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
On 30 March 2013 21:32, aaaa aaaa indefix1@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(${EXTEN},b) exten => _1XX,102,Hangup
(You will need to reload/restart Asterisk after making these changes)
The issue you are having is an Asterisk one, not Kamailio and would be better asked in the Asterisk list if you dont get it working fr m the above..
Hope this helps.
-Barry
Barry, It solved with your guidance. Thanx!
________________________________ Απο: Barry Flanagan barry@flanagan.ie Προς: aaaa aaaa indefix1@yahoo.gr; Kamailio (SER) - Users Mailing List sr-users@lists.sip-router.org Στάλθηκε: 10:35 μ.μ. Κυριακή, 31 Μαρτίου 2013 Θέμα: Re: [SR-Users] Problem with Realtime Kamailio-Asterisk integration
On 30 March 2013 21:32, aaaa aaaa indefix1@yahoo.gr wrote:
Hello,
I followed the step by step guide (http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb) that describe the realtime integration between Kamailio and Asterisk. I have no problem with registration but when I try a call from 101 to 102 I get the followng error: [Mar 31 01:18:44] NOTICE[32330][C-00000006]: chan_sip.c:25195 handle_request_invite: Call from '101' (192.168.1.100:5060) to extension '103' rejected because extension not found in context 'DEFAULT NULL'.
Looks like you have an issue with your database table. The context being returned from the database appears to be "DEFAULT NULL" which is definitely not what you want. I don't know how you ended up with that value in your db by following the instructions.
Quick fix might be to change the value of the field sipusers.context for all entries to something - say "mycontext" and then in your asterisk extensions.conf make sure that you are defining your extensions in this context, like so:
[mycontext] ; our phones use 3 digit extensions, starting with 1 exten => _1XX,1,Dial(SIP/${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
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
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sip.conf
-- WBR, Victor JID: coyote@bks.tv JID: coyote@bryansktel.ru I use FREE operation system: 3.8.4-calculate GNU/Linux
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.
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Hello,
The port is configured in the sip.conf file.
Gj -----Original Message----- From: sr-users-bounces@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@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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