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  • Different Linux message queues have the same id?

    - by Halo
    I open a mesage queue in a .c file, and upon success it says the message queue id is 3. While that program is still running, in another terminal I start another program (of another .c file), that creates a new message queue with a different mqd_t. But its id also appears as 3. Is this a problem? server file goes like this: void server(char* req_mq) { struct mq_attr attr; mqd_t mqdes; struct request* msgptr; int n; char *bufptr; int buflen; pid_t apid; //attr.mq_maxmsg = 300; //attr.mq_msgsize = 1024; mqdes = mq_open(req_mq, O_RDWR | O_CREAT, 0666, NULL); if (mqdes == -1) { perror("can not create msg queue\n"); exit(1); } printf("server mq created, mq id = %d\n", (int) mqdes); and the client goes like: void client(char* req_mq, int min, int max, char* dir_path_name, char* outfile) { pid_t pid; /* get the process id */ if ((pid = getpid()) < 0) { perror("unable to get client pid"); } mqd_t mqd, dq; char pfx[50] = DQ_PRFX; char suffix[50]; // sprintf(suffix, "%d", pid); strcat(pfx, suffix); dq = mq_open(pfx, O_RDWR | O_CREAT, 0666, NULL); if (dq == -1) { perror("can not open data queue\n"); exit(1); } printf("data queue created, mq id = %d\n", (int) dq); mqd = mq_open(req_mq, O_RDWR); if (mqd == -1) { perror("can not open msg queue\n"); exit(1); } mqdes and dq seem to share the same id 3.

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  • Windows Azure Queues, WCF, MSMQ integration

    - by user104295
    Hi there, I have a scenario where I need a desktop console app to communicate with a Windows Azure Queue... the most important thing is that the message is received by the server eventually. Also, the desktop app may be disconnected from the Internet sometimes. In the traditional WCF+MSMQ approach you'd be able to send a message which would be cached in MSMQ until MSMQ could reach the Server's MSMQ and send the message. What's the equivalent when Windows Azure is the server-side? Is it possible for the same approach to be used, where MSMQ just communicates with a Windows Azure Queue rather than an MSMQ on a Windows Server? Maybe Windows Azure Queue is the wrong approach? I have heard about something called message buffer, but don't know what this is (yet!). thanks for your help Kris

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  • Basic question about request queues in IIS / ASP.Net

    - by larryq
    I have an ASP.Net application running under IIS 6. A simple page has two radio buttons and a submit button. If I select radio button "A" and submit the page, a lengthy PDF file is generated, which takes about a minute to build. If I select radio button "B", a small PDF is generated. (In both cases the PDF is written out to the Response object and it opens in my browser.) If I select radio button "A" and submit, then hit the red X in my browser to stop the current request, then select radio button "B" and resubmit, the page still takes a long time to process my request. No doubt my first request is still being processed on the server, but I was wondering how IIS and/or ASP.Net are queuing my requests so that fair server use is guaranteed among all users. Am I roughly correct in assuming something like this happens, and if so, how is it done?

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  • Conflating queues

    - by oo
    Are there any good examples of conflating queue in dotnet. I have thousands of messages per second coming in from another system and i wanted to see if this was the best solution and see some implementation examples

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  • persistant message queues

    - by Will
    I have several services on different machines and a message-passing system suits my problem. Reliability - sent messages always delivered, even if one end goes down - is the key concern, although it should also be fast and reasonably bandwidth-efficient. So which message queue should I use?

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  • Applications of concurrent queues and stacks in .NET 4

    - by Jon Harrop
    .NET 4 includes new concurrent data structures. The Bag and Dictionary collections have obvious applications but I cannot see any use for the Queue and Stack data structures. What are people using these for? Also, I've noticed that the design based upon linked lists incurs a lot of allocation and that destroys scalability. This is surprising given that the sole purpose of these collections is multicore programming. Is this an inherent limitation or are they just badly implemented?

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  • Queues And Wait Handles in C#

    - by Michael Covelli
    I've had the following code in my application for some years and have never seen an issue from it. while ((PendingOrders.Count > 0) || (WaitHandle.WaitAny(CommandEventArr) != 1)) { lock (PendingOrders) { if (PendingOrders.Count > 0) { fbo = PendingOrders.Dequeue(); } else { fbo = null; } } // Do Some Work if fbo is != null } Where CommandEventArr is made up of the NewOrderEvent (an auto reset event) and the ExitEvent (a manual reset event). But I just realized today that its not thread safe at all. If this thread gets interrupted right after the first (PendingOrder.Count 0) check has returned false. And then the other thread both enqueues an order and sets the NewOrderEvent before I get a chance to wait on it, the body of the while loop will never run. What's the usual pattern used with a Queue and an AutoResetEvent to fix this and do what I'm trying to do with the code above?

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  • Using an extension to block a caller

    - by Trewq
    I have a couple of SIP phones and use callcentric. I get a lot of junk calls. I'd like to implement the following feature and would like some suggestions on how to do this: Once I get a junk call, I typically hang up. I think want to dial some number (like *23 or something) and I'd like the last number that was received to be put in a database. Any future call from that number will be directed to VM or a busy tone. I'd appreciate some pointers on how I'd go about doing this.. I prefer an open source solution.

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  • not able to register sip user on red5server, using red5phone

    - by sunil221
    I start the red5, and then i start red5phone i try to register sip user , details i provide are username = 999999 password = **** ip = asteriskserverip and i got --- Registering contact -- sip:[email protected]:5072 the right contact could be --- sip :99999@asteriskserverip this is the log: SipUserAgent - listen -> Init... Red5SIP register [SIPUser] register RegisterAgent: Registering contact <sip:[email protected]:5072> (it expires in 3600 secs) RegisterAgent: Registration failure: No response from server. [SIPUser] SIP Registration failure Timeout RegisterAgent: Failed Registration stop try. Red5SIP Client leaving app 1 Red5SIP Client closing client 35C1B495-E084-1651-0C40-559437CAC7E1 Release ports: sip port 5072 audio port 3002 Release port number:5072 Release port number:3002 [SIPUser] close1 [SIPUser] hangup [SIPUser] closeStreams RTMPUser stopStream [SIPUser] unregister RegisterAgent: Unregistering contact <sip:[email protected]:5072> SipUserAgent - hangup -> Init... SipUserAgent - closeMediaApplication -> Init... [SIPUser] provider.halt RegisterAgent: Registration failure: No response from server. [SIPUser] SIP Registration failure Timeout please let me know if i am doing anything wrong. regards Sunil

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  • Linux DHCPD Mac-Address based Groups

    - by GruffTech
    Our Current DHCPD.conf looks like the following. subnet 10.0.32.0 netmask 255.255.255.0 { range 10.0.32.100 10.0.32.254; option subnet-mask 255.255.255.0; option broadcast-address 10.0.32.255; option domain-name-servers 208.67.222.222,208.67.220.220; option routers 10.0.32.5; host Dev-ABaird-W { hardware ethernet 00:1D:09:3E:49:13; fixed-address 10.0.32.94; } ... more static hosts .... } About as basic as it gets. The old router is 10.0.32.1, our company wanted to implement a squid proxy to better monitor web traffic while at work, and if necessary block large time-wasters, IE Facebook.com. However, we've quickly realized that this change has played a mean prank on our Polycom SIP Phones. Occasionally our phones will not ring, the end recipient hears ringing (this is artificially created by our PBX) however the handset never rings. The ONLY thing that has changed in our network is the option routers line. So, Since all Polycom MAC addresses begin with 00:04:F2 would it be possible in DHCP to say any 00:04:F2:::* MAC addresses get option routers 10.0.32.1, and anything else must talk with our Gateway?

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  • What is a fairly accepted endpoints to concurrent calls ratio for a PBX?

    - by Logan Bibby
    Hey guys! Not too positive about the relevance of this question to Server Fault, but I'll let you guys poke around at it anyway. :) I'm trying to figure out what an accepted ratio of endpoints to concurrent calls on the "average" office PBX. I'm defining concurrent calls as any call being placed, including internal calls. I'm not going to be considering call centers into this ratio, I'll be looking at those differently, anyway. To give you guys a little history behind the question: I need this ratio to be able to figure out decent specs for a virtual PBX service I'm going to be rolling out within the upcoming months. The ratio will determine the number of trunks needed per PBX system. -- Logan

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  • not able to register sip user on red5server, using red5phone

