[Gelöst] Eingehende Anrufe gehen nicht und Verbindung bricht nach genau 30min ab

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tio_morlix

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Hallo zusammen,

jetzt habe ich schon eine Weile versucht die letzten Probleme die ich mit Asterisk habe selbst zu lösen, aber jetzt komme ich irgendwie nicht mehr weiter.

Ich habe (soweit ich weiß) noch 2 Probleme.

* Eingehende Anrufe

Das erste Problem ist, das seit einiger Zeit eingehende Anrufe gar nicht mehr funktionieren. Am Anfang waren es nur andere externe SIP User die mich nicht mehr anrufen konnten. Ich konnte Sie aber ohne Probleme anrufen.
Seit ich letztens mein Asterisk über die Paketverwaltung von Gentoo aktualisiert habe, scheinen jetzt keine eingehenden Anrufe mehr zu funktionieren. Bis dato hatte ich nichts an der Konfiguration verändert. Erst heute als ich dem Problem auf die Schliche kommen wollte habe ich wieder einiges an der Konfiguration von Asterisk verändert (ich poste die aktuelle Config und nicht die alte).

Update: Hier sollte ich vielleicht dazu sagen, das es am Telefon klingelt und sobald man abnimmt ist das Gespräch nach einer sekunde beendet.

Hier mal das sip debug log wenn ein externer anruf kommt. (Leider kann ich nicht 100%ig sagen was davon wichtig ist, daher alles...)

Code: 
<--- SIP read from UDP:217.0.18.16:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Max-Forwards: 51
Via: SIP/2.0/UDP 217.0.18.16:5060;branch=z9hG4bKg3Zqkv7irljpq3ed5b5slo3vufygjk994
To: <sip:[email protected]:5060;user=phone>
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_2c2243ae
Call-ID: [email protected]
CSeq: 610230 INVITE
Contact: <sip:[email protected];transport=udp>
Record-Route: <sip:217.0.18.16;transport=udp;lr>
Min-Se: 900
P-Asserted-Identity: <sip:[email protected];user=phone;rn_source=1>
Session-Expires: 3600
Supported: histinfo
Supported: early-session
Supported: 199
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 312
Session-ID: 5154423F-00F35C6C@ydxc1pcu-2
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REGISTER, UPDATE

v=0
o=hiQ9200 3988620130228141439 1781137431 IN IP4 10.0.99.135
s=Phone Call via hiQ9200 SIPCA
c=IN IP4 217.0.2.128
t=0 0
m=audio 20404 RTP/AVP 8 100
b=AS:87
b=RS:1375
b=RR:4125
a=rtpmap:8 PCMA/8000
a=fmtp:8 vad=no
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=sqn: 0
a=sendrecv
a=ptime:20
<------------->
--- (20 headers 16 lines) ---
Sending to 217.0.18.16:5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer 'telekom-in' for '+491736719999' from 217.0.18.16:5060
Found RTP audio format 8
Found RTP audio format 100
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 100
Capabilities: us - (gsm|ulaw|alaw|ilbc), peer - audio=(alaw)/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 217.0.2.128:20404
Looking for 071419569111 in from-telekom (domain 87.147.170.176)
list_route: hop: <sip:217.0.18.16;transport=udp;lr>

<--- Transmitting (no NAT) to 217.0.18.16:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.0.18.16:5060;branch=z9hG4bKg3Zqkv7irljpq3ed5b5slo3vufygjk994;received=217.0.18.16
Record-Route: <sip:217.0.18.16;transport=udp;lr>
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_2c2243ae
To: <sip:[email protected]:5060;user=phone>
Call-ID: [email protected]
CSeq: 610230 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>  
Audio is at 10060
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100010 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.99.60:2666:
INVITE sip:[email protected]:2666;transport=udp;line=3rbn5n SIP/2.0
Via: SIP/2.0/UDP 192.168.99.10:5060;branch=z9hG4bK31b8a945
Max-Forwards: 70
From: <sip:[email protected]>;tag=as578fad10
To: <sip:[email protected]:2666;transport=udp;line=3rbn5n>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.2.1
Date: Thu, 28 Mar 2013 13:14:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 356

v=0
o=root 2014298794 2014298794 IN IP4 192.168.99.10
s=Asterisk PBX 11.2.1
c=IN IP4 192.168.99.10
t=0 0
m=audio 10060 RTP/AVP 8 0 3 97 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16 
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.99.60:2666 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.99.10:5060;branch=z9hG4bK31b8a945
From: <sip:[email protected]>;tag=as578fad10
To: <sip:[email protected]:2666;transport=udp;line=3rbn5n>;tag=hnhu8e
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Contact: <sip:[email protected]:2666;transport=udp;line=3rbn5n>
Supported: 100rel, replaces, norefersub
Content-Length: 0

<-------------> 
--- (9 headers 0 lines) ---

<--- SIP read from UDP:192.168.99.60:2666 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.99.10:5060;branch=z9hG4bK31b8a945
From: <sip:[email protected]>;tag=as578fad10
To: <sip:[email protected]:2666;transport=udp;line=3rbn5n>;tag=hnhu8e
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Contact: <sip:[email protected]:2666;transport=udp;line=3rbn5n>
Supported: 100rel, replaces, norefersub
Content-Length: 0

<-------------> 
--- (9 headers 0 lines) ---
list_route: hop: <sip:[email protected]:2666;transport=udp;line=3rbn5n>

<--- Transmitting (no NAT) to 217.0.18.16:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 217.0.18.16:5060;branch=z9hG4bKg3Zqkv7irljpq3ed5b5slo3vufygjk994;received=217.0.18.16
Record-Route: <sip:217.0.18.16;transport=udp;lr>
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_2c2243ae
To: <sip:[email protected]:5060;user=phone>;tag=as13e9c253
Call-ID: [email protected]
CSeq: 610230 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>  

