[Blink] Blink Digest, Vol 60, Issue 6

Adrian Georgescu ag at ag-projects.com
Mon Oct 27 10:45:11 CET 2014


Screenshots for the logs don't help much. Send the logs exported in text format. Instructions can be found here: 

Sending A Crash Report

http://projects.ag-projects.com/projects/blinkc/wiki/Help_For_Blink_Pro#Sending-A-Crash-Report

Regards,
Adrian


On 21 Oct 2014, at 12:35, Daniel Guevara <dguevara at nxtview.com> wrote:

> I add the console when the blink crash.
> 
> Regards
> 
> Daniel Guevara
> socio-director ingeniería
> 
> 
> 
> d Amsterdam No. 12, PH Col. Condesa México D.F. 06140
> m dguevara at nxtview.com
> t  12090480
> w nxtview.com
> 
> 
> De: blink-request at lists.ag-projects.com
> Para: blink at lists.ag-projects.com
> Enviados: Martes, 21 de Octubre 2014 5:00:01
> Asunto: Blink Digest, Vol 60, Issue 6
> 
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>    1. Re: Bkink pro crash (Adrian Georgescu)
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> 
> ----------------------------------------------------------------------
> 
> Message: 1
> Date: Mon, 20 Oct 2014 18:47:30 -0200
> From: Adrian Georgescu <ag at ag-projects.com>
> To: Blink Support Forum <blink at lists.ag-projects.com>
> Subject: Re: [Blink] Bkink pro crash
> Message-ID: <6BD1D069-9100-4D22-AE6F-DAC2272EED28 at ag-projects.com>
> Content-Type: text/plain; charset="us-ascii"
> 
> Can you send the crash report?
> 
> Adrian
> 
> On 20 Oct 2014, at 18:45, Daniel Guevara <dguevara at clearcom.mx> wrote:
> 
> > Hello , 
> > 
> > I am experimenting Blink crashes since latest version 4.0.1 in MACOS 10.9.4 .
> > To reproduced the problem this is the scenario : 
> > i Have an account to an internal Asterisk V1.8 , when i receive a call from an another internal extension the blink closed without reason.
> > I have no problem in outgoing calls to the asterisk only inbound.
> > This is the trace i had in the sip debug asterisk.
> > 
> > nxtphone*CLI> 
> >   == Using SIP RTP CoS mark 5
> >     -- Executing [1030 at anexos:1] Gosub("SIP/1004-00000261", "std-exten,~~s~~,1(1030,"SIP")") in new stack
> >     -- Executing [~~s~~@std-exten:1] MSet("SIP/1004-00000261", "LOCAL(ext)=1030") in new stack
> >     -- Executing [~~s~~@std-exten:2] MSet("SIP/1004-00000261", "LOCAL(dev)="SIP"") in new stack
> >     -- Executing [~~s~~@std-exten:3] MSet("SIP/1004-00000261", "LOCAL(~~EXTEN~~)=~~s~~") in new stack
> >     -- Executing [~~s~~@std-exten:4] MSet("SIP/1004-00000261", "LOCAL(~~EXTEN~~)=~~s~~") in new stack
> >     -- Executing [~~s~~@std-exten:5] Set("SIP/1004-00000261", "CHANNEL(language)=es") in new stack
> >     -- Executing [~~s~~@std-exten:6] MSet("SIP/1004-00000261", "DYNAMIC_FEATURES=automon") in new stack
> >     -- Executing [~~s~~@std-exten:7] Dial("SIP/1004-00000261", "SIP/1030,20,TrtWw") in new stack
> >   == Using SIP RTP CoS mark 5
> > Audio is at 18978
> > Adding codec 0x100 (g729) to SDP
> > Adding codec 0x8 (alaw) to SDP
> > Adding codec 0x2 (gsm) to SDP
> > Adding codec 0x4 (ulaw) to SDP
> > Adding codec 0x1000 (g722) to SDP
> > Adding non-codec 0x1 (telephone-event) to SDP
> > Reliably Transmitting (no NAT) to 192.168.1.188:55605:
> > INVITE sip:86302491 at 192.168.1.188:55605 SIP/2.0
> > Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK4e269302
> > Max-Forwards: 70
> > From: "1004" <sip:1004 at 192.168.1.3>;tag=as41241883
> > To: <sip:86302491 at 192.168.1.188:55605>
> > Contact: <sip:1004 at 192.168.1.3:5060>
> > Call-ID: 1c48ed916eefbe597874385a268ec938 at 192.168.1.3:5060
> > CSeq: 102 INVITE
> > User-Agent: Asterisk PBX 1.8.15-cert1
> > Date: Mon, 20 Oct 2014 19:49:33 GMT
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> > Supported: replaces, timer
> > Content-Type: application/sdp
> > Content-Length: 382
> > 
> > v=0
> > o=root 225996422 225996422 IN IP4 192.168.1.3
> > s=Asterisk PBX 1.8.15-cert1
> > c=IN IP4 192.168.1.3
> > t=0 0
> > m=audio 18978 RTP/AVP 18 8 3 0 9 101
> > a=rtpmap:18 G729/8000
> > a=fmtp:18 annexb=no
> > a=rtpmap:8 PCMA/8000
> > a=rtpmap:3 GSM/8000
> > a=rtpmap:0 PCMU/8000
> > a=rtpmap:9 G722/8000
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-16
> > a=silenceSupp:off - - - -
