302 Moved Temporally Callerid Behavior
Hello!
I have a Polycom phone and sometimes I need to transfer calls without picking them up to local extensions. Polycom has a transfer button which sends SIP 302 packet to asterisk. Another local extension, receiving the call, sees not the original number but the local number that was transferring the call. I would like that the original number is shown. I am stuck at this point. I see messages like “Not accepting call completion offers from call-forward recipient” in the logs but I’m not sure if it’s somehow related to the problem. Can anybody help?
Asterisk 13.1.0 Ubuntu 16
10 thoughts on - 302 Moved Temporally Callerid Behavior
Your doing an attended transfer what you want to do is a blind transfer.
Surely “transfer calls without picking them up” is a blind transfer?
Antony.
—
“If I’ve told you once, I’ve told you a million times – stop exaggerating!”
Please reply to the list;
please *don’t* CC me.
—
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. The forwarded call showed my cell phone number, so I cannot reproduce.
Doug
—
The call is not actually picked up, there is a “Forward” button on the phone. After pressing it the phone sends a 302 Moved Temporally to asterisk and the call goes to another extension. I guess attended transfer is something else. Anyway, how is it connected with transferring the real callerid?
вт, 25 июн. 2019 г. в 17:35, Antony Stone < Antony.Stone@asterisk.open.source.it>:
Thanks for trying, what asterisk version do you use?
вт, 25 июн. 2019 г. в 17:50, Doug Lytle:
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.
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.
—
That would be correct.
The forward button on the polycom phones just do a redirect to the destination extension or external phone number.
Doug
—
core show version
Asterisk 13.26.0 built by doug @ asterisk on a x86_64 running Linux on 2019-04-05 11:41:43 UTC
Built from source,
Douh
—
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”;tag=as24ef1afd To: “Test Peer” ;tag=93AFFFD9-7DF89662
CSeq: 102 INVITE
Call-ID: 6143ff1e2dc860f04ebf7dc518fcb00d@1.2.3.4:5060
Contact:
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”;tag=as24ef1afd To: “Test Peer” ;tag=93AFFFD9-7DF89662;reason=deflection Content-Length: 0
CSeq: 102 INVITE
Call-ID: 6143ff1e2dc860f04ebf7dc518fcb00d@1.2.3.4:5060
Contact:
User-Agent: PolycomSoundPointIP-SPIP_650-UA/4.0.11.0583
Accept-Language: en Diversion: “Test Peer”
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:
Lol, everything was too simple. It was just a macro with app Dial with ‘f’
option configured. Normally I don’t use ‘f’, so I haven’t checked that 🙂
вт, 25 июн. 2019 г. в 19:05, Doug Lytle: