Hallo,
ich habe ein Problem mit der Faxgeschichte. Ich habe das t38modem drei mal mit einem dahinter geschalteten Hylafax Server auf der Asterisk Maschine laufen. Diese t38modem stellen ihre Verbindung zum Asterisk her.
Zusätzlich habe ich eine Fritzbox an den Asterisk mittels SIP verbunden. An dieser Fritzbox hängt über eine ISDN Telefonanlage ein analoges Faxgerät dran. Soweit so gut.
Möchte ich nun von dem t38modem aus ein Fax an das Faxgerät senden, kommt es beim Aufbau der Verbindung zu eine Fehler.
Im folgenden die SIP Debug Log: (Das ganze sieht aus als würde das t38modem auf einmal nicht mehr auf die SIP Nachrichten reagieren (Transmitting). Die Fritzbox scheint aber T38 zu erkennen und auch ordentlich zu antworten.
Hier noch meine SIP Konfiguration:
Habt ihr eine Idee? Danke schonmal im Voraus.
Gruß
MeisterM
ich habe ein Problem mit der Faxgeschichte. Ich habe das t38modem drei mal mit einem dahinter geschalteten Hylafax Server auf der Asterisk Maschine laufen. Diese t38modem stellen ihre Verbindung zum Asterisk her.
Zusätzlich habe ich eine Fritzbox an den Asterisk mittels SIP verbunden. An dieser Fritzbox hängt über eine ISDN Telefonanlage ein analoges Faxgerät dran. Soweit so gut.
Möchte ich nun von dem t38modem aus ein Fax an das Faxgerät senden, kommt es beim Aufbau der Verbindung zu eine Fehler.
Im folgenden die SIP Debug Log: (Das ganze sieht aus als würde das t38modem auf einmal nicht mehr auf die SIP Nachrichten reagieren (Transmitting). Die Fritzbox scheint aber T38 zu erkennen und auch ordentlich zu antworten.
Code:
<--- SIP read from UDP:127.0.0.1:57977 --->
INVITE sip:[email protected] SIP/2.0
Date: Wed, 13 Jun 2012 08:41:02 GMT
CSeq: 1 INVITE
Via: SIP/2.0/UDP 127.0.0.1:57977;branch=z9hG4bKe8ed502d-a1b3-e111-8e41-000c6ef5a49f;rport
User-Agent: OPAL/2.0
From: "root" <sip:[email protected]>;tag=6cd4362d-a1b3-e111-8e41-000c6ef5a49f
Call-ID: e8c6362d-a1b3-e111-8e41-000c6ef5a49f@telefonanlage
To: <sip:[email protected]>
Contact: <sip:[email protected]:57977;transport=udp>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE,INFO,PING,PUBLISH
Content-Type: application/sdp
Content-Length: 299
Max-Forwards: 70
v=0
o=- 1339576862 1339576862 IN IP4 127.0.0.1
s=Opal SIP Session
c=IN IP4 127.0.0.1
t=0 0
m=audio 5000 RTP/AVP 0 8 101 100
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,32-49
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
m=image 5004 udptl t38
<------------->
[2012-06-13 10:41:02.986] VERBOSE[10414] chan_sip.c: --- (13 headers 13 lines) ---
[2012-06-13 10:41:02.987] VERBOSE[10414] chan_sip.c: Sending to 127.0.0.1:57977 (no NAT)
[2012-06-13 10:41:02.987] VERBOSE[10414] chan_sip.c: Using INVITE request as basis request - e8c6362d-a1b3-e111-8e41-000c6ef5a49f@telefonanlage
[2012-06-13 10:41:02.987] VERBOSE[10414] chan_sip.c: Found peer 'T38modem0' for 'T38modem0' from 127.0.0.1:57977
[2012-06-13 10:41:02.987] VERBOSE[10414] netsock2.c: == Using SIP RTP CoS mark 5
[2012-06-13 10:41:02.987] VERBOSE[10414] chan_sip.c: Found RTP audio format 0
[2012-06-13 10:41:02.987] VERBOSE[10414] chan_sip.c: Found RTP audio format 8
[2012-06-13 10:41:02.987] VERBOSE[10414] chan_sip.c: Found RTP audio format 101
[2012-06-13 10:41:02.987] VERBOSE[10414] chan_sip.c: Found RTP audio format 100
[2012-06-13 10:41:02.987] VERBOSE[10414] chan_sip.c: Found audio description format PCMU for ID 0
[2012-06-13 10:41:02.987] VERBOSE[10414] chan_sip.c: Found audio description format PCMA for ID 8
[2012-06-13 10:41:02.987] VERBOSE[10414] chan_sip.c: Found audio description format telephone-event for ID 101
