Re: [asterisk-users] 302 moved temporally callerid behavior
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 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 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" ;reason=deflection Content-Length: 0 I would like that the peer at number is receiving the real number 123456789, but it is receiving test-peer internal number. вт, 25 июн. 2019 г. в 18:05, Doug Lytle : > >>> 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 > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 302 moved temporally callerid behavior
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 302 moved temporally callerid behavior
>>> 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 302 moved temporally callerid behavior
Thanks for trying, what asterisk version do you use? вт, 25 июн. 2019 г. в 17:50, Doug Lytle : > 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 > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 302 moved temporally callerid behavior
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.st...@asterisk.open.source.it>: > On Tuesday 25 June 2019 at 15:06:55, Dovid Bender wrote: > > > 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. > > > On Tue, Jun 25, 2019 at 8:41 AM Kseniya Blashchuk wrote: > > > 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 > > -- > "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. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 302 moved temporally callerid behavior
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 302 moved temporally callerid behavior
On Tuesday 25 June 2019 at 15:06:55, Dovid Bender wrote: > 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. > On Tue, Jun 25, 2019 at 8:41 AM Kseniya Blashchuk wrote: > > 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 -- "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. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 302 moved temporally callerid behavior
Your doing an attended transfer what you want to do is a blind transfer. On Tue, Jun 25, 2019 at 8:41 AM Kseniya Blashchuk wrote: > 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 > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
