“Caller rejected because extension not found” in asterisk
I'm an asterisk newbie and I've just installed it on my ubuntu 12.04 server with sudo apt-get install asterisk. I'm following the tutorial for asterisk from here: https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality
which describes setting up the most simplest PBX with two sip phones demo-alice and demo-bob. I've followed the instruction to the dot, however, when I make a call from demo-bob to demo-alice I get:
[Feb 5 13:23:03] NOTICE[13667]: chan_sip.c:22622 handle_request_invite:
Call from 'demo-bob' (192.168.1.2:5060) to extension '6001' rejected
because extension not found in context 'users'.
However, the dialplan gives me:
*CLI> dialplan show users
[ Context 'users' created by 'pbx_config' ]
'6001' => 1. Dial(SIP/demo-alice) [pbx_config]
'6002' => 1. Dial(SIP/demo-bob) [pbx_config]
-= 2 extensions (2 priorities) in 1 context. =-
So I clearly HAVE extension 6001 in context users. What am I doing wrong? Please help?
Some more info:
I've reloaded the config and dialplan with sip reload and dialplan reload. I'm using asterisk 1.8. This is the output of sip show peers:
*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
demo-alice/demo-alice 192.168.1.12 D N A 5060 Unmonitored
demo-bob/demo-bob 192.168.1.2 D N A 5060 Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
ubuntu-12.04 voip asterisk
asked Feb 5 '14 at 13:29
hogliuxhogliux
116112
add a comment |
I'm an asterisk newbie and I've just installed it on my ubuntu 12.04 server with sudo apt-get install asterisk. I'm following the tutorial for asterisk from here: https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality
which describes setting up the most simplest PBX with two sip phones demo-alice and demo-bob. I've followed the instruction to the dot, however, when I make a call from demo-bob to demo-alice I get:
[Feb 5 13:23:03] NOTICE[13667]: chan_sip.c:22622 handle_request_invite:
Call from 'demo-bob' (192.168.1.2:5060) to extension '6001' rejected
because extension not found in context 'users'.
However, the dialplan gives me:
*CLI> dialplan show users
[ Context 'users' created by 'pbx_config' ]
'6001' => 1. Dial(SIP/demo-alice) [pbx_config]
'6002' => 1. Dial(SIP/demo-bob) [pbx_config]
-= 2 extensions (2 priorities) in 1 context. =-
So I clearly HAVE extension 6001 in context users. What am I doing wrong? Please help?
Some more info:
I've reloaded the config and dialplan with sip reload and dialplan reload. I'm using asterisk 1.8. This is the output of sip show peers:
*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
demo-alice/demo-alice 192.168.1.12 D N A 5060 Unmonitored
demo-bob/demo-bob 192.168.1.2 D N A 5060 Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
ubuntu-12.04 voip asterisk
asked Feb 5 '14 at 13:29
hogliuxhogliux
116112
try something likechannel originate SIP/thufir extension 18003569377@outboundfrom the console. You'll have to to modify for your system.
– Thufir
Jan 11 '17 at 12:17
add a comment |
I'm an asterisk newbie and I've just installed it on my ubuntu 12.04 server with sudo apt-get install asterisk. I'm following the tutorial for asterisk from here: https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality
which describes setting up the most simplest PBX with two sip phones demo-alice and demo-bob. I've followed the instruction to the dot, however, when I make a call from demo-bob to demo-alice I get:
[Feb 5 13:23:03] NOTICE[13667]: chan_sip.c:22622 handle_request_invite:
Call from 'demo-bob' (192.168.1.2:5060) to extension '6001' rejected
because extension not found in context 'users'.
However, the dialplan gives me:
*CLI> dialplan show users
[ Context 'users' created by 'pbx_config' ]
'6001' => 1. Dial(SIP/demo-alice) [pbx_config]
'6002' => 1. Dial(SIP/demo-bob) [pbx_config]
-= 2 extensions (2 priorities) in 1 context. =-
So I clearly HAVE extension 6001 in context users. What am I doing wrong? Please help?
Some more info:
I've reloaded the config and dialplan with sip reload and dialplan reload. I'm using asterisk 1.8. This is the output of sip show peers:
*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
demo-alice/demo-alice 192.168.1.12 D N A 5060 Unmonitored
demo-bob/demo-bob 192.168.1.2 D N A 5060 Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
ubuntu-12.04 voip asterisk
asked Feb 5 '14 at 13:29
hogliuxhogliux
116112
I'm an asterisk newbie and I've just installed it on my ubuntu 12.04 server with sudo apt-get install asterisk. I'm following the tutorial for asterisk from here: https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality
which describes setting up the most simplest PBX with two sip phones demo-alice and demo-bob. I've followed the instruction to the dot, however, when I make a call from demo-bob to demo-alice I get:
[Feb 5 13:23:03] NOTICE[13667]: chan_sip.c:22622 handle_request_invite:
Call from 'demo-bob' (192.168.1.2:5060) to extension '6001' rejected
because extension not found in context 'users'.