[2012-06-13 10:41:02.987] VERBOSE[10414] chan_sip.c: Found unknown media description format NSE for ID 100
[2012-06-13 10:41:02.988] VERBOSE[10414] netsock.c: == Using UDPTL CoS mark 5
[2012-06-13 10:41:02.988] VERBOSE[10414] chan_sip.c: Got T.38 offer in SDP in dialog e8c6362d-a1b3-e111-8e41-000c6ef5a49f@telefonanlage
[2012-06-13 10:41:02.988] VERBOSE[10414] chan_sip.c: Capabilities: us - (ulaw|alaw|g729), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
[2012-06-13 10:41:02.988] VERBOSE[10414] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2012-06-13 10:41:02.988] VERBOSE[10414] chan_sip.c: Peer audio RTP is at port 127.0.0.1:5000
[2012-06-13 10:41:02.988] VERBOSE[10414] chan_sip.c: Looking for 20007 in fax-out (domain 127.0.0.1)
[2012-06-13 10:41:02.988] VERBOSE[10414] chan_sip.c: list_route: hop: <sip:[email protected]:57977;transport=udp>
[2012-06-13 10:41:02.988] VERBOSE[10414] chan_sip.c:
<--- Transmitting (no NAT) to 127.0.0.1:57977 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 127.0.0.1:57977;branch=z9hG4bKe8ed502d-a1b3-e111-8e41-000c6ef5a49f;received=127.0.0.1;rport=57977
From: "root" <sip:[email protected]>;tag=6cd4362d-a1b3-e111-8e41-000c6ef5a49f
To: <sip:[email protected]>
Call-ID: e8c6362d-a1b3-e111-8e41-000c6ef5a49f@telefonanlage
CSeq: 1 INVITE
Server: Asterisk PBX 10.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------>
[2012-06-13 10:41:02.996] VERBOSE[2701] pbx.c: -- Executing [20007@fax-out:1] Dial("SIP/T38modem0-00000016", "SIP/20007") in new stack
[2012-06-13 10:41:02.997] VERBOSE[2701] netsock2.c: == Using SIP RTP CoS mark 5
[2012-06-13 10:41:02.998] VERBOSE[2701] chan_sip.c: Audio is at 14616
[2012-06-13 10:41:02.999] VERBOSE[2701] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[2012-06-13 10:41:02.999] VERBOSE[2701] chan_sip.c: Adding codec 100008 (g729) to SDP
[2012-06-13 10:41:02.999] VERBOSE[2701] chan_sip.c: Adding codec 100004 (alaw) to SDP
[2012-06-13 10:41:02.999] VERBOSE[2701] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2012-06-13 10:41:03.000] VERBOSE[2701] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.2.19:5060:
INVITE sip:[email protected];uniq=62C14 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK5a1bce51
Max-Forwards: 70
From: "root" <sip:[email protected]>;tag=as3932d449
To: <sip:[email protected];uniq=62C14>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.4.0
Date: Wed, 13 Jun 2012 08:41:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: "root" <sip:[email protected]>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 331
v=0
o=root 441915051 441915051 IN IP4 192.168.1.15
s=Asterisk PBX 10.4.0
c=IN IP4 192.168.1.15
t=0 0
m=audio 14616 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[2012-06-13 10:41:03.000] VERBOSE[2701] app_dial.c: -- Called SIP/20007
[2012-06-13 10:41:03.043] VERBOSE[10414] chan_sip.c:
<--- SIP read from UDP:192.168.2.19:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK5a1bce51
From: "root" <sip:[email protected]>;tag=as3932d449
To: <sip:[email protected];uniq=62C14>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.05.20 (Feb 27 2012)
Content-Length: 0
<------------->
[2012-06-13 10:41:03.043] VERBOSE[10414] chan_sip.c: --- (8 headers 0 lines) ---
[2012-06-13 10:41:04.302] VERBOSE[10414] chan_sip.c:
<--- SIP read from UDP:192.168.2.19:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK5a1bce51
From: "root" <sip:[email protected]>;tag=as3932d449
To: <sip:[email protected];uniq=62C14>;tag=01E53AFC4324749B
