertjan Wolzak -
Victor V. Kustov -
zhengyw