Hallo Leute,
ich bin momentan am Testen mit 2 M-ATAs und einer Smartnode 4638. Dazwischen steht ein Asterisk in Version 1.2.18.
Leider funktioniert bei mir kein re-invite. Ausgehend funktioniert es, jedoch bekomme ich eingehend von den ATAs ein "482 Loop Detected" zurück. Auf der Smartnode läuft Smartware Version 3.21, auf den ATAs 0412.
Leider ist ein re-invite für mich unumgänglich, da die ATAs Faxe übertragen sollen. Das Faxen funktioniert ausgehend nach etwas spielen mit dem Jitter Buffer und Input/Output Gain absolut einwandfrei. Im Log eines IAXModem mit Hylafax auf der anderen Seite tauchen inzwischen keine HDLC Fehler mehr auf. Wenn jedoch kein re-invite stattfindet sieht das ganze schon wieder ganz anders aus.
Ich kann wirklich absolut keinen Fehler im Trace sehen und kann auch keine Option auf den ATAs finden. Das Packet Bridging mit SNOMs funktioniert ohne Probleme.
Hier mal ein SIP Trace
Schon jetzt vielen Dank für eure Anregungen...!
ich bin momentan am Testen mit 2 M-ATAs und einer Smartnode 4638. Dazwischen steht ein Asterisk in Version 1.2.18.
Leider funktioniert bei mir kein re-invite. Ausgehend funktioniert es, jedoch bekomme ich eingehend von den ATAs ein "482 Loop Detected" zurück. Auf der Smartnode läuft Smartware Version 3.21, auf den ATAs 0412.
Leider ist ein re-invite für mich unumgänglich, da die ATAs Faxe übertragen sollen. Das Faxen funktioniert ausgehend nach etwas spielen mit dem Jitter Buffer und Input/Output Gain absolut einwandfrei. Im Log eines IAXModem mit Hylafax auf der anderen Seite tauchen inzwischen keine HDLC Fehler mehr auf. Wenn jedoch kein re-invite stattfindet sieht das ganze schon wieder ganz anders aus.
Ich kann wirklich absolut keinen Fehler im Trace sehen und kann auch keine Option auf den ATAs finden. Das Packet Bridging mit SNOMs funktioniert ohne Probleme.
Hier mal ein SIP Trace
Code:
=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2007.06.13 23:27:29 =~=~=~=~=~=~=~=~=~=~=~=
Destroying call '[email protected]'
SRV-AsteriskHB*CLI>
<-- SIP read from 192.168.0.169:5060:
BYE sip:[email protected]:5060 SIP/2.0
From: sip:[email protected];tag=iX3f2-F8Yo*
To: 06361994030<sip:[email protected]>;tag=as7f253838
Call-ID: [email protected]
CSeq: 96 BYE
Via: SIP/2.0/UDP 192.168.0.169:5060;branch=z9hG4bKiG0f2-VK3vF*mg0
Contact: sip:[email protected]:5060
Max-Forwards: 70
User-Agent: Patton Smartlink MATA <4.01.001 OE EN MA (0412)><00a0ba01a787>
Content-Length: 0
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 192.168.0.169:5060:
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 192.168.0.169:5060;branch=z9hG4bKiG0f2-VK3vF*mg0;received=192.168.0.169
From: sip:[email protected];tag=iX3f2-F8Yo*
To: 06361994030<sip:[email protected]>;tag=as7f253838
Call-ID: [email protected]
CSeq: 96 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
SRV-AsteriskHB*CLI>
SRV-AsteriskHB*CLI>
SRV-AsteriskHB*CLI>
SRV-AsteriskHB*CLI>
-- Executing Dial("SIP/pattonHB-082714a0", "SIP/ata-h3") in new stack
SRV-AsteriskHB*CLI>
We're at 192.168.0.2 port 18280
SRV-AsteriskHB*CLI>
Adding codec 0x8 (alaw) to SDP
SRV-AsteriskHB*CLI>
Adding non-codec 0x1 (telephone-event) to SDP
SRV-AsteriskHB*CLI>
13 headers, 10 lines
SRV-AsteriskHB*CLI>
Reliably Transmitting (no NAT) to 192.168.0.169:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK191d1c3c;rport
From: "06361994030" <sip:[email protected]>;tag=as25bba62b
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 13 Jun 2007 21:27:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 212
v=0
o=root 4194 4194 IN IP4 192.168.0.2
s=session
c=IN IP4 192.168.0.2
t=0 0
m=audio 18280 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Called ata-h3
SRV-AsteriskHB*CLI>
<-- SIP read from 192.168.0.169:5060:
SIP/2.0 100 Trying
From: 06361994030<sip:[email protected]>;tag=as25bba62b
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK191d1c3c
Content-Length: 0
--- (7 headers 0 lines) ---
SRV-AsteriskHB*CLI>
<-- SIP read from 192.168.0.169:5060:
SIP/2.0 180 Ringing
From: 06361994030<sip:[email protected]>;tag=as25bba62b
To: sip:[email protected];tag=iX3f2-E3sDQ
Call-ID: [email protected]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK191d1c3c
Contact: sip:[email protected]:5060
User-Agent: Patton Smartlink MATA <4.01.001 OE EN MA (0412)><00a0ba01a787>
Content-Length: 0
--- (9 headers 0 lines) ---
SRV-AsteriskHB*CLI>
-- SIP/ata-h3-08277798 is ringing
SRV-AsteriskHB*CLI>
12 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.0.169:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK46b202a8;rport
From: "asterisk" <sip:[email protected]>;tag=as14743271
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 13 Jun 2007 21:27:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
SRV-AsteriskHB*CLI>
<-- SIP read from 192.168.0.169:5060:
SIP/2.0 200 OK
From: asterisk<sip:[email protected]>;tag=as14743271
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK46b202a8
Allow: INVITE,BYE,CANCEL,OPTIONS,PRACK,NOTIFY,UPDATE,REFER
Supported: timer,replaces
User-Agent: Patton Smartlink MATA <4.01.001 OE EN MA (0412)><00a0ba01a787>
Accept: application/sdp
Content-Length: 0
--- (11 headers 0 lines) ---
Destroying call '[email protected]'
SRV-AsteriskHB*CLI>
<-- SIP read from 192.168.0.169:5060:
SIP/2.0 200 OK
From: 06361994030<sip:[email protected]>;tag=as25bba62b
To: sip:[email protected];tag=iX3f2-E3sDQ
Call-ID: [email protected]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK191d1c3c
Contact: sip:[email protected]:5060
User-Agent: Patton Smartlink MATA <4.01.001 OE EN MA (0412)><00a0ba01a787>
Allow: INVITE,BYE,CANCEL,OPTIONS,PRACK,NOTIFY,UPDATE,REFER
Supported: timer,replaces
Content-Type: application/sdp
Content-Length: 212
v=0
o=ata-h3 17768 1 IN IP4 192.168.0.169
s=-
c=IN IP4 192.168.0.169
t=0 0
m=audio 8000 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
--- (12 headers 10 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.169:8000
Found description format PCMA
Found description format telephone-event
Found description format telephone-event
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:[email protected]:5060>
set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
set_destination: set destination to 192.168.0.169, port 5060
Transmitting (no NAT) to 192.168.0.169:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK4de0faa1;rport
From: "06361994030" <sip:[email protected]>;tag=as25bba62b
To: <sip:[email protected]:5060>;tag=iX3f2-E3sDQ
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
SRV-AsteriskHB*CLI>
-- SIP/ata-h3-08277798 answered SIP/pattonHB-082714a0
-- Attempting native bridge of SIP/pattonHB-082714a0 and SIP/ata-h3-08277798
set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
set_destination: set destination to 192.168.0.169, port 5060
We're at 192.168.0.2 port 18280
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 192.168.0.169:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK1af9bf57;rport
From: "06361994030" <sip:[email protected]>;tag=as25bba62b
To: <sip:[email protected]:5060>;tag=iX3f2-E3sDQ
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 215
v=0
o=root 4194 4195 IN IP4 192.168.0.250
s=session
c=IN IP4 192.168.0.250