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Contact: <sip:[email protected];uniq=62C14AD77C330A1DB00441795750B>
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.05.20 (Feb 27 2012)
Content-Length: 0
<------------->
[2012-06-13 10:41:04.302] VERBOSE[10414] chan_sip.c: --- (9 headers 0 lines) ---
[2012-06-13 10:41:04.302] VERBOSE[10414] chan_sip.c: list_route: hop: <sip:[email protected];uniq=62C14AD77C330A1DB00441795750B>
[2012-06-13 10:41:04.302] VERBOSE[2701] app_dial.c: -- SIP/20007-00000017 is ringing
[2012-06-13 10:41:04.303] VERBOSE[2701] chan_sip.c:
<--- Transmitting (no NAT) to 127.0.0.1:57977 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 127.0.0.1:57977;branch=z9hG4bKe8ed502d-a1b3-e111-8e41-000c6ef5a49f;received=127.0.0.1;rport=57977
From: "root" <sip:[email protected]>;tag=6cd4362d-a1b3-e111-8e41-000c6ef5a49f
To: <sip:[email protected]>;tag=as0a283501
Call-ID: e8c6362d-a1b3-e111-8e41-000c6ef5a49f@telefonanlage
CSeq: 1 INVITE
Server: Asterisk PBX 10.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------>
[2012-06-13 10:41:05.929] VERBOSE[10414] chan_sip.c:
<--- SIP read from UDP:192.168.2.19:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK5a1bce51
From: "root" <sip:[email protected]>;tag=as3932d449
To: <sip:[email protected];uniq=62C14>;tag=01E53AFC4324749B
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Contact: <sip:[email protected];uniq=62C14AD77C330A1DB00441795750B>
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.05.20 (Feb 27 2012)
Supported: 100rel,replaces
Allow-Events: telephone-event,refer
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH
Content-Type: application/sdp
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 211
v=0
o=user 10589442 10589442 IN IP4 192.168.2.19
s=Asterisk PBX 10.4.0
c=IN IP4 192.168.2.19
t=0 0
m=audio 7078 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=sendrecv
a=rtcp:7079
a=ptime:30
<------------->
[2012-06-13 10:41:05.929] VERBOSE[10414] chan_sip.c: --- (15 headers 11 lines) ---
[2012-06-13 10:41:05.930] VERBOSE[10414] chan_sip.c: Found RTP audio format 0
[2012-06-13 10:41:05.930] VERBOSE[10414] chan_sip.c: Found RTP audio format 8
[2012-06-13 10:41:05.930] VERBOSE[10414] chan_sip.c: Found audio description format PCMU for ID 0
[2012-06-13 10:41:05.930] VERBOSE[10414] chan_sip.c: Found audio description format PCMA for ID 8
[2012-06-13 10:41:05.930] VERBOSE[10414] chan_sip.c: Capabilities: us - (ulaw|alaw|g729), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
[2012-06-13 10:41:05.930] VERBOSE[10414] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
[2012-06-13 10:41:05.930] VERBOSE[10414] chan_sip.c: Peer audio RTP is at port 192.168.2.19:7078
[2012-06-13 10:41:05.930] VERBOSE[10414] chan_sip.c: list_route: hop: <sip:[email protected];uniq=62C14AD77C330A1DB00441795750B>
[2012-06-13 10:41:05.932] VERBOSE[10414] chan_sip.c: set_destination: Parsing <sip:[email protected];uniq=62C14AD77C330A1DB00441795750B> for address/port to send to
[2012-06-13 10:41:05.932] VERBOSE[10414] chan_sip.c: set_destination: set destination to 192.168.2.19:5060
[2012-06-13 10:41:05.932] VERBOSE[10414] chan_sip.c: Transmitting (no NAT) to 192.168.2.19:5060:
ACK sip:[email protected];uniq=62C14AD77C330A1DB00441795750B SIP/2.0
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK509d17f3
Max-Forwards: 70
From: "root" <sip:[email protected]>;tag=as3932d449
To: <sip:[email protected];uniq=62C14>;tag=01E53AFC4324749B
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 10.4.0