However, the dialplan gives me:
*CLI> dialplan show users
[ Context 'users' created by 'pbx_config' ]
'6001' => 1. Dial(SIP/demo-alice) [pbx_config]
'6002' => 1. Dial(SIP/demo-bob) [pbx_config]
-= 2 extensions (2 priorities) in 1 context. =-
So I clearly HAVE extension 6001 in context users. What am I doing wrong? Please help?
Some more info:
I've reloaded the config and dialplan with sip reload and dialplan reload. I'm using asterisk 1.8. This is the output of sip show peers:
*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
demo-alice/demo-alice 192.168.1.12 D N A 5060 Unmonitored
demo-bob/demo-bob 192.168.1.2 D N A 5060 Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
ubuntu-12.04 voip asterisk
ubuntu-12.04 voip asterisk
asked Feb 5 '14 at 13:29
hogliuxhogliux
116112
asked Feb 5 '14 at 13:29
hogliuxhogliux
116112
asked Feb 5 '14 at 13:29
hogliuxhogliux
116112
asked Feb 5 '14 at 13:29
hogliuxhogliux
116112
asked Feb 5 '14 at 13:29
hogliuxhogliux
116112
116112
try something likechannel originate SIP/thufir extension 18003569377@outboundfrom the console. You'll have to to modify for your system.
– Thufir
Jan 11 '17 at 12:17
add a comment |
try something likechannel originate SIP/thufir extension 18003569377@outboundfrom the console. You'll have to to modify for your system.
– Thufir
Jan 11 '17 at 12:17
try something like
channel originate SIP/thufir extension 18003569377@outbound from the console. You'll have to to modify for your system.– Thufir
Jan 11 '17 at 12:17
try something like
channel originate SIP/thufir extension 18003569377@outbound from the console. You'll have to to modify for your system.– Thufir
Jan 11 '17 at 12:17
add a comment |
2 Answers
2
active
oldest
votes
Check context by
asterisk -rx "dialplan show 6001@users"
If not help, enable sip debug or general debug to ensure you are calling 6001.
answered Feb 6 '14 at 5:20
arheopsarheops
6452716
add a comment |
I had the issue as well, it was related to the Class of Service not containing the Outgoing line. When adding the newly made outgoing line to the default Class of Service, the issue disappeared.
answered Jan 5 '17 at 9:26
SaeXSaeX
294414
add a comment |
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2 Answers
2
active
oldest
votes
2 Answers
2
active
oldest
votes
active
oldest
votes
active
oldest
votes
Check context by
asterisk -rx "dialplan show 6001@users"
If not help, enable sip debug or general debug to ensure you are calling 6001.
answered Feb 6 '14 at 5:20
arheopsarheops
6452716
add a comment |
Check context by
asterisk -rx "dialplan show 6001@users"
If not help, enable sip debug or general debug to ensure you are calling 6001.
answered Feb 6 '14 at 5:20
arheopsarheops
6452716
add a comment |
Check context by
asterisk -rx "dialplan show 6001@users"
If not help, enable sip debug or general debug to ensure you are calling 6001.
answered Feb 6 '14 at 5:20
arheopsarheops
6452716
Check context by
asterisk -rx "dialplan show 6001@users"
If not help, enable sip debug or general debug to ensure you are calling 6001.
answered Feb 6 '14 at 5:20
arheopsarheops
6452716
answered Feb 6 '14 at 5:20
arheopsarheops
6452716
answered Feb 6 '14 at 5:20
arheopsarheops
6452716
answered Feb 6 '14 at 5:20
arheopsarheops
6452716
6452716
add a comment |
add a comment |
I had the issue as well, it was related to the Class of Service not containing the Outgoing line. When adding the newly made outgoing line to the default Class of Service, the issue disappeared.
answered Jan 5 '17 at 9:26
SaeXSaeX
294414
add a comment |
I had the issue as well, it was related to the Class of Service not containing the Outgoing line. When adding the newly made outgoing line to the default Class of Service, the issue disappeared.
answered Jan 5 '17 at 9:26
SaeXSaeX
294414
add a comment |
I had the issue as well, it was related to the Class of Service not containing the Outgoing line. When adding the newly made outgoing line to the default Class of Service, the issue disappeared.
answered Jan 5 '17 at 9:26
SaeXSaeX
294414
I had the issue as well, it was related to the Class of Service not containing the Outgoing line. When adding the newly made outgoing line to the default Class of Service, the issue disappeared.
answered Jan 5 '17 at 9:26
SaeXSaeX
294414
answered Jan 5 '17 at 9:26
SaeXSaeX
294414
answered Jan 5 '17 at 9:26
SaeXSaeX
294414
answered Jan 5 '17 at 9:26
SaeXSaeX
294414
294414
add a comment |
add a comment |
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try something like
channel originate SIP/thufir extension 18003569377@outboundfrom the console. You'll have to to modify for your system.– Thufir
Jan 11 '17 at 12:17