<--- SIP read from UDP:192.168.99.60:2666 --->
SIP/2.0 200 Ok  
Via: SIP/2.0/UDP 192.168.99.10:5060;branch=z9hG4bK31b8a945
From: <sip:[email protected]>;tag=as578fad10
To: <sip:[email protected]:2666;transport=udp;line=3rbn5n>;tag=hnhu8e
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Contact: <sip:[email protected]:2666;transport=udp;line=3rbn5n>
Supported: 100rel, replaces, norefersub
User-Agent: snom-m9/9.6.2-a
Content-Type: application/sdp
Content-Length: 205

v=0
o=root 572104435 572104436 IN IP4 192.168.99.60
s=-
c=IN IP4 192.168.99.60
t=0 0
m=audio 61334 RTP/AVP 8 101
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15 
a=sendrecv
<-------------> 
--- (11 headers 10 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Found audio description format pcma for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|ilbc), peer - audio=(alaw)/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 192.168.99.60:61334
list_route: hop: <sip:[email protected]:2666;transport=udp;line=3rbn5n>
set_destination: Parsing <sip:[email protected]:2666;transport=udp;line=3rbn5n> for address/port to send to
set_destination: set destination to 192.168.99.60:2666
Transmitting (no NAT) to 192.168.99.60:2666:
ACK sip:[email protected]:2666;transport=udp;line=3rbn5n SIP/2.0
Via: SIP/2.0/UDP 192.168.99.10:5060;branch=z9hG4bK2361127c
Max-Forwards: 70
From: <sip:[email protected]>;tag=as578fad10
To: <sip:[email protected]:2666;transport=udp;line=3rbn5n>;tag=hnhu8e
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 ACK   
User-Agent: Asterisk PBX 11.2.1
Content-Length: 0


---
Audio is at 10026
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 217.0.18.16:5060 --->
SIP/2.0 200 OK  
Via: SIP/2.0/UDP 217.0.18.16:5060;branch=z9hG4bKg3Zqkv7irljpq3ed5b5slo3vufygjk994;received=217.0.18.16
Record-Route: <sip:217.0.18.16;transport=udp;lr>
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_2c2243ae
To: <sip:[email protected]:5060;user=phone>;tag=as13e9c253
Call-ID: [email protected]
CSeq: 610230 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 266

v=0
o=root 1191822783 1191822783 IN IP4 87.147.170.176
s=Asterisk PBX 11.2.1
c=IN IP4 87.147.170.176
t=0 0
m=audio 10026 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16 
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>  
set_destination: Parsing <sip:[email protected]:2666;transport=udp;line=3rbn5n> for address/port to send to
set_destination: set destination to 192.168.99.60:2666
Audio is at 10060
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.99.60:2666:
INVITE sip:[email protected]:2666;transport=udp;line=3rbn5n SIP/2.0
Via: SIP/2.0/UDP 192.168.99.10:5060;branch=z9hG4bK36c78e04
Max-Forwards: 70
From: <sip:[email protected]>;tag=as578fad10
To: <sip:[email protected]:2666;transport=udp;line=3rbn5n>;tag=hnhu8e
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 2014298794 2014298795 IN IP4 217.0.2.128
s=Asterisk PBX 11.2.1
c=IN IP4 217.0.2.128
t=0 0
m=audio 20404 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16 
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:217.0.18.16:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Max-Forwards: 51
Via: SIP/2.0/UDP 217.0.18.16:5060;branch=z9hG4bKg3Zqkv7iugrnnex1mpnrekuj2i9ixlzy3
To: <sip:[email protected]:5060;user=phone>;tag=as13e9c253
From: <sip:[email protected];user=phone>;tag=h7g4Esbg_2c2243ae
Call-ID: [email protected]
CSeq: 610230 ACK
Contact: <sip:[email protected];transport=udp>
Content-Length: 0
Session-ID: 5154423F-00F35C6C@ydxc1pcu-2
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REGISTER, UPDATE

<-------------> 
--- (11 headers 0 lines) ---
set_destination: Parsing <sip:217.0.18.16;transport=udp;lr> for address/port to send to
set_destination: set destination to 217.0.18.16:5060
Audio is at 10026
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 217.0.18.16:5060:
INVITE sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP 87.147.170.176:5060;branch=z9hG4bK4352ce43
Route: <sip:217.0.18.16;transport=udp;lr>
Max-Forwards: 70
From: <sip:[email protected]:5060;user=phone>;tag=as13e9c253
To: <sip:[email protected];user=phone>;tag=h7g4Esbg_2c2243ae
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 1191822783 1191822784 IN IP4 192.168.99.60
s=Asterisk PBX 11.2.1
c=IN IP4 192.168.99.60
t=0 0
m=audio 61334 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16 
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Retransmitting #1 (no NAT) to 192.168.99.60:2666:
INVITE sip:[email protected]:2666;transport=udp;line=3rbn5n SIP/2.0
Via: SIP/2.0/UDP 192.168.99.10:5060;branch=z9hG4bK36c78e04
Max-Forwards: 70
From: <sip:[email protected]>;tag=as578fad10
To: <sip:[email protected]:2666;transport=udp;line=3rbn5n>;tag=hnhu8e
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 2014298794 2014298795 IN IP4 217.0.2.128
s=Asterisk PBX 11.2.1
c=IN IP4 217.0.2.128
t=0 0
m=audio 20404 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16 
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:217.0.18.16:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 87.147.170.176:5060;branch=z9hG4bK4352ce43
To: <sip:[email protected];user=phone>;tag=h7g4Esbg_2c2243ae
From: <sip:[email protected];user=phone>;tag=as13e9c253
Call-ID: [email protected]
CSeq: 102 INVITE
Content-Length: 0

