Caller ID over internal SIP Trunk

siptrunk
Tags: #<Tag:0x00007f7021da0de8>

(Iso Dan) #1

I’m having an issue with Caller ID over a SIP Trunk in my setup. The setup is, I have a client with remote FreePBXs that I need to route calls from. They are moving from a different VoIP service provider to our VoIP system. Because these FreePBX systems are remote and numerous I would prefer to not make changes to them if possible. So far I have been able to build SIP trunks between my “HUB” FreePBX and a few test remotes. The “HUB” PBX then has the trunks out to Sangoma Sipstation. The swap is possible using the configuration already on the remotes and changing a DNS entry that would then tie them to us. Calls are now working in and out without having to touch the remotes. The issue I am having is, when I call from a remote PBX I get the general number I configured with my Sipstation trunk as my caller ID on the receiving. Not the number configured as “Outbound CallerID” on the outbound route and trunk on the remote (these are already configured on the remotes so I don’t have to change them).

My trunk configurations on the “HUB” side are:

Trunk Name: Remote

Peer details:
username=HUB01
type=friend
secret=xxxxxxxxx
qualify=yes
nat=yes
host=dynamic
context=from-trunk

and remote side:

Trunk name: HUB01

Peer Details:

username=Remote
type=friend
sendrpid=yes
secret=xxxxx
nat=yes
keepalive=30
insecure=invite,port
host=placeholder.voip.blah.com
fromuser=Remote
fromdomain=placeholder.voip.blah.com
dtmfmode=rfc2833
context=from-trunk
canreinvite=no

I’m not sure is there is a way to do what I am asking. I wish I could just do IAX2 trunks between the devices but that would be a challenge and wouldn’t work on the non-asterisk PBXs the clients also has so I would like to find a way to do it with SIP if possible. If there is any more information that might help please let me know. This is the last thing I need to sort for this project and its been driving me nuts.


#2

On the hub PBX trunk from remote, set
trustrpid=yes
and test. If you still have trouble, at the Asterisk command prompt type
sip set debug on
and have the remote make a test call. If it’s working properly, the Asterisk log should show the caller ID in the incoming Remote-Party-ID header and it should be propagated to the outbound trunk. If so, but the called party still sees your trunk CID, check that the number is correctly formatted (2125551212 vs. 12125551212 vs. +12125551212).


(Iso Dan) #3

Boy oh Boy that worked perfect! You are the best!
Thank you so much!


(system) closed #4

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