Cloud hosted FreePBX on a public IP.
Few IP phones connected to the home router (Ubiquiti USG). Always work properly. No issues with IP phones.
I’m trying to connect an Adtran TA924 device to be able to connect some analog phones to the PBX.
FXS Port 1 on the Adtran is configured as Analog Station and pointed to a SIP trunk.
SIP trunk is pointed to the public IP of FreePBX and is showing registered on both Adtran and FreePBX.
I can make calls from my Buttset to any IP phone, but no call comes through from IP phone to the Buttset.
These are a few of the last lines from the CLI when I make a call to the Buttset.
Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio TOS bits 184 in TCLASS field.
== Using SIP RTP Audio CoS mark 5
== Using SIP RTP Video TOS bits 136
== Using SIP RTP Video TOS bits 136 in TCLASS field.
== Using SIP RTP Video CoS mark 4
– Connected line update to PJSIP/201-00000206 prevented.
The log fragment you posted is not relevant. Paste the complete log for a failed call into http://pastebin.freepbx.org/ and post the link here.
What does the caller hear (error message, voicemail, ring-no-answer, etc.)?
Why are you using trunks? I would assume that each FXS port could register as an extension on the PBX and calling that extension number would ring the corresponding FXS port. If you have a good reason for the trunks, please explain how they are configured.
I think it’s just what Adtran is calling a registration of the Adtran to the PBX. Because in the Adtran you have to create a Trunk once, and point it to the PBX IP address. Then in Adtran extensions, you always use that trunk, and just put in username and password for each extension. I shouldn’t probably even mention trunk then.
The thing is, both FreePBX and Adtran show this extension as registered. When I make a call from an IP phone to the buttset, the FreePBX just tries to ring the extension (since for FreePBX is an available peer) and the IP phone rings until it gets a message that the person is unavailable (just not picking up the phone). But you don’t receive anything on the buttset, and in the CLI the very last line says this:
– Connected line update to PJSIP/201-00000206 prevented.
When I call from IP phone to IP phone, the call goes through and the next line after this one says Ringing. In the above situation the CLI does not get to the line when it says Ringing.
Sorry, I’m traveling today and don’t have time to answer. Use SIP debug, a packet capture or the Adtran’s syslog to see whether it’s being sent an incoming INVITE and what response is given.
If no luck, at the Asterisk command prompt, type pjsip set logger on
or sip set debug on
according to whether Adtran entensions are pjsip or chan_sip. The SIP traffic will then appear in the regular Asterisk log.
Or, run tcpdump on the PBX and move the capture file to your PC for analysis with Wireshark.
I finally resolved the issue by switching everything to a default port 5060. Both FreePBX and Adtran. I usually never use the default port that’s why I started configuring the Adtran using the custom port.
Looks like I need to find where else I need to change the port from default to custom in order for the Adtran to work with a custom SIP port.