<-------------> 
--- (7 headers 0 lines) ---
set_destination: Parsing <sip:217.0.18.16;transport=udp;lr> for address/port to send to
set_destination: set destination to 217.0.18.16:5060
Transmitting (no NAT) to 217.0.18.16:5060:
ACK sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP 87.147.170.176:5060;branch=z9hG4bK4352ce43
Route: <sip:217.0.18.16;transport=udp;lr>
Max-Forwards: 70
From: <sip:[email protected]:5060;user=phone>;tag=as13e9c253
To: <sip:[email protected];user=phone>;tag=h7g4Esbg_2c2243ae
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 ACK   
User-Agent: Asterisk PBX 11.2.1
Content-Length: 0


---
[Mar 28 14:14:37] WARNING[10029][C-00000001]: chan_sip.c:22718 handle_response_invite: Received response: "Forbidden" from '<sip:[email protected]:5060;user=phone>;tag=as13e9c253'
set_destination: Parsing <sip:217.0.18.16;transport=udp;lr> for address/port to send to
set_destination: set destination to 217.0.18.16:5060
Audio is at 10026
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 217.0.18.16:5060:
INVITE sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP 87.147.170.176:5060;branch=z9hG4bK710ae31a
Route: <sip:217.0.18.16;transport=udp;lr>
Max-Forwards: 70
From: <sip:[email protected]:5060;user=phone>;tag=as13e9c253
To: <sip:[email protected];user=phone>;tag=h7g4Esbg_2c2243ae
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 266

v=0
o=root 1191822783 1191822785 IN IP4 87.147.170.176
s=Asterisk PBX 11.2.1
c=IN IP4 87.147.170.176
t=0 0
m=audio 10026 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16 
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.99.60:2666 --->
SIP/2.0 200 Ok  
Via: SIP/2.0/UDP 192.168.99.10:5060;branch=z9hG4bK36c78e04
From: <sip:[email protected]>;tag=as578fad10
To: <sip:[email protected]:2666;transport=udp;line=3rbn5n>;tag=hnhu8e
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Contact: <sip:[email protected]:2666;transport=udp;line=3rbn5n>
Supported: 100rel, replaces, norefersub
User-Agent: snom-m9/9.6.2-a
Content-Type: application/sdp
Content-Length: 205

v=0
o=root 572104435 572104437 IN IP4 192.168.99.60
s=-
c=IN IP4 192.168.99.60
t=0 0
m=audio 61334 RTP/AVP 8 101
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15 
a=sendrecv
<-------------> 
--- (11 headers 10 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Found audio description format pcma for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|ilbc), peer - audio=(alaw)/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 192.168.99.60:61334
set_destination: Parsing <sip:[email protected]:2666;transport=udp;line=3rbn5n> for address/port to send to
set_destination: set destination to 192.168.99.60:2666
Transmitting (no NAT) to 192.168.99.60:2666:
ACK sip:[email protected]:2666;transport=udp;line=3rbn5n SIP/2.0
Via: SIP/2.0/UDP 192.168.99.10:5060;branch=z9hG4bK4e23eb8c
Max-Forwards: 70
From: <sip:[email protected]>;tag=as578fad10
To: <sip:[email protected]:2666;transport=udp;line=3rbn5n>;tag=hnhu8e
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 103 ACK   
User-Agent: Asterisk PBX 11.2.1
Content-Length: 0


---
set_destination: Parsing <sip:[email protected]:2666;transport=udp;line=3rbn5n> for address/port to send to
set_destination: set destination to 192.168.99.60:2666
Audio is at 10060
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.99.60:2666:
INVITE sip:[email protected]:2666;transport=udp;line=3rbn5n SIP/2.0
Via: SIP/2.0/UDP 192.168.99.10:5060;branch=z9hG4bK5ea892ad
Max-Forwards: 70
From: <sip:[email protected]>;tag=as578fad10
To: <sip:[email protected]:2666;transport=udp;line=3rbn5n>;tag=hnhu8e
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 104 INVITE
User-Agent: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 2014298794 2014298796 IN IP4 192.168.99.10
s=Asterisk PBX 11.2.1
c=IN IP4 192.168.99.10
t=0 0
m=audio 10060 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16 
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.99.60:2666 --->
SIP/2.0 200 Ok  
Via: SIP/2.0/UDP 192.168.99.10:5060;branch=z9hG4bK36c78e04
From: <sip:[email protected]>;tag=as578fad10
To: <sip:[email protected]:2666;transport=udp;line=3rbn5n>;tag=hnhu8e
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Contact: <sip:[email protected]:2666;transport=udp;line=3rbn5n>
Supported: 100rel, replaces, norefersub
User-Agent: snom-m9/9.6.2-a
Content-Type: application/sdp
Content-Length: 205

v=0
o=root 572104435 572104437 IN IP4 192.168.99.60
s=-
c=IN IP4 192.168.99.60
t=0 0
m=audio 61334 RTP/AVP 8 101
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15 
a=sendrecv
<-------------> 
--- (11 headers 10 lines) ---
set_destination: Parsing <sip:[email protected]:2666;transport=udp;line=3rbn5n> for address/port to send to
set_destination: set destination to 192.168.99.60:2666
Transmitting (no NAT) to 192.168.99.60:2666:
ACK sip:[email protected]:2666;transport=udp;line=3rbn5n SIP/2.0
Via: SIP/2.0/UDP 192.168.99.10:5060;branch=z9hG4bK7215d0de
Max-Forwards: 70
From: <sip:[email protected]>;tag=as578fad10
To: <sip:[email protected]:2666;transport=udp;line=3rbn5n>;tag=hnhu8e
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 103 ACK   
User-Agent: Asterisk PBX 11.2.1
Content-Length: 0


