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• How do I install Kodos in 12.10?

  - by Brutus

  In previous Ubuntu version I used Kodos extensively. But the package has been dropped in 12.10. It's a neat regular expression testing tool. It's hard to find an alternative that is not Windows only (or Air, or uses action script which bails on more complicated RegEx, or works in Wine - well kinda, but not really..., etc.). So I tried to install Kodos from source, which seems to work (download sourceball and setup.py) but it won't run because of missing PyQT dependencies (that I can't manage to fulfill with anything install-able trough standard packages). I then tried to install PyQT manually (which seems to require manual install of SIP and QT too) and instead of pip I have to use configure.py and qmake. It throws error after error at me. I tried to overcome one after another for over an hour but no luck. It even managed to break Calibre and Music Brains Picard. So I purged all the stuff, reinstalled python-sipand python-qt4 from the standard packages and gave up. Has anyone managed to get Kodos running on 12.10? Or any hints on how to do it?

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• h264 RTP timestamp

  - by user269090

  Hi Guys, I have a confusion about the timestamp of h264 RTP packet. I know the wall clock rate of video is 90KHz which I defined in the SIP SDP. The frame rate of my encoder is not exactly 30 FPS, it is variable. It varies from 15 FPS to 30 FPS on the fly. So, I cannot use any fixed timestamp. Could any one tell me the timestamp of the following encoded packet. After 0 milisecond encoded RTP timestamp = 0 (Let the starting timestamp 0) After 50 milisecond encoded RTP timestamp = ? After 40 milisecond encoded RTP timestamp = ? After 33 milisecond encoded RTP timestamp = ? What is the formula when the encoded frame rate is variable? Thank you in advance.

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• NSAlert doesn't work

  - by Marco

  Hello i have implemented the following NSAlert: NSString *title = [NSString stringWithFormat:@"Keine Internetverbindung"]; NSString *alertMessage = [NSString stringWithFormat:@"Es konnte keine Verbindung zu www.sip.de aufgebaut werden!"]; NSString *ok = [NSString stringWithFormat:@"Ok"]; UIAlertView *alert = [[UIAlertView alloc] initWithTitle:title message:alertMessage delegate:self cancelButtonTitle:ok otherButtonTitles:nil]; [alert show]; [alert release]; and this is the delegate method to close the app: - (void)alertView:(UIAlertView *)alertView didDisMissWithButtonIndex:(NSInteger)buttonIndex{ exit(3); } But why the programm doesn't go into the method, what is my mistake? greetings Marco

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• Visual Studio Code Metrics and the Maintainability index of switch case

  - by pee2002

  Hi there! As a person who loves to follow the best practices, If i run code metrics (right click on project name in solution explorer and select "Calculate Code Metrics" - Visual Studio 2010) on: public static string GetFormFactor(int number) { string formFactor = string.Empty; switch (number) { case 1: formFactor = "Other"; break; case 2: formFactor = "SIP"; break; case 3: formFactor = "DIP"; break; case 4: formFactor = "ZIP"; break; case 5: formFactor = "SOJ"; break; } return formFactor; } It Gives me a Maintainability index of 61 (of course this is insignificant if you have only this, but if you use an utility like class whos philosophy is doing stuff like that, your utility class will have the maintainability index much worst..) Whats the solution for this?

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• iPhone: NSAlert Delegate Method Never Called

  - by Marco

  Hello, I have implemented an NSAlert but the delegate method didDissmissWithButton is never called. This code evokes the NSAlert: NSString *title = [NSString stringWithFormat:@"Keine Internetverbindung"]; NSString *alertMessage = [NSString stringWithFormat:@"Es konnte keine Verbindung zu www.sip.de aufgebaut werden!"]; NSString *ok = [NSString stringWithFormat:@"Ok"]; UIAlertView *alert = [[UIAlertView alloc] initWithTitle:title message:alertMessage delegate:self cancelButtonTitle:ok otherButtonTitles:nil]; [alert show]; [alert release]; and this is the NSAlert delegate method: - (void)alertView:(UIAlertView *)alertView didDisMissWithButtonIndex:(NSInteger)buttonIndex{ exit(3); } The method is never called, what is my mistake?

