[Asterisk-Users] SPA-2000 Dropped calls

Mike Benoit ipso at snappymail.ca
Tue Nov 30 11:31:36 MST 2004 

This sounds similar to the issue I have been dealing with for the past
several months. 

Are the calls just being dropped, or is the SPA rebooting itself? One
way to find out is to setup a syslog server, and configure the SPA to
send all its debug output to that (make sure you increase the debug
level to highest as well). 

Monitor it for a few hours, and see if you get messages like this:

Nov 29 08:20:56 192.168.1.185 System started: ip at 192.168.1.185, reboot
reason:H7372014f
Nov 29 08:21:04 192.168.1.190 System started: ip at 192.168.1.190, reboot
reason:H0
Nov 29 08:21:04 192.168.1.190 System started: ip at 192.168.1.190, reboot
reason:H0
Nov 29 08:21:04 192.168.1.189 System started: ip at 192.168.1.189, reboot
reason:H0
Nov 29 08:21:04 192.168.1.189 System started: ip at 192.168.1.189, reboot
reason:H0

If you do, let me know, as I can probably help you out. 

On Mon, 2004-11-29 at 22:48 -0600, Tim Lewis wrote:
> Been having a problem with my two Sipura 2000's dropping calls from the
> SPA-2000 side. Seems the calls are dropped right before the "Next
> Registration" time. Calls drop about ever 60 minutes or so. I have
> dialed from one port to the other and let it sit. After about 60 minutes
> or so the calls get dropped.
> 
> System details are below
> 
> Asterisk ver. CVS-HEAD-11/27/04-23:42:45
> 
> RHEL 3
> 
> 10/100 LAN
> 
> Static IP address
> 
> SPA-2000 Software Ver. 2.0.11(g)
> SPA-2000 Hardware Ver. 2.0.1(563f)
> 
> sip.conf
> 
> [8445983]
> type=friend
> username=8445983
> secret=mypassword
> nat=0
> context=toll-access
> host=dynamic
> canreinvite=no
> reinvite=no
> allow=ulaw
> ;allow=alaw
> mailbox=5983
> 
> 
> output from sip debug
> 
> Sip read:
> REGISTER sip:192.168.0.5 SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK-b1938413
> From: 8445985 <sip:8445985 at 192.168.0.5>;tag=c864004bd9b6bbbdo0
> To: 8445985 <sip:8445985 at 192.168.0.5>
> Call-ID: 76662903-a6afea65 at 192.168.0.20
> CSeq: 1 REGISTER
> Max-Forwards: 70
> Contact: 8445985 <sip:8445985 at 192.168.0.20:5060>;expires=9999
> User-Agent: Sipura/SPA2000-2.0.11(g)
> Content-Length: 0
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> Supported: x-sipura
> 
> 
> 12 headers, 0 lines
> Using latest request as basis request
> Sending to 192.168.0.20 : 5060 (non-NAT)
> Transmitting (no NAT):
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK-b1938413
> From: 8445985 <sip:8445985 at 192.168.0.5>;tag=c864004bd9b6bbbdo0
> To: 8445985 <sip:8445985 at 192.168.0.5>;tag=as174ef08c
> Call-ID: 76662903-a6afea65 at 192.168.0.20
> CSeq: 1 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:8445985 at 192.168.0.5>
> Content-Length: 0
> 
> 
>  to 192.168.0.20:5060
> Transmitting (no NAT):
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK-b1938413
> From: 8445985 <sip:8445985 at 192.168.0.5>;tag=c864004bd9b6bbbdo0
> To: 8445985 <sip:8445985 at 192.168.0.5>;tag=as174ef08c
> Call-ID: 76662903-a6afea65 at 192.168.0.20
> CSeq: 1 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:8445985 at 192.168.0.5>
> WWW-Authenticate: Digest realm="asterisk", nonce="47af5efb"
> Content-Length: 0
> 
> 
>  to 192.168.0.20:5060
> Scheduling destruction of call '76662903-a6afea65 at 192.168.0.20' in 15000
> ms
> pbx*CLI>
> 
> Sip read:
