Ciao ragazzi,
ich bin ein Stück weitergekommen. Ich habe einen SIP-Trunk-User "29286#" mit authname "29286" eingerichtet
Code: Alles auswählen
[SIP-Connection] 2018/07/22 22:15:51,287 [SIP UDP Transport (192.168.106.16:5060)]: Processing new inbound SIP message
[SIP-Packet] 2018/07/22 22:15:51,287 [Packet]:
Receiving datagram (410 Bytes) at 192.168.106.5:5060 from 192.168.106.16:5060 using UDP (RtgTag 0):
REGISTER sip:192.168.106.5 SIP/2.0\r\n
Via: SIP/2.0/UDP 192.168.106.16:5060;branch=z9hG4bK7345e8bb\r\n
Max-Forwards: 70\r\n
From: <sip:29286@192.168.106.5>;tag=as54bc5fdb\r\n
To: <sip:29286@192.168.106.5>\r\n
Call-ID: 141483486008753f3ac7778f438adf99@192.168.101.1\r\n
CSeq: 137 REGISTER\r\n
Supported: replaces, timer\r\n
User-Agent: Asterisk PBX 13.20.0\r\n
Expires: 120\r\n
Contact: <sip:29286@192.168.106.16:5060>\r\n
Content-Length: 0\r\n
\r\n
[SIP-Connection] 2018/07/22 22:15:51,287 [SIP UDP Transport (192.168.106.16:5060)]: VCM registrar proceeds with REGISTER request
[Callmanager] 2018/07/22 22:15:51,288 [SIP Registration]: 29286@lancom.intern: cSipRegistration::UpdateRegistration -- iter-Domain: 192.168.106.16, Contact-Name: 192.168.106.16
[Callmanager] 2018/07/22 22:15:51,288 [SIP Binding]: <sip:29286@192.168.106.16:5060>: New expiration time: 120s (was 47s)
[SIP-Connection] 2018/07/22 22:15:51,288 [SIP UDP Transport (192.168.106.16:5060)]: Discarding (local socket was 192.168.106.5:5060, Tag 0)
[Callmanager] 2018/07/22 22:15:51,288 [SIP Binding]: <sip:29286@192.168.106.16:5060>: Transport is valid, remote address: 192.168.106.16:5060
[SIP-Packet] 2018/07/22 22:15:51,288 [Packet]:
Sending datagram (428 Bytes) from 192.168.106.5:5060 to 192.168.106.16:5060 using UDP (RtgTag 0):
SIP/2.0 200 OK\r\n
Via: SIP/2.0/UDP 192.168.106.16:5060;branch=z9hG4bK7345e8bb;received=192.168.106.16\r\n
From: <sip:29286@192.168.106.5>;tag=as54bc5fdb\r\n
To: <sip:29286@192.168.106.5>;tag=1457660387-1777993383\r\n
Call-ID: 141483486008753f3ac7778f438adf99@192.168.101.1\r\n
CSeq: 137 REGISTER\r\n
User-Agent: LANCOM 1781EF+ / 10.12.0382 / 12.06.2018\r\n
Server: Lancom\r\n
Contact: <sip:29286@192.168.106.16:5060>;expires=120\r\n
Content-Length: 0\r\n
\r\n
und das ganze registriert sich auch am Lancom:
Code: Alles auswählen
root@1781efplus:/Status/Voice-Call-Manager/Users/7
> dir
Index INFO: 7
Number/Name INFO: 29286#
Type INFO: SIP-User
Ifc INFO: none
Address INFO: <sip:29286@192.168.106.16:5060>
Display-Name INFO:
Domain INFO: lancom.intern
Local-register INFO: Registered
Remote-register INFO: Not-possible
CFU-Active INFO: No
CFU-Target INFO:
CFNR-Active INFO: No
CFNR-Target INFO:
CFNR-Timeout INFO: 0
CFB-Active INFO: No
CFB-Target INFO:
Access-from-WAN INFO: No
Expires INFO: 110
Unter Lines im Sip-Mapping ist jetzt als "Internal number" 29286# eingetragen. Damit funktionieren jetzt zur Abwechslung die eingehenden Gespräche ...
