[Gelöst] Voxeo -> Asterisk (direkt ohne Provider): Wie richtig im Asterisk?

[MET]

Vielleicht habe ich meine Frage hier am falschen Ort gestellt.

Nochmals, im Prinzip funktionieren die eingehenden Anrufe. Es ist nur unschön, dass sie erst im extension.conf unter [default] abgefangen werden und, dass der port=5060 sein muss. Gibt es eine Möglichkeit solche eingehenden Anrufe (mit bekannter IP) "normal" schon im sip.conf abzufangen und an [incoming] weiterzuleiten? Ich habe es in der sip.conf damit probiert:

[Voxeo_in]
type=friend
host=IP_Voxeo
...
context=incoming

aber dies geht nich; die Anrufe kommen immer noch via [default] rein.

Hier noch ein debug mit dem derzeitigen Verbindungsaufbau.

<------------>
[Oct  9 17:28:35] VERBOSE[5854] logger.c: 
<--- SIP read from IP_Voxeo:45672 --->
INVITE sip:Irgendwas@IPmyAsterisk SIP/2.0
From: <sip:Restricted@IP_Voxeo:45672>;tag=201320a8-0-13c4-6009-12d-2a1467ea-12d
To: <sip:Irgendwas@IPmyAsterisk>
Call-ID: 1444404515093-2aaab4253b20-20a65320-0000000f@IP_Local
CSeq: 1 INVITE
Via: SIP/2.0/UDP IP_Voxeo:45672;rport;branch=z9hG4bK-12d-499f4-c38c9f4-1febf670
x-accountid: AccNo
x-appid: 804b6b431f5d4cb3a0bdefac0fca266d
x-dialogid: f3908cd4fda3871e04354f1419082e95-0
x-joinsid: 31cc5ee944b0fbca37bd932512588eb4
x-psid: 76d9862e5d35998721ea5b773d5f3b39
x-sid: 60575430153fb8387f6ea8d674d1f9da
x-voxeo-romeo: true
x-voxeo-to: <sip:Irgendwas@IPmyAsterisk>
x-voxeo-type: bridge
Max-Forwards: 70
User-Agent: VCS14.0.10.111.82985
Contact: <sip:Restricted@IP_Voxeo:45672>
Content-Type: application/sdp
Content-Length: 290

v=0
o=- 1 1 IN IP4 IP_Voxeo
s=voxeo.14.0.10.111.82985
c=IN IP4 IP_Voxeo
t=0 0
m=audio 10108 RTP/AVP 101 0 8 104 106
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:104 iSAC/16000
a=rtpmap:106 OPUS/48000/2
a=ptime:20

<------------->
[Oct  9 17:28:35] VERBOSE[5854] logger.c: --- (20 headers 13 lines) ---
[Oct  9 17:28:35] VERBOSE[5854] logger.c: Sending to IP_Voxeo : 45672 (no NAT)
[Oct  9 17:28:35] VERBOSE[5854] logger.c: Using INVITE request as basis request - 1444404515093-2aaab4253b20-20a65320-0000000f@IP_Local
[Oct  9 17:28:35] VERBOSE[5854] logger.c: Found no matching peer or user for 'IP_Voxeo:45672'
[Oct  9 17:28:35] VERBOSE[5854] logger.c: Found RTP audio format 101
[Oct  9 17:28:35] VERBOSE[5854] logger.c: Found RTP audio format 0
[Oct  9 17:28:35] VERBOSE[5854] logger.c: Found RTP audio format 8
[Oct  9 17:28:35] VERBOSE[5854] logger.c: Found RTP audio format 104
[Oct  9 17:28:35] VERBOSE[5854] logger.c: Found RTP audio format 106
[Oct  9 17:28:35] VERBOSE[5854] logger.c: Found audio description format telephone-event for ID 101
[Oct  9 17:28:35] VERBOSE[5854] logger.c: Found audio description format PCMU for ID 0
[Oct  9 17:28:35] VERBOSE[5854] logger.c: Found audio description format PCMA for ID 8
[Oct  9 17:28:35] VERBOSE[5854] logger.c: Found unknown media description format iSAC for ID 104
[Oct  9 17:28:35] VERBOSE[5854] logger.c: Found unknown media description format OPUS for ID 106
[Oct  9 17:28:35] VERBOSE[5854] logger.c: Capabilities: us - 0xe0e (gsm|ulaw|alaw|g726|speex|ilbc), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), 

combined - 0xc (ulaw|alaw)
[Oct  9 17:28:35] VERBOSE[5854] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 

(telephone-event)
[Oct  9 17:28:35] DEBUG[5854] chan_sip.c: Our T38 capability = (0), peer T38 capability (0), joint T38 capability (0)
[Oct  9 17:28:35] VERBOSE[5854] logger.c: Peer audio RTP is at port IP_Voxeo:10108
[Oct  9 17:28:35] VERBOSE[5854] logger.c: Looking for Irgendwas in default (domain IPmyAsterisk)
[Oct  9 17:28:35] VERBOSE[5854] logger.c: list_route: hop: <sip:Restricted@IP_Voxeo:45672>
[Oct  9 17:28:35] VERBOSE[5854] logger.c: 
<--- Transmitting (NAT) to IP_Voxeo:45672 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP IP_Voxeo:45672;branch=z9hG4bK-12d-499f4-c38c9f4-1febf670;received=IP_Voxeo;rport=45672
From: <sip:Restricted@IP_Voxeo:45672>;tag=201320a8-0-13c4-6009-12d-2a1467ea-12d
To: <sip:Irgendwas@IPmyAsterisk>
Call-ID: 1444404515093-2aaab4253b20-20a65320-0000000f@IP_Local
CSeq: 1 INVITE
User-Agent: MyDevice
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:Irgendwas@IPmyAsterisk>
Content-Length: 0


