DROP CALL AFTER 6sec on Soho66 & Mutitel but all my other Provider working perfectly


(CptKing) #1

Hi there

Could anyone help me please! For some reason when I get an incoming call and answer them the call DROP after 6sec on Soho66 & Mutitel but all my other Providers working perfectly!

These are the setting: for soho66
authuser=1000******
username=1000******
fromuser=1000*******
fromdomain=sbc.soho66.co.uk
secret=********
host=sbc.soho66.co.uk
dtmfmode=info
disallow=all
allow=alaw&ulaw&gsm
type=peer
nat=yes
canreinvite=no
qualify=yes
port=8060
insecure=very
context=from-trunk

1000********:**8@sbc.soho66.co.uk:8060/1000

These are the setting: for multitel
username=1789*********
type=peer
secret=************
port=5060
outboundproxy=sbc-uk.multitel.net
insecure=port,invite
host=sbc-uk.multitel.net
fromuser=1789******
fromdomain=sbc-uk.multitel.net
dtmfmode=rfc2833
disallow=all
canreinvite=no
allow=alaw&ulaw
qualify=yes
nat=yes

:@sbc-uk.multitel.net/*****


(Itzik) #2

Why is reinvite disabled?

Please enable sip debug, sip set debug on reproduce this issue, paste the output to pastebin.freepbx.org and share the link here.


(CptKing) #3

Why is reinvite disabled?

they give me that setting. is it suppose to be yes?

i have sip set debug on how do I get the log?


(CptKing) #4

(post withdrawn by author, will be automatically deleted in 24 hours unless flagged)


#5

These logs are useless. It appears that you selected lines containing ‘soho66’ and ‘multitel’.

We need everything logged for a failing call. It should start with a line containing 'INVITE ’ and go through a line containing ‘no reply to our critical packet’ Please do this for a call delivered by each provider.


(CptKing) #6

ok how do I get the information from that


#7

Log is in /var/log/asterisk/full or in the GUI at Reports -> Asterisk Logfiles


(CptKing) #8

soho66 trunk

https://pastebin.freepbx.org/view/5b80ed5f

mulititel trunk

https://pastebin.freepbx.org/view/618f816b