---
Retransmitting #1 (no NAT) to 217.0.18.16:5060:
INVITE sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP 87.147.170.176:5060;branch=z9hG4bK710ae31a
Route: <sip:217.0.18.16;transport=udp;lr>
Max-Forwards: 70
From: <sip:[email protected]:5060;user=phone>;tag=as13e9c253
To: <sip:[email protected];user=phone>;tag=h7g4Esbg_2c2243ae
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 266

v=0
o=root 1191822783 1191822785 IN IP4 87.147.170.176
s=Asterisk PBX 11.2.1
c=IN IP4 87.147.170.176
t=0 0
m=audio 10026 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16 
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Retransmitting #1 (no NAT) to 192.168.99.60:2666:
INVITE sip:[email protected]:2666;transport=udp;line=3rbn5n SIP/2.0
Via: SIP/2.0/UDP 192.168.99.10:5060;branch=z9hG4bK5ea892ad
Max-Forwards: 70
From: <sip:[email protected]>;tag=as578fad10
To: <sip:[email protected]:2666;transport=udp;line=3rbn5n>;tag=hnhu8e
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 104 INVITE
User-Agent: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 2014298794 2014298796 IN IP4 192.168.99.10
s=Asterisk PBX 11.2.1
c=IN IP4 192.168.99.10
t=0 0
m=audio 10060 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16 
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:217.0.18.16:5060 --->
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP 87.147.170.176:5060;branch=z9hG4bK710ae31a
To: <sip:[email protected];user=phone>;tag=h7g4Esbg_2c2243ae
From: <sip:[email protected];user=phone>;tag=as13e9c253
Call-ID: [email protected]
CSeq: 103 INVITE
Contact: <sip:[email protected];transport=udp>
Content-Length: 0

<-------------> 
--- (8 headers 0 lines) ---
set_destination: Parsing <sip:217.0.18.16;transport=udp;lr> for address/port to send to
set_destination: set destination to 217.0.18.16:5060
Transmitting (no NAT) to 217.0.18.16:5060:
ACK sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP 87.147.170.176:5060;branch=z9hG4bK710ae31a
Route: <sip:217.0.18.16;transport=udp;lr>
Max-Forwards: 70
From: <sip:[email protected]:5060;user=phone>;tag=as13e9c253
To: <sip:[email protected];user=phone>;tag=h7g4Esbg_2c2243ae
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 103 ACK   
User-Agent: Asterisk PBX 11.2.1
Content-Length: 0


---
[Mar 28 14:14:37] WARNING[10029][C-00000001]: chan_sip.c:22807 handle_response_invite: just did sched_add waitid(180) for sip_reinvite_retry for dialog [email protected] in handle_response_invite

<--- SIP read from UDP:217.0.18.16:5060 --->
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP 87.147.170.176:5060;branch=z9hG4bK710ae31a
To: <sip:[email protected];user=phone>;tag=h7g4Esbg_2c2243ae
From: <sip:[email protected];user=phone>;tag=as13e9c253
Call-ID: [email protected]
CSeq: 103 INVITE
Contact: <sip:[email protected];transport=udp>
Content-Length: 0

<-------------> 
--- (8 headers 0 lines) ---
set_destination: Parsing <sip:217.0.18.16;transport=udp;lr> for address/port to send to
set_destination: set destination to 217.0.18.16:5060
Transmitting (no NAT) to 217.0.18.16:5060:
ACK sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP 87.147.170.176:5060;branch=z9hG4bK710ae31a
Route: <sip:217.0.18.16;transport=udp;lr>
Max-Forwards: 70
From: <sip:[email protected]:5060;user=phone>;tag=as13e9c253
To: <sip:[email protected];user=phone>;tag=h7g4Esbg_2c2243ae
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 103 ACK   
User-Agent: Asterisk PBX 11.2.1
Content-Length: 0


---
Scheduling destruction of SIP dialog '[email protected]:5060' in 6400 ms (Method: INVITE)
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: ACK)
set_destination: Parsing <sip:217.0.18.16;transport=udp;lr> for address/port to send to
set_destination: set destination to 217.0.18.16:5060
Reliably Transmitting (no NAT) to 217.0.18.16:5060:
BYE sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP 87.147.170.176:5060;branch=z9hG4bK5af148ed
Route: <sip:217.0.18.16;transport=udp;lr>
Max-Forwards: 70
From: <sip:[email protected]:5060;user=phone>;tag=as13e9c253
To: <sip:[email protected];user=phone>;tag=h7g4Esbg_2c2243ae
Call-ID: [email protected]
CSeq: 104 BYE   
User-Agent: Asterisk PBX 11.2.1
X-Asterisk-HangupCause: Interworking, unspecified
X-Asterisk-HangupCauseCode: 127
Content-Length: 0


---

<--- SIP read from UDP:192.168.99.60:2666 --->
SIP/2.0 200 Ok  
Via: SIP/2.0/UDP 192.168.99.10:5060;branch=z9hG4bK5ea892ad
From: <sip:[email protected]>;tag=as578fad10
To: <sip:[email protected]:2666;transport=udp;line=3rbn5n>;tag=hnhu8e
Call-ID: [email protected]:5060
CSeq: 104 INVITE
Contact: <sip:[email protected]:2666;transport=udp;line=3rbn5n>
Supported: 100rel, replaces, norefersub
User-Agent: snom-m9/9.6.2-a
Content-Type: application/sdp
Content-Length: 205

v=0
o=root 572104435 572104438 IN IP4 192.168.99.60
s=-
c=IN IP4 192.168.99.60
t=0 0
m=audio 61334 RTP/AVP 8 101
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15 
a=sendrecv
<-------------> 
--- (11 headers 10 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Found audio description format pcma for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|ilbc), peer - audio=(alaw)/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 192.168.99.60:61334
set_destination: Parsing <sip:[email protected]:2666;transport=udp;line=3rbn5n> for address/port to send to
set_destination: set destination to 192.168.99.60:2666
Transmitting (no NAT) to 192.168.99.60:2666:
ACK sip:[email protected]:2666;transport=udp;line=3rbn5n SIP/2.0
Via: SIP/2.0/UDP 192.168.99.10:5060;branch=z9hG4bK60314ced
Max-Forwards: 70
From: <sip:[email protected]>;tag=as578fad10
To: <sip:[email protected]:2666;transport=udp;line=3rbn5n>;tag=hnhu8e
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 104 ACK   
User-Agent: Asterisk PBX 11.2.1
Content-Length: 0


