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Disconnecting call 'SIP/XXX--XXX' for lack of RTP activity
Posted:
Wed Oct 26, 2016 4:31 pm
by Stinger554
Hello,
I'm managing some Vicidial-Asterisk servers that are getting several disconnecting call 'SIP/XXX--XXXXXX' for lack of RTP activity in 61 seconds errors. The servers are not behind the firewall/NAT they are sitting in the WAN and connect to the LAN so it shouldn't be a NAT/firewall issue.
The versions are:
Vicidial version
VERSION: 2.10-451a
BUILD: 140902-0816
Asterisk version
Asterisk 1.8.32.2-vici
OS version
openSUSE 13.1 (Bottle) (x86_64)
Any assistance would be greatly appreciated.
Thanks,
Stinger554
P.S. I can post a SIP.conf if you want me to as well as a SIP debug capture.
Re: Disconnecting call 'SIP/XXX--XXX' for lack of RTP activi
Posted:
Wed Oct 26, 2016 8:38 pm
by Rogger
Hello,
Post you carrier configuration carrier and Asterisk CLI.
A good tip is to test your service provider in a softphone.
Re: Disconnecting call 'SIP/XXX--XXX' for lack of RTP activi
Posted:
Thu Oct 27, 2016 9:57 am
by Stinger554
here is the SIP.conf which includes the carrier config. What did you want me to post from the Asterisk CLI?
[general]
context=trunkinbound
allowguest=no
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=ulaw
allow=gsm
mohinterpret=default
mohsuggest=default
language=en
relaxdtmf=yes
trustrpid = no
sendrpid = yes
progressinband=no
dtmfmode = rfc2833
videosupport=no
callevents=yes
rtptimeout=60
notifyringing = yes
notifyhold = yes
limitonpeers = yes
externip = xxx.xxx.xxx.xxx
localnet=192.168.1.0/255.255.255.0
localnet=10.0.0.0/255.0.0.0
localnet=172.16.0.0/12
localnet=169.254.0.0/255.255.0.0
nat=no
canreinvite=no
jbenable = yes
jbforce = no
jbmaxsize = 100
jbresyncthreshold = 1000
jbimpl = adaptive
jblog = no
qualify=yes
limitonpeer = yes
[general]
localnet=192.168.0.0/255.255.255.0
localnet=192.168.1.0/255.255.255.0
localnet=192.168.2.0/255.255.255.0
localnet=192.168.3.0/255.255.255.0
[vitel-inbound]
type=friend
context=trunkinbound
usecallerid=yes
trustrpid=yes
sendrpid=yes
dtmfmode=rfc2833
host=xxxxx.xxxx.xxxx
qualify=60
insecure=port,invite
allow=all
qualify=yes
allowguest=no
[xcast-outbound]
insecure=port,invite
secret=no
qualify=yes
host=xxxxx.xxxxx.xxxxx.
disallow=all
allow=ulaw
dtmfmode=rfc2833
externip=xxxxx.xxxx.xxx
nat=no
canreinvite=no
context=from-internal
type=peer
[100]
username=100
secret=secert
accountcode=100
callerid="" <>
mailbox=100
type=friend
host=dynamic
canreinvite=no
context=default
nat=no
qualify=yes
rtpkeepalive=0
disallow=all
allow=ulaw,alaw
Re: Disconnecting call 'SIP/XXX--XXX' for lack of RTP activi
Posted:
Thu Oct 27, 2016 5:47 pm
by Rogger
Hello,
Put your conf inside the VICIdial. ADMIN > CARRIERS and Asterisk CLI the error shown by Asterisk.