Am 28.12.2021 um 20:24 schrieb Antony Stone: > No, you want to look at the "180 Ringing" response in both cases - what goes > in to Asterisk, and what comes out of it.
OK > No, data FROM Deutsche Telekom. They are the ones sending the "180 Ringing" > back to you once they think the external telephone is ringing. OK. So I sniffed data from internal network and from DSL, then I started the call using the web management system of the SNOM. I see Asterisk sends to the phone: Via: SIP/2.0/UDP 192.168.60.53:3072;branch=z9hG4bK-igym6msxn88p;received=192.168.60.53;rport=3072 From: "Sekretariat" <sip:74@192.168.60.1>;tag=ts2ye4krhs To: <sip:01773218409@192.168.60.1;user=phone>;tag=as32fe51ba Call-ID: 313634303731393637343630373636-ex7145moy1mt CSeq: 2 INVITE Server: Asterisk PBX 16.2.1~dfsg-1+deb10u2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Contact: <sip:01773218409@192.168.60.1:5060> Content-Length: 0 After about 6 seconds I get from the Telekom: Via: SIP/2.0/UDP 87.191.224.158:5060;received=87.191.224.158;rport=5060;branch=z9hG4bKPj43be873a-cf55-4348-8867-5c2bb97bd76a To: <sip:01773218...@tel.t-online.de>;tag=h7g4Esbg_p65544t1640719676m169304c9321s1_3514393582-932943693 From: <sip:03529529...@tel.t-online.de>;tag=4781eb96-b155-421e-8206-593d44c9f7c4 Call-ID: 478ba582-946c-46ac-984d-6f1835e3391b CSeq: 15716 INVITE Contact: <sip:sgc_c@217.0.27.161;transport=udp> Record-Route: <sip:217.0.27.161;transport=udp;lr> P-Early-Media: sendrecv, gated Require: 100rel RSeq: 2 Content-Type: application/sdp Content-Length: 281 Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, PUBLISH, MESSAGE, UPDATE, PRACK, INFO, INVITE, ACK, OPTIONS, CANCEL, BYE Then I see Asterisk sends this to the phone: Via: SIP/2.0/UDP 192.168.60.53:3072;branch=z9hG4bK-igym6msxn88p;received=192.168.60.53;rport=3072 From: "Sekretariat" <sip:74@192.168.60.1>;tag=ts2ye4krhs To: <sip:01773218409@192.168.60.1;user=phone>;tag=as32fe51ba Call-ID: 313634303731393637343630373636-ex7145moy1mt CSeq: 2 INVITE Server: Asterisk PBX 16.2.1~dfsg-1+deb10u2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces Contact: <sip:01773218409@192.168.60.1:5060> P-Asserted-Identity: "03529529874" <sip:03529529874@192.168.60.1> Content-Length: 0 So, it seems Asterisk receives from Deutsche Telekom _one_ "Ringing" and sends the phone _two_ "Ringing", the second one with the P-Asserted-Identity... Maybe help it to identify the problem? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _____________________________________________________________________ -- 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