    - by sunil221
    I start the red5, and then i start red5phone i try to register sip user , details i provide are username = 999999 password = **** ip = asteriskserverip and i got --- Registering contact -- sip:[email protected]:5072 the right contact could be --- sip :99999@asteriskserverip this is the log: SipUserAgent - listen -> Init... Red5SIP register [SIPUser] register RegisterAgent: Registering contact <sip:[email protected]:5072> (it expires in 3600 secs) RegisterAgent: Registration failure: No response from server. [SIPUser] SIP Registration failure Timeout RegisterAgent: Failed Registration stop try. Red5SIP Client leaving app 1 Red5SIP Client closing client 35C1B495-E084-1651-0C40-559437CAC7E1 Release ports: sip port 5072 audio port 3002 Release port number:5072 Release port number:3002 [SIPUser] close1 [SIPUser] hangup [SIPUser] closeStreams RTMPUser stopStream [SIPUser] unregister RegisterAgent: Unregistering contact <sip:[email protected]:5072> SipUserAgent - hangup -> Init... SipUserAgent - closeMediaApplication -> Init... [SIPUser] provider.halt RegisterAgent: Registration failure: No response from server. [SIPUser] SIP Registration failure Timeout please let me know if i am doing anything wrong. regards Sunil

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  • Cheap Solution for Routing a Toll Free Number to a Standard POTS Number

    - by VxJasonxV
    I do some technical work for an Internet Radio Show/Podcast, and need to fix something that has been broken for a while. The hosts have a Skype-In number to take listener calls, and for convenience sake, I bought and paid for a toll free number for a period of time. I used to use Asterlink for routing calls, but they folded and sent my number to OneBox, but they're ridiculously expensive by comparison. I'm looking for a cheap solution for this one simple task. Forward toll free calls to a skype-in number. The definition of cheap is as cheap or cheaper than Asterlink was. I paid something like $2 a month, and then the termination/call rate, which was a fraction of a sent for termination, and only whole cents after some serious time on the call. A $20 preload lasted me months at a time. I don't want to be upsold too, I want a simple web based management screen (CDR/stats are fun!), and obviously, it needs to be reliable. What vendors out there are you a fan of that solves this need?

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  • Provisioning SIP Phones over the internet

    - by Jorge Fernandez
    I have a few SIP Phones that are located of site and connect to my PBX over the internet to make calls. For some reason one of these phones has become unprovisioned. In my office phones get provisioned by the server via TFTP. The ones that I have off site I pre-provisioned manually before I sent them off-site (I'm in Florida the phone is in New Jersey). Whats the best way to provision these over the internet? TFTP is very insecure. Sending the plain text profiles with the SIP Account and Password over the internet is out of the question. The phones have been off-site for about 6 months without any issues. Im using Trixbox and Cisco 7940 Phones.

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  • Cheap Solution for Routing a Toll Free Number to a Standard POTS Number

    - by VxJasonxV
    I do some technical work for an Internet Radio Show/Podcast, and need to fix something that has been broken for a while. The hosts have a Skype-In number to take listener calls, and for convenience sake, I bought and paid for a toll free number for a period of time. I used to use Asterlink for routing calls, but they folded and sent my number to OneBox, but they're ridiculously expensive by comparison. I'm looking for a cheap solution for this one simple task. Forward toll free calls to a skype-in number. The definition of cheap is as cheap or cheaper than Asterlink was. I paid something like $2 a month, and then the termination/call rate, which was a fraction of a sent for termination, and only whole cents after some serious time on the call. A $20 preload lasted me months at a time. I don't want to be upsold too, I want a simple web based management screen (CDR/stats are fun!), and obviously, it needs to be reliable. What vendors out there are you a fan of that solves this need?

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  • How to configure three IP address into single server

    - by user1363308
    I have Cisco device for call forwarding and three different system,I want to configure 15 and 16 server IP into 192.168.53.197 means eth0 --> 192.168.53.197 eth1 --> 192.168.16.15 eth2 --> 192.168.16.16 which work i have done with 15 and 16 individual , I will do some work on 197 after configuration eth1 and eth2. Means one system have three IP address but base IP address is 192.168.53.197

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  • SpeechBackground

    - by abinila
    Hai everyone, I have used the SpeechBackground application in asterisk. I used the version 1.6.0.6. I have a entry like, ;;SpeechCreate exten => s,1,SpeechCreate() exten => s,2,SpeechActivateGrammar(yesno) exten => s,3,SpeechStart() exten => s,4,SpeechBackground(demo-instruct) exten => s,5,SpeechDeactivateGrammar(yesno) I don't know which file I meed to give in SpeechBackground application. Please give me any idea. I have given the sound file from /sounds directory. If I call to 's' the call will be immediately released.I didn't get any audio sound. Please any one help me...