> REGISTER sip:192.168.0.5 SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.20:5061;branch=z9hG4bK-9723ca95
> From: 8445983 <sip:8445983 at 192.168.0.5>;tag=342babdb37a0856do1
> To: 8445983 <sip:8445983 at 192.168.0.5>
> Call-ID: 5039486b-2415121d at 192.168.0.20
> CSeq: 1 REGISTER
> Max-Forwards: 70
> Contact: 8445983 <sip:8445983 at 192.168.0.20:5061>;expires=3600
> User-Agent: Sipura/SPA2000-2.0.11(g)
> Content-Length: 0
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> Supported: x-sipura
> 
> 
> 12 headers, 0 lines
> Using latest request as basis request
> Sending to 192.168.0.20 : 5061 (non-NAT)
> Transmitting (no NAT):
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 192.168.0.20:5061;branch=z9hG4bK-9723ca95
> From: 8445983 <sip:8445983 at 192.168.0.5>;tag=342babdb37a0856do1
> To: 8445983 <sip:8445983 at 192.168.0.5>;tag=as7ec12bce
> Call-ID: 5039486b-2415121d at 192.168.0.20
> CSeq: 1 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:8445983 at 192.168.0.5>
> Content-Length: 0
> 
> 
>  to 192.168.0.20:5061
> Transmitting (no NAT):
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP 192.168.0.20:5061;branch=z9hG4bK-9723ca95
> From: 8445983 <sip:8445983 at 192.168.0.5>;tag=342babdb37a0856do1
> To: 8445983 <sip:8445983 at 192.168.0.5>;tag=as7ec12bce
> Call-ID: 5039486b-2415121d at 192.168.0.20
> CSeq: 1 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:8445983 at 192.168.0.5>
> WWW-Authenticate: Digest realm="asterisk", nonce="5a8a4fbd"
> Content-Length: 0
> 
> 
>  to 192.168.0.20:5061
> Scheduling destruction of call '5039486b-2415121d at 192.168.0.20' in 15000
> ms
> pbx*CLI>
> 
> Sip read:
> REGISTER sip:192.168.0.5 SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK-46bc8b4
> From: 8445985 <sip:8445985 at 192.168.0.5>;tag=c864004bd9b6bbbdo0
> To: 8445985 <sip:8445985 at 192.168.0.5>
> Call-ID: 76662903-a6afea65 at 192.168.0.20
> CSeq: 2 REGISTER
> Max-Forwards: 70
> Authorization: Digest
> username="8445985",realm="asterisk",nonce="47af5efb",uri="
> sip:8445985 at 192.168.0.5",algorithm=MD5,response="1a025630f2e7b5a94fa227ae79fef7a
> 1"
> Contact: 8445985 <sip:8445985 at 192.168.0.20:5060>;expires=9999
> User-Agent: Sipura/SPA2000-2.0.11(g)
> Content-Length: 0
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> Supported: x-sipura
> 
> 
> 13 headers, 0 lines
> Using latest request as basis request
> Sending to 192.168.0.20 : 5060 (non-NAT)
> Transmitting (no NAT):
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK-46bc8b4
> From: 8445985 <sip:8445985 at 192.168.0.5>;tag=c864004bd9b6bbbdo0
> To: 8445985 <sip:8445985 at 192.168.0.5>;tag=as174ef08c
> Call-ID: 76662903-a6afea65 at 192.168.0.20
> Seq: 2 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:8445985 at 192.168.0.5>
> Content-Length: 0
> 
> 
>  to 192.168.0.20:5060
> Transmitting (no NAT):
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK-46bc8b4
> From: 8445985 <sip:8445985 at 192.168.0.5>;tag=c864004bd9b6bbbdo0
> To: 8445985 <sip:8445985 at 192.168.0.5>;tag=as174ef08c
> Call-ID: 76662903-a6afea65 at 192.168.0.20
> CSeq: 2 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Expires: 3600
> Contact: <sip:8445985 at 192.168.0.20:5060>;expires=3600
> Date: Tue, 30 Nov 2004 04:29:03 GMT
> Content-Length: 0
> 
> 
>  to 192.168.0.20:5060
> Scheduling destruction of call '76662903-a6afea65 at 192.168.0.20' in 15000
