Hallo,
das Thema ist nicht neu, aber Hilfe habe ich bisher noch keine passende gefunden.
Das Problem ist, dass Gespräche, die von aussen kommen nach ca 20 Sekunden abgebrochen werden. Für den Anrufer scheint es , als ob der Gesprächspartner einfach nichts mehr sagt. Der Angerufene (mein SIP Telefon BT201 am Asterisk) hört ein Besetzzeichen.
Hier meine SIP.conf:
Mit sip debug ehalte ich folgenden Log:
Kann jemand helfen ?
Ich nutze Asterisk 1.4.25 hinter einem NAT.
das Thema ist nicht neu, aber Hilfe habe ich bisher noch keine passende gefunden.
Das Problem ist, dass Gespräche, die von aussen kommen nach ca 20 Sekunden abgebrochen werden. Für den Anrufer scheint es , als ob der Gesprächspartner einfach nichts mehr sagt. Der Angerufene (mein SIP Telefon BT201 am Asterisk) hört ein Besetzzeichen.
Hier meine SIP.conf:
Code:
[general]
port=5060
bindaddr=0.0.0.0
externhost=checkip.dyndns.org
externrefresh=10
localnet=192.168.2.0/255.255.255.0
disallow=all
allow=alaw
allow=gsm
srvlookup = yes
; Deutsche Sprachbausteine aktivieren
language=de
; bei Sipgate registrieren
register => uuuu:[email protected]/uuuu
[2000]
type=friend
secret=1234
host=dynamic
vmexten = 2999
mailbox = 2000
[sip-account]
type = friend
context = sipgate-in
username=uuuu
fromuser=uuuu
secret=pppp
host=sipgate.de
fromdomain=sigate.de
qualify=yes
insecure=port,invite
nat=yes
canreinvite = noMit sip debug ehalte ich folgenden Log:
Code:
-- Executing [7589073@sipgate-in:1] Verbose("SIP/sip-account-081fb680", "3|### Eingehender SIP - Anruf von 01727395202 an 7589073") in new stack
-- ### Eingehender SIP - Anruf von 01727395202 an 7589073
-- Executing [7589073@sipgate-in:2] Dial("SIP/sip-account-081fb680", "SIP/2000|20") in new stack
Audio is at 192.168.2.201 port 17558
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.2.102:5060:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK1dc8b5e8;rport
From: "01727395202" <sip:[email protected]>;tag=as3bedd72e
To: <sip:[email protected]:5060;transport=udp>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 18 Jun 2009 06:17:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 27427 27427 IN IP4 192.168.2.201
s=session
c=IN IP4 192.168.2.201
t=0 0
m=audio 17558 RTP/AVP 8 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called 2000
<--- SIP read from 192.168.2.102:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK1dc8b5e8;rport
From: "01727395202" <sip:[email protected]>;tag=as3bedd72e
To: <sip:[email protected]:5060;transport=udp>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Grandstream BT200 1.1.6.37
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from 192.168.2.102:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK1dc8b5e8;rport
From: "01727395202" <sip:[email protected]>;tag=as3bedd72e
To: <sip:[email protected]:5060;transport=udp>;tag=56e523039c097df8
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Grandstream BT200 1.1.6.37
Contact: <sip:[email protected]:5060;transport=udp>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
-- SIP/2000-081ff690 is ringing
<--- Transmitting (NAT) to 217.10.79.9:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK5c6b.85c175a6.0;received=217.10.79.9
Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK5c6b.85c175a6.0
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK116f45c0
Via: SIP/2.0/UDP 217.10.67.141:5060;branch=z9hG4bK116f45c0;rport=5060
Record-Route: <sip:217.10.79.9;lr=on;ftag=as2d9c75fe>
Record-Route: <sip:172.20.40.2;lr=on>
Record-Route: <sip:217.10.79.9;lr=on;ftag=as2d9c75fe>
From: "01727395202" <sip:[email protected]>;tag=as2d9c75fe
To: <sip:[email protected]>;tag=as2916f91d
Call-ID: [email][email protected][/email]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Length: 0
<------------>
<--- SIP read from 192.168.2.102:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK1dc8b5e8;rport
From: "01727395202" <sip:[email protected]>;tag=as3bedd72e
To: <sip:[email protected]:5060;transport=udp>;tag=56e523039c097df8
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Grandstream BT200 1.1.6.37