<------------>
[Oct  9 17:28:35] VERBOSE[23015] logger.c:     -- Executing [Irgendwas@default:1] Macro("SIP/45672-00000038", "ruf|SIP|30") in new stack
[Oct  9 17:28:35] VERBOSE[23015] logger.c:     -- Executing [s@macro-ruf:1] NoOp("SIP/45672-00000038", "Wir sind im Macro ruf gelandet") in new stack
[Oct  9 17:28:35] DEBUG[23015] app_macro.c: Executed application: NoOp
[Oct  9 17:28:35] VERBOSE[23015] logger.c:     -- Executing [s@macro-ruf:2] Macro("SIP/45672-00000038", "callforwarding|30") in new stack
[Oct  9 17:28:35] DEBUG[23015] func_db.c: DB: CFI/30 not found in database.
[Oct  9 17:28:35] VERBOSE[23015] logger.c:     -- Executing [s@macro-callforwarding:1] Set("SIP/45672-00000038", "temp=") in new stack
[Oct  9 17:28:35] DEBUG[23015] app_macro.c: Executed application: Set
[Oct  9 17:28:35] VERBOSE[23015] logger.c:     -- Executing [s@macro-callforwarding:2] GotoIf("SIP/45672-00000038", "?cfi:nocfi") in new stack
[Oct  9 17:28:35] VERBOSE[23015] logger.c:     -- Goto (macro-callforwarding,s,4)
[Oct  9 17:28:35] DEBUG[23015] app_macro.c: Executed application: GotoIf
[Oct  9 17:28:35] VERBOSE[23015] logger.c:     -- Executing [s@macro-callforwarding:4] NoOp("SIP/45672-00000038", "") in new stack
[Oct  9 17:28:35] DEBUG[23015] app_macro.c: Executed application: NoOp
[Oct  9 17:28:35] DEBUG[23015] app_macro.c: Executed application: Macro
[Oct  9 17:28:35] VERBOSE[23015] logger.c:     -- Executing [s@macro-ruf:3] Dial("SIP/45672-00000038", "SIP/30&IAX2/40|30|r") in new stack
[Oct  9 17:28:35] VERBOSE[23015] logger.c: Audio is at IPmyAsterisk port 18352
[Oct  9 17:28:35] VERBOSE[23015] logger.c: Adding codec 0x4 (ulaw) to SDP
[Oct  9 17:28:35] VERBOSE[23015] logger.c: Adding codec 0x8 (alaw) to SDP
[Oct  9 17:28:35] VERBOSE[23015] logger.c: Adding codec 0x2 (gsm) to SDP
[Oct  9 17:28:35] VERBOSE[23015] logger.c: Adding codec 0x800 (g726) to SDP
[Oct  9 17:28:35] VERBOSE[23015] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[Oct  9 17:28:35] VERBOSE[23015] logger.c: Reliably Transmitting (NAT) to IPmyLinksys:5062:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP IPmyAsterisk:5060;branch=z9hG4bK1a286ce6;rport
From: "Restricted" <sip:Restricted@IPmyAsterisk>;tag=as601a3b47
To: <sip:[email protected]:5060>
Contact: <sip:Restricted@IPmyAsterisk>
Call-ID: 19821cef2286d1743a4c71f74efa8162@IPmyAsterisk
CSeq: 102 INVITE
User-Agent: MyDevice
Max-Forwards: 70
Date: Fri, 09 Oct 2015 15:28:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 314

v=0
o=root 5335 5335 IN IP4 IPmyAsterisk
s=session
c=IN IP4 IPmyAsterisk
t=0 0
m=audio 18352 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Oct  9 17:28:35] VERBOSE[23015] logger.c:     -- Called 30
[Oct  9 17:28:35] DEBUG[23015] chan_iax2.c: prepending 4 to prefs
[Oct  9 17:28:35] VERBOSE[23015] logger.c:     -- Called 40
[Oct  9 17:28:35] VERBOSE[23015] logger.c: 
<--- Transmitting (NAT) to IP_Voxeo:45672 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP IP_Voxeo:45672;branch=z9hG4bK-12d-499f4-c38c9f4-1febf670;received=IP_Voxeo;rport=45672
From: <sip:Restricted@IP_Voxeo:45672>;tag=201320a8-0-13c4-6009-12d-2a1467ea-12d
To: <sip:Irgendwas@IPmyAsterisk>;tag=as20198a65
Call-ID: 1444404515093-2aaab4253b20-20a65320-0000000f@IP_Local
CSeq: 1 INVITE
User-Agent: MyDevice
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:Irgendwas@IPmyAsterisk>
Content-Length: 0