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• Java JDK Source Code?? Where to find it?

  - by Martijn Courteaux

  Hi, I like to see what a method in the Java API does. So I want the JDK Source Code. Before I re-installed Linux I had the scr.sip file with all the code in it. I just had to tell Eclipse this file and I could see the code. But now I haven't the file anymore... So the question is: Where can I find it? Please don't paste the Google results here. I searched already long, but I can't find it... I need just that file. Please be sure if it is the correct file before you answer. Answering by giving an URL is enough for me. Thanks in advance Martijn

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• Allowing threads from python after calling a blocking i/o code in a python extension generated using

  - by SS

  I have written a python extension wrapping an existing C++ library live555 (wrapping RTSP client interface to be specific) in SWIG. The extension works when it is operated in a single thread, but as soon as I call the event loop function of the library, python interpreter never gets the control back. So if I create a scheduled task using threading.Timer right before calling the event loop, that task never gets executed once event loop starts. To fix this issue, I added Py_BEGIN_ALLOW_THREADS and Py_END_ALLOW_THREADS macros manually in the SWIG auto generated wrapper cxx file around every doEventLoop() function call. But now, I want to do the same (i.e. allow threads) when SWIG generates the code itself and not to change any code manually. Has anyone done something similar in SWIG? P.S. - I would also consider switching to any other framework (like SIP) to get this working. I selected SWIG over any other technology is because writing SWIG interface was really very easy and I just had to include the existing header files.

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• Brew install pyqt mavericks

  - by user3722876

  I have some trouble installing PyQt on my Mac. HOMEBREW_VERSION: 0.9.5 ORIGIN: https://github.com/Homebrew/homebrew.git HEAD: d8af29d63a5b94ffee863788210c3a895315035f HOMEBREW_PREFIX: /usr/local HOMEBREW_CELLAR: /usr/local/Cellar CPU: quad-core 64-bit sandybridge OS X: 10.9.3-x86_64 Xcode: 5.1.1 CLT: 5.1.0.0.1.1396320587 Clang: 5.1 build 503 MacPorts/Fink: /opt/local/bin/port X11: 2.7.6 => /opt/X11 System Ruby: 2.0.0-451 Perl: /usr/bin/perl Python: /opt/local/bin/python => /opt/local/Library/Frameworks/Python.framework/Versions/2.7/bin/python2.7 Ruby: /usr/bin/ruby sip installation ok qt installation ok brew install pyqt => make 1 error generated. make[1]: *** [qtlib.o] Error 1 1 error generated. make[1]: *** [siplib.o] Error 1 make: *** [all] Error 2 No idea what's happening...

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• php push 2d array into mysql

  - by john

  Hay All, I cant seem to get my head around this dispite the number to examples i read. Basically I have a 2d array and want to insert it into MySQL. The array contains a few strings. I cant get the following to work... $value = addslashes(serialize($temp3));//temp3 is my 2d array, do i need to use keys? (i am not at the moment) $query = "INSERT INTO table sip (id,keyword,data,flags) VALUES(\"$value\")"; mysql_query($query) or die("Failed Query"); Thanks Guys,

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• Best protocol for client/server communication, from PHP/Perl to C++/Qt4

  - by Kyle

  I'm the author of an Open Source kiosk management system, Libki. The current version, though functional, was very much a learning experience for me. I'm working on a complete rewrite and am having a hard time deciding what protocol to use. The server will be written in PHP or Perl. Most likely PHP because I need to support some uncommon protocols that Library software use, ( SIP and NCIP ). So far I've only found a SIP2 library in PHP. The client is written in C++/Qt4. I'm looking at RPC and REST for client/server communication. I've found RPC client libraries for Qt4, and REST is already part of the Qt4 libraries. Is there an alternative I've missed? So far, REST seems to be the winner.