Contact: <sip:[email protected]:5060;transport=udp>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Supported: replaces, timer
Content-Length: 213
v=0
o=2000 8000 8000 IN IP4 192.168.2.102
s=SIP Call
c=IN IP4 192.168.2.102
t=0 0
m=audio 5004 RTP/AVP 8 101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
<------------->
--- (12 headers 11 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.2.102:5004
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0xa (gsm|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.2.102:5004
list_route: hop: <sip:[email protected]:5060;transport=udp>
set_destination: Parsing <sip:[email protected]:5060;transport=udp> for address/port to send to
set_destination: set destination to 192.168.2.102, port 5060
Transmitting (no NAT) to 192.168.2.102:5060:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK7dba3a57;rport
From: "01727395202" <sip:[email protected]>;tag=as3bedd72e
To: <sip:[email protected]:5060;transport=udp>;tag=56e523039c097df8
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/2000-081ff690 answered SIP/sip-account-081fb680
Audio is at 208.78.69.70 port 15184
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK5c6b.85c175a6.0;received=217.10.79.9
Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK5c6b.85c175a6.0
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK116f45c0
Via: SIP/2.0/UDP 217.10.67.141:5060;branch=z9hG4bK116f45c0;rport=5060
Record-Route: <sip:217.10.79.9;lr=on;ftag=as2d9c75fe>
Record-Route: <sip:172.20.40.2;lr=on>
Record-Route: <sip:217.10.79.9;lr=on;ftag=as2d9c75fe>
From: "01727395202" <sip:[email protected]>;tag=as2d9c75fe
To: <sip:[email protected]>;tag=as2916f91d
Call-ID: [email][email protected][/email]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 263
v=0
o=root 27427 27427 IN IP4 208.78.69.70
s=session
c=IN IP4 208.78.69.70
t=0 0
m=audio 15184 RTP/AVP 8 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
-- Packet2Packet bridging SIP/sip-account-081fb680 and SIP/2000-081ff690
Retransmitting #1 (NAT) to 217.10.79.9:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK5c6b.85c175a6.0;received=217.10.79.9
Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK5c6b.85c175a6.0
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK116f45c0
Via: SIP/2.0/UDP 217.10.67.141:5060;branch=z9hG4bK116f45c0;rport=5060
Record-Route: <sip:217.10.79.9;lr=on;ftag=as2d9c75fe>
Record-Route: <sip:172.20.40.2;lr=on>
Record-Route: <sip:217.10.79.9;lr=on;ftag=as2d9c75fe>
From: "01727395202" <sip:[email protected]>;tag=as2d9c75fe
To: <sip:[email protected]>;tag=as2916f91d
Call-ID: [email][email protected][/email]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 263
v=0
o=root 27427 27427 IN IP4 208.78.69.70
s=session
c=IN IP4 208.78.69.70
t=0 0
m=audio 15184 RTP/AVP 8 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #2 (NAT) to 217.10.79.9:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK5c6b.85c175a6.0;received=217.10.79.9
Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK5c6b.85c175a6.0
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK116f45c0
Via: SIP/2.0/UDP 217.10.67.141:5060;branch=z9hG4bK116f45c0;rport=5060
Record-Route: <sip:217.10.79.9;lr=on;ftag=as2d9c75fe>
Record-Route: <sip:172.20.40.2;lr=on>
Record-Route: <sip:217.10.79.9;lr=on;ftag=as2d9c75fe>
From: "01727395202" <sip:[email protected]>;tag=as2d9c75fe
To: <sip:[email protected]>;tag=as2916f91d
Call-ID: [email][email protected][/email]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 263
v=0
o=root 27427 27427 IN IP4 208.78.69.70
s=session
c=IN IP4 208.78.69.70
t=0 0
m=audio 15184 RTP/AVP 8 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #3 (NAT) to 217.10.79.9:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK5c6b.85c175a6.0;received=217.10.79.9
Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK5c6b.85c175a6.0
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK116f45c0
Via: SIP/2.0/UDP 217.10.67.141:5060;branch=z9hG4bK116f45c0;rport=5060
