Hi All,
I’m a newbie here in asterisk.
just want to forward a call from myhotline# to to-some-cellphone-#
details of configuration below.
########################
sip.conf
[general]
allowguest=no
srvlookup=no
port=5060
bindaddr = 0.0.0.0
disallow=all
allow=alaw
allow=g722
context=internal
nat=yes
qualify=yes
externip=to-some-public-ip
localnet=172.31.14.119/20
[didlogic]
host=sip.sg.didlogic.net
user=12345
username=12345
fromuser=12345
authname=12345
secret=54321
insecure=port,invite
type=friend
qualify=yes
disallow=all
allow=ulaw
##############################
extensions.conf
[general]
static=yes
writeprotect=yes
priorityjumping=no
autofallthrough=no
[internal]
exten => myhotline#,1,Dial(SIP/didlogic/to-some-cellphone-#,30)
###############################
I’m getting this error in asterisk using sip set debug enable
== Using SIP RTP CoS mark 5
– Executing [myhotine#@internal:1] Dial(“SIP/didlogic-00000000”, “SIP/didlogic/to-some-cellphone-#,30”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/didlogic/to-some-cellphone-#
== Everyone is busy/congested at this time (1:0/0/1)
[Jun 2 11:42:16] WARNING[31435][C-00000000]: pbx.c:6724 __ast_pbx_run: Don’t know what to do with ‘SIP/didlogic-00000000’
I tried to google this but im afraid their are less information about this. has anyone encountered this problem?
Can someone point out what am i missing?
thanks
Your output is incomplete if you have sip set debug on.
You are missing the error message about the didlogic being unavailable.
The fact that the last count is 1 means that you got an unavailable status which generally means that a register or qualify failed. As you have qualify=yes, my guess is that didlogic are not responding to the OPTIONS packets that this sends.
type=friend is bad practice.
insecure=port is usually unnecessarily insecure.
Hi David55
thanks for the reply.
I’ve change in sip.conf
type=peer
but when I remove insecure=port,invite I’m getting a busy tone.
Here’s the full sip debug output
Connected to Asterisk 11.10.0 currently running on ip-172-31-14-119 (pid = 31203)
== Using SIP RTP CoS mark 5
– Executing [myhotline#@DID:1] Dial(“SIP/didlogic-00000000”, “SIP/to-some-cellphone-#@didlogic,30”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/to-some-cellphone-#@didlogic
== Everyone is busy/congested at this time (1:0/0/1)
[Jun 4 16:00:36] WARNING[31250][C-00000000]: pbx.c:6724 __ast_pbx_run: Don’t know what to do with ‘SIP/didlogic-00000000’
== Using SIP RTP CoS mark 5
– Executing [myhotline#@DID:1] Dial(“SIP/didlogic-00000002”, “SIP/to-some-cellphone-#@didlogic,30”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/to-some-cellphone-#@didlogic
== Everyone is busy/congested at this time (1:0/0/1)
[Jun 4 16:00:45] WARNING[31252][C-00000001]: pbx.c:6724 __ast_pbx_run: Don’t know what to do with ‘SIP/didlogic-00000002’
as for /var/log/asterisk/messages
WARNING[31530][C-00000000] pbx.c: Don’t know what to do with 'SIP/didlogic-00000000’
WARNING[31534][C-00000001] pbx.c: Don’t know what to do with ‘SIP/didlogic-00000002’
I said insecure=port was too insecure. Insecure=invite is a necessary relaxation as ITSPs don’t normally authenticate with their clients. (remotesecret, rather than secret, is a cleaner solution on recent versions).
Being too insecure doesn’t stop things working.
In any case, it only affects incoming calls.
As already noted, the most probable cause is that the ITSP doesn’t support qualify=yes.