[gelöst] Packet Bridging (re-invite) mit Patton M-ATA

Tweety · 14 Jun 2007

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

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...!

Tweety · 14 Jun 2007

So, glaube der Fehler ist gefunden. Die M-ATAs mögen wohl keine re-invites wenn bei der Callerid kein Name gesetzt ist.

Wird mit "Set(CALLERID(name)=Fax)" der Name gesetzt, geht der re-invite durch. :gruebel: :blonk:
Was meint ihr? Ist das ein Bug Report wert?

So.. und jetzt: gute Nacht

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[gelöst] Packet Bridging (re-invite) mit Patton M-ATA

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