search for: 30303

Displaying 20 results from an estimated 45 matches for "30303".

Did you mean: 3003
2003 Sep 08
2
cannot allocate vector of size...
Hi, I know that this question has been asked several times before but I haven't been able to find an answer. I'm running R on a P3 933 with 384 MB of RAM. The system is running Windows XP Home. I'm trying to read a 1054 row by 30303 column matrix from a tab delimited text file ("file.txt") into R using the following command: m <- matrix(scan(file ="file.txt"), nrow = 1054, ncol = 30303, byrow = TRUE) When I use this command I get the following output: Read 31939362 items Error: cannot allocate v...
2016 Mar 23
4
Setting up replication?
...vail = 1 unix_listener replicator-doveadm { mode = 0600 } } service aggregator { fifo_listener replication-notify-fifo { mode = 0666 } unix_listener replication-notify { mode = 0666 } } service doveadm { inet_listener { port = 30303 } } #doveadm_port = 30303 doveadm_password = secret plugin { mail_replica = tcp:knute2.frazmtn.com:30303 } replication_dsync_parameters = -d -N -l 30 -U --------------------- knute at knute2:/etc/dovecot/conf.d $ dovecot -n # 2.2.13: /etc/dovecot/dovecot.conf # OS: Linux 4.1.19-v7+...
2011 Nov 27
6
Does Asterisk alter the Headers of INVITE Message
Hi all, I am trying to send an extra header in SIP INVITE Message , i.e (email="me at me.com") but when I check the Message at the target that header is not there So I is Askterisk altering the Message and Is there away to include extra headers for SIP INVITE Message? Thank u
2011 Jul 04
4
stream rtp from asterisk
Hi! Anybody familiar with streaming rtp from asterisk. Preferably with the xorcom asterisk patch which streams rtp from asterisk to oreka audio server. Any ideas will do just fine though! Regards / Marcus
2003 Aug 26
1
Mann-Whitney U Table
...ugh 30. I could use Monte Carlo simulation or the normal approximation when n1 and n2 are greater than, 10, but I figured someone may know how to calculate these exactly. Thanks. Sincerely yours, Mark J. Lamias Statistical Consultant Grizzard Agency 229 Peachtree Street - 12th Floor Atlanta, GA 30303
2011 May 16
2
Reporting Tool: To show who is login, queue, ... etc
Hi All; It look like there are some free (open source) tools that are used for Asterisk reporting special for call center (to see number of agents logged in, number of calls now, .. etc), and to be used as dashboard. Can someone direct me for something really is suitable and stable? Regards Bilal
2019 Jun 11
2
Re: blockcommit of domain not successfull
...:719 : internal error: End of file from qemu monitor 2019-06-08 03:59:17.690+0000: 30299: error : qemuMonitorIO:719 : internal error: End of file from qemu monitor 2019-06-08 03:59:26.145+0000: 30300: warning : qemuGetProcessInfo:1461 : cannot parse process status data 2019-06-08 03:59:26.191+0000: 30303: warning : qemuGetProcessInfo:1461 : cannot parse process status data 2019-06-08 03:59:56.095+0000: 27956: warning : qemuDomainObjBeginJobInternal:4865 : Cannot start job (destroy, none) for domain severin; current job is (modify, none) owned by (13061 remoteDispatchDomainBlockJobAbort, 0 <null&...
2011 May 16
2
AMI perl daemon
Would anybody know how to run a perl script as a daemon that would stay connected to asterisk via AMI? Right now, my AMI script connects to the manager interface, originates a call, disconnects. The script will be run maybe 20+ per minute. It would make more sense to me to have the script run as a daemon and have a persistent connection to asterisk's AMI. Thank you in advance for your input.
2011 Sep 19
1
SIP OPTIONS... Error?
I know over time SIP OPTIONS message handling has changed and I've seen some write ups that seem to indicate that an s extension in the default context is needed now to get them to work. It's probably my error in any case. So, what am I doing wrong or what do I need to do to get the sip ping to work? Bruce Ferrell Just for fun, I created a sip peer called ping at a fixed address
2011 Jun 21
1
: Re: ITSP failover for PRI
...Yes, that's what I thought but you never know ;-) (Maybe SS7 offers such redundancy but I've got no experience of any king in this domain). > > -- > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ > > -- > ______________________________**______________________________**_________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for...
