callerID on remote extensions


#1

Hi,

I 've some extensions which are remote to FreePBX system.
I ve VPN to all these remote sites.
When i am calling these phones from my LAN i do not see any caller id on the screen.

Do i ve to configure anything ?


#2

Do you notice any other issues with making and receiving calls from the remote extensions? If they are properly registering, you usually don’t need to do any special configuration for callerID on remote phones when calling from other extensions on your LAN. What make and model of phones are these? Do you have any trouble seeing the callerId on your LAN phones when one of the remote phones calls them? There could be some issues if the remote phones are on a different subnet, and that network is not covered in Settings->Asterisk SIP Settings->Local Networks, but this usually leads to bigger issues. A sip packet trace of a call might be needed for further diagnosis.


#3

hi,

they are Sangoma S505. They are on different subnets but i ve defined them on Asterisk SIP Settings->Local Networks.


#4

Hi,

This a trace from a call from a phone which is on the main site to the remote office

https://pastebin.freepbx.org/view/c8f43977

5000 is the main phone 113 is the remote, in this call no caller id is show to 113.

Here is the trace from the call from 113 to 5000

https://pastebin.freepbx.org/view/1047426a

Here the caller-id is show normally

Do you see any difference ?test.tgz (2,4 KB)

I ve upload also my pcap trace


(Lorne Gaetz) #5

Call traces look identical to me. The pcap shows Caller ID headers (both From and PAI) in the INVITE going to phone 113. Not sure what is going on here.


#6

Hi,

What else should i check ?


#7

I’m not seeing anything from the logs jump out at me either. Does callerID not show up on remote extensions if the caller is from an external number as well, or is it just from other internal extensions?

One thing you can try is changing the Send RPID setting in Ext113’s Extension Settings, in the Advanced section. Does trying the other options make a difference? I don’t think this would cause issues normally, but this is a strange situation. Can you show us the extensions asterisk options? You can get this from the asterisk cli by typing ‘sip show peer 113’, assuming this is a chan_sip extension. Also, is there any particular reason you’re using chan_sip over pjsip for these extensions? It shouldn’t matter for such a basic call scenario but it might help to know if you were experiencing issues with pjsip.