<------------>
[Oct  9 17:28:35] VERBOSE[5854] logger.c: 
<--- SIP read from IPmyLinksys:5062 --->
SIP/2.0 100 Trying
To: <sip:[email protected]:5060>
From: "Restricted" <sip:Restricted@IPmyAsterisk>;tag=as601a3b47
Call-ID: 19821cef2286d1743a4c71f74efa8162@IPmyAsterisk
CSeq: 102 INVITE
Via: SIP/2.0/UDP IPmyAsterisk:5060;branch=z9hG4bK1a286ce6
Server: Linksys/SPA3102-5.1.5(GWa)
Content-Length: 0


<------------->
[Oct  9 17:28:35] VERBOSE[5854] logger.c: --- (8 headers 0 lines) ---
[Oct  9 17:28:35] VERBOSE[5854] logger.c: 
<--- SIP read from IPmyLinksys:5062 --->
SIP/2.0 180 Ringing
To: <sip:[email protected]:5060>;tag=8ac01b9e64deceb4i0
From: "Restricted" <sip:Restricted@IPmyAsterisk>;tag=as601a3b47
Call-ID: 19821cef2286d1743a4c71f74efa8162@IPmyAsterisk
CSeq: 102 INVITE
Via: SIP/2.0/UDP IPmyAsterisk:5060;branch=z9hG4bK1a286ce6
Server: Linksys/SPA3102-5.1.5(GWa)
Remote-Party-ID: "+MyPhoneNo" <sip:30@IPmyAsterisk>;screen=yes;party=called
Content-Length: 0


<------------->
[Oct  9 17:28:35] VERBOSE[5854] logger.c: --- (9 headers 0 lines) ---
[Oct  9 17:28:35] VERBOSE[23015] logger.c:     -- SIP/30-00000039 is ringing
[Oct  9 17:28:35] VERBOSE[5810] logger.c:     -- Call accepted by IPmyLinksys (format ulaw)
[Oct  9 17:28:35] VERBOSE[5810] logger.c:     -- Format for call is ulaw
[Oct  9 17:28:35] VERBOSE[23015] logger.c:     -- IAX2/40-1533 is ringing
[Oct  9 17:28:38] VERBOSE[5854] logger.c: 
<--- SIP read from IPmyLinksys:5062 --->
NOTIFY sip:IPmyAsterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.1.33:5060;branch=z9hG4bK-aa212f8c
From: "+MyPhoneNo" <sip:30@IPmyAsterisk>;tag=720758a2bffbea7bo0
To: <sip:IPmyAsterisk>
Call-ID: [email protected]
CSeq: 1353 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Linksys/SPA3102-5.1.5(GWa)
Content-Length: 0


<------------->
[Oct  9 17:28:38] VERBOSE[5854] logger.c: --- (10 headers 0 lines) ---
[Oct  9 17:28:38] VERBOSE[5854] logger.c: Sending to 192.168.1.33 : 5060 (no NAT)
[Oct  9 17:28:38] VERBOSE[5854] logger.c: 
<--- Transmitting (no NAT) to 192.168.1.33:5060 --->
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP 192.168.1.33:5060;branch=z9hG4bK-aa212f8c;received=IPmyLinksys
From: "+MyPhoneNo" <sip:30@IPmyAsterisk>;tag=720758a2bffbea7bo0
To: <sip:IPmyAsterisk>;tag=as2dc7fa9d
Call-ID: [email protected]
CSeq: 1353 NOTIFY
User-Agent: MyDevice
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>
[Oct  9 17:28:42] VERBOSE[5854] logger.c: 
<--- SIP read from IP_Voxeo:45672 --->
CANCEL sip:Irgendwas@IPmyAsterisk SIP/2.0
From: <sip:Restricted@IP_Voxeo:45672>;tag=201320a8-0-13c4-6009-12d-2a1467ea-12d
To: <sip:Irgendwas@IPmyAsterisk>
Call-ID: 1444404515093-2aaab4253b20-20a65320-0000000f@IP_Local
CSeq: 1 CANCEL
Via: SIP/2.0/UDP IP_Voxeo:45672;rport;branch=z9hG4bK-12d-499f4-c38c9f4-1febf670
Max-Forwards: 70
x-accountid: AccNo
x-appid: 804b6b431f5d4cb3a0bdefac0fca266d
x-dialogid: f3908cd4fda3871e04354f1419082e95-0
x-joinsid: 31cc5ee944b0fbca37bd932512588eb4
x-psid: 76d9862e5d35998721ea5b773d5f3b39
x-sid: 60575430153fb8387f6ea8d674d1f9da
x-voxeo-romeo: true
x-voxeo-to: <sip:Irgendwas@IPmyAsterisk>
x-voxeo-type: bridge
Content-Length: 0

 

[xrated]

Found no matching peer or user for 'IP_Voxeo:45672'

Problem mit Namensauflösung?

 

[rentier-s]

Für so etwas bitte immer type=peer benutzen.
Die ... aus Deinem sip.conf Auszug wären jetzt noch interessant. Wenn kein secret gesetzt ist, und die Anrufe immer von der gleichen IP Adresse kommen, sollte das schon funktionieren.

