External call dont show the Caller-ID of the caller

Hi,

I ve a few sangoma phones S505 and ive the follwing issue.
all incoming calls are going to a ring group with 3 phones. (reception)

and then the reception phones are tranfering the calls to the appropriate department.

The problem is that i cannot see the caller-id of the external caller but it sends the caller id of the recetption phone i.e the extension number

How can i overide these, because with previous Grandstream phones i didnt have any problem with the caller - id

Thanks

How are you initiating the transfer? SIP native transfers are indistinguishable from a new call on a second line until the transfer is completed.

If your phones are capable of connected line presentation updates, and you enable it (sendrpid not no for chan_sip), the number will update when the transfer is completed.

I haven’t personally used features transfers.

Hi,

I am not sure if i ve understand your question.
But i make an attended transfer.
The problem is that even in the history of the Sangoma phone in the details section it is not showing the callerid of the caller.

it is showing only the caller id of the extension

I hope that helps

I meant did you press the transfer button/soft key on the phone, or did you dial the FreePBX transfer code/used a soft key configured to do that.

The phone isn’t really involved in the latter case, but in the former case, the way that SIP works is that the phone starts a separate call to the destination, as though it were a new call. None of the SIP phones I’ve used have sent remote party ID information for the associated incoming call. Only when the attended transfer is completed does a SIP phone tell the PBX that the incoming and outgoing calls are related.

There is more on this topic, with a different SIP phone, at Caller ID with Attended vs Unattended Transfers

I press the tranfer button on the phone

sangoma phone have the following options for caller display source

From-only
PAI-FROM
PAI-RPID-FROM
RPID-PAI-FROM
RPID-FROM

Any idea what all of these means ?

Just use the SIP From: header.

Prefer the P-Asserted-Identity header, falling back to Remote-Party-ID, and finally to From, etc.

All except the first, when used, at the destination, with the appropriate Asterisk setting to send P-Asserted-Identity, or Remote-Party-ID, should result in the caller ID updating, on that destination phone, when the operator completes the transfer.

OK and in Asterisk settings in SIP sendrpid what option to choose
YES
PAI
NO

the dafault i think it was PAI

I’d expect PAI to work with all except From-Only and RPID-FROM.

Please note that connected line update only happens when the transfer is completed, not when the operator is announcing the call.

OK i ll give it a try

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