Content-Length: 0
---
[2012-06-13 10:41:05.932] VERBOSE[2701] app_dial.c: -- SIP/20007-00000017 answered SIP/T38modem0-00000016
[2012-06-13 10:41:05.933] VERBOSE[2701] chan_sip.c: Audio is at 14782
[2012-06-13 10:41:05.933] VERBOSE[2701] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[2012-06-13 10:41:05.933] VERBOSE[2701] chan_sip.c: Adding codec 100004 (alaw) to SDP
[2012-06-13 10:41:05.933] VERBOSE[2701] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2012-06-13 10:41:05.933] VERBOSE[2701] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 127.0.0.1:57977 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:57977;branch=z9hG4bKe8ed502d-a1b3-e111-8e41-000c6ef5a49f;received=127.0.0.1;rport=57977
From: "root" <sip:[email protected]>;tag=6cd4362d-a1b3-e111-8e41-000c6ef5a49f
To: <sip:[email protected]>;tag=as0a283501
Call-ID: e8c6362d-a1b3-e111-8e41-000c6ef5a49f@telefonanlage
CSeq: 1 INVITE
Server: Asterisk PBX 10.4.0
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: 307
v=0
o=root 1156100772 1156100772 IN IP4 X.X.X.X
s=Asterisk PBX 10.4.0
c=IN IP4 X.X.X.X
t=0 0
m=audio 14782 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=image 0 udptl t38
<------------>
[2012-06-13 10:41:05.933] VERBOSE[2701] rtp_engine.c: -- Locally bridging SIP/T38modem0-00000016 and SIP/20007-00000017
[2012-06-13 10:41:05.960] VERBOSE[10414] chan_sip.c:
<--- SIP read from UDP:192.168.2.19:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.19:5060;rport;branch=z9hG4bK9EB4A477B6D2D716
From: <sip:[email protected];uniq=62C14>;tag=01E53AFC4324749B
To: "root" <sip:[email protected]>;tag=as3932d449
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Contact: <sip:[email protected];uniq=62C14AD77C330A1DB00441795750B>
Max-Forwards: 70
X-Designated-Service: fax/t38
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.05.20 (Feb 27 2012)
Supported: 100rel,replaces
Allow-Events: telephone-event,refer
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH
Content-Type: application/sdp
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 337
v=0
o=user 10589442 10589443 IN IP4 192.168.2.19
s=call
c=IN IP4 192.168.2.19
t=0 0
m=image 7078 udptl t38
a=T38FaxVersion:1
a=T38MaxBitRate:14400
a=T38FaxTranscodingMMR
a=T38FaxTranscodingJBIG
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxUdpEC:t38UDPFEC
a=T38FaxMaxDatagram:512
a=sendrecv
<------------->
[2012-06-13 10:41:05.960] VERBOSE[10414] chan_sip.c: --- (17 headers 15 lines) ---
[2012-06-13 10:41:05.961] VERBOSE[10414] chan_sip.c: Sending to 192.168.2.19:5060 (no NAT)
[2012-06-13 10:41:05.961] VERBOSE[10414] netsock.c: == Using UDPTL CoS mark 5
[2012-06-13 10:41:05.961] VERBOSE[10414] chan_sip.c: Got T.38 offer in SDP in dialog [email protected]:5060
[2012-06-13 10:41:05.962] VERBOSE[10414] chan_sip.c: Capabilities: us - (ulaw|alaw|g729), peer - audio=(nothing)/video=(nothing)/text=(nothing), combined - (nothing)
[2012-06-13 10:41:05.962] VERBOSE[10414] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
[2012-06-13 10:41:05.962] VERBOSE[10414] chan_sip.c: Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
[2012-06-13 10:41:05.963] VERBOSE[10414] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.2.19:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.19:5060;branch=z9hG4bK9EB4A477B6D2D716;received=192.168.2.19;rport=5060
From: <sip:[email protected];uniq=62C14>;tag=01E53AFC4324749B
To: "root" <sip:[email protected]>;tag=as3932d449
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Server: Asterisk PBX 10.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------>