2006 Mar 26
1
Newbie clustering/classification question
...oop, or all at once) or how to save the results to a file. Any suggestions? Example input (tab delimited) condition protA protB protC protD protE protF protG protH healthy1 11111 22222 33333 70681 61735 66666 77777 88888 healthy1 12121 21111 32132 57230 69715 67890 87878 98989 healthy1 10101 20202 30303 67223 51967 65656 78900 111111 healthy2 12345 23111 32100 65931 67650 60001 80001 101010 healthy2 13333 21231 34111 58761 54086 60002 80002 122222 healthy2 13232 20101 30009 68752 70360 60003 80003 91919 asthma 32132 19889 30733 59959 71783 60237 65603 20374 asthma 34344 20483 31182 70531 59630 404...
2019 Jun 13
0
Re: blockcommit of domain not successfull
...om qemu monitor So this looks like qemu crashed. Or at least it's the usual symptom we get. Is there anything in /var/log/libvirt/qemu/$VMNAME.log? > 2019-06-08 03:59:26.145+0000: 30300: warning : qemuGetProcessInfo:1461 : cannot parse process status data > 2019-06-08 03:59:26.191+0000: 30303: warning : qemuGetProcessInfo:1461 : cannot parse process status data > 2019-06-08 03:59:56.095+0000: 27956: warning : qemuDomainObjBeginJobInternal:4865 : Cannot start job (destroy, none) for domain severin; current job is (modify, none) owned by (13061 remoteDispatchDomainBlockJobAbort, 0 <...
2006 Aug 16
0
Customizing Gnome
This is not really CentOS specific, but I'm using and supporting Gnome on CentOS 4.3 systems, and there are a few Gnome issues that I haven't been able to resolve that I thought this list might be able to help with. On my home CentOS 4.3 system, which was originally a CentOS 3.X system (i.e., I ran Gnome from the same user account then as I do now), when I log in to the Gnome
2011 May 03
1
Asterisk 1.6 Questions
I have a couple of questions about asterisk 1.6: Can you change codecs mid-call upon re-invite? Can you handle the SDP offer-answer in the 200-ACK instead of the usual INVITE-200? Thanks in advance for any insight. Gary -------------- next part -------------- An HTML attachment was scrubbed... URL:
2011 May 05
1
asterisk for g729 to g711
Hi, Does anyone know if Asterisk is a good tool to be used for a large quantity of g711 and g729 transcoding? What is the best alternative for that? -- Woody Dickson woodydickson at gmail.com <woody.dickson at gmail.com> US and Worldwide Termination ============ Contact me for the following offering ============ USA Onnet - 0.0049/min USA Offnet - 0.011/min USA Mobile starting
2011 May 09
3
asterisk syntax highlighting for gedit
Hi, Apologies if this is a duplicate - been having mail server issues and I don't think I managed to send it when I tried this morning. It seems there is no .conf syntax highlighting script available for gedit. I'm thinking of putting one together myself, but don't want to reinvent the wheel. So I'm just enquiring if anyone knows of one that already exists that i've missed.
2011 May 11
2
Asterisk SIP Trunking with Cisco UC 560
Hello, I'm interested in knowing if anyone out there has successfully connected Asterisk to a Cisco UC 560 via SIP trunking? We have a client of ours that we put in an Asterisk install, one of their sister companies who we don't control is putting in a Cisco UC 560. From my looking I think it can be done, but the vendor is telling them it can't. Thought I'd ask around here and see
2011 May 16
1
AMD tweaking
Hi, long time ago, I came up with an optimal configuration set for my environment - good detection and little false positives. Unfortunately some people are always being detected as Answering Machines. I'm not up to re-adjust my precious balance of initial_silence/max_words/... , so I'm thinking about to check if the pickup time is equal to the pickup time when the same phone number was
2011 May 23
1
SIP-T to SIP Gateway
Hello, There are some parameters in the ISUP data (coming into the network via SIP-T packets) that need to be translated into SIP headers to be used by asterisk for proper call routing. What gateways are available to accomplish this? Thanks, Elliot -------------- next part -------------- An HTML attachment was scrubbed... URL:
2011 May 31
1
queuemetrics with 1.8 queue_log
Hi Guys! We were using queuemetrics since long time with asterisk 1.2 but recently we have install 1.8 asterisk and but there is a big different in queue_log its saying SIP/XXXX instead of Agent/XXXX that is obvious behaviors. so do i need to change Agent/XXXX to SIP/XXXX in queuemetrics ? or is there any workaround to keep business running same like it was before. -S --------------