Freepbx rejecting calls before reaching asterisk

FreePBX Providers
1

Hi All,

Hope everyone is doing well. I’m in a little bit of a pickle. Today our PBX system stopped receiving calls from our provider (Plivo). When doing some digging, CDR doesn’t show any calls, logging into the CLI and putting asterisk in debug mode, I don’t see anything on the screen. However, when I run a packet capture on the PBX system and make a call, I see the following:

488 Not Acceptable Here.

When I look at plivos end, I see the follwoing:

Hangup Cause sdp_not_acceptable_here_by_customer
Hangup Code 4640

I’m the sole administrator for this PBX and I have not changed/updated the system for at least a month (until today after the issue as a troubleshooting step).
I read a few forum posts but nothing has seemed to work.

I should note, calls to and from Twilio are working.

Below is the SIP info:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 13.52.9.100:5060;branch=z9hG4bKad1.64c9a76d32f6e9dd2259b854810af3a4.0
From: sip:[email protected]:5060;isup-oli=62;tag=gK02288be6
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 16624 INVITE
Max-Forwards: 64
Content-Length: 277
Content-Disposition: session; handling=required
Content-Type: application/sdp
User-Agent: Zentrunk
Accept: application/sdp
Allow: INVITE,ACK,CANCEL,BYE,UPDATE
P-Asserted-Identity: sip:[email protected]
Contact: sip:[email protected]

v=0
o=Sonus_UAC 23680 24208 IN IP4 18.214.109.221
s=SIP Media Capabilities
c=IN IP4 18.214.109.221
t=0 0
m=audio 15690 RTP/AVPF 0 101
a=maxptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=rtcp:15691 IN IP4 18.214.109.221
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 13.52.9.100:5060;rport=5060;received=13.52.9.100;branch=z9hG4bKad1.64c9a76d32f6e9dd2259b854810af3a4.0
Call-ID: [email protected]
From: <sip:[email protected];isup-oli=62>;tag=gK02288be6
To: <sip:[email protected]>
CSeq: 16624 INVITE
Server: FPBX-15.0.16.49(16.9.0)
Content-Length:  0

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 13.52.9.100:5060;rport=5060;received=13.52.9.100;branch=z9hG4bKad1.64c9a76d32f6e9dd2259b854810af3a4.0
Call-ID: [email protected]
From: <sip:[email protected];isup-oli=62>;tag=gK02288be6
To: <sip:[email protected]>;tag=9d596abe-ad7c-4fbd-bf36-a4a7ccf2c1f5
CSeq: 16624 INVITE
Server: FPBX-15.0.16.49(16.9.0)
Content-Length:  0

Any help is greatly appreciated

2

The SDP contains RTP/AVPF which is not what is normally used. Normally RTP/AVP is used. In the chan_sip module there is a configuration option called “avpf” to enable it. In the chan_pjsip module there is a configuration option named “use_avpf” to enable it. I don’t know where those are in the FreePBX GUI, but that’s likely your underlying problem. It’s possible they changed it on their side somewhere.

3

Looked around and didn’t see it. I’m guessing I’m going to need to go into the CLI and enable it and see if it works. I’ll let you know.

4

I added it to the plivo trunk under pjsip.endpoint.conf, restart asterisk using fwconsole restart, made sure it stayed and made a test call, no luck. I am going to remove it tho.

5

Hi @jcolp,

You were right. It was on the providers side. The issue is now resolved.
I’ve asked for a RCA, hopefully it sheds some light onto what happened.

6

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