t=0 0
m=audio 5104 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
SRV-AsteriskHB*CLI>
Retransmitting #1 (no NAT) to 192.168.0.169:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK1af9bf57;rport
From: "06361994030" <sip:[email protected]>;tag=as25bba62b
To: <sip:[email protected]:5060>;tag=iX3f2-E3sDQ
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 215
v=0
o=root 4194 4195 IN IP4 192.168.0.250
s=session
c=IN IP4 192.168.0.250
t=0 0
m=audio 5104 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
SRV-AsteriskHB*CLI>
<-- SIP read from 192.168.0.169:5060:
SIP/2.0 100 Trying
From: 06361994030<sip:[email protected]>;tag=as25bba62b
To: sip:[email protected];tag=iX3f2-E3sDQ
Call-ID: [email protected]
CSeq: 103 INVITE
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK1af9bf57
Content-Length: 0
--- (7 headers 0 lines) ---
SRV-AsteriskHB*CLI>
<-- SIP read from 192.168.0.169:5060:
SIP/2.0 200 OK
From: 06361994030<sip:[email protected]>;tag=as25bba62b
To: sip:[email protected];tag=iX3f2-E3sDQ
Call-ID: [email protected]
CSeq: 103 INVITE
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK1af9bf57
Contact: sip:[email protected]:5060
User-Agent: Patton Smartlink MATA <4.01.001 OE EN MA (0412)><00a0ba01a787>
Allow: INVITE,BYE,CANCEL,OPTIONS,PRACK,NOTIFY,UPDATE,REFER
Supported: timer,replaces
Content-Type: application/sdp
Content-Length: 212
v=0
o=ata-h3 17768 2 IN IP4 192.168.0.169
s=-
c=IN IP4 192.168.0.169
t=0 0
m=audio 8000 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
--- (12 headers 10 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.169:8000
Found description format PCMA
Found description format telephone-event
Found description format telephone-event
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
set_destination: set destination to 192.168.0.169, port 5060
Transmitting (no NAT) to 192.168.0.169:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK370602e2;rport
From: "06361994030" <sip:[email protected]>;tag=as25bba62b
To: <sip:[email protected]:5060>;tag=iX3f2-E3sDQ
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
SRV-AsteriskHB*CLI>
<-- SIP read from 192.168.0.169:5060:
SIP/2.0 100 Trying
From: 06361994030<sip:[email protected]>;tag=as25bba62b
To: sip:[email protected];tag=iX3f2-E3sDQ
Call-ID: [email protected]
CSeq: 103 INVITE
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK1af9bf57
Content-Length: 0
--- (7 headers 0 lines) ---
SRV-AsteriskHB*CLI>
<-- SIP read from 192.168.0.169:5060:
SIP/2.0 482 Loop Detected
From: 06361994030<sip:[email protected]>;tag=as25bba62b
To: sip:[email protected];tag=iX3f2-E3sDQ
Call-ID: [email protected]
CSeq: 103 INVITE
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK1af9bf57
User-Agent: Patton Smartlink MATA <4.01.001 OE EN MA (0412)><00a0ba01a787>
Content-Length: 0
--- (8 headers 0 lines) ---
-- Got SIP response 482 "Loop Detected" back from 192.168.0.169
set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
set_destination: set destination to 192.168.0.169, port 5060
Transmitting (no NAT) to 192.168.0.169:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK370602e2;rport
From: "06361994030" <sip:[email protected]>;tag=as25bba62b
To: <sip:[email protected]:5060>;tag=iX3f2-E3sDQ
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
SRV-AsteriskHB*CLI>
== Spawn extension (Extern, 602717, 1) exited non-zero on 'SIP/pattonHB-082714a0'
SRV-AsteriskHB*CLI>
Destroying call '[email protected]'
SRV-AsteriskHB*CLI> sip no debug
SRV-AsteriskHB*CLI>
SIP Debugging Disabled
SRV-AsteriskHB*CLI>
Schon jetzt vielen Dank für eure Anregungen...!