Hallo SF - Welt!
Wer hat Sip Connect schon einmal in der Starface eingebunden?
Ich habe es heute versucht. Leider gibt es keinen Wiki Eintrag. Ziel ist es, mit diesem Eintrag und wenn wir die Lösung haben einen zu machen.
Ich werde den dann Starface senden, mit der bitte den ins Wiki aufzunehmen:
Also folgendes habe ich versucht:
A) Konto im Skype Manager erstellt (https://manager.skype.com/)
B) Sip Leitung gekauft
C) User der Sip Leitung zugefügt
D) SF -> Leitung -> Skype Connect SIP zugefügt und Daten eingegeben. (Leitung erscheint Grün!!)
Anrufe gehen weder rein noch raus!
SIP Debug ergibt folgendes:
Code
=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2015.05.23 16:19:55 =~=~=~=~=~=~=~=~=~=~=~=asterisk -rvvvvls -allasterisk -rvvvvAsterisk 11.11.0, Copyright (C) 1999 - 2013 Digium, Inc. and others.Created by Mark Spencer <markster@digium.com>Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.This is free software, with components licensed under the GNU General PublicLicense version 2 and other licenses; you are welcome to redistribute it undercertain conditions. Type 'core show license' for details.=========================================================================Connected to Asterisk 11.11.0 currently running on pbx (pid = 1676)pbx*CLI> sip set debug on ip 63.209.144.201
<------------->--- (8 headers 0 lines) ---
pbx*CLI>
<--- SIP read from UDP:77.59.196.120:5060 ---><------------->
pbx*CLI>
-- Remote UNIX connection
pbx*CLI>
-- Remote UNIX connection disconnected
pbx*CLI>
<--- SIP read from UDP:127.0.0.1:37926 --->OPTIONS sip:127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:37926;branch=z9hG4bK.43e3a6e8;rport;alias
From: sip:sipsak@127.0.0.1:37926;tag=6c3946d8
To: sip:127.0.0.1:5060
Call-ID: 1815693016@127.0.0.1
CSeq: 1 OPTIONS
Contact: sip:sipsak@127.0.0.1:37926
Content-Length: 0
Max-Forwards: 70
User-Agent: sipsak 0.9.6
Accept: text/plain
<------------->--- (11 headers 0 lines) ---
pbx*CLI>
Sending to 127.0.0.1:37926 (NAT)Looking for s in default (domain 127.0.0.1)<--- Transmitting (NAT) to 127.0.0.1:37926 --->SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:37926;branch=z9hG4bK.43e3a6e8;alias;received=127.0.0.1;rport=37926
From: sip:sipsak@127.0.0.1:37926;tag=6c3946d8
To: sip:127.0.0.1:5060;tag=as40e85b31
Call-ID: 1815693016@127.0.0.1
CSeq: 1 OPTIONS
Server: STARFACE PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:127.0.0.1:5060>
Accept: application/sdp
Content-Length: 0
<------------>Scheduling destruction of SIP dialog '1815693016@127.0.0.1' in 32000 ms (Method: OPTIONS)
pbx*CLI>
<--- SIP read from UDP:78.141.179.70:5060 --->INVITE sip:99051000266489@192.168.200.5:5060;transport=udp SIP/2.0
From: "itconsultingbraunwalder" <sip:Anonymous@sip.skype.com:19000>;tag=46b38d4e-13c4-55608ca1-2b75d3b5-629aedc8
To: <sip:99051000266489@sip.skype.com>
Call-ID: CXC-587-66801f68-46b38d4e-13c4-55608ca1-2b75d3b5-7e4a8fba
CSeq: 1 INVITE
Via: SIP/2.0/UDP 78.141.179.70:5060;branch=z9hG4bK-aea57-55608ca1-2b75d3b5-611a5e61
Max-Forwards: 30
User-Agent: SipGW 17.1.0.9
Privacy: id
P-Asserted-Identity: "itconsultingbraunwalder"<sip:Anonymous@10.203.239.208:19000>
Remote-Party-ID: "itconsultingbraunwalder" <sip:Anonymous@sip.skype.com>;party=calling;screen=yes;privacy=full
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE
Contact: <sip:Anonymous@78.141.179.70:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 263
v=0
o=Anonymous 1432390817 1432390817 IN IP4 78.141.179.70
s=Skype call
c=IN IP4 78.141.179.70
t=0 0
m=audio 28696 RTP/AVP 18 8 0 100
a=rtpmap:18 g729/8000
a=rtpmap:100 telephone-event/8000
a=rtpmap:8 pcma/8000
a=rtpmap:0 pcmu/8000
a=fmtp:18 annexb=no
<------------->--- (15 headers 11 lines) ---
pbx*CLI>
Sending to 78.141.179.70:5060 (NAT)
pbx*CLI>