---
set_destination: Parsing <sip:[email protected]:2666;transport=udp;line=3rbn5n> for address/port to send to
set_destination: set destination to 192.168.99.60:2666
Reliably Transmitting (no NAT) to 192.168.99.60:2666:
BYE sip:[email protected]:2666;transport=udp;line=3rbn5n SIP/2.0
Via: SIP/2.0/UDP 192.168.99.10:5060;branch=z9hG4bK4bcc0299
Max-Forwards: 70
From: <sip:[email protected]>;tag=as578fad10
To: <sip:[email protected]:2666;transport=udp;line=3rbn5n>;tag=hnhu8e
Call-ID: [email protected]:5060
CSeq: 105 BYE   
User-Agent: Asterisk PBX 11.2.1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
Scheduling destruction of SIP dialog '[email protected]:5060' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.99.60:2666 --->
SIP/2.0 200 Ok  
Via: SIP/2.0/UDP 192.168.99.10:5060;branch=z9hG4bK5ea892ad
From: <sip:[email protected]>;tag=as578fad10
To: <sip:[email protected]:2666;transport=udp;line=3rbn5n>;tag=hnhu8e
Call-ID: [email protected]:5060
CSeq: 104 INVITE
Contact: <sip:[email protected]:2666;transport=udp;line=3rbn5n>
Supported: 100rel, replaces, norefersub
User-Agent: snom-m9/9.6.2-a
Content-Type: application/sdp
Content-Length: 205

v=0
o=root 572104435 572104438 IN IP4 192.168.99.60
s=-
c=IN IP4 192.168.99.60
t=0 0
m=audio 61334 RTP/AVP 8 101
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15 
a=sendrecv
<-------------> 
--- (11 headers 10 lines) ---
set_destination: Parsing <sip:[email protected]:2666;transport=udp;line=3rbn5n> for address/port to send to
set_destination: set destination to 192.168.99.60:2666
Transmitting (no NAT) to 192.168.99.60:2666:
ACK sip:[email protected]:2666;transport=udp;line=3rbn5n SIP/2.0
Via: SIP/2.0/UDP 192.168.99.10:5060;branch=z9hG4bK18433cc0
Max-Forwards: 70
From: <sip:[email protected]>;tag=as578fad10
To: <sip:[email protected]:2666;transport=udp;line=3rbn5n>;tag=hnhu8e
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 104 ACK   
User-Agent: Asterisk PBX 11.2.1
Content-Length: 0


---

<--- SIP read from UDP:192.168.99.60:2666 --->
SIP/2.0 200 Ok  
Via: SIP/2.0/UDP 192.168.99.10:5060;branch=z9hG4bK4bcc0299
From: <sip:[email protected]>;tag=as578fad10
To: <sip:[email protected]:2666;transport=udp;line=3rbn5n>;tag=hnhu8e
Call-ID: [email protected]:5060
CSeq: 105 BYE   
Contact: <sip:[email protected]:2666;transport=udp;line=3rbn5n>
Supported: 100rel, replaces, norefersub
Content-Length: 0

<-------------> 
--- (9 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: INVITE

<--- SIP read from UDP:217.0.18.16:5060 --->
SIP/2.0 200 OK  
Via: SIP/2.0/UDP 87.147.170.176:5060;branch=z9hG4bK5af148ed
To: <sip:[email protected];user=phone>;tag=h7g4Esbg_2c2243ae
From: <sip:[email protected]:5060;user=phone>;tag=as13e9c253
Call-ID: [email protected]
CSeq: 104 BYE   
Content-Length: 0

* Ausgehende Gespräche werden nach genau 30min beendet

Dieses Problem ist mir erst vorgestern aufgefallen als ich mit einem Freund telefoniert habe und dies 2 mal hintereinander passiert ist. Heute morgen habe ich es mit einem anderen Kollegen verifiziert.

Hier das sip debug log von dem Moment wo die Verbindung unterbrochen wird:

Code: 
Scheduling destruction of SIP dialog '54cdfe4573f98ffdcafa6f2cfaa3a934@0:0:0:0:0:0:0:0' in 32000 ms (Method: OPTIONS)
set_destination: Parsing <sip:217.0.18.16;transport=udp;lr> for address/port to send to
set_destination: set destination to 217.0.18.16:5060
Audio is at 10048
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 217.0.18.16:5060:
INVITE sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP 87.147.170.176:5060;branch=z9hG4bK6e6399fa
Route: <sip:217.0.18.16;transport=udp;lr>
Max-Forwards: 70
From: "071419569111" <sip:[email protected]>;tag=as2a4fc61b
To: <sip:[email protected]>;tag=h7g4Esbg_89165a87
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.2.1
Session-Expires: 3600;refresher=uac
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (Session-Timers)
Content-Type: application/sdp
Content-Length: 266