[2012-06-13 10:41:06.033] VERBOSE[10414] chan_sip.c: Retransmitting #1 (no NAT) to 127.0.0.1:57977:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:57977;branch=z9hG4bKe8ed502d-a1b3-e111-8e41-000c6ef5a49f;received=127.0.0.1;rport=57977
From: "root" <sip:[email protected]>;tag=6cd4362d-a1b3-e111-8e41-000c6ef5a49f
To: <sip:[email protected]>;tag=as0a283501
Call-ID: e8c6362d-a1b3-e111-8e41-000c6ef5a49f@telefonanlage
CSeq: 1 INVITE
Server: Asterisk PBX 10.4.0
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: 307
v=0
o=root 1156100772 1156100772 IN IP4 X.X.X.X
s=Asterisk PBX 10.4.0
c=IN IP4 X.X.X.X
t=0 0
m=audio 14782 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=image 0 udptl t38
---
[2012-06-13 10:41:06.233] VERBOSE[10414] chan_sip.c: Retransmitting #2 (no NAT) to 127.0.0.1:57977:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:57977;branch=z9hG4bKe8ed502d-a1b3-e111-8e41-000c6ef5a49f;received=127.0.0.1;rport=57977
From: "root" <sip:[email protected]>;tag=6cd4362d-a1b3-e111-8e41-000c6ef5a49f
To: <sip:[email protected]>;tag=as0a283501
Call-ID: e8c6362d-a1b3-e111-8e41-000c6ef5a49f@telefonanlage
CSeq: 1 INVITE
Server: Asterisk PBX 10.4.0
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: 307
v=0
o=root 1156100772 1156100772 IN IP4 X.X.X.X
s=Asterisk PBX 10.4.0
c=IN IP4 X.X.X.X
t=0 0
m=audio 14782 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=image 0 udptl t38
---
[2012-06-13 10:41:06.633] VERBOSE[10414] chan_sip.c: Retransmitting #3 (no NAT) to 127.0.0.1:57977:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:57977;branch=z9hG4bKe8ed502d-a1b3-e111-8e41-000c6ef5a49f;received=127.0.0.1;rport=57977
From: "root" <sip:[email protected]>;tag=6cd4362d-a1b3-e111-8e41-000c6ef5a49f
To: <sip:[email protected]>;tag=as0a283501
Call-ID: e8c6362d-a1b3-e111-8e41-000c6ef5a49f@telefonanlage
CSeq: 1 INVITE
Server: Asterisk PBX 10.4.0
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: 307
v=0
o=root 1156100772 1156100772 IN IP4 X.X.X.X
s=Asterisk PBX 10.4.0
c=IN IP4 X.X.X.X
t=0 0
m=audio 14782 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=image 0 udptl t38
---
[2012-06-13 10:41:07.433] VERBOSE[10414] chan_sip.c: Retransmitting #4 (no NAT) to 127.0.0.1:57977:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:57977;branch=z9hG4bKe8ed502d-a1b3-e111-8e41-000c6ef5a49f;received=127.0.0.1;rport=57977
From: "root" <sip:[email protected]>;tag=6cd4362d-a1b3-e111-8e41-000c6ef5a49f
To: <sip:[email protected]>;tag=as0a283501
Call-ID: e8c6362d-a1b3-e111-8e41-000c6ef5a49f@telefonanlage
CSeq: 1 INVITE
Server: Asterisk PBX 10.4.0
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: 307
v=0
o=root 1156100772 1156100772 IN IP4 X.X.X.X
s=Asterisk PBX 10.4.0
c=IN IP4 X.X.X.X
t=0 0
m=audio 14782 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=image 0 udptl t38
---
[2012-06-13 10:41:08.496] VERBOSE[10414] chan_sip.c: Reliably Transmitting (no NAT) to 127.0.0.1:6062:
OPTIONS sip:127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK01ea1b29
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as562f4d6d
To: <sip:127.0.0.1>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.4.0
Date: Wed, 13 Jun 2012 08:41:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
[2012-06-13 10:41:08.500] VERBOSE[10414] chan_sip.c:
<--- SIP read from UDP:127.0.0.1:6062 --->
SIP/2.0 200 OK
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK01ea1b29;received=127.0.0.1
From: "asterisk" <sip:[email protected]>;tag=as562f4d6d
