asterisk users - Jun 2019 - 302 moved temporally callerid behavior

On Tuesday 25 June 2019 at 16:49:23, Doug Lytle wrote:
> We have Polycom phones (I'm using a VVX601, the destination is a
VVX301).
> We're also on Asterisk 13.
> 
> I forwarded my call to the VVX301 and then dialed my phones DID.
Surely that is "call forwarding", which is quite different from either
a blind
or attended transfer?

A transfer involves a call coming in to phone A, which rings, a person at 
phone A transferring the call to phone B, and B answering it.

If the person at A speaks to B, it is an attended transfer; if A transfers the 
call without speaking to B (ie: B does not answer the call until A has 
completed the transfer), it is a blind transfer.
> The forwarded call showed my cell phone number, so I cannot reproduce.
Maybe the OP can outline precisely what is being done on the first phone which 
rings with the inbound call, so that we all know we're talking about the
same
situation?


Antony.

-- 
I still maintain the point that designing a monolithic kernel in 1991 is a 
fundamental error.  Be thankful you are not my student.  You would not get a 
high grade for such a design :-)
 - Andrew Tanenbaum to Linus Torvalds

                                                   Please reply to the list;
                                                         please *don't* CC
me.

>>> Surely that is "call forwarding", which is quite
different from either a blind or attended transfer?
That would be correct.

The forward button on the polycom phones just do a redirect to the destination
extension or external phone number.

Doug

This is what is actually going on:

Call is made to test-peer from number 123456789

 SIP/2.0 180 Ringing
 Via: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK4baae0d7;rport
 From: "Empty" <sip:123456789 at 1.2.3.4>;tag=as24ef1afd
 To: "Test Peer" <sip:test-peer at
4.3.2.1>;tag=93AFFFD9-7DF89662
 CSeq: 102 INVITE
 Call-ID: 6143ff1e2dc860f04ebf7dc518fcb00d at 1.2.3.4:5060
 Contact: <sip:test-peer at 4.3.2.1>
 User-Agent: PolycomSoundPointIP-SPIP_650-UA/4.0.11.0583
 Allow-Events: conference,talk,hold
 Accept-Language: en
 Content-Length: 0

Polycom redirects it to number 9999

 SIP/2.0 302 Moved Temporarily
 Via: SIP/2.0/UDP 1.2.3.4:5060;branch=z9hG4bK4baae0d7;rport
 From: "Empty" <sip:123456789 at 1.2.3.4>;tag=as24ef1afd
 To: "Test Peer" <sip:test-peer at
4.3.2.1>;tag=93AFFFD9-7DF89662
 CSeq: 102 INVITE
 Call-ID: 6143ff1e2dc860f04ebf7dc518fcb00d at 1.2.3.4:5060
 Contact: <sip:9999 at asterisk.example.com;user=phone>
 User-Agent: PolycomSoundPointIP-SPIP_650-UA/4.0.11.0583
 Accept-Language: en
 Diversion: "Test Peer" <sip:test-peer at
4.3.2.1>;reason=deflection
 Content-Length: 0

I would like that the peer at number 9999 is receiving the real number
123456789, but it is receiving test-peer internal number.

вт, 25 июн. 2019 г. в 18:05, Doug Lytle <support at drdos.info>:
> >>> Surely that is "call forwarding", which is quite
different from either
> a blind or attended transfer?
>
> That would be correct.
>
> The forward button on the polycom phones just do a redirect to the
> destination extension or external phone number.
>
> Doug
>
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