v=0
o=root 1118042216 1118042216 IN IP4 87.147.170.176
s=Asterisk PBX 11.2.1
c=IN IP4 87.147.170.176
t=0 0
m=audio 10048 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16 
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Retransmitting #1 (no NAT) to 217.0.18.16:5060:
INVITE sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP 87.147.170.176:5060;branch=z9hG4bK6e6399fa
Route: <sip:217.0.18.16;transport=udp;lr>
Max-Forwards: 70
From: "071419569111" <sip:[email protected]>;tag=as2a4fc61b
To: <sip:[email protected]>;tag=h7g4Esbg_89165a87
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.2.1
Session-Expires: 3600;refresher=uac
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (Session-Timers)
Content-Type: application/sdp
Content-Length: 266

v=0
o=root 1118042216 1118042216 IN IP4 87.147.170.176
s=Asterisk PBX 11.2.1
c=IN IP4 87.147.170.176
t=0 0
m=audio 10048 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16 
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:217.0.18.16:5060 --->
SIP/2.0 488 SDP Parameter Error In SIP Request
Via: SIP/2.0/UDP 87.147.170.176:5060;branch=z9hG4bK6e6399fa
To: <sip:[email protected]>;tag=h7g4Esbg_89165a87
From: "071419569111" <sip:[email protected]>;tag=as2a4fc61b
Call-ID: [email protected]
CSeq: 103 INVITE
Content-Length: 0

<-------------> 
--- (7 headers 0 lines) ---
set_destination: Parsing <sip:217.0.18.16;transport=udp;lr> for address/port to send to
set_destination: set destination to 217.0.18.16:5060
Transmitting (no NAT) to 217.0.18.16:5060:
ACK sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP 87.147.170.176:5060;branch=z9hG4bK6e6399fa
Route: <sip:217.0.18.16;transport=udp;lr>
Max-Forwards: 70
From: "071419569111" <sip:[email protected]>;tag=as2a4fc61b
To: <sip:[email protected]>;tag=h7g4Esbg_89165a87
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 103 ACK   
User-Agent: Asterisk PBX 11.2.1
Content-Length: 0


---

<--- SIP read from UDP:217.0.18.16:5060 --->
SIP/2.0 488 SDP Parameter Error In SIP Request
Via: SIP/2.0/UDP 87.147.170.176:5060;branch=z9hG4bK6e6399fa
To: <sip:[email protected]>;tag=h7g4Esbg_89165a87
From: "071419569111" <sip:[email protected]>;tag=as2a4fc61b
Call-ID: [email protected]
CSeq: 103 INVITE
Content-Length: 0

<-------------> 
--- (7 headers 0 lines) ---
set_destination: Parsing <sip:217.0.18.16;transport=udp;lr> for address/port to send to
set_destination: set destination to 217.0.18.16:5060
Transmitting (no NAT) to 217.0.18.16:5060:
ACK sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP 87.147.170.176:5060;branch=z9hG4bK6e6399fa
Route: <sip:217.0.18.16;transport=udp;lr>
Max-Forwards: 70
From: "071419569111" <sip:[email protected]>;tag=as2a4fc61b
To: <sip:[email protected]>;tag=h7g4Esbg_89165a87
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 103 ACK   
User-Agent: Asterisk PBX 11.2.1
Content-Length: 0


---
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:217.0.18.16;transport=udp;lr> for address/port to send to
set_destination: set destination to 217.0.18.16:5060
Reliably Transmitting (no NAT) to 217.0.18.16:5060:
BYE sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP 87.147.170.176:5060;branch=z9hG4bK5515a256
Route: <sip:217.0.18.16;transport=udp;lr>
Max-Forwards: 70
From: "071419569111" <sip:[email protected]>;tag=as2a4fc61b
To: <sip:[email protected]>;tag=h7g4Esbg_89165a87
Call-ID: [email protected]
CSeq: 104 BYE   
User-Agent: Asterisk PBX 11.2.1
X-Asterisk-HangupCause: Bearer capability not available
X-Asterisk-HangupCauseCode: 58
Content-Length: 0


---
Scheduling destruction of SIP dialog '3n3pslrj8a' in 6400 ms (Method: ACK)
set_destination: Parsing <sip:[email protected]:2666;transport=udp;line=3rbn5n> for address/port to send to
set_destination: set destination to 192.168.9.60:2666
Reliably Transmitting (no NAT) to 192.168.9.60:2666:
BYE sip:[email protected]:2666;transport=udp;line=3rbn5n SIP/2.0
Via: SIP/2.0/UDP 192.168.9.10:5060;branch=z9hG4bK0e6ca433;rport
Max-Forwards: 70
From: "+4970719288888" <sip:[email protected];user=phone>;tag=as3dbc7897
To: "tut nix zur sache" <sip:[email protected]>;tag=57zjwz
Call-ID: 3n3pslrj8a
CSeq: 102 BYE   
User-Agent: Asterisk PBX 11.2.1
Proxy-Authorization: Digest username="40", realm="asterisk", algorithm=MD5, uri="sip:hostname.internal-domain.local", nonce="39a38ab7", response="bc54339402a8d2a35f3bbc6c2c21e1e6"
X-Asterisk-HangupCause: Bearer capability not available
X-Asterisk-HangupCauseCode: 58
Content-Length: 0


---

<--- SIP read from UDP:192.168.9.60:2666 --->
SIP/2.0 200 Ok  
Via: SIP/2.0/UDP 192.168.9.10:5060;branch=z9hG5bK0e6ca433;rport=5060
From: "+4970719288888" <sip:[email protected];user=phone>;tag=as3dbc7897
To: "tut nix zur sache" <sip:[email protected]>;tag=57zjwz
Call-ID: 3n3pslrj8a
CSeq: 102 BYE   
Contact: <sip:[email protected]:2666;transport=udp;line=3rbn5n>
Supported: 100rel, replaces, norefersub
Content-Length: 0

