I have recently switched to another provider. Earlier I used to call someone subsequently transferring calls with microsip to an external number, with the number called first shown as the caller id during the second, transferred called. Now instead of the right caller id there is the “unavailable” word. What could change?
Here is the trace of a single call.
00042 1646842055 * ==> XXX.30.38.XXX:5060 INVITE sip:83952248110@XXX.30.38.XXX:5060 SIP/2.0
00043 1646842055 * <== XXX.30.38.XXX:5060 SIP/2.0 100 Trying
00044 1646842055 * <== XXX.30.38.XXX:5060 SIP/2.0 180 Ringing
00045 1646842055 * ==> XXX.30.38.XXX:5060 PRACK sip:83952248110@XXX.30.38.XXX:5060;transport=udp SIP/2.0
00046 1646842055 * ==> 10.137.148.41:61763 SIP/2.0 180 Ringing
00047 1646842055 * <== XXX.30.38.XXX:5060 SIP/2.0 200 OK
00048 1646842056 * <== XXX.30.38.XXX:5060 SIP/2.0 200 OK
00049 1646842056 * ==> XXX.30.38.XXX:5060 ACK sip:unavailable@XXX.30.38.XXX:5060;transport=udp SIP/2.0
The address 83952248110 eventually becomes unavailable.
Are you saying that Asterisk is sending “unavailable”, or that the final recipient sees it? In the latter case it will be because of the introduction of STIR/SHAKEN, to stop fraud, the alternative being to mark the numbers as probable spam.
In the former case, you will need to provide your, hopefully, pjsip configuration and dialplan.
Can I turn off STIR/SHAKEN?
No. It is a legal requirement on the service providers. In any case, the effect of turning it off would be to mark all calls from you as probable spam.
The fact that you even asked makes me think that you are so unaware of it that you didn’t really understand what I was saying.
Thank you. I have understood.
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