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  • How to receive the text which was sent by SendText

    - by thillai-selvan
    In asterisk I have sent a text message using SendText as follows I have two registered users in sip.conf file. sip.conf details [thillai] username=thillai secret=thillai host=dynamic type=friend allow=all context=test [selvan] username=selvan secret=selvan allow=all host=dynamic type=friend context=test Then I have created the necessary extensions like this extensions.conf file: [test] exten = 677,1,BackGround(thankyou) exten = 677,n,Dial(SIP/thillai) exten = 677,n,SendText('this is for testing') So when a caller is trying to call to extension 677 this text information will be sent. My question is how can I receive this text in the caller side? Any help will be much appreciated.

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  • Using flash for videoconferencing

    - by PHP
    Hi I'd like to set up a videoconferencing (1-1 initially) system using Flash as the client. I've tried some web searches but couldn't find much; any pointers on the following would be great: Does the standard browser Flash 10 plugin contain everything needed for P2P video conferencing, or is server support needed? Does anyone have any experience of opensource server videoconferencing products to be used with Flash on the browser side? E.g. would Asterisk work? Something like http://www.red5chat.com/ ? (ideally I'd like to be able to record videoconferences also). Any licencing issues? Many thanks

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  • Centreon setup (web part) config not writeable

    - by Evgeniy
    I have a problem with Centreon installation. At end of install process, I stopped at 10th step: Writable Centreon Configuration File (centreon.conf.php) Critical: Not Writeable /etc/centreon/ asterisk:asterisk (755) Should be : (755) Generate Centreon configuration file OK /etc/centreon/centreon.conf.php asterisk:asterisk (755) Should be : (755) Generate Centstorage configuration file OK /etc/centreon//conf.pm asterisk:asterisk (755) Should be : (755) But permissions and owner are OK. And I tried chmod -R 777 /etc/centreon Please help to understand what's wrong. UPD: Screenshot: http://gyazo.com/82701d119a5e0d98947ad35406a5b8f3

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  • Ubuntu-one syncs single files, but not directories [closed]

