Displaying 13 results from an estimated 13 matches for "histinfo".
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listinfo
2011 Dec 31
1
Histogram omitting/collapsing groups
I have two large datasets (156K and 2.06M records). Each row has the
hour that an event happened, represented by an integer from 0 to 23.
R's histogram is combining some data.
Here's the command I ran to get the histogram:
> histinfo <- hist(crashes$hour, right=FALSE)
Here's histinfo:
> histinfo
$breaks
?[1] ?0 ?1 ?2 ?3 ?4 ?5 ?6 ?7 ?8 ?9 10 11 12 13 14 15 16 17 18 19 20 21 22 23
$counts
?[1] ?4755 ?4618 ?5959 ?3292 ?2378 ?2715 ?4592 ?6144 ?6860 ?5598 ?5601
?6596 ?7152 ?7490 ?8166
[16] ?9758 11301 11745 ?9943 ?7494 ?...
2015 Feb 23
2
Asterisk does not listed to port 5060
...o: <sip:+91712442211 at unknown.invalid>.
Contact: <sip:+91711189078 at 10.85.0.24:5060;transport=udp>.
From: <sip:+91711189078 at 10.25.84.3;user=phone>;tag=5BD23246313536415F1CF602.
P-Asserted-Identity: <sip:+91711189078 at 10.25.84.3;user=phone>.
Privacy: none.
Supported: histinfo,100rel.
Request-Disposition: no-fork.
P-Early-Media: supported.
History-Info: <sip:+91712442211 at unknown.invalid;user=phone?Privacy=history&Reason=SIP%3Bcause%3D302%3Btext%3D%22Moved%20Temporarily%22>;index=1,
<sip:+91712442211 at unknown.invalid;user=phone>;index=1.1.
Max-Forward...
2011 May 05
4
Insert values to histogram
I'm trying to add the exact value on top of each column of an Histogram, i
have been trying with the text function but it doesn't work.
The problem is that the program it self decides the exact value to give to
each column, and ther is not like in a bar-plot that I know exactly which
values are been plotting.
If anyone have any new idea on how to do this
Thanks
Matias
--
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2011 Feb 04
1
standalone NOTIFY message handling for Asterisk
...: <sip:a.b.c.d:5060;transport=udp;lr>^M
Event: message-summary^M
Accept: application/sdp,application/media_control+xml,message/sipfrag^M
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,INFO,PRACK,UPDATE^M
Contact: <sip:a.b.c.d:5060;transport=udp>^M
Max-Forwards: 70^M
Supported: 100rel,timer,histinfo^M
Subscription-State: active^M
MIME-Version: 1.0^M
Content-Type: application/simple-message-summary^M
Content-Length: 40^M
^M
Messages-Waiting: yes^M
None: 5/0 (0/0)^M
Thanks a lot for your help!
Felton
2014 Feb 26
1
SIP 603 Declined error message
...9ce3124b652973b4b00
To: <sip:51104 at edj.devjones.com>
Call-ID: 8066eb6f589ce3125b652973b4b00
CSeq: 1 INVITE
Max-Forwards: 71
Via: SIP/2.0/TCP 172.17.184.46;branch=z9hG4bK8066eb6f589ce3126b652973b4b00
Via: SIP/2.0/TCP 172.18.78.67;branch=z9hG4bK8066eb6f589ce3126b652973b4b00
Supported: 100rel,histinfo,join,replaces,sdp-anat,timer
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,INFO,PRACK,PUBLISH,UPDATE
User-Agent: Avaya CM/R016x.02.0.823.0
Contact: "Haley, Scott" <sip:3145152244 at 172.17.184.46;transport=tcp>
Route: <sip:192.168.122.51;transport=tcp;lr;phase=termi...
2009 Jan 08
0
SIP message routed back to mysql
...nch=z9hG4bK796c2124;rport=5060
From: "+435555550002"
<sip:05555550002~+43555666 at test.domain.tld>;tag=as3077f211
To: <sip:055555500011 at test.domain.tld>
Call-ID: 4218d05b096c3b45278f73e8146561f7 at test.domain.tld
CSeq: 103 INVITE
Max-Forwards: 68
Supported: replaces,timer,histinfo
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Contact: <sip:05555550002 at a.b.c.151>
Content-Length: 291
Content-Type: application/sdp
Record-Route: <sip:a.b.c.130:5084;lr>
Record-Route:
<sip:a.b.c.131;lr;ftag=as3077f211;vsf=AAAAAAAAAAAAAAAAAAAAPl9RQEUdXlRfRV
VcbhURc3Q...
