It's entirely possible that something is not set up correctly, though I have checked and rechecked that anywhere that I have a chance to control the CID of a call passed through my system I maintain the CID of the originator of the call.
jlg84: It's entirely possible that something is not set up correctly, though I have checked and rechecked that anywhere that I have a chance to control the CID of a call passed through my system I maintain the CID of the originator of the call.
It is impossible for this to work without using a SIP302 redirect header, which is what the code above sets. Even then it requires support for this from your VoIP provider. FreePBX has the option to enable diversion headers which can work in some cases.
If you simply include your mobile in a ring group in a stock standard Asterisk install the call will originate from your PBX and will only show the CID of your system, as setting the CID to A party party isn't possible as it would be CID spoofing and will be blocked by your VoIP provider.
The minute your call is answered by your PBX however SIP302 redirects can't be used. In cases like this where SIP302 can't be used the solution I've deployed in the past has been to announce the CID to the mobile user when they answer the call "the following call is from 04 1234567" and then bridge the channels. I also created a solution giving the mobile user an IVR option or 1 or 2 to bridge the call or send it to the timeout destination in the ring group or queue.
I believe my provider allws this, since before installing Freepbx this worked find. If its just a matter of finding the place within Freepbx to choose diversion header, then I should be golden.
I have found the setting for "Generate Diversion Headers"and have clicked "true" (it was set to "false") so will check if this has worked after our guests for tonight finish their meal and retire to their rooms. Fingers crossed!
Acrobits is just a soft phone, and as such gets a caller id for the incoming call. It shows that caller ID. If your caller ID is enabled on the line, it will show on the SIP phones. If it all wars as it is meant to, you don't really need to do anything special, I don't think...
it worked because your phone was registered directly to "2 talk", as it is now registered to yr freepbx, it wont work unless you add the code as mentioned above, as at present it is the caller calling yr pbx, then yr pbx calling you and tranfering the original A party on yr answer.
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