NO JOY Odd one way issue

[RTOMikey]

Ok guys, this one has me stumped. I am running FPBX 12.0.65 all with *1.8, some upgraded from versions past, some using the ubuntu14 incredpbx. Terminations is great a around, however, after bringing everything to 12.0.65, origination is working if a call comes from a cell or other VoIP provider, but it a call comes in from an analog provider (Verizon, AT&T, etc). The caller gets silence, either on a direct did to an extension or an IVR. When it rings to an extension, the receiving extension CAN Hear the caller. Any thoughts on what would cause only the analog called to get one way audio?

 

[MGD4me]

What device is being used to connect to the POTS line?

 

[RTOMikey]

No pots lines...... The problem is when callers call FROM AN ANALOG SYTEM and Verizon hands it off th my VoIP provider

 

[atsak]

Nothing else changed by chance did it? Have you tried from more than one carrier for origination just for interest sake? This really feels like a fail2ban or other type of firewall problem.

 

[RTOMikey]

Nope nothing else changed......and of course VI is telling me that they dont know what is going on....... soo need to find a new upstream with the hosted billing they offer, but with better support

 

[briankelly63]

some providers proxy their RTP others don't so that may be why you see a difference depending on the call. Sounds to me like an Asterisk NAT situation. Check NAT setting on extensions assuming all other setting are correct.

 

[RTOMikey]

NAT is not an issue, the PBX is internet facing, and it makes no nevermind wether it is going to an internal, external extension, or even an IVR. Asterisk/full shows the call getting picked up for IVR and sent where it belongs, but the caller hears nothing past the pbx answering.

 

[RTOMikey]

also seeing all of a sudden tongiht, one of my servers is now refusing all inbound traffic, SIP Trace giving me WARNING[7693]: chan_sip.c:4136 retrans_pkt: Hanging up call [email protected] - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). lost on that one because the calls come in on IP 64.136.173.31 yet VI tells me they have not changed anything...... it just started using the IP in the CALLID header all by itself overnight I guess ughhhhhhhhhhhhhhh

 

[atsak]

Voip innovations hasn't changed anything - or I'd see what you see. Try turning off Fail2ban for a bit and see what happens.

 

[SkykingOH]

Somehow the call is getting reinvited by an SBC or other proxy. Asterisk 1.8 was the beginning of deep support for RPID and can display these issues. Try turning trust RPID off and see what happens.

 

[RTOMikey]

That di the Trick SkykingOH, Thank you so much, now if UI could just find out why two exchanges are not getting through to ANY of my servers, I would be ecstatic.... That problem started at the exact same time, and I can seem to resolve it.... all of a sudden calls from 570-644 and 570-648 cannot get through VI to my server, but if someone calls from one of those numbers to a DID hosted by either D4S or Alcazar, they go right through with no issue