<-------------> 
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '3n3pslrj8a' Method: ACK

<--- SIP read from UDP:217.0.18.16:5060 --->
SIP/2.0 200 OK  
Via: SIP/2.0/UDP 87.147.170.176:5060;branch=z9hG4bK5515a256
To: <sip:[email protected]>;tag=h7g4Esbg_89165a87
From: "071419569111" <sip:[email protected]>;tag=as2a4fc61b
Call-ID: [email protected]
CSeq: 104 BYE   
Content-Length: 0

<-------------> 
--- (7 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: INVITE
Reliably Transmitting (no NAT) to 192.168.9.124:5060:
OPTIONS sip:[email protected]:5060;transport=udp;registering_acc=hostname.internal-domain.local SIP/2.0
Via: SIP/2.0/UDP 192.168.9.10:5060;branch=z9hG4bK49290921
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as1774ee36
To: <sip:[email protected]:5060;transport=udp;registering_acc=hostname.internal-domain.local>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.2.1
Date: Thu, 28 Mar 2013 14:41:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.9.124:5060 --->
SIP/2.0 200 OK  
To: <sip:[email protected]:5060;transport=udp;registering_acc=hostname.internal-domain.local>;tag=ae754fe6
Via: SIP/2.0/UDP 192.168.9.10:5060;branch=z9hG4bK49290921
CSeq: 102 OPTIONS
Call-ID: [email protected]:5060
From: "asterisk" <sip:[email protected]>;tag=as1774ee36
Contact: "morlix" <sip:[email protected]:5060;transport=udp;registering_acc=hostname.internal-domain.local>
User-Agent: Jitsi2.0.4506.10553Linux
Allow: INFO,OPTIONS,MESSAGE,BYE,REFER,SUBSCRIBE,ACK,CANCEL,PUBLISH,NOTIFY,INVITE
Allow-Events: refer
Content-Length: 0

<-------------> 
--- (11 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
Really destroying SIP dialog '2ea28ff307a425579cd46b6c8435d526@0:0:0:0:0:0:0:0' Method: OPTIONS



So hier erstmal meine aktuelle Konfiguration:

Asterisk 11.2.1 (vormals 1.8.X)
Der Asterisk Server läuft auf einem Gentoo Linux System hinter einem Ipfire Router von dem die Ports UDP 5060 und UDP 10000-10100 auf den Asterisk Server weitergeleitet (port forwarding) werden.

rtp.conf
Code: 
[general]
rtpstart=10000
rtpend=10100

sip.conf
Code: 
[general]
context=default
allowguest=no
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
transport=udp
srvlookup=yes
defaultexpiry=600
language=de
registertimeout=60

register => 071419569111:secret:[email protected]/071419569111
register => 071413880222:secret:[email protected]/071413880222
register => 071416493333:secret:[email protected]/071416493333
register => 3228444:[email protected]/3228444

localnet=192.168.9.0/24
externhost=hostname.dyndns.org
externrefresh=180
nat=auto_force_rport,auto_comedia
domain=hostname.internal-domain.local
autodomain=yes

[authentication]
[basic-options](!)                ; a template
        dtmfmode=rfc2833
        context=from-office
        type=friend
[natted-phone](!,basic-options)   ; another template inheriting basic-options
        directmedia=no
        host=dynamic
[public-phone](!,basic-options)   ; another template inheriting basic-options
        directmedia=yes
[my-codecs](!)                    ; a template for my preferred codecs
        disallow=all
        allow=ilbc
        allow=g729
        allow=gsm
        allow=g723
        allow=ulaw
[ulaw-phone](!)                   ; and another one for ulaw-only
        disallow=all
        allow=ulaw

[telekom-in]
type=peer
fromdomain=tel.t-online.de
host=tel.t-online.de
qualify=yes
disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=ilbc
context=from-telekom

[telekom-out-071419569111]
type=peer
username=username
secret=secret
host=tel.t-online.de
fromdomain=tel.t-online.de
fromuser=071419569111
qualify=yes
canreinvite=no
disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=ilbc
insecure=port,invite

[telekom-out-071413880222]
type=peer
username=username
secret=secret
host=tel.t-online.de
fromdomain=tel.t-online.de
fromuser=071413880222
qualify=yes
canreinvite=no
disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=ilbc
insecure=port,invite

[telekom-out-071416493333]
type=peer
username=username
secret=secret
host=tel.t-online.de
fromdomain=tel.t-online.de
fromuser=071416493333
qualify=yes
canreinvite=no
disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=ilbc
insecure=port,invite

[sipgate-in]
type=peer
fromdomain=sipgate.de
host=sipgate.de
qualify=yes
disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=ilbc
context=from-sipgate

[sipgate-out-3228444]
type=peer
username=3228444
fromuser=3228444
secret=secret
host=sipgate.de
fromdomain=sipgate.de
qualify=yes
insecure=port,invite
canreinvite=no
disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=ilbc
dtmfmode=rfc2833

[40]
type=friend
user=40
secret=secret
host=dynamic
nat=no
qualify=yes
context=internal-phones
callerid="tut nix zur sache" <40>
mailbox=40
deny=0.0.0.0/0.0.0.0
permit=192.168.9.0/255.255.255.0
disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=ilbc

[morlix]
type=friend
user=morlix
secret=secret
host=dynamic
nat=no
qualify=yes
context=internal-phones
callerid="morlix" <morlix>
mailbox=
deny=0.0.0.0/0.0.0.0
permit=192.168.9.0/255.255.255.0
disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=ilbc

extensions.conf
Code: 
[general]
static=yes
writeprotect=no
clearglobalvars=no

[globals]
CONSOLE=Console/dsp                             ; Console interface for demo
IAXINFO=guest                                   ; IAXtel username/password
TRUNK=DAHDI/G2                                  ; Trunk interface
AREA_CODE=07141
TRUNKMSD=1                                      ; MSD digits to strip (usually 1 or 0)

