Sip Trunk unregistered by it self

[Wanted8000 (Jose)]

the pbx it uregisters this is the message.

[2018-07-27 18:52:48] NOTICE[25894]: chan_sip.c:30180 sip_poke_noanswer: Peer ‘Outgoing_ATTx’ is now UNREACHABLE! Last qualify: 27
[2018-07-27 18:53:03] NOTICE[25894]: chan_sip.c:15906 sip_reg_timeout: – Registration for ‘[email protected]’ timed out, trying again (Attempt #2)

then it gets to the 180th attempt and registers or if i do an amportal restart it registers right away and let me do call and receive them for a period of time then it unregisters again

my current conf is:

host=172.31.255.254
username=7877081350
remotesecret=xxxxxxxx
context=from-trunk
qualify=yes
nat=yes
disallow=all
allow=ulaw&g729
type=peer
insecure=very
sendrpid=yes
trustrpid=yes
canreinvite=yes
dtmfmode=rfc2833
faxdetect=yes

7877081350:[email protected]
Is my configuration wrong ?

[VoIPTek]

PLEASE MAKE SURE YOUR USERNAME IS NOT THE REAL USER NAME OR PEOPLE WILL START USING YOUR ACCOUNT!!!
Never use real IP’s or user names.

[VoIPTek]

Overall the settings look OK to me.
I would run a continuous ping to lets say google to maintain an internet connection.
This accomplishes 2 things in my mind.

1. prevents network from dropping since there is activity and if your trunk doesn’t drop, it may be some kind of timeout.
2. It tells you if maybe your internet provider is having an issue.

I hope that give you something to test.

[yois]

I had a similar issue that was caused by over-restrictive firewall. Make sure you’ve whitelisted any IP addresses your sip provider has given you. Many sip providers today will expect to send sip invites and acks from any of their IPs, not just the one you’ve registered to. PJSIP handles this much better as well in the “match” field.

[Wanted8000 (Jose)]

here is how is behaving now right before unregister send out this message

[2018-07-30 13:08:34] WARNING[9212]: chan_sip.c:4068 retrans_pkt: Retransmission timeout reached on transmission BW [email protected] for seqno 900127680 (Critical Response) – See https://wiki.asterisk.org/wiki/ display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response
[2018-07-30 13:08:34] WARNING[9212]: chan_sip.c:4092 retrans_pkt: Hanging up call [email protected] 1.1.4 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).

[VoIPTek]

Goot point @yois
Honestly you just have to check all your communications settings , firewall, and IP’s that everything is expecting and configured to.
It’s clearly not communicating.

[cynjut (Dave Burgess)]

Every time I’ve seen this is was a NAT settings problem on the trunk and/or a firewall not routing incoming traffic for the PBX correctly. I’d start with making sure you NAT is set up correctly. and then double check the settings in your router to make sure the inbound SYN packets for the system are getting routed correctly.

[Wanted8000 (Jose)]

here is my net diagram

[yois]

You’re saying you have nothing between freepbx and the internet? Then grab some packet captures from your firewall and make sure communication is taking place the way you want it to. Your sip traffic may be on the wrong NIC since you’re multihomed.

Make sure you have no gateway set on your internal nic.

[Wanted8000 (Jose)]

so no nat problem should exist in this in the diagram?

[Wanted8000 (Jose)]

change of tactics the telco At&t will do the registration for the pbx in their equipment an i get to delete all the registarion conf in the trunk now it works.