Incoming calls closes 10 second duration


(Estados Unidos) #1

Hello, I have this problem, incoming calls are closed for 10 seconds connection duration.
I have freepbx distro 15, with two (2) network cards.

eth0: phones lines,
ADDRESS= 10.0.0.210
NETMASK=255.255.255.0
GATEWARE=10.0.0.1

eth1: Trunk line:.
ADDRESS= 192.168.1.100
NETMASK=255.255.255.0
GATEWARE=192.168.1.254

I need to know how to solve this nat problem, any help please?


(Communication Technologies) #2

https://wiki.freepbx.org/display/SUP/Providing+Great+Debug#ProvidingGreatDebug-AsteriskLogs-PartII

Also need to turn on SIP Debug so you can see what messages are being sent.


(Estados Unidos) #3

trunk is canal PJSIP/anonymous-000000ad, I not use PSJSIP ?

jue., oct. 1 2020 10:46 AM CHAN_START 8xxx DEFAULT 8xxx from-sip-external PJSIP/anonymous-000000ac
jue., oct. 1 2020 10:46 AM ANSWER 8xxx 8xxx 8xxx 8xxxx DEFAULT s ivr-1 Answer PJSIP/anonymous-000000ac
jue., oct. 1 2020 10:47 AM HANGUP 8xxx 8xxx 8xxx 8xxxx DEFAULT h ivr-1 PJSIP/anonymous-000000ac
jue., oct. 1 2020 10:47 AM CHAN_END 8xxx 8xxx 8xxx 8xxx DEFAULT h ivr-1 PJSIP/anonymous-000000ac
jue., oct. 1 2020 10:47 AM LINKEDID_END 8xxx 8xxx 8xxx 8xxx DEFAULT h ivr-1 PJSIP/anonymous-000000ac


(Communication Technologies) #4

This does not look like the instructions provided above. Without a true log it would be difficult for anyone to assist you.


(Estados Unidos) #5

I need know this:
Also need to turn on SIP Debug s

I try grep 1601566831.883 /var/log/asterisk/full*, but does nothing the asteristk skips a blank


(Communication Technologies) #6

Is the log issue related to this?


(system) closed #7

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