Problem mit Telefonat

[anonymous]

Seid einem Update auf die neuste Version ist von aussen kein Anruf mehr möglich:



15:48:06.130 Active calls counted toward license limit: []
15:47:36.128 Active calls counted toward license limit: []
15:47:04.126 Active calls counted toward license limit: []
15:46:32.124 Active calls counted toward license limit: []
15:46:00.123 Active calls counted toward license limit: []
15:45:28.121 Active calls counted toward license limit: []
15:44:56.119 Active calls counted toward license limit: []
15:44:26.117 Active calls counted toward license limit: []
15:43:54.115 Active calls counted toward license limit: []
15:43:38.621 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
INVITE sip:[email protected]:61156;rinstance=29469376c2be94df SIP/2.0
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bKf1c4.02a1d5b6.0
Via: SIP/2.0/UDP 172.20.40.1;branch=z9hG4bKf1c4.02a1d5b6.0
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.226;branch=z9hG4bK41d3bfaa
Via: SIP/2.0/UDP 217.10.67.11:5060;branch=z9hG4bK41d3bfaa;rport=5060
Max-Forwards: 67
Record-Route:
Record-Route:
Record-Route:
Contact:
To:
From: "059719486801";tag=as5b76fde1
Call-ID: [email protected]
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

 

[stefanw]

Schau mal hier: (Auszug aus: https://www.3cx.com/blog/voip-howto/nat-stun-network-configuration/)


“Request Line URI: User Part” can be matched against the “LineNumber external number of line”. This kind of strategy would suit cases where a VoIP provider inserts the DID number or external number of a line in the Request Line instead of the account’s Authentication
IDINVITE sip: 0015552368@ 72.14.207.99:5060;rinstance=800f2f60c40d1d10 SIP/2.0

Das sollte es schon ein für dich, alternativ geht auch:


“TO: User Part” can be matched against the “LineNumber external number of line”. This strategy would look for a line’s external number e.g. “0015552368” in the user part of the “To” header of the SIP URI in an INVITE from the VoIP provider in question.
To:

“From: Host Part” can be matched against the “GWHostPort gateway/provider host/port”. This strategy would look for the VoIP provider’s Proxy address and port (myexamplprovider.nl) in the host part of the SIP URI in an INVITE from the VoIP provider in question.
From: “00315552368″ ;tag=as33577b1c
Via and record route

 

[stefanw]

ps es ist da bestimmt was nicht mit deinem SIP Settings an sich oder dem Account richtig, denn wenn SipGate nicht gehen würde müsste das Forum davon voll sein... (erinnere mich da nur an das Record Route Problem)