Call-ID: [email protected]:5060
To: <sip:127.0.0.1>
Contact: <sip:127.0.0.1:6062;transport=udp>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE,INFO,PING,PUBLISH
Content-Length: 0
<------------->
[2012-06-13 10:41:08.500] VERBOSE[10414] chan_sip.c: --- (9 headers 0 lines) ---
[2012-06-13 10:41:08.500] VERBOSE[10414] chan_sip.c: Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
[2012-06-13 10:41:08.600] VERBOSE[10414] chan_sip.c: Reliably Transmitting (no NAT) to 127.0.0.1:6061:
OPTIONS sip:127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK15af476c
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as6c20349b
To: <sip:127.0.0.1>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.4.0
Date: Wed, 13 Jun 2012 08:41:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
[2012-06-13 10:41:08.600] VERBOSE[10414] chan_sip.c: Reliably Transmitting (no NAT) to 127.0.0.1:6060:
OPTIONS sip:127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK6f31e81b
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as43fe7d9e
To: <sip:127.0.0.1>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.4.0
Date: Wed, 13 Jun 2012 08:41:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
[2012-06-13 10:41:08.606] VERBOSE[10414] chan_sip.c:
<--- SIP read from UDP:127.0.0.1:6060 --->
SIP/2.0 200 OK
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK6f31e81b;received=127.0.0.1
From: "asterisk" <sip:[email protected]>;tag=as43fe7d9e
Call-ID: [email protected]:5060
To: <sip:127.0.0.1>
Contact: <sip:127.0.0.1:6060;transport=udp>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE,INFO,PING,PUBLISH
Content-Length: 0
<------------->
[2012-06-13 10:41:08.606] VERBOSE[10414] chan_sip.c: --- (9 headers 0 lines) ---
[2012-06-13 10:41:08.607] VERBOSE[10414] chan_sip.c: Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
[2012-06-13 10:41:08.608] VERBOSE[10414] chan_sip.c:
<--- SIP read from UDP:127.0.0.1:6061 --->
SIP/2.0 200 OK
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK15af476c;received=127.0.0.1
From: "asterisk" <sip:[email protected]>;tag=as6c20349b
Call-ID: [email protected]:5060
To: <sip:127.0.0.1>
Contact: <sip:127.0.0.1:6061;transport=udp>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE,INFO,PING,PUBLISH
Content-Length: 0
<------------->
[2012-06-13 10:41:08.608] VERBOSE[10414] chan_sip.c: --- (9 headers 0 lines) ---
[2012-06-13 10:41:08.608] VERBOSE[10414] chan_sip.c: Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
[2012-06-13 10:41:09.033] VERBOSE[10414] chan_sip.c: Retransmitting #5 (no NAT) to 127.0.0.1:57977:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:57977;branch=z9hG4bKe8ed502d-a1b3-e111-8e41-000c6ef5a49f;received=127.0.0.1;rport=57977
From: "root" <sip:[email protected]>;tag=6cd4362d-a1b3-e111-8e41-000c6ef5a49f
To: <sip:[email protected]>;tag=as0a283501
Call-ID: e8c6362d-a1b3-e111-8e41-000c6ef5a49f@telefonanlage
CSeq: 1 INVITE
Server: Asterisk PBX 10.4.0
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: 307
v=0
o=root 1156100772 1156100772 IN IP4 X.X.X.X
s=Asterisk PBX 10.4.0
c=IN IP4 X.X.X.X
t=0 0
m=audio 14782 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=image 0 udptl t38
---
[2012-06-13 10:41:10.963] VERBOSE[10414] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 192.168.2.19:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.2.19:5060;branch=z9hG4bK9EB4A477B6D2D716;received=192.168.2.19;rport=5060
From: <sip:[email protected];uniq=62C14>;tag=01E53AFC4324749B
To: "root" <sip:[email protected]>;tag=as3932d449