 

[MET]

Bin endlich dazu gekommen hier etwas mehr zu testen. Habe jetzt herausgefunden, dass es via sip.conf und [incoming] mit folgendem Eintrag in der sip.conf funktionierte, jedoch nur "zufällig" einmal dann wieder nicht:

[Voxeo_in]
type=peer
host=IP_Voxeo
bindport=5060
nat=yes
disallow=all
allow=ulaw
allow=alaw
allow=ilbc
allow=g723
allow=g729
allow=g726
context=incoming

Der folgende Debug zeigt den Verbindungsaufbau von drei Versuchen die hintereinander erfolgten und bei denen nichts verändert wurde. Beim ersten Versuch kommt der Anruf mit 'IP_Voxeo:5060' rein und funktioniert. Habe dann unterbrochen und zwei Mal neu versucht. Bei diesen erneuten Versuchen ist der port dann nicht mehr 5060 und endet deshalb mit einem Fehler.

<------------>
[Oct 24 13:28:13] VERBOSE[5854] logger.c: 
<--- SIP read from IP_Voxeo:5060 --->
INVITE sip:MyVoxeoTelNr@IP_Asterisk SIP/2.0
From: <sip:Restricted@IP_Voxeo>;tag=2271b8c0-0-13c4-6009-eeb1-6b167da2-eeb1
To: <sip:MyVoxeoTelNr@IP_Asterisk>
Call-ID: [email protected]
CSeq: 1 INVITE
Via: SIP/2.0/UDP IP_Voxeo:5060;rport;branch=z9hG4bK-eeb1-3a4648c-1ed4083-224f38b0
x-accountid: MyAccID
x-appid: 804b6b431f5d4cb3a0bdefac0fca266d
x-dialogid: f3af8eed7110eb2d88b9211d087b369d-0
x-joinsid: dec40c138dab5e5011ae6445ddd2b6c8
x-psid: 22a7f3995a5121e172f096a5d112d20f
x-sid: 61f8674d310a74dd5af4574da7a99598
x-voxeo-romeo: true
x-voxeo-to: <sip:MyVoxeoTelNr@IP_Asterisk>
x-voxeo-type: bridge
Max-Forwards: 70
User-Agent: VCS11.7.71818.0
Contact: <sip:Restricted@IP_Voxeo:5060>
Content-Type: application/sdp
Content-Length: 331

v=0
o=- 1 1 IN IP4 IP_Voxeo
s=voxeo.11.7.71818.0
c=IN IP4 IP_Voxeo
t=0 0
m=audio 13126 RTP/AVP 0 8 101 116 9 3 104
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:116 SPEEX/16000
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=rtpmap:104 iSAC/16000
a=ptime:20

<------------->
[Oct 24 13:28:13] VERBOSE[5854] logger.c: --- (20 headers 15 lines) ---
[Oct 24 13:28:13] VERBOSE[5854] logger.c: Sending to IP_Voxeo : 5060 (no NAT)
[Oct 24 13:28:13] VERBOSE[5854] logger.c: Using INVITE request as basis request - [email protected]
[Oct 24 13:28:13] VERBOSE[5854] logger.c: Found peer 'Voxeo_in'
[Oct 24 13:28:13] VERBOSE[5854] logger.c: Found RTP audio format 0
[Oct 24 13:28:13] VERBOSE[5854] logger.c: Found RTP audio format 8
[Oct 24 13:28:13] VERBOSE[5854] logger.c: Found RTP audio format 101
[Oct 24 13:28:13] VERBOSE[5854] logger.c: Found RTP audio format 116
[Oct 24 13:28:13] VERBOSE[5854] logger.c: Found RTP audio format 9
[Oct 24 13:28:13] VERBOSE[5854] logger.c: Found RTP audio format 3
[Oct 24 13:28:13] VERBOSE[5854] logger.c: Found RTP audio format 104
[Oct 24 13:28:13] VERBOSE[5854] logger.c: Found audio description format PCMU for ID 0
[Oct 24 13:28:13] VERBOSE[5854] logger.c: Found audio description format PCMA for ID 8
[Oct 24 13:28:13] VERBOSE[5854] logger.c: Found audio description format telephone-event for ID 101
[Oct 24 13:28:13] VERBOSE[5854] logger.c: Found audio description format SPEEX for ID 116
[Oct 24 13:28:13] VERBOSE[5854] logger.c: Found audio description format G722 for ID 9
[Oct 24 13:28:13] VERBOSE[5854] logger.c: Found audio description format GSM for ID 3
[Oct 24 13:28:13] VERBOSE[5854] logger.c: Found unknown media description format iSAC for ID 104
[Oct 24 13:28:13] VERBOSE[5854] logger.c: Capabilities: us - 0xd0d (g723|ulaw|alaw|g726|g729|ilbc), peer - audio=0x120e (gsm|ulaw|alaw|speex|

g722)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
[Oct 24 13:28:13] VERBOSE[5854] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 