Sending to 78.141.179.70:5060 (NAT)Using INVITE request as basis request - CXC-587-66801f68-46b38d4e-13c4-55608ca1-2b75d3b5-7e4a8fbaFound peer '99051000266489' for 'Anonymous' from 78.141.179.70:5060 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5Found RTP audio format 18Found RTP audio format 8Found RTP audio format 0Found RTP audio format 100Found audio description format g729 for ID 18Found audio description format telephone-event for ID 100Found audio description format pcma for ID 8Found audio description format pcmu for ID 0Capabilities: us - (alaw), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw)Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)Peer audio RTP is at port 78.141.179.70:28696
pbx*CLI>
Looking for 99051000266489 in Skype-incoming (domain 192.168.200.5)
pbx*CLI>
list_route: hop: <sip:Anonymous@78.141.179.70:5060;transport=udp>
pbx*CLI>
<--- Transmitting (NAT) to 78.141.179.70:5060 --->SIP/2.0 100 Trying
Via: SIP/2.0/UDP 78.141.179.70:5060;branch=z9hG4bK-aea57-55608ca1-2b75d3b5-611a5e61;received=78.141.179.70;rport=5060
From: "itconsultingbraunwalder" <sip:Anonymous@sip.skype.com:19000>;tag=46b38d4e-13c4-55608ca1-2b75d3b5-629aedc8
To: <sip:99051000266489@sip.skype.com>
Call-ID: CXC-587-66801f68-46b38d4e-13c4-55608ca1-2b75d3b5-7e4a8fba
CSeq: 1 INVITE
Server: STARFACE PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:99051000266489@192.168.200.5:5060>
Content-Length: 0
<------------>
pbx*CLI>
-- Executing [99051000266489@Skype-incoming:1] Set("SIP/99051000266489-00000abc", "channelname=Skype-incoming") in new stack
pbx*CLI>
-- Executing [99051000266489@Skype-incoming:2] Set("SIP/99051000266489-00000abc", "lineconfigid=2294") in new stack
pbx*CLI>
-- Executing [99051000266489@Skype-incoming:3] Goto("SIP/99051000266489-00000abc", "Skype-incoming-manuell,99051000266489,1") in new stack
pbx*CLI>
-- Goto (Skype-incoming-manuell,99051000266489,1)
pbx*CLI>
-- Executing [99051000266489@Skype-incoming-manuell:1] Goto("SIP/99051000266489-00000abc", "incoming,99051000266489,1") in new stack
pbx*CLI>
-- Goto (incoming,99051000266489,1)
pbx*CLI>
-- Executing [99051000266489@incoming:1] Goto("SIP/99051000266489-00000abc", "calling,99051000266489,1") in new stack
pbx*CLI>
-- Goto (calling,99051000266489,1)
pbx*CLI>
-- Executing [99051000266489@calling:1] GotoIf("SIP/99051000266489-00000abc", "0?4") in new stack
pbx*CLI>
-- Executing [99051000266489@calling:2] AGI("SIP/99051000266489-00000abc", "agi:async,,2294,,,,,,,,,,,") in new stack
pbx*CLI>
-- Manager 'actionasterisk' from 127.0.0.1, hanging up channel: SIP/99051000266489-00000abc
pbx*CLI>
== Spawn extension (calling, 99051000266489, 2) exited non-zero on 'SIP/99051000266489-00000abc'
pbx*CLI>
-- Executing [h@calling:1] NoOp("SIP/99051000266489-00000abc", "HC 21") in new stack
pbx*CLI>
-- Executing [h@calling:2] Goto("SIP/99051000266489-00000abc", "_exit_,0") in new stack
pbx*CLI>
-- Goto (calling,_exit_,0)
pbx*CLI>
Scheduling destruction of SIP dialog 'CXC-587-66801f68-46b38d4e-13c4-55608ca1-2b75d3b5-7e4a8fba' in 32000 ms (Method: INVITE)
pbx*CLI>
<--- Reliably Transmitting (NAT) to 78.141.179.70:5060 --->SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 78.141.179.70:5060;branch=z9hG4bK-aea57-55608ca1-2b75d3b5-611a5e61;received=78.141.179.70;rport=5060
From: "itconsultingbraunwalder" <sip:Anonymous@sip.skype.com:19000>;tag=46b38d4e-13c4-55608ca1-2b75d3b5-629aedc8
To: <sip:99051000266489@sip.skype.com>;tag=as019a1a63
Call-ID: CXC-587-66801f68-46b38d4e-13c4-55608ca1-2b75d3b5-7e4a8fba
CSeq: 1 INVITE
Server: STARFACE PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
pbx*CLI>
<--- SIP read from UDP:78.141.179.70:5060 --->ACK sip:99051000266489@192.168.200.5:5060;transport=udp SIP/2.0
From: "itconsultingbraunwalder" <sip:Anonymous@sip.skype.com:19000>;tag=46b38d4e-13c4-55608ca1-2b75d3b5-629aedc8
To: <sip:99051000266489@sip.skype.com>;tag=as019a1a63
Call-ID: CXC-587-66801f68-46b38d4e-13c4-55608ca1-2b75d3b5-7e4a8fba
CSeq: 1 ACK
Via: SIP/2.0/UDP 78.141.179.70:5060;branch=z9hG4bK-aea57-55608ca1-2b75d3b5-611a5e61
Max-Forwards: 70
Contact: <sip:Anonymous@78.141.179.70:5060;transport=udp>
Content-Length: 0
Alles anzeigen
Ein TCP Dump das folgende:
skype.jpg
Grosse Frage:
Wiso wird das Gespräch abgelehnt?
Kann mir da jemand helfen?
Danke fürs Feedback.
Gruss
CH