2016 Feb 10
2
Unexpected termination of the call when pick up (res_pjsip)
...at stc.euroset.ru>;tag=0ea59f7e-817c-48a1-8e44-6e896322609a
To: <sip:+7980xxxxxxx at stc.euroset.ru>;tag=B955C4E4-606476-16E1127B
Call-ID: 5ac4642d-007b-4e29-908a-1b06417148c7
CSeq: 3711 INVITE
Contact: <sip:signode-606476-16E1127B at 83.143.192.141>
Supported: 100rel,timer,replaces,histinfo,precondition
User-Agent: CommuniGatePro-callLeg/5.4.10
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER
Content-Type: application/sdp
Content-Length: 376
v=0
o=CGPLeg606476 1366433634 683216818 IN IP4 83.143.192.141
s=SIP Call
c=IN IP4 83.143.192.141
t=0 0
m=au...
2015 Apr 02
2
Update peer IP address
...62525
Call-ID: af71bbfbf269b895 at 62.155.0.75
CSeq: 3950540 INVITE
Contact: <sip:sgc_c at 217.0.23.68;transport=udp>
Record-Route: <sip:217.0.23.68;transport=udp;lr>
Min-Se: 900
P-Asserted-Identity: <sip:+49123456789 at tel.t-online.de;user=phone>
Session-Expires: 3600
Supported: histinfo
Supported: timer
Supported: norefersub
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 204
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, REGISTER, UPDATE
v=0
o=- 0 0 IN IP4 217.0.23.68
s=-
c=IN IP4 217.0.4.134
t=0 0
m=audio 36480 RTP/AVP 9 8 102
a=rtpmap:...
2023 Jun 21
1
PJSIP not performing outbound authentication
I didn't use pjsip_wizard, I'm kind of crafting this by hand as I learn. I actually have a plain asterisk, and a FreePBX, system to help me learn. I sometimes create something in FreePBX to see what it does to the config files. So that's how I modelled my pjsip.X.conf files
If I issue the command "pjsip show endpoint Twilio" it does show that outbound_auth=Twilio
Does
2015 Apr 02
0
Update peer IP address
...62.155.0.75
> CSeq: 3950540 INVITE
> Contact: <sip:sgc_c at 217.0.23.68;transport=udp>
> Record-Route: <sip:217.0.23.68;transport=udp;lr>
> Min-Se: 900
> P-Asserted-Identity: <sip:+49123456789 at tel.t-online.de;user=phone>
> Session-Expires: 3600
> Supported: histinfo
> Supported: timer
> Supported: norefersub
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 204
> Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, REGISTER,
> UPDATE
>
> v=0
> o=- 0 0 IN IP4 217.0.23.68
> s=-
> c=IN IP...
2019 Jul 06
4
unsolved: Re: solved: how to create a working certificate for using TLS?
On 7/6/19 10:40 AM, Michael Maier wrote:
> On 05.07.19 at 22:02 hw wrote:
>>
>> openssl verify -CAfile ca.pem asterisk.pem
>> asterisk.pem: OK
>>
>>
>> When I set tlsdontverifyserver=yes, it works (i. e. asterisk registers
>> to the SIP provider and there is no error message). Otherwise I'm
>> getting the error message and asterisk does not
2015 Apr 02
3
Update peer IP address
...540 INVITE
> Contact: <sip:sgc_c at 217.0.23.68;transport=udp <>>
> Record-Route: <sip:217.0.23.68;transport=udp;lr <>>
> Min-Se: 900
> P-Asserted-Identity: <sip:+49123456789 at tel.t-online.de;user=phone <>>
> Session-Expires: 3600
> Supported: histinfo
> Supported: timer
> Supported: norefersub
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 204
> Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, REGISTER, UPDATE
>
> v=0
> o=- 0 0 IN IP4 217.0.23.68
> s=-
> c=IN IP4 21...
2015 Apr 02
3
Update peer IP address
Scott, I have changed the configuration as said it and will test it. I?m curious.
Can you briefly explain what insecure=invite,port does?
;insecure=port ; Allow matching of peer by IP address without
; matching port number
;insecure=invite ; Do not require authentication of incoming INVITEs
;insecure=port,invite ; (both)
Do I understand correctly that