(telephone-event)
[Oct 24 13:28:13] DEBUG[5854] chan_sip.c: Our T38 capability = (0), peer T38 capability (0), joint T38 capability (0)
[Oct 24 13:28:13] VERBOSE[5854] logger.c: Peer audio RTP is at port IP_Voxeo:13126
[Oct 24 13:28:13] VERBOSE[5854] logger.c: Looking for MyVoxeoTelNr in incoming (domain IP_Asterisk)
[Oct 24 13:28:13] VERBOSE[5854] logger.c: list_route: hop: <sip:Restricted@IP_Voxeo:5060>
[Oct 24 13:28:13] VERBOSE[5854] logger.c: 
<--- Transmitting (NAT) to IP_Voxeo:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP IP_Voxeo:5060;branch=z9hG4bK-eeb1-3a4648c-1ed4083-224f38b0;received=IP_Voxeo;rport=5060
From: <sip:Restricted@IP_Voxeo>;tag=2271b8c0-0-13c4-6009-eeb1-6b167da2-eeb1
To: <sip:MyVoxeoTelNr@IP_Asterisk>
Call-ID: [email protected]
CSeq: 1 INVITE
User-Agent: MyDevice
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:MyVoxeoTelNr@IP_Asterisk>
Content-Length: 0


<------------>
[Oct 24 13:28:13] VERBOSE[3618] logger.c:     -- Executing [MyVoxeoTelNr@incoming:1] Macro("SIP/Voxeo_in-0000005d", "ruf|SIP|30") in new stack
[Oct 24 13:28:13] VERBOSE[3618] logger.c:     -- Executing [s@macro-ruf:1] NoOp("SIP/Voxeo_in-0000005d", "Wir sind im Macro ruf gelandet") in new stack
[Oct 24 13:28:13] DEBUG[3618] app_macro.c: Executed application: NoOp
[Oct 24 13:28:13] VERBOSE[3618] logger.c:     -- Executing [s@macro-ruf:2] Macro("SIP/Voxeo_in-0000005d", "callforwarding|30") in new stack
[Oct 24 13:28:13] DEBUG[3618] func_db.c: DB: CFI/30 not found in database.
[Oct 24 13:28:13] VERBOSE[3618] logger.c:     -- Executing [s@macro-callforwarding:1] Set("SIP/Voxeo_in-0000005d", "temp=") in new stack
[Oct 24 13:28:13] DEBUG[3618] app_macro.c: Executed application: Set
[Oct 24 13:28:13] VERBOSE[3618] logger.c:     -- Executing [s@macro-callforwarding:2] GotoIf("SIP/Voxeo_in-0000005d", "?cfi:nocfi") in new stack
[Oct 24 13:28:13] VERBOSE[3618] logger.c:     -- Goto (macro-callforwarding,s,4)
[Oct 24 13:28:13] DEBUG[3618] app_macro.c: Executed application: GotoIf
[Oct 24 13:28:13] VERBOSE[3618] logger.c:     -- Executing [s@macro-callforwarding:4] NoOp("SIP/Voxeo_in-0000005d", "") in new stack
[Oct 24 13:28:13] DEBUG[3618] app_macro.c: Executed application: NoOp
[Oct 24 13:28:13] DEBUG[3618] app_macro.c: Executed application: Macro
[Oct 24 13:28:13] VERBOSE[3618] logger.c:     -- Executing [s@macro-ruf:3] Dial("SIP/Voxeo_in-0000005d", "SIP/30&IAX2/40|30|r") in new stack
[Oct 24 13:28:13] VERBOSE[3618] logger.c: Audio is at IP_Asterisk port 10010
[Oct 24 13:28:13] VERBOSE[3618] logger.c: Adding codec 0x4 (ulaw) to SDP
[Oct 24 13:28:13] VERBOSE[3618] logger.c: Adding codec 0x8 (alaw) to SDP
[Oct 24 13:28:13] VERBOSE[3618] logger.c: Adding codec 0x2 (gsm) to SDP
[Oct 24 13:28:13] VERBOSE[3618] logger.c: Adding codec 0x800 (g726) to SDP
[Oct 24 13:28:13] VERBOSE[3618] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[Oct 24 13:28:13] VERBOSE[3618] logger.c: Reliably Transmitting (NAT) to IP_Linksys:5062:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP IP_Asterisk:5060;branch=z9hG4bK67855cd7;rport
From: "Restricted" <sip:Restricted@IP_Asterisk>;tag=as48d39ae4
To: <sip:[email protected]:5060>
Contact: <sip:Restricted@IP_Asterisk>
Call-ID: 25b7f05e3ebfdcf45db986284eb4d246@IP_Asterisk
CSeq: 102 INVITE
User-Agent: MyDevice
Max-Forwards: 70
Date: Sat, 24 Oct 2015 11:28:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 314

v=0
o=root 5335 5335 IN IP4 IP_Asterisk
s=session
c=IN IP4 IP_Asterisk
t=0 0
m=audio 10010 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Oct 24 13:28:13] VERBOSE[3618] logger.c:     -- Called 30
[Oct 24 13:28:13] DEBUG[3618] chan_iax2.c: prepending 4 to prefs
[Oct 24 13:28:13] VERBOSE[3618] logger.c:     -- Called 40
[Oct 24 13:28:13] VERBOSE[3618] logger.c: 
<--- Transmitting (NAT) to IP_Voxeo:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP IP_Voxeo:5060;branch=z9hG4bK-eeb1-3a4648c-1ed4083-224f38b0;received=IP_Voxeo;rport=5060
From: <sip:Restricted@IP_Voxeo>;tag=2271b8c0-0-13c4-6009-eeb1-6b167da2-eeb1
To: <sip:MyVoxeoTelNr@IP_Asterisk>;tag=as5f47941c
Call-ID: [email protected]
CSeq: 1 INVITE
User-Agent: MyDevice
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:MyVoxeoTelNr@IP_Asterisk>
Content-Length: 0


* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * *

<------------>
[Oct 24 13:28:54] VERBOSE[5854] logger.c: 
<--- SIP read from IP_Voxeo:63132 --->
INVITE sip:MyVoxeoTelNr@IP_Asterisk SIP/2.0
From: <sip:Restricted@IP_Voxeo:63132>;tag=2b58c7d0-0-13c4-6009-169d-32cf7873-169d
To: <sip:MyVoxeoTelNr@IP_Asterisk>
Call-ID: [email protected]
CSeq: 1 INVITE
Via: SIP/2.0/UDP IP_Voxeo:63132;rport;branch=z9hG4bK-169d-5857ab-76c1b980-2b347d60
x-accountid: MyAccID
x-appid: 804b6b431f5d4cb3a0bdefac0fca266d
x-dialogid: f5caae185e6aebafa1a60169fbb139eb-0
x-joinsid: a7e64a50843428b2611354e4c66f2998
x-psid: 92e95438d87aa58a4095b56c8a6e2cb8
x-sid: 15ee3236c0d923d0df315538c7270c60
x-voxeo-romeo: true
x-voxeo-to: <sip:MyVoxeoTelNr@IP_Asterisk>
x-voxeo-type: bridge
Max-Forwards: 70
User-Agent: VCS14.0.10.111.82985
Contact: <sip:Restricted@IP_Voxeo:63132>
Content-Type: application/sdp
Content-Length: 290

v=0
o=- 1 1 IN IP4 IP_Voxeo
s=voxeo.14.0.10.111.82985
c=IN IP4 IP_Voxeo
t=0 0
m=audio 11620 RTP/AVP 101 0 8 104 106
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:104 iSAC/16000
a=rtpmap:106 OPUS/48000/2
a=ptime:20

<------------->
[Oct 24 13:28:54] VERBOSE[5854] logger.c: --- (20 headers 13 lines) ---
[Oct 24 13:28:54] VERBOSE[5854] logger.c: Sending to IP_Voxeo : 63132 (no NAT)
[Oct 24 13:28:54] VERBOSE[5854] logger.c: Using INVITE request as basis request - [email protected]
[Oct 24 13:28:54] VERBOSE[5854] logger.c: Found no matching peer or user for 'IP_Voxeo:63132'
[Oct 24 13:28:54] VERBOSE[5854] logger.c: Found RTP audio format 101
[Oct 24 13:28:54] VERBOSE[5854] logger.c: Found RTP audio format 0
[Oct 24 13:28:54] VERBOSE[5854] logger.c: Found RTP audio format 8
[Oct 24 13:28:54] VERBOSE[5854] logger.c: Found RTP audio format 104
[Oct 24 13:28:54] VERBOSE[5854] logger.c: Found RTP audio format 106
[Oct 24 13:28:54] VERBOSE[5854] logger.c: Found audio description format telephone-event for ID 101
[Oct 24 13:28:54] VERBOSE[5854] logger.c: Found audio description format PCMU for ID 0
[Oct 24 13:28:54] VERBOSE[5854] logger.c: Found audio description format PCMA for ID 8
[Oct 24 13:28:54] VERBOSE[5854] logger.c: Found unknown media description format iSAC for ID 104
[Oct 24 13:28:54] VERBOSE[5854] logger.c: Found unknown media description format OPUS for ID 106
[Oct 24 13:28:54] VERBOSE[5854] logger.c: Capabilities: us - 0xe0e (gsm|ulaw|alaw|g726|speex|ilbc), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), 

combined - 0xc (ulaw|alaw)
[Oct 24 13:28:54] VERBOSE[5854] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 

(telephone-event)
[Oct 24 13:28:54] DEBUG[5854] chan_sip.c: Our T38 capability = (0), peer T38 capability (0), joint T38 capability (0)
[Oct 24 13:28:54] VERBOSE[5854] logger.c: Peer audio RTP is at port IP_Voxeo:11620
[Oct 24 13:28:54] VERBOSE[5854] logger.c: Looking for MyVoxeoTelNr in default (domain IP_Asterisk)
[Oct 24 13:28:54] VERBOSE[5854] logger.c: 
<--- Reliably Transmitting (NAT) to IP_Voxeo:63132 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP IP_Voxeo:63132;branch=z9hG4bK-169d-5857ab-76c1b980-2b347d60;received=IP_Voxeo;rport=63132
From: <sip:Restricted@IP_Voxeo:63132>;tag=2b58c7d0-0-13c4-6009-169d-32cf7873-169d
To: <sip:MyVoxeoTelNr@IP_Asterisk>;tag=as5c90db0d
Call-ID: [email protected]
CSeq: 1 INVITE
User-Agent: MyDevice
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>
[Oct 24 13:28:54] NOTICE[5854] chan_sip.c: Call from '' to extension 'MyVoxeoTelNr' rejected because extension not found.
[Oct 24 13:28:54] VERBOSE[5854] logger.c: Scheduling destruction of SIP dialog '[email protected]' in 32000 ms 

