SOLVED Incredible PBX 13.0.192.19 -- Call will not go through .

[phonebuff]

Asterisk 13.21.1,
Built back in 2018.
Hosted on RentPBX --
Useragent : Cisco ATA 186 v3.2.1 atasip (050616A)
Reg. Contact : sip:[email protected]:5060;user=phone;transport=udp

Fighting with an older system - Chan SIP - ATA on the end.. Calls from the Extension work, calls to the extension get a strange failure message that I don't remember ever having encountered before -- Reading the URL referenced does not point me in the right direction. I could do a wireshark I guess but SIP debug attached. Wondering if anyone had fought this error before.

    -- Connected line update to SIP/2101-0000015e prevented.
[2020-06-18 13:30:34] WARNING[24326]: chan_sip.c:4069 retrans_pkt: Retransmission timeout reached on transmission [email protected]:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 22593ms with no response
[2020-06-18 13:30:34] WARNING[24326]: chan_sip.c:4093 retrans_pkt: Hanging up call [email protected]:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
  == Everyone is busy/congested at this time (1:0/0/1)

Is the solution just to move to Vultr and a "new" build ??

 

[ostridge]

Cisco ATA 186

Just wondering if you have the sip firmware files for the 186 ??

 

[phonebuff]

Useragent : Cisco ATA 186 v3.2.1 atasip (050616A)

As it says == atasip --

 

[atsak]

[2020-06-18 13:30:34] WARNING[24326]: chan_sip.c:4069 retrans_pkt: Retransmission timeout reached on transmission [email protected]:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 22593ms with no response
[2020-06-18 13:30:34] WARNING[24326]: chan_sip.c:4093 retrans_pkt: Hanging up call [email protected]:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
== Everyone is busy/congested at this time (1:0/0/1)

This looks like a NAT problem - the ATA is not replying to the right place or something like that.

 

[phonebuff]

That was my first thought -- But the unit is registered and works fine when it originates a call, just will not work when a call is placed to it -

 

[atsak]

That was my first thought -- But the unit is registered and works fine when it originates a call, just will not work when a call is placed to it -

That can be though depending on how the NAT is being done. Some devices use a NAT setting for the registration server and some don't, some send the private IP instead of the public one when replying to SIP invite, or your firewall/router might be using an ALG that is doing SIP NAT on outbound but not inbound . . . or your firewall router might not be handling the inbound properly. Is the firmware up to date? I found the Cisco SPA's buggy years ago when I first used them (just after Sipura was folded in) but much better with newer firmware now . . .

 

[ostridge]

Useragent : Cisco ATA 186 v3.2.1 atasip (050616A) As it says == atasip --

ok, I saw that but I was needing to flash to sip and can't find the files?

 

[atsak]

I didn't realize how old the 186 is. I think you might want to invest in a newer one.

 

[tbrummell]

So you're on RentPBX, do you have a VPN link to your server, or is it public? The Contact is 10.0.0.3, which is a private address. If that's not routeable via a VPN, you have a NAT problem with that ATA.

 

[phonebuff]

Thank all for responding -- The situation has been solved. Had to do with a Default value that was changed and not overridden at the specific extension which is a literal ocean away from the server..

An yes the ATA-186 is question could order drinks in most US bars.