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:[email protected]>;tag=ts2ye4krhs
To: <sip:[email protected];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:[email protected]: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:[email protected]>;tag=h7g4Esbg_p65544t1640719676m169304c9321s1_3514393582-932943693
From:
<sip:[email protected]>;tag=4781eb96-b155-421e-8206-593d44c9f7c4
Call-ID: 478ba582-946c-46ac-984d-6f1835e3391b
CSeq: 15716 INVITE
Contact: <sip:[email protected];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:[email protected]>;tag=ts2ye4krhs
To: <sip:[email protected];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:[email protected]:5060>
P-Asserted-Identity: "03529529874" <sip:[email protected]>
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
([email protected])

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