(Method: INVITE)
[Oct 24 13:28:54] VERBOSE[5854] logger.c: 
<--- SIP read from IP_Voxeo:63132 --->
ACK sip:MyVoxeoTelNr@IP_Asterisk SIP/2.0
From: <sip:Restricted@IP_Voxeo:63132>;tag=2b58c7d0-0-13c4-6009-169d-32cf7873-169d
To: <sip:MyVoxeoTelNr@IP_Asterisk>;tag=as5c90db0d
Call-ID: [email protected]
CSeq: 1 ACK
Via: SIP/2.0/UDP IP_Voxeo:63132;rport;branch=z9hG4bK-169d-5857ab-76c1b980-2b347d60
Max-Forwards: 70
Contact: <sip:Restricted@IP_Voxeo:63132>
Content-Length: 0


<------------->
[Oct 24 13:28:54] VERBOSE[5854] logger.c: --- (9 headers 0 lines) ---
[Oct 24 13:28:54] VERBOSE[5854] logger.c: Really destroying SIP dialog '[email protected]' Method: ACK
[Oct 24 13:28:56] VERBOSE[5854] logger.c: 
<--- SIP read from IP_Linksys:5062 --->
NOTIFY sip:IP_Asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.1.33:5060;branch=z9hG4bK-eee8d247
From: "+MyTelNr" <sip:30@IP_Asterisk>;tag=3fb0a9ad0ef20e6o0
To: <sip:IP_Asterisk>
Call-ID: [email protected]
CSeq: 529 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Linksys/SPA3102-5.1.5(GWa)
Content-Length: 0



* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * *


<------------>
[Oct 24 13:29:21] VERBOSE[5854] logger.c: 
<--- SIP read from IP_Voxeo:63132 --->
INVITE sip:MyVoxeoTelNr@IP_Asterisk SIP/2.0
From: <sip:Restricted@IP_Voxeo:63132>;tag=2b58ee68-0-13c4-6009-16b8-26c68cf5-16b8
To: <sip:MyVoxeoTelNr@IP_Asterisk>
Call-ID: [email protected]
CSeq: 1 INVITE
Via: SIP/2.0/UDP IP_Voxeo:63132;rport;branch=z9hG4bK-16b8-58bf7b-7dc110da-2b34aca0
x-accountid: MyAccID
x-appid: 804b6b431f5d4cb3a0bdefac0fca266d
x-dialogid: dcdeda000e05dcd5d8dd420b49a1ab06-0
x-joinsid: fce249c3b0a073ec0e830c29c53c380c
x-psid: 9747c82ee3e748eb7ea6c41bf85c2da3
x-sid: 44867c94a9cb5b37e74b637daf98b10e
x-voxeo-romeo: true
x-voxeo-to: <sip:MyVoxeoTelNr@IP_Asterisk>
x-voxeo-type: bridge
Max-Forwards: 70
User-Agent: VCS14.0.10.111.82985
Contact: <sip:Restricted@IP_Voxeo:63132>
Content-Type: application/sdp
Content-Length: 290

v=0
o=- 1 1 IN IP4 IP_Voxeo
s=voxeo.14.0.10.111.82985
c=IN IP4 IP_Voxeo
t=0 0
m=audio 11640 RTP/AVP 101 0 8 104 106
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:104 iSAC/16000
a=rtpmap:106 OPUS/48000/2
a=ptime:20

<------------->
[Oct 24 13:29:21] VERBOSE[5854] logger.c: --- (20 headers 13 lines) ---
[Oct 24 13:29:21] VERBOSE[5854] logger.c: Sending to IP_Voxeo : 63132 (no NAT)
[Oct 24 13:29:21] VERBOSE[5854] logger.c: Using INVITE request as basis request - [email protected]
[Oct 24 13:29:21] VERBOSE[5854] logger.c: Found no matching peer or user for 'IP_Voxeo:63132'
[Oct 24 13:29:21] VERBOSE[5854] logger.c: Found RTP audio format 101
[Oct 24 13:29:21] VERBOSE[5854] logger.c: Found RTP audio format 0
[Oct 24 13:29:21] VERBOSE[5854] logger.c: Found RTP audio format 8
[Oct 24 13:29:21] VERBOSE[5854] logger.c: Found RTP audio format 104
[Oct 24 13:29:21] VERBOSE[5854] logger.c: Found RTP audio format 106
[Oct 24 13:29:21] VERBOSE[5854] logger.c: Found audio description format telephone-event for ID 101
[Oct 24 13:29:21] VERBOSE[5854] logger.c: Found audio description format PCMU for ID 0
[Oct 24 13:29:21] VERBOSE[5854] logger.c: Found audio description format PCMA for ID 8
[Oct 24 13:29:21] VERBOSE[5854] logger.c: Found unknown media description format iSAC for ID 104
[Oct 24 13:29:21] VERBOSE[5854] logger.c: Found unknown media description format OPUS for ID 106
[Oct 24 13:29:21] VERBOSE[5854] logger.c: Capabilities: us - 0xe0e (gsm|ulaw|alaw|g726|speex|ilbc), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), 

combined - 0xc (ulaw|alaw)
[Oct 24 13:29:21] VERBOSE[5854] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 

