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Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: Teviot on September 20, 2010, 01:36:33 PM
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[Sep 20 21:33:42] ERROR[4511]: chan_iax2.c:4330 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 192.168.76.2 in the calltokenoptional list or setting user saintsme8b6 requirecalltoken=no
sme8b6*CLI>
Being new to the latest version of asterisk, what does this mean and how do I rectify it?
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This is not related to SME8 nor Sail2.6:
Add this line to your trunk:
requirecalltoken=no
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Tried that on both machines. I have a test machine and a production machine
here is the console output
[root@sme8b6 ~]# asterisk -rvvv
Asterisk 1.4.36, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
Connected to Asterisk 1.4.36 currently running on sme8b6 (pid = 3294)
Verbosity was 0 and is now 3
-- Executing [401@internal:1] AGI("SIP/5000-00000001", "selintra|OutCluster|401") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
-- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=default)
-- AGI Script selintra completed, returning 0
-- Executing [401@qrxvtmny:1] AGI("SIP/5000-00000001", "selintra|OutRoute|Free") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
-- AGI Script Executing Application: (Set) Options: (TIMEOUT(absolute)=3600)
-- Channel will hangup at 2010-09-23 18:11:08 UTC.
-- AGI Script Executing Application: (Dial) Options: (IAX2/sme8b6saint@saintsme8b6/401||T)
-- Called sme8b6saint@saintsme8b6/401
[Sep 24 03:11:08] NOTICE[3378]: chan_iax2.c:9435 socket_process: Host 192.168.76.3 failed to authenticate as sme8b6saint
[Sep 24 03:11:08] WARNING[3370]: chan_iax2.c:9197 socket_process: Call rejected by 192.168.76.3: No authority found
-- Hungup 'IAX2/saintsme8b6-22298'
== Everyone is busy/congested at this time (1:0/0/1)
-- AGI Script Executing Application: (Background) Options: (were-sorry)
-- <SIP/5000-00000001> Playing 'were-sorry' (language 'en')
-- AGI Script Executing Application: (Background) Options: (call-cannot-complete)
[Sep 24 03:11:10] WARNING[30248]: file.c:665 ast_openstream_full: File call-cannot-complete does not exist in any format
[Sep 24 03:11:10] WARNING[30248]: file.c:992 ast_streamfile: Unable to open call-cannot-complete (format 0x2 (gsm)): No such file or directory
[Sep 24 03:11:10] WARNING[30248]: pbx.c:5757 pbx_builtin_background: ast_streamfile failed on SIP/5000-00000001 for call-cannot-complete
-- AGI Script Executing Application: (Background) Options: (please-hang-up-and-try-again)
-- <SIP/5000-00000001> Playing 'please-hang-up-and-try-again' (language 'en')
-- AGI Script selintra completed, returning 0
== Auto fallthrough, channel 'SIP/5000-00000001' status is 'CHANUNAVAIL'
-- Executing [h@qrxvtmny:1] Hangup("SIP/5000-00000001", "") in new stack
== Spawn extension (qrxvtmny, h, 1) exited non-zero on 'SIP/5000-00000001'
sme8b6*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
[root@sme8b6 ~]#
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That's a different issue and not due to the lack of token.
See:
http://voip-info.linuxsys.com/wiki/view/Asterisk+No+authority+found.html
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I still can't get it to work
Machine 1 is called sme8b6
Machine 2 is called saint
The password on the 2 machines are the same password for both.
Franco can you tell me what I should have as the usernames on each machine?
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When trying to call from an extension on machine 2 (192.168.76.3) (production machine) is called "saint" to an extension on machine 1 (192.168.76.2)(test machine) called "sme8b6"
CLI on Machine 2
saint*CLI>
-- Got SIP response 400 "Bad Request" back from 192.168.76.45
-- Executing [5000@internal:1] AGI("SIP/401-b7d38f70", "selintra|OutCluster|5000") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
-- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=default)
-- AGI Script selintra completed, returning 0
-- Executing [5000@qrxvtmny:1] AGI("SIP/401-b7d38f70", "selintra|OutRoute|sme8b6") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
-- AGI Script Executing Application: (Set) Options: (__filename=1285264818-5000-401.wav)
-- AGI Script Executing Application: (Set) Options: (__channame=SIP/401-b7d38f70)
-- AGI Script Executing Application: (Monitor) Options: (wav|1285264818-5000-401|mb)
-- AGI Script Executing Application: (Set) Options: (TIMEOUT(absolute)=3600)
-- Channel will hangup at 2010-09-23 19:00:18 UTC.
-- AGI Script Executing Application: (Dial) Options: (IAX2/saintsme8b6@sme8b6saint/5000||T)
-- Called saintsme8b6@sme8b6saint/5000
[Sep 24 04:00:18] NOTICE[20156]: chan_iax2.c:7540 socket_process: Rejected connect attempt from 192.168.76.2, who was trying to reach '5000@'
[Sep 24 04:00:18] WARNING[20155]: chan_iax2.c:7750 socket_process: Call rejected by 192.168.76.2: No authority found
-- Hungup 'IAX2/sme8b6saint-16392'
== Everyone is busy/congested at this time (1:0/0/1)
-- AGI Script Executing Application: (Background) Options: (were-sorry)
-- <SIP/401-b7d38f70> Playing 'were-sorry' (language 'en')
-- AGI Script Executing Application: (Background) Options: (call-cannot-complete)
-- <SIP/401-b7d38f70> Playing 'call-cannot-complete' (language 'en')
== Spawn extension (qrxvtmny, 5000, 1) exited non-zero on 'SIP/401-b7d38f70'
-- Executing [h@qrxvtmny:1] Hangup("SIP/401-b7d38f70", "") in new stack
== Spawn extension (qrxvtmny, h, 1) exited non-zero on 'SIP/401-b7d38f70'
-- Got SIP response 400 "Bad Request" back from 192.168.76.45
saint*CLI>
Machine 1 type=peer
host=192.168.76.2
qualify=3000
canreinvite=no
requirecalltoken=no
username=sme8b6saint
fromuser=sme8b6saint
secret=**********
trunk=yes
disallow=all
allow=g729
allow=alaw
allow=ulaw
allow=gsm
and type=user
secret=**********
context=internal
requirecalltoken=no
Machine 2 type=peer
host=192.168.76.2
qualify=3000
canreinvite=no
username=saintsme8b6
fromuser=saintsme8b6
secret=**********
trunk=yes
disallow=all
allow=g729
allow=alaw
allow=ulaw
allow=gsm
and type=user
secret=**********
context=internal
requirecalltoken=no
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Teviot,
I just started playing with this edition, and so far I have seen a few bugs.
Let me dig a bit more before I give you a answer.
Best,
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You have coded the same host IP address for both sides of the link Teviot.
I suspect it may not work
S
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You have coded the same host IP address for both sides of the link Teviot.
I suspect it may not work
S
Well spotted S
I was tired when I set this up and have missed it everytime since. All works now.
Lesson to Self
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MAKE SURE YOR HAVE THE CORRECT IP ADDRESS or HOST NAME IN PLACE