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Server: Asterisk PBX 10.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
<------------>
[2012-06-13 10:41:10.981] VERBOSE[10414] chan_sip.c:
<--- SIP read from UDP:192.168.2.19:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.19:5060;rport;branch=z9hG4bK9EB4A477B6D2D716
From: <sip:[email protected];uniq=62C14>;tag=01E53AFC4324749B
To: "root" <sip:[email protected]>;tag=as3932d449
Call-ID: [email protected]:5060
CSeq: 103 ACK
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.05.20 (Feb 27 2012)
Content-Length: 0
<------------->
[2012-06-13 10:41:10.982] VERBOSE[10414] chan_sip.c: --- (8 headers 0 lines) ---
[2012-06-13 10:41:11.728] VERBOSE[10414] chan_sip.c: Really destroying SIP dialog '[email protected]' Method: REGISTER
[2012-06-13 10:41:12.233] VERBOSE[10414] chan_sip.c: Retransmitting #6 (no NAT) to 127.0.0.1:57977:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:57977;branch=z9hG4bKe8ed502d-a1b3-e111-8e41-000c6ef5a49f;received=127.0.0.1;rport=57977
From: "root" <sip:[email protected]>;tag=6cd4362d-a1b3-e111-8e41-000c6ef5a49f
To: <sip:[email protected]>;tag=as0a283501
Call-ID: e8c6362d-a1b3-e111-8e41-000c6ef5a49f@telefonanlage
CSeq: 1 INVITE
Server: Asterisk PBX 10.4.0
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: 307
v=0
o=root 1156100772 1156100772 IN IP4 X.X.X.X
s=Asterisk PBX 10.4.0
c=IN IP4 X.X.X.X
t=0 0
m=audio 14782 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=image 0 udptl t38
---
[2012-06-13 10:41:12.332] WARNING[10414] chan_sip.c: Retransmission timeout reached on transmission e8c6362d-a1b3-e111-8e41-000c6ef5a49f@telefonanlage for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response
[2012-06-13 10:41:12.333] WARNING[10414] chan_sip.c: Hanging up call e8c6362d-a1b3-e111-8e41-000c6ef5a49f@telefonanlage - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
[2012-06-13 10:41:12.333] VERBOSE[2701] chan_sip.c: Scheduling destruction of SIP dialog '[email protected]:5060' in 32000 ms (Method: ACK)
[2012-06-13 10:41:12.334] VERBOSE[2701] chan_sip.c: set_destination: Parsing <sip:[email protected];uniq=62C14AD77C330A1DB00441795750B> for address/port to send to
[2012-06-13 10:41:12.334] VERBOSE[2701] chan_sip.c: set_destination: set destination to 192.168.2.19:5060
[2012-06-13 10:41:12.334] VERBOSE[2701] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.2.19:5060:
BYE sip:[email protected];uniq=62C14AD77C330A1DB00441795750B SIP/2.0
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK0f06a92d;rport
Max-Forwards: 70
From: "root" <sip:[email protected]>;tag=as3932d449
To: <sip:[email protected];uniq=62C14>;tag=01E53AFC4324749B
Call-ID: [email protected]:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 10.4.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
[2012-06-13 10:41:12.335] VERBOSE[2701] pbx.c: == Spawn extension (fax-out, 20007, 1) exited non-zero on 'SIP/T38modem0-00000016'
[2012-06-13 10:41:12.335] VERBOSE[2701] chan_sip.c: Scheduling destruction of SIP dialog 'e8c6362d-a1b3-e111-8e41-000c6ef5a49f@telefonanlage' in 6400 ms (Method: INVITE)
[2012-06-13 10:41:12.335] VERBOSE[2701] chan_sip.c: set_destination: Parsing <sip:[email protected]:57977;transport=udp> for address/port to send to
[2012-06-13 10:41:12.336] VERBOSE[2701] chan_sip.c: set_destination: set destination to 127.0.0.1:57977
[2012-06-13 10:41:12.336] VERBOSE[2701] chan_sip.c: Reliably Transmitting (no NAT) to 127.0.0.1:57977:
BYE sip:[email protected]:57977;transport=udp SIP/2.0
Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK3968c35c;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as0a283501
To: "root" <sip:[email protected]>;tag=6cd4362d-a1b3-e111-8e41-000c6ef5a49f
Call-ID: e8c6362d-a1b3-e111-8e41-000c6ef5a49f@telefonanlage
CSeq: 102 BYE
User-Agent: Asterisk PBX 10.4.0
X-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
Content-Length: 0
---
[2012-06-13 10:41:12.340] VERBOSE[10414] chan_sip.c:
<--- SIP read from UDP:127.0.0.1:57977 --->
SIP/2.0 481 Call Leg/Transaction Does Not Exist
CSeq: 102 BYE
Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK3968c35c;rport=5060;received=127.0.0.1
From: <sip:[email protected]>;tag=as0a283501
Call-ID: e8c6362d-a1b3-e111-8e41-000c6ef5a49f@telefonanlage
To: "root" <sip:[email protected]>;tag=6cd4362d-a1b3-e111-8e41-000c6ef5a49f
Contact: <sip:[email protected]:57977;transport=udp>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE,INFO,PING,PUBLISH
Content-Length: 0
<------------->
[2012-06-13 10:41:12.340] VERBOSE[10414] chan_sip.c: --- (9 headers 0 lines) ---
[2012-06-13 10:41:12.340] VERBOSE[10414] chan_sip.c: Really destroying SIP dialog 'e8c6362d-a1b3-e111-8e41-000c6ef5a49f@telefonanlage' Method: INVITE
[2012-06-13 10:41:12.369] VERBOSE[10414] chan_sip.c:
<--- SIP read from UDP:192.168.2.19:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK0f06a92d;rport=5060
From: "root" <sip:[email protected]>;tag=as3932d449
To: <sip:[email protected];uniq=62C14>;tag=01E53AFC4324749B
Call-ID: [email protected]:5060
CSeq: 103 BYE
X-RTP-Stat: CS=0;PS=83;ES=214;OS=19920;SP=0/0;SO=0;QS=-;PR=0;ER=322;OR=0;CR=0;SR=0;QR=-;PL=0,0;BL=0;LS=0;RB=0/0;SB=0/0;EN=PCMU,FAX;DE=;JI=0,0;DL=0,0,0;IP=192.168.2.19:7078,192.168.1.15:14616
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.05.20 (Feb 27 2012)
Supported: 100rel,replaces
Allow-Events: telephone-event,refer
Content-Length: 0
<------------->
[2012-06-13 10:41:12.369] VERBOSE[10414] chan_sip.c: --- (11 headers 0 lines) ---
[2012-06-13 10:41:12.369] VERBOSE[10414] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived
[2012-06-13 10:41:12.369] VERBOSE[10414] chan_sip.c: Really destroying SIP dialog '[email protected]:5060' Method: ACK
Hier noch meine SIP Konfiguration:
Code:
[general]
context = default
faxdetect = yes ; Erkennt automatisch Faxübertragungen und springt in fax Context
t38pt_udptl = yes ; Aktiviert T38 für Faxübertragung
bindport=5060 ; Bindet den Asterisk Server auf Port 5060
srvlookup=yes ; Löst Hostnamen auf
disallow=all ; Codecs abschalten
allow = g729 ; Codec G.729 erlauben
allow = ulaw ; Codec ulaw erlauben
allow = alaw ; Codec alaw erlauben
language=de ; Standardsprache deutsch
trustrpid = yes ; Soll der Remote-Party-ID vertraut werden
sendrpid = yes ; P-Asserted-Identitiy header benutzen
nat=no ; Nat Standardmäßig aus
externip=X.X.X.X ; Externe Adresse
localnet=192.168.1.0/24 ; Lokales Netz
localnet=192.168.2.0/24
localnet=192.168.3.0/24
localnet=192.168.4.0/24
call-limit=3 ; Anruferlimit auf 3
directmedia=nonat ; Verhinert das Direkte Verbinden außerhalb des lokalen Netzes
rtcachefriends=yes ; Cacht SIP User aus der Datenbank
[globals]
DYNAMIC_FEATURES=testfeature
[t38modem-options](!)
type = friend
host = 127.0.0.1
context = fax-out
disallow = all
allow = g729
allow = ulaw
allow = alaw
t38pt_udptl = yes
dtmfmode = rfc2833
qualify = yes
nat = no
directmedia=no
[T38modem0](t38modem-options)
port = 6060
[T38modem1](t38modem-options)
port = 6061
[T38modem2](t38modem-options)
port = 6062
Habt ihr eine Idee? Danke schonmal im Voraus.
Gruß
MeisterM