Record-Route: <sip:217.10.79.9;lr=on;ftag=as2d9c75fe>
Record-Route: <sip:172.20.40.2;lr=on>
Record-Route: <sip:217.10.79.9;lr=on;ftag=as2d9c75fe>
From: "01727395202" <sip:[email protected]>;tag=as2d9c75fe
To: <sip:[email protected]>;tag=as2916f91d
Call-ID: [email][email protected][/email]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 263
v=0
o=root 27427 27427 IN IP4 208.78.69.70
s=session
c=IN IP4 208.78.69.70
t=0 0
m=audio 15184 RTP/AVP 8 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #4 (NAT) to 217.10.79.9:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK5c6b.85c175a6.0;received=217.10.79.9
Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK5c6b.85c175a6.0
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK116f45c0
Via: SIP/2.0/UDP 217.10.67.141:5060;branch=z9hG4bK116f45c0;rport=5060
Record-Route: <sip:217.10.79.9;lr=on;ftag=as2d9c75fe>
Record-Route: <sip:172.20.40.2;lr=on>
Record-Route: <sip:217.10.79.9;lr=on;ftag=as2d9c75fe>
From: "01727395202" <sip:[email protected]>;tag=as2d9c75fe
To: <sip:[email protected]>;tag=as2916f91d
Call-ID: [email][email protected][/email]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 263
v=0
o=root 27427 27427 IN IP4 208.78.69.70
s=session
c=IN IP4 208.78.69.70
t=0 0
m=audio 15184 RTP/AVP 8 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #5 (NAT) to 217.10.79.9:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK5c6b.85c175a6.0;received=217.10.79.9
Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK5c6b.85c175a6.0
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK116f45c0
Via: SIP/2.0/UDP 217.10.67.141:5060;branch=z9hG4bK116f45c0;rport=5060
Record-Route: <sip:217.10.79.9;lr=on;ftag=as2d9c75fe>
Record-Route: <sip:172.20.40.2;lr=on>
Record-Route: <sip:217.10.79.9;lr=on;ftag=as2d9c75fe>
From: "01727395202" <sip:[email protected]>;tag=as2d9c75fe
To: <sip:[email protected]>;tag=as2916f91d
Call-ID: [email][email protected][/email]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 263
v=0
o=root 27427 27427 IN IP4 208.78.69.70
s=session
c=IN IP4 208.78.69.70
t=0 0
m=audio 15184 RTP/AVP 8 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
<--- SIP read from 217.10.79.9:5060 --->
<------------->
Retransmitting #6 (NAT) to 217.10.79.9:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK5c6b.85c175a6.0;received=217.10.79.9
Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK5c6b.85c175a6.0
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK116f45c0
Via: SIP/2.0/UDP 217.10.67.141:5060;branch=z9hG4bK116f45c0;rport=5060
Record-Route: <sip:217.10.79.9;lr=on;ftag=as2d9c75fe>
Record-Route: <sip:172.20.40.2;lr=on>
Record-Route: <sip:217.10.79.9;lr=on;ftag=as2d9c75fe>
From: "01727395202" <sip:[email protected]>;tag=as2d9c75fe
To: <sip:[email protected]>;tag=as2916f91d
Call-ID: [email][email protected][/email]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 263
v=0
o=root 27427 27427 IN IP4 208.78.69.70
s=session
c=IN IP4 208.78.69.70
t=0 0
m=audio 15184 RTP/AVP 8 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Jun 18 08:17:27] WARNING[27452]: [COLOR="Red"]chan_sip.c:1980 retrans_pkt: Maximum retries exceeded on transmission [/COLOR][email protected] for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt.
[Jun 18 08:17:27] [COLOR="Red"]WARNING[27452]: chan_sip.c:2002 retrans_pkt: Hanging up call [/COLOR][email protected] - no reply to our critical packet (see doc/sip-retransmit.txt).
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:[email protected]:5060;transport=udp> for address/port to send to
set_destination: set destination to 192.168.2.102, port 5060
Reliably Transmitting (no NAT) to 192.168.2.102:5060:
BYE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK7eb98f65;rport
From: "01727395202" <sip:[email protected]>;tag=as3bedd72e
To: <sip:[email protected]:5060;transport=udp>;tag=56e523039c097df8
Call-ID: [email protected]
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0Kann jemand helfen ?
Ich nutze Asterisk 1.4.25 hinter einem NAT.
Zuletzt bearbeitet:
ff - - - - 
