[Blink] Bkink pro crash

Daniel Guevara dguevara at clearcom.mx
Mon Oct 20 22:45:13 CEST 2014


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 
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