[default]

[internal-phones]
include => internal-calls
include => voicemail-comfort
include => voicemail-general
include => telekom-out
include => sipgate-out
include => test
include => unknown_exten

[test]
exten => 80,1,Answer()
exten => 80,n,Playback(hello-world)
exten => 80,n,Hangup()
exten => 81,1,Answer()
exten => 81,n,Wait(1)
exten => 81,n,Playback(demo-echotest)
exten => 81,n,Echo()
exten => 81,n,Playback(demo-echodone)
exten => 81,n,Hangup()

[internal-calls]
exten => 40,1,Dial(SIP/${EXTEN},60)
exten => 40,n,VoiceMail(${EXTEN},u)
exten => 40,n,Hangup()

[voicemail-comfort]   
exten => 99,1,VoiceMailMain(${CALLERID(num)},s)
exten => 99,n,Hangup()
exten => M,1,VoiceMailMain(${CALLERID(num)},s)
exten => M,n,Hangup() 

[voicemail-general]   
exten => 98,1,VoiceMailMain()
exten => 98,n,Hangup()

[unknown_exten]
exten => _X.,1,NoOp(Undefined number ${INVALID_EXTEN} called.)
exten => _X.,n,Answer()
exten => _X.,n,Playback(that-is-not-rec-phn-num)
exten => _X.,n,Hangup()

[telekom-out]
exten => _0Z.,1,Set(CALLERID(number)=071419569111)
exten => _0Z.,n,Set(CALLERID(name)=071419569111)
exten => _0Z.,n,Dial(SIP/${EXTEN}@telekom-out-071419569111)
exten => _0Z.,n,Hangup()
exten => _ZXXX.,1,Set(CALLERID(number)=071419569111)
exten => _ZXXX.,n,Set(CALLERID(name)=071419569111)
exten => _ZXXX.,n,Dial(SIP/${AREA_CODE}${EXTEN}@telekom-out-071419569111)
exten => _ZXXX.,n,Hangup()
exten => _+49[1-9].,1,Set(CALLERID(number)=071419569111)
exten => _+49[1-9].,n,Set(CALLERID(name)=071419569111)
exten => _+49[1-9].,n,Dial(SIP/${EXTEN}@telekom-out-071419569111)
exten => _+49[1-9].,n,Hangup()
exten => _0049[1-9].,1,Set(CALLERID(number)=071419569111)
exten => _0049[1-9].,n,Set(CALLERID(name)=071419569111)
exten => _0049[1-9].,n,Dial(SIP/${EXTEN}@telekom-out-071419569111)
exten => _0049[1-9].,n,Hangup()
exten => _+XX[1-9].,1,Set(CALLERID(number)=071419569111)
exten => _+XX[1-9].,n,Set(CALLERID(name)=071419569111)
exten => _+XX[1-9].,n,Dial(SIP/${EXTEN}@telekom-out-071419569111)
exten => _+XX[1-9].,n,Hangup()
exten => _00XX.,1,Set(CALLERID(number)=071419569111)
exten => _00XX.,n,Set(CALLERID(name)=071419569111)
exten => _00XX.,n,Dial(SIP/${EXTEN}@telekom-out-071419569111)
exten => _00XX.,n,Hangup()

[sipgate-out]
exten => _*1.,1,Set(CALLERID(number)=3228444)
exten => _*1.,n,Set(CALLERID(name)=4971413099444)
exten => _*1.,n,Dial(SIP/${EXTEN:2}@sipgate-out-3228444)
exten => _*1.,n,Hangup()

[from-telekom]
exten => 071419569111,1,Dial(SIP/40,60)
exten => 071413880222,1,Macro(from-external,SIP/40,40)
exten => 071416493333,1,Dial(SIP/morlix,60)

[from-sipgate]
exten => 3228444,1,Macro(from-external,SIP/40,40)
exten => 3228444,1,Macro(from-external,SIP/40,40)
exten => 3228444,1,Macro(from-external,SIP/40,40)

[macro-from-external] 
exten => s,1,Dial(${ARG1},30)
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,VoiceMail(${ARG2},u)
exten => s-NOANSWER,n,Hangup()
exten => s-BUSY,1,VoiceMail(${ARG2},b)
exten => s-BUSY,n,Hangup()
exten => s-ANSWER,1,Hangup()
exten => _s-.,1,Goto(s-NOANSWER,1)

Danke im Voraus für jeden Hinweis!

Gruß morlix

PS: Ich habe die Telefonnummern, Benutzernamen, Passwörter und sonstiges verändert, bringt also nichts die Nummern anzurufen oder die Passwörter zu testen.
 
Zuletzt bearbeitet:
So mein zweites Problem konnte ich jetzt doch selbst lösen.

Und zwar lag es an der Telekom, welche keine SIP Session Timers supportet.

Ich habe jetzt für die ausgehenden Peers der Telekom in der sip.conf folgenden Parameter hinzugefügt und damit ist das Problem erledigt.

Code: 
session-timers=refuse

Fehlt nur noch die Lösung für das Problem mit den ankommenden Anrufen.
 
So jetzt habe ich auch mein erstes Problem gelöst.

Der Parameter "canreinvite" wurde umbenannt in "directmedia". (Echt toll das asterisk keine Meldung über deprecated parameter ausgibt).

Nachdem ich allen externen Peers "directmedia=no" hinzugefügt habe, die "nat=yes" Parameter entfernt und global "nat=auto_force_rport,auto_comedia" hinzugefügt habe, geht alles wieder.
 
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