(telephone-event)
[Oct 24 13:29:21] DEBUG[5854] chan_sip.c: Our T38 capability = (0), peer T38 capability (0), joint T38 capability (0)
[Oct 24 13:29:21] VERBOSE[5854] logger.c: Peer audio RTP is at port IP_Voxeo:11640
[Oct 24 13:29:21] VERBOSE[5854] logger.c: Looking for MyVoxeoTelNr in default (domain IP_Asterisk)
[Oct 24 13:29:21] VERBOSE[5854] logger.c: 
<--- Reliably Transmitting (NAT) to IP_Voxeo:63132 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP IP_Voxeo:63132;branch=z9hG4bK-16b8-58bf7b-7dc110da-2b34aca0;received=IP_Voxeo;rport=63132
From: <sip:Restricted@IP_Voxeo:63132>;tag=2b58ee68-0-13c4-6009-16b8-26c68cf5-16b8
To: <sip:MyVoxeoTelNr@IP_Asterisk>;tag=as030f9a6d
Call-ID: [email protected]
CSeq: 1 INVITE
User-Agent: MyDevice
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>
[Oct 24 13:29:21] NOTICE[5854] chan_sip.c: Call from '' to extension 'MyVoxeoTelNr' rejected because extension not found.
[Oct 24 13:29:21] VERBOSE[5854] logger.c: Scheduling destruction of SIP dialog '[email protected]' in 32000 ms 

(Method: INVITE)
[Oct 24 13:29:21] VERBOSE[5854] logger.c: 
<--- SIP read from IP_Voxeo:63132 --->
ACK sip:MyVoxeoTelNr@IP_Asterisk SIP/2.0
From: <sip:Restricted@IP_Voxeo:63132>;tag=2b58ee68-0-13c4-6009-16b8-26c68cf5-16b8
To: <sip:MyVoxeoTelNr@IP_Asterisk>;tag=as030f9a6d
Call-ID: [email protected]
CSeq: 1 ACK
Via: SIP/2.0/UDP IP_Voxeo:63132;rport;branch=z9hG4bK-16b8-58bf7b-7dc110da-2b34aca0
Max-Forwards: 70
Contact: <sip:Restricted@IP_Voxeo:63132>
Content-Length: 0


<------------->
[Oct 24 13:29:21] VERBOSE[5854] logger.c: --- (9 headers 0 lines) ---
[Oct 24 13:29:21] VERBOSE[5854] logger.c: Really destroying SIP dialog '[email protected]' Method: ACK
[Oct 24 13:29:31] VERBOSE[5854] logger.c: 
<--- SIP read from IP_Linksys:5062 --->
NOTIFY sip:IP_Asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.1.33:5060;branch=z9hG4bK-9532b2f7
From: "+MyTelNr" <sip:30@IP_Asterisk>;tag=3fb0a9ad0ef20e6o0
To: <sip:IP_Asterisk>
Call-ID: [email protected]
CSeq: 531 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Linksys/SPA3102-5.1.5(GWa)
Content-Length: 0

Aus irgend einem Grund ist der port nicht immer 5060. Liegt das Problem jetzt nur bei Voxeo oder doch irgendwie auch am Asterisk?

 

[xrated]

bindport gehört in general, ist aber eigentlich eh default 5060

Dieses voxeo (was auch immer das ist) schickt dir den invite von Port 63132
From: <sip:Restricted@IP_Voxeo:63132>;tag=2b58ee68-0-13c4-6009-16b8-26c68cf5-16b8
To: <sip:MyVoxeoTelNr@IP_Asterisk>
Call-ID: [email protected]
CSeq: 1 INVITE
Via: SIP/2.0/UDP IP_Voxeo:63132;rport;branch=z9hG4bK-16b8-58bf7b-7dc110da-2b34aca0

 

[MET]

bindport gehört in general, ist aber eigentlich eh default 5060

Habe ich auch in [general]. Da trotzdem manchmal ein anderer port verwendet wird, habe ich es dann zusätzlich noch damit probiert. Frage mich nachträglich, ob der port 5060 war, weil ich beim Testen vorher möglicherweise einen Versuch via [defualt] in extension.conf machte und er sich dies vieleicht auch noch "gemerkt" hatte ???

Dieses voxeo (was auch immer das ist)

Siehe hier bzw. hier, für USA (oder Skype) sehr praktisch.

 

[MET]

So, habe jetzt eine Lösung gefunden. Mit insecure=port gelangen diese Anrufe jetzt regelmässig von sip.conf an [incoming] in der extension.conf.

[Voxeo_in]
type=peer
insecure=port
host=IP_Voxeo
nat=yes
disallow=all
allow=ulaw
allow=alaw
allow=ilbc
allow=g723
allow=g729
allow=g726
context=incoming

Unter [general] muss bindport=5060 sein, mit einem anderen port funktioniert es nicht, auch nicht wenn unter [Voxeo_in] dann noch bindport=5060 gesetzt wird. Das nat=yes ist ein Überbleibsel der Tests; bin mir nicht sicher, ob es tatsächlich notwendig ist. Endlich kann ich jetzt Voxeo auch benutzen wenn allowguest=no gesetzt ist.

Frage: Bezieht sich insecure=port nur auf IP_Voxeo aus oder auch auf andere IPs in der sip.conf?

 

[xrated]

Ja insecure ist generell notwendig bei Peers von SIP Anbietern, sonst erfolgt authentification.
Normalerweise setzt man insecure=invite,port