    - by Luiz Cláudio Duarte
    I'm using Ubuntu 10.10, fully updated. I have tried to sync my ~/Documents and ~/Pictures folders; U1 replicates the directory structure, but no files are uploaded. Next I tried to sync a single file inside ~/Ubuntu One and it was synced. Then I tried to put a directory inside ~/Ubuntu One and, again, the directory structure was replicated, but no files were synced. All the files have the "syncing" icon, however. The latest syncdaemon.log is below: 2011-03-30 07:41:50,752 - ubuntuone.SyncDaemon.fsm - INFO - loading updated metadata 2011-03-30 07:41:55,081 - ubuntuone.SyncDaemon.fsm - INFO - initialized: idx_path: 266, idx_node_id: 266, shares: 1 2011-03-30 07:41:55,082 - ubuntuone.SyncDaemon.GeneralINotProc - INFO - Ignoring files: ['\\A#.*\\Z', '\\A.*~\\Z', '\\A.*\\.py[oc]\\Z', '\\A.*\\.sw[nopx]\\Z', '\\A.*\\.swpx\\Z', '\\A\\..*\\.tmp\\Z'] 2011-03-30 07:41:55,083 - ubuntuone.SyncDaemon.HQ - INFO - HashQueue: _hasher started 2011-03-30 07:41:55,902 - ubuntuone.SyncDaemon.DBus - INFO - DBusInterface initialized. 2011-03-30 07:41:55,903 - ubuntuone.SyncDaemon.Main - INFO - Using '/home/l_claudius/Ubuntu One' as root dir 2011-03-30 07:41:55,903 - ubuntuone.SyncDaemon.Main - INFO - Using '/home/l_claudius/.local/share/ubuntuone/syncdaemon' as data dir 2011-03-30 07:41:55,903 - ubuntuone.SyncDaemon.Main - INFO - Using '/home/l_claudius/.local/share/ubuntuone/shares' as shares root dir 2011-03-30 07:41:55,903 - ubuntuone.SyncDaemon.Main - NOTE - ---- MARK (state: <State: 'INIT' (queues IDLE connection 'Not User Not Network')>; queues: metadata: 0; content: 0; hash: 0, fsm-cache: hit=1 miss=266) ---- 2011-03-30 07:41:55,904 - ubuntuone.SyncDaemon.Main - NOTE - Local rescan starting... 2011-03-30 07:41:55,904 - ubuntuone.SyncDaemon.local_rescan - INFO - start scan all volumes 2011-03-30 07:41:55,906 - ubuntuone.SyncDaemon.local_rescan - INFO - processing trash 2011-03-30 07:41:56,044 - ubuntuone.SyncDaemon.local_rescan - INFO - processing move limbo 2011-03-30 07:41:56,491 - ubuntuone.SyncDaemon.Main - NOTE - Local rescan finished! 2011-03-30 07:41:56,492 - ubuntuone.SyncDaemon.Main - INFO - hash queue empty. We are ready! 2011-03-30 07:42:15,583 - ubuntuone.SyncDaemon.DBus - INFO - u'CredentialsFound': callbacking with credentials (token_name: None). 2011-03-30 07:42:15,584 - ubuntuone.SyncDaemon.DBus - INFO - connect: credential request was successful, pushing SYS_USER_CONNECT. 2011-03-30 07:42:15,617 - ubuntuone.SyncDaemon.ActionQueue - INFO - Connection started to host fs-1.one.ubuntu.com, port 443. 2011-03-30 07:42:15,977 - ubuntuone.SyncDaemon.ActionQueue - INFO - Connection made. 2011-03-30 07:42:15,978 - ubuntuone.SyncDaemon.StorageClient - INFO - Connection made. 2011-03-30 07:42:16,581 - ubuntuone.SyncDaemon.ActionQueue - INFO - The request 'protocol_version' finished OK. 2011-03-30 07:42:16,774 - ubuntuone.SyncDaemon.ActionQueue - INFO - The request 'caps_raising_if_not_accepted' finished OK. 2011-03-30 07:42:16,966 - ubuntuone.SyncDaemon.ActionQueue - INFO - The request 'caps_raising_if_not_accepted' finished OK. 2011-03-30 07:42:17,722 - ubuntuone.SyncDaemon.ActionQueue - INFO - The request 'oauth_authenticate' finished OK. 2011-03-30 07:42:17,723 - ubuntuone.SyncDaemon.ActionQueue - NOTE - Session ID: '563bc960-35fa-4f44-b9b6-125819656dc3' 2011-03-30 07:42:19,258 - ubuntuone.SyncDaemon.ActionQueue - INFO - The request 'list_volumes' finished OK. 2011-03-30 07:43:55,903 - ubuntuone.SyncDaemon.Main - NOTE - ---- MARK (state: <State: 'QUEUE_MANAGER' (queues IDLE connection 'With User With Network')>; queues: metadata: 0; content: 0; hash: 0, fsm-cache: hit=1059 miss=266) ---- 2011-03-30 07:45:55,903 - ubuntuone.SyncDaemon.Main - NOTE - ---- MARK (state: <State: 'QUEUE_MANAGER' (queues IDLE connection 'With User With Network')>; queues: metadata: 0; content: 0; hash: 0, fsm-cache: hit=1059 miss=266) ---- 2011-03-30 07:47:55,903 - ubuntuone.SyncDaemon.Main - NOTE - ---- MARK (state: <State: 'QUEUE_MANAGER' (queues IDLE connection 'With User With Network')>; queues: metadata: 0; content: 0; hash: 0, fsm-cache: hit=1059 miss=266) ---- 2011-03-30 07:49:55,903 - ubuntuone.SyncDaemon.Main - NOTE - ---- MARK (state: <State: 'QUEUE_MANAGER' (queues IDLE connection 'With User With Network')>; queues: metadata: 0; content: 0; hash: 0, fsm-cache: hit=1059 miss=266) ---- 2011-03-30 07:51:55,903 - ubuntuone.SyncDaemon.Main - NOTE - ---- MARK (state: <State: 'QUEUE_MANAGER' (queues IDLE connection 'With User With Network')>; queues: metadata: 0; content: 0; hash: 0, fsm-cache: hit=1059 miss=266) ----

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  • How to use regex to match ASTERISK in awk

    - by Ken Chen
    I'm stil pretty new to regular expression and just started learning to use awk. What I am trying to accomplish is writing a ksh script to read-in lines from text, and and for every lines that match the following: *RECORD 0000001 [some_serial_#] to replace $2 (i.e. 000001) with a different number. So essentially the script read in batch record dump, and replace the record number with date+record#, and write to separate file. So this is what I'm thinking the format should be: awk 'match($0,"/*FTR")!=0{$2="$DATE-n++"; print $0} match($0,"/*FTR")==0{print $0}' $BATCH > $OUTPUT but obviously "/*FTR" is not going to work, and I'm not sure if changing $2 and then write the whole line is the correct way to do this. So I am in need of some serious enlightenment.

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