> ms
> 
> 
> Sip read:
> REGISTER sip:192.168.0.5 SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.20:5061;branch=z9hG4bK-572e66a
> From: 8445983 <sip:8445983 at 192.168.0.5>;tag=342babdb37a0856do1
> To: 8445983 <sip:8445983 at 192.168.0.5>
> Call-ID: 5039486b-2415121d at 192.168.0.20
> CSeq: 2 REGISTER
> Max-Forwards: 70
> Authorization: Digest
> username="8445983",realm="asterisk",nonce="5a8a4fbd",uri="
> sip:8445983 at 192.168.0.5",algorithm=MD5,response="31e27e17f7a0f434cc636ce16a3e48e
> 9"
> Contact: 8445983 <sip:8445983 at 192.168.0.20:5061>;expires=3600
> User-Agent: Sipura/SPA2000-2.0.11(g)
> Content-Length: 0
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> Supported: x-sipura
> 
> 
> 13 headers, 0 lines
> Using latest request as basis request
> Sending to 192.168.0.20 : 5061 (non-NAT)
> Transmitting (no NAT):
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 192.168.0.20:5061;branch=z9hG4bK-572e66a
> From: 8445983 <sip:8445983 at 192.168.0.5>;tag=342babdb37a0856do1
> To: 8445983 <sip:8445983 at 192.168.0.5>;tag=as7ec12bce
> Call-ID: 5039486b-2415121d at 192.168.0.20
> CSeq: 2 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:8445983 at 192.168.0.5>
> Content-Length: 0
> 
> 
>  to 192.168.0.20:5061
> Transmitting (no NAT):
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.0.20:5061;branch=z9hG4bK-572e66a
> From: 8445983 <sip:8445983 at 192.168.0.5>;tag=342babdb37a0856do1
> To: 8445983 <sip:8445983 at 192.168.0.5>;tag=as7ec12bce
> Call-ID: 5039486b-2415121d at 192.168.0.20
> CSeq: 2 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Expires: 3600
> Contact: <sip:8445983 at 192.168.0.20:5061>;expires=3600
> Date: Tue, 30 Nov 2004 04:29:03 GMT
> Content-Length: 0
> 
> 
>  to 192.168.0.20:5061
> Scheduling destruction of call '5039486b-2415121d at 192.168.0.20' in 15000
> ms
> 11 headers, 2 lines
> Reliably Transmitting:
> NOTIFY sip:8445983 at 192.168.0.20:5061 SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK40a138cf
> From: "asterisk" <sip:asterisk at 192.168.0.5>;tag=as40e64aad
> To: <sip:8445983 at 192.168.0.20:5061>
> Contact: <sip:asterisk at 192.168.0.5>
> Call-ID: 794d086c252c042d7ae8378c22a5dde9 at 192.168.0.5
> CSeq: 102 NOTIFY
> User-Agent: Asterisk PBX
> Event: message-summary
> Content-Type: application/simple-message-summary
> Content-Length: 42
> 
> Messages-Waiting: no
> Voice-Message: 0/1
>  (no NAT) to 192.168.0.20:5061
> Scheduling destruction of call
> '794d086c252c042d7ae8378c22a5dde9 at 192.168.0.5' in                      
> 15000 ms
> pbx*CLI>
> 
> Sip read:
> SIP/2.0 200 OK
> To: <sip:8445983 at 192.168.0.20:5061>;tag=a2f4e6f3d386c535i1
> From: "asterisk" <sip:asterisk at 192.168.0.5>;tag=as40e64aad
> Call-ID: 794d086c252c042d7ae8378c22a5dde9 at 192.168.0.5
> CSeq: 102 NOTIFY
> Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK40a138cf
> Server: Sipura/SPA2000-2.0.11(g)
> Content-Length: 0
> 
> 
> 8 headers, 0 lines
> Destroying call '794d086c252c042d7ae8378c22a5dde9 at 192.168.0.5'
> Destroying call '76662903-a6afea65 at 192.168.0.20'
> Destroying call '5039486b-2415121d at 192.168.0.20'
> pbx*CLI>
> 
> Any ideas would be of GREAT help
> 
> Thanks
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
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-- 
Mike Benoit <ipso at snappymail.ca>
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