> > a=ptime:20
> > a=sendrecv
> > 
> > ---
> >     -- Called SIP/1030
> > Retransmitting #1 (no NAT) to 192.168.1.188:55605:
> > INVITE sip:86302491 at 192.168.1.188:55605 SIP/2.0
> > Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK4e269302
> > Max-Forwards: 70
> > From: "1004" <sip:1004 at 192.168.1.3>;tag=as41241883
> > To: <sip:86302491 at 192.168.1.188:55605>
> > Contact: <sip:1004 at 192.168.1.3:5060>
> > Call-ID: 1c48ed916eefbe597874385a268ec938 at 192.168.1.3:5060
> > CSeq: 102 INVITE
> > User-Agent: Asterisk PBX 1.8.15-cert1
> > Date: Mon, 20 Oct 2014 19:49:33 GMT
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> > Supported: replaces, timer
> > Content-Type: application/sdp
> > Content-Length: 382
> > 
> > v=0
> > o=root 225996422 225996422 IN IP4 192.168.1.3
> > s=Asterisk PBX 1.8.15-cert1
> > c=IN IP4 192.168.1.3
> > t=0 0
> > m=audio 18978 RTP/AVP 18 8 3 0 9 101
> > a=rtpmap:18 G729/8000
> > a=fmtp:18 annexb=no
> > a=rtpmap:8 PCMA/8000
> > a=rtpmap:3 GSM/8000
> > a=rtpmap:0 PCMU/8000
> > a=rtpmap:9 G722/8000
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-16
> > a=silenceSupp:off - - - -
> > a=ptime:20
> > a=sendrecv
> > 
> > ---
> > 
> > <--- SIP read from UDP:192.168.1.188:55605 --->
> > SIP/2.0 500 Internal Server Error
> > Via: SIP/2.0/UDP 192.168.1.3:5060;received=192.168.1.3;branch=z9hG4bK4e269302
> > Call-ID: 1c48ed916eefbe597874385a268ec938 at 192.168.1.3:5060
> > From: "1004" <sip:1004 at 192.168.1.3>;tag=as41241883
> > To: <sip:86302491 at 192.168.1.188>;tag=z9hG4bK4e269302
> > CSeq: 102 INVITE
> > Server: Blink Pro 4.0.1 (MacOSX)
> > Content-Length: 0
> > 
> > <------------->
> > --- (8 headers 0 lines) ---
> >     -- Got SIP response 500 "Internal Server Error" back from 192.168.1.188:55605
> > Transmitting (no NAT) to 192.168.1.188:55605:
> > ACK sip:86302491 at 192.168.1.188:55605 SIP/2.0
> > Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK4e269302
> > Max-Forwards: 70
> > From: "1004" <sip:1004 at 192.168.1.3>;tag=as41241883
> > To: <sip:86302491 at 192.168.1.188:55605>;tag=z9hG4bK4e269302
> > Contact: <sip:1004 at 192.168.1.3:5060>
> > Call-ID: 1c48ed916eefbe597874385a268ec938 at 192.168.1.3:5060
> > CSeq: 102 ACK
> > User-Agent: Asterisk PBX 1.8.15-cert1
> > Content-Length: 0
> > 
> > 
> > ---
> >     -- SIP/1030-00000262 is circuit-busy
> >   == Everyone is busy/congested at this time (1:0/1/0)
> >     -- Executing [~~s~~@std-exten:8] Goto("SIP/1004-00000261", "sw_36_CONGESTION,10") in new stack
> >     -- Goto (std-exten,sw_36_CONGESTION,10)
> >     -- Executing [sw_36_CONGESTION at std-exten:10] VoiceMail("SIP/1004-00000261", "1030 at default,ug(11)") in new stack
> > Really destroying SIP dialog '1c48ed916eefbe597874385a268ec938 at 192.168.1.3:5060' Method: INVITE
> >     -- <SIP/1004-00000261> Playing 'vm-theperson.ulaw' (language 'es')
> >     -- <SIP/1004-00000261> Playing 'digits/1.ulaw' (language 'es')
> >     -- <SIP/1004-00000261> Playing 'digits/0.ulaw' (language 'es')
> >     -- <SIP/1004-00000261> Playing 'digits/3.ulaw' (language 'es')
> >     -- <SIP/1004-00000261> Playing 'digits/0.ulaw' (language 'es')
> >   == Spawn extension (std-exten, sw_36_CONGESTION, 10) exited non-zero on 'SIP/1004-00000261'
> >     -- Executing [h at std-exten:1] Goto("SIP/1004-00000261", "9991") in new stack
> >     -- Goto (std-exten,h,9991)
> >     -- Executing [h at std-exten:9991] Set("SIP/1004-00000261", "~~parentcxt~~=anexos") in new stack
> >     -- Executing [h at std-exten:9992] GotoIf("SIP/1004-00000261", "0?9996") in new stack
> >     -- Executing [h at std-exten:9993] GotoIf("SIP/1004-00000261", "0?9994:9996") in new stack
> >     -- Goto (std-exten,h,9996)
> >     -- Executing [h at std-exten:9996] NoOp("SIP/1004-00000261", "") in new stack
> > 
> > Daniel Guevara
> > _______________________________________________
> > Blink mailing list
> > Blink at lists.ag-projects.com
> > http://lists.ag-projects.com/mailman/listinfo/blink
> 
> --
> Adrian
> 
> 
> 
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> End of Blink Digest, Vol 60, Issue 6
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--
Adrian



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