Mehrere GXP-2000´s am Asterisk nach unbestimmter Zeit nicht mehr anrufbar.

wildzero

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Hallo, ich habe hier vier GXP-2000 an einem 1.2.7.1 Asterisk am laufen. Soweit geht alles. Nur weigern sich die GXP´s nach einer unbestimmten Zeit Anrufe anzunehemen wie folgt:


Code:
Scheduling destruction of call '[email protected]' in 32000 ms
accessvoip*CLI>
<-- SIP read from 192.168.202.121:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.202.3:5060;branch=z9hG4bK57bea36b;rport
From: "J. Fritz" <sip:[email protected]>;tag=as31487c93
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Grandstream GXP2000 1.1.0.13
Content-Length: 0
 

--- (8 headers 0 lines)---
accessvoip*CLI>
<-- SIP read from 192.168.202.121:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.202.3:5060;branch=z9hG4bK57bea36b;rport
From: "J. Fritz" <sip:[email protected]>;tag=as31487c93
To: <sip:[email protected]:5060>;tag=6f69876048d1f00d
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Grandstream GXP2000 1.1.0.13
Contact: <sip:[email protected]:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0
 

--- (10 headers 0 lines)---
Retransmitting #1 (no NAT) to 192.168.202.121:5060:
CANCEL sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.202.3:5060;branch=z9hG4bK57bea36b;rport
From: "J. Fritz" <sip:[email protected]>;tag=as31487c93
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 CANCEL
User-Agent: Asterisk
Max-Forwards: 70
Content-Length: 0
 

---
accessvoip*CLI>
<-- SIP read from 192.168.202.121:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.202.3:5060;branch=z9hG4bK57bea36b;rport
From: "J. Fritz" <sip:[email protected]>;tag=as31487c93
To: <sip:[email protected]:5060>;tag=6f69876048d1f00d
Call-ID: [email protected]
CSeq: 102 CANCEL
User-Agent: Grandstream GXP2000 1.1.0.13
Contact: <sip:[email protected]:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Supported: replaces, timer
Content-Length: 0
 

--- (11 headers 0 lines)---
accessvoip*CLI>
<-- SIP read from 192.168.202.121:5060:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 192.168.202.3:5060;branch=z9hG4bK57bea36b;rport
From: "J. Fritz" <sip:[email protected]>;tag=as31487c93
To: <sip:[email protected]:5060>;tag=6f69876048d1f00d
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Grandstream GXP2000 1.1.0.13
Content-Length: 0
 

--- (8 headers 0 lines)---
Transmitting (no NAT) to 192.168.202.121:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.202.3:5060;branch=z9hG4bK57bea36b;rport
From: "J. Fritz" <sip:[email protected]>;tag=as31487c93
To: <sip:[email protected]:5060>;tag=6f69876048d1f00d
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk
Max-Forwards: 70
Content-Length: 0
 

---
Destroying call '[email protected]'
accessvoip*CLI>
<-- SIP read from 192.168.202.121:5060:
SIP/2.0 481 No Such Call
Via: SIP/2.0/UDP 192.168.202.3:5060;branch=z9hG4bK57bea36b;rport
From: "J. Fritz" <sip:[email protected]>;tag=as31487c93
To: <sip:[email protected]:5060>;tag=933b0a9ad858a725
Call-ID: [email protected]
CSeq: 102 CANCEL
User-Agent: Grandstream GXP2000 1.1.0.13
Content-Length: 0

Neustart des GXP-2000: Weiterhin gleicher Fehler

Neustart Asterisk per: asterisk -rx'restart now': Behebt den Fehler wieder für ca. ein paar Stunden / einen Tag ca. und die GXP´s nehmen ganz braf Ihre Anrufe an. Dann wieder die gleiche Fehlermeldung nach einer unbestimmten Zeit bzw. ubnbestimmter Zahl an Anrufen.

Vorallem auch: "SIP/2.0 481 No Such Call"

Code:
481 Call leg/transaction does not exist:

Indicates the server is ignoring the request because it was either a BYE for which there was no matching legID or a CANCEL for which there was no
matching transaction
sagt mir so ziemlich nix.

Für jeden Tipp dankbar was hier los sein könnte.
 
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