Hi All,
I am getting Congestion error when i am calling from 100 to 101 extension. Need your help and suggestion.
– Executing [0135153358@testing:1] Dial(“SIP/100-00000010”, “SIP/101”) in new stack
== Using SIP RTP CoS mark 5
– Called 101
– SIP/101-00000011 is ringing
– SIP/101-00000011 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Auto fallthrough, channel ‘SIP/0133429331-00000010’ status is ‘CONGESTION’
Hi shazissuet!
The dialplan is also needed for us to see what is wrong!
The fact that you get ringing followed by busy suggets that the problem lies with the destination device, not Asterisk.
Hi Nypon,
Here is the sip and extension configuration of this extension.
***sip.conf
[101]
type=friend
context=testing
host=dynamic
port=8080
username=101
secret=101
canreinvite=no
dtmfmode=rfc2833
insecure=port,invite
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
allow=ilbc
allow=h263
allow=speex
allow=g722
callerid=“101” <101>
nat=yes
qualify=4000
language=en
***extension.conf
exten => 101,1,Dial(SIP/101)
Hi David,
I have change the sip client but still getting the same issue.thanks
You need to provide SIP traces (sip set debug on).
Although none of these should make a difference for this issue, you probably want:
type=peer
delete insecure (the current setting makes toll fraud rather easy).
replace canreinvite = no (obsolete and usually sub-optimal for an extension) with directmedia=yes.
delete the quotes around 101
nat=no
qualify=yes (4000 seems excessive to me)
Does it work to call from 101 to 100?
Have you tried another phone at extension 100?