Jetzt machen nur noch die ausgehenden Gespräche Probleme (das war der Teil, der zuvor per SIP-PBX-Eintrag gelöst war - dieser musste deaktiviert werden, sonst konnte sich Asterisk nicht anmelden):
Code: Alles auswählen
[SIP-Connection] 2018/07/22 22:18:22,861 [SIP UDP Transport (192.168.106.16:5060)]: Processing new inbound SIP message
[SIP-Packet] 2018/07/22 22:18:22,861 [Packet]:
Receiving datagram (879 Bytes) at 192.168.106.5:5060 from 192.168.106.16:5060 using UDP (RtgTag 0):
INVITE sip:08003301000@192.168.106.5 SIP/2.0\r\n
Via: SIP/2.0/UDP 192.168.106.16:5060;branch=z9hG4bK01c089a2\r\n
Max-Forwards: 70\r\n
From: "PG" <sip:2928660@192.168.106.16>;tag=as243ed0e8\r\n
To: <sip:08003301000@192.168.106.5>\r\n
Contact: <sip:2928660@192.168.106.16:5060>\r\n
Call-ID: 0908d2a063760d8c37b4823605e59545@192.168.106.16:5060\r\n
CSeq: 102 INVITE\r\n
User-Agent: Asterisk PBX 13.20.0\r\n
Date: Sun, 22 Jul 2018 20:18:23 GMT\r\n
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE\r\n
Supported: replaces, timer\r\n
Content-Type: application/sdp\r\n
Content-Length: 280\r\n
\r\n
v=0\r\n
o=root 1866538796 1866538796 IN IP4 192.168.106.16\r\n
s=Asterisk PBX 13.20.0\r\n
c=IN IP4 192.168.106.16\r\n
t=0 0\r\n
m=audio 10504 RTP/AVP 8 9 101\r\n
a=rtpmap:8 PCMA/8000\r\n
a=rtpmap:9 G722/8000\r\n
a=rtpmap:101 telephone-event/8000\r\n
a=fmtp:101 0-16\r\n
a=ptime:20\r\n
a=maxptime:150\r\n
a=sendrecv\r\n
[SIP-Connection] 2018/07/22 22:18:22,861 [SIP UDP Transport (192.168.106.16:5060)]: VCM user agent server proceeds with non-REGISTER request
[Callmanager] 2018/07/22 22:18:22,861 [Sip-UA] : -----[ INVITE INDICATION, call-id=0908d2a063760d8c37b4823605e59545@192.168.106.16:5060, Configured CLIR=no
[Callmanager] 2018/07/22 22:18:22,861 [Sip-UA] : FindBinding -- OrigName: "2928660", OrigDomain: "192.168.106.16", ContactName: "2928660"
[Callmanager] 2018/07/22 22:18:22,861 [Registration Registry]: Registration Registry: FindRegistration -- Name: "2928660", Domain: "192.168.106.16"
[SIP-Packet] 2018/07/22 22:18:22,862 [Packet]:
Sending datagram (476 Bytes) from 192.168.106.5:5060 to 192.168.106.16:5060 using UDP (RtgTag 0):
SIP/2.0 404 Not Found\r\n
Via: SIP/2.0/UDP 192.168.106.16:5060;branch=z9hG4bK01c089a2;received=192.168.106.16\r\n
From: "PG"<sip:2928660@192.168.106.16>;tag=as243ed0e8\r\n
To: <sip:08003301000@192.168.106.5>\r\n
Call-ID: 0908d2a063760d8c37b4823605e59545@192.168.106.16:5060\r\n
CSeq: 102 INVITE\r\n
User-Agent: LANCOM 1781EF+ / 10.12.0382 / 12.06.2018\r\n
Server: Lancom\r\n
Allow: REGISTER, INVITE, ACK, CANCEL, BYE, REFER, NOTIFY, OPTIONS, PRACK, UPDATE, SUBSCRIBE\r\n
Content-Length: 0\r\n
\r\n
[SIP-Connection] 2018/07/22 22:18:22,862 [SIP UDP Transport (192.168.106.16:5060)]: Discarding (local socket was 192.168.106.5:5060, Tag 0)
[SIP-Connection] 2018/07/22 22:18:22,863 [SIP UDP Transport (192.168.106.16:5060)]: Processing new inbound SIP message
[SIP-Packet] 2018/07/22 22:18:22,863 [Packet]:
Receiving datagram (403 Bytes) at 192.168.106.5:5060 from 192.168.106.16:5060 using UDP (RtgTag 0):
ACK sip:08003301000@192.168.106.5 SIP/2.0\r\n
Via: SIP/2.0/UDP 192.168.106.16:5060;branch=z9hG4bK01c089a2\r\n
Max-Forwards: 70\r\n
From: "PG" <sip:2928660@192.168.106.16>;tag=as243ed0e8\r\n
To: <sip:08003301000@192.168.106.5>\r\n
Contact: <sip:2928660@192.168.106.16:5060>\r\n
Call-ID: 0908d2a063760d8c37b4823605e59545@192.168.106.16:5060\r\n
CSeq: 102 ACK\r\n
User-Agent: Asterisk PBX 13.20.0\r\n
Content-Length: 0\r\n
\r\n
[SIP-Connection] 2018/07/22 22:18:22,864 [SIP UDP Transport (192.168.106.16:5060)]: VCM user agent server proceeds with non-REGISTER request
[SIP-Connection] 2018/07/22 22:18:22,864 [SIP UDP Transport (192.168.106.16:5060)]: Discarding (local socket was 192.168.106.5:5060, Tag 0)
Ich nehme an, dass meine Default Route im Callmanager noch angepasst werden muss.
Kann mir jemand einen Tip geben?
Tanti saluti
Florian