[Blink] Blink Digest, Vol 60, Issue 6

Tue Oct 21 16:35:13 CEST 2014

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 

----- Mensaje original -----

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 

Send Blink mailing list submissions to 
blink at lists.ag-projects.com 

To subscribe or unsubscribe via the World Wide Web, visit 
http://lists.ag-projects.com/mailman/listinfo/blink 
or, via email, send a message with subject or body 'help' to 
blink-request at lists.ag-projects.com 

You can reach the person managing the list at 
blink-owner at lists.ag-projects.com 

When replying, please edit your Subject line so it is more specific 
than "Re: Contents of Blink digest..." 

Today's Topics: 

1. Re: Bkink pro crash (Adrian Georgescu) 

---------------------------------------------------------------------- 

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 

-------------- next part -------------- 
An HTML attachment was scrubbed... 
URL: <http://lists.ag-projects.com/pipermail/blink/attachments/20141020/0d91c0fb/attachment-0001.html> 
-------------- next part -------------- 
A non-text attachment was scrubbed... 
Name: signature.asc 
Type: application/pgp-signature 
Size: 203 bytes 
Desc: Message signed with OpenPGP using GPGMail 
URL: <http://lists.ag-projects.com/pipermail/blink/attachments/20141020/0d91c0fb/attachment-0001.pgp> 

------------------------------ 

_______________________________________________ 
Blink mailing list 
Blink at lists.ag-projects.com 
http://lists.ag-projects.com/mailman/listinfo/blink 

End of Blink Digest, Vol 60, Issue 6 
************************************ 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.ag-projects.com/pipermail/blink/attachments/20141021/7687d103/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: Captura de pantalla 2014-10-20 a la(s) 19.11.29.png
Type: image/png
Size: 278854 bytes
Desc: not available
URL: <http://lists.ag-projects.com/pipermail/blink/attachments/20141021/7687d103/attachment-0001.png>