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Forums » Asterisk Support » rejected because extension not found
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| Author | Topic: rejected because extension not found | 818 Views |

7 July 2009 at 3:57pm
I recently installed asterisk and trying to get it working. My X-Lite phones are registered but i cant make calls between two phones. I only modified sip.conf & extensions.conf do i need to modify any other any other file as well.
[root@asterisk1 asterisk]# cat /etc/asterisk/extensions.conf
[globals]
PHONE1=SIP/1234
PHONE2=SIP/1111
[macro-oneline]
exten => s,1,Dial(${ARG1},20,t)
exten => s,2,Voicemail(u${MACRO_EXTEN})
exten => s,3,Hangup
exten => s,102,Voicemail(b${MACRO_EXTEN})
exten => s,103,Hangup
[local]
exten => 1234,1,Macro(oneline,${PHONE1})
exten => 1111,1,Macro(oneline,${PHONE2})
Sip.conf
[PHONE1]
type=friend
context=default
regexten=1234
callerid="User1" <1234>
host=dynamic
disallow=all
allow=ulaw
[PHONE2]
type=friend
context=default
regexten=1111
callerid="User2" <1111>
host=dynamic
disallow=all
allow=ulaw
I am getting the following error:
[Jul 7 16:54:52] NOTICE[27657] chan_sip.c: Call from 'PHONE1' to extension '1111' rejected because extension not found.
SIP Debug:
--- SIP read from UDP://10.16.103.244:35672 --->
INVITE sip:1111@10.16.103.173 SIP/2.0
Via: SIP/2.0/UDP 10.16.103.244:35672;branch=z9hG4bK-d8754z-443f1d4698659d5a-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:PHONE1@10.16.103.244:35672>
To: "1111"<sip:1111@10.16.103.173>
From: "1234"<sip:PHONE1@10.16.103.173>;tag=5c117172
Call-ID: ZGQyMTVkYmEzODI4MjI2YTQwZDUwZTAxZGRhZmIwODc.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1103d stamp 53117
Content-Length: 267
v=0
o=- 6 2 IN IP4 10.16.103.244
s=CounterPath X-Lite 3.0
c=IN IP4 10.16.103.244
t=0 0
m=audio 26956 RTP/AVP 107 0 8 101
a=alt:1 1 : RbbV/Xv3 90+Yl1m3 10.16.103.244 26956
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
--- (12 headers 11 lines) ---
== Using SIP RTP CoS mark 5
Sending to 10.16.103.244 : 35672 (NAT)
Using INVITE request as basis request - ZGQyMTVkYmEzODI4MjI2YTQwZDUwZTAxZGRhZmIwODc.
Found peer 'PHONE1' for 'PHONE1' from 10.16.103.244:35672
Found RTP audio format 107
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 10.16.103.244:26956
Found unknown media description format BV32 for ID 107
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.16.103.244:26956
Looking for 1111 in default (domain 10.16.103.173)
<--- Reliably Transmitting (no NAT) to 10.16.103.244:35672 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.16.103.244:35672;branch=z9hG4bK-d8754z-443f1d4698659d5a-1---d8754z-;received=10.16.103.244;rport=35672
From: "1234"<sip:PHONE1@10.16.103.173>;tag=5c117172
To: "1111"<sip:1111@10.16.103.173>;tag=as1dc2c5a7
Call-ID: ZGQyMTVkYmEzODI4MjI2YTQwZDUwZTAxZGRhZmIwODc.
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0
<------------>
[Jul 7 18:31:22] NOTICE[27657]: chan_sip.c:18160 handle_request_invite: Call from 'PHONE1' to extension '1111' rejected because ext
ension not found.
Scheduling destruction of SIP dialog 'ZGQyMTVkYmEzODI4MjI2YTQwZDUwZTAxZGRhZmIwODc.' in 32000 ms (Method: INVITE)
<--- SIP read from UDP://10.16.103.244:35672 --->
ACK sip:1111@10.16.103.173 SIP/2.0
Via: SIP/2.0/UDP 10.16.103.244:35672;branch=z9hG4bK-d8754z-443f1d4698659d5a-1---d8754z-;rport
To: "1111"<sip:1111@10.16.103.173>;tag=as1dc2c5a7
From: "1234"<sip:PHONE1@10.16.103.173>;tag=5c117172
Call-ID: ZGQyMTVkYmEzODI4MjI2YTQwZDUwZTAxZGRhZmIwODc.
CSeq: 1 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog 'ZGQyMTVkYmEzODI4MjI2YTQwZDUwZTAxZGRhZmIwODc.' Method: ACK

1 August 2010 at 2:33am
Dear
It is easy to start with Web GUI of asterisk server that is FreePBX....using that you dont have to manuly configuration into file but configure it through GUI interface

14 September 2010 at 11:14am
Sip.conf
[PHONE1]
type=friend
context=default
regexten=1234
callerid="User1" <1234>
host=dynamic
disallow=all
allow=ulaw
Context default does not exist in your extensions.conf so change it to context=local. If it does not work you may come back here.
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