I have my 13-13 system working, except that a remote phone (Yealink T46G) isn't connecting correctly.
The phone was working fine under PBX Green, and I am using the same extension and password - so those two items aren't the issue.
The phone will show as registered, but there is no audio - either in or out. The no audio is true for extension to extension and outgoing calls.
I set sip set debug peer 5000 (the extension on)
This is what I get:
<--- SIP read from UDP:66.XX.XX.XX:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.40.29:5060;branch=z9hG4bK0989280f;rport =5060
From: "Unknown" <sip:[email protected]>;tag=as7ce55944
To: <sip:[email protected]:5060>;tag=689071486
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Yealink SIP-T46G 28.83.0.50
Content-Length: 0
66.XX.XX.XX is his public IP
192.168.40.29 is the IP of the PBX in the basement
5000 is his extension
192.168.0.28 is his private IP on his network
I have PJSIP disabled. I have every NAT setting I can find set to yes. Did I miss something?
What settings should I look at and what am I looking for if I try to capture a call between extensions?
Thanks all.
Andrew
The phone was working fine under PBX Green, and I am using the same extension and password - so those two items aren't the issue.
The phone will show as registered, but there is no audio - either in or out. The no audio is true for extension to extension and outgoing calls.
I set sip set debug peer 5000 (the extension on)
This is what I get:
<--- SIP read from UDP:66.XX.XX.XX:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.40.29:5060;branch=z9hG4bK0989280f;rport =5060
From: "Unknown" <sip:[email protected]>;tag=as7ce55944
To: <sip:[email protected]:5060>;tag=689071486
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Yealink SIP-T46G 28.83.0.50
Content-Length: 0
66.XX.XX.XX is his public IP
192.168.40.29 is the IP of the PBX in the basement
5000 is his extension
192.168.0.28 is his private IP on his network
I have PJSIP disabled. I have every NAT setting I can find set to yes. Did I miss something?
What settings should I look at and what am I looking for if I try to capture a call between extensions?
Thanks all.
Andrew