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• Adding or reading some of the contact fields in sybian using j2me

  - by learn

  I want to add or read the fields of cotact like i am getting the telephone home no ContactList clist; Contact con; String no; if(cList.isSupportedAttribute(Contact.TEL, Contact.ATTR_HOME)) { con.addString(Contact.TEL, Contact.ATTR_HOME, no); } and mobile no if(cList.isSupportedAttribute(Contact.TEL, Contact.ATTR_MOBILE)) { con.addString(Contact.TEL, Contact.ATTR_MOBILE, mb); } now i want to get the fields internet telephone, push to talk, mobile(home), mobile(business), dtmf, shareview, sip, children, spouse and some more fields please help me.. Thanks in advance

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• Official definition of CSCI (Computer Software Configuration Item)

  - by Andreas_D

  I'm looking for the most official definition of CSCI / Configuration Item - not just what it is but what we have to deliver / can expect when a contract defines subsystems which shall be developed as configuration items. I spend some time with my famous search tool and found a lot of explanations for CSCI (wikipedia, acronym directories, ...) but I haven't found a standard or a pointer to a standard (like ISO-xxx) yet which tells (1) what it is and (2) what has to be done from a QM/CM point of view. I just ask, because a contractors QM representative stated during an acceptance test, that CI only requires to not forget the CI in the configuration plan and to assign a serial number ... I expected to see some SRS, SDD, ICD, SVD, SIP, ... documents and acceptance test documentation for those subsystems...

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• Java: How to clear socket bindings

  - by Matt1776

  I am having a few issues with sockets within my Java SIP client. When I bind to an address and port, if something goes wrong I have to attempt to reconnect, usually after I've stopped and restarted the process. Problem with that is then the port is bound and I am forced to increment the local port. How can I remove the binding to the port I am targeting before binding to it? If that isnt possible, then how can I trap the process just before it ends so that I can locate the socket binding and close it manually?

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• Whats the best way to parse a file in Java

  - by chitresh

  Hi All, I have a text file with Tag - Value format data. I want to parse this file to form a Trie. What will be the best approach? Sample of File: (String inside "" is a tag and '#' is used to comment the line.) #Hi, this is a sample file. "abcd" = 12; "abcde" = 16; "http" = 32; "sip" = 21;

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• QoS for Cisco Router to Prioritize Voice and Interactive Traffic

  - by TJ Huffington

  I have a Cisco 891W NATing Voice and Data to the internet over a 10mbit/2mbit connection. Voice traffic gets degraded when I upload large files. Pings time out as well. I tried to configure a QoS policy but it's basically not doing anything. Voice traffic still degrades when upload bandwidth gets saturated. Here is my current configruation: class-map match-any QoS-Transactional match protocol ssh match protocol xwindows class-map match-any QoS-Voice match protocol rtp audio class-map match-any QoS-Bulk match protocol secure-nntp match protocol smtp match protocol tftp match protocol ftp class-map match-any QoS-Management match protocol snmp match protocol dns match protocol secure-imap class-map match-any QoS-Inter-Video match protocol rtp video class-map match-any QoS-Voice-Control match access-group name Voice-Control policy-map QoS-Priority-Output class QoS-Voice priority percent 25 set dscp ef class QoS-Inter-Video bandwidth remaining percent 10 set dscp af41 class QoS-Transactional bandwidth remaining percent 25 random-detect dscp-based set dscp af21 class QoS-Bulk bandwidth remaining percent 5 random-detect dscp-based set dscp af11 class QoS-Management bandwidth remaining percent 1 set dscp cs2 class QoS-Voice-Control priority percent 5 set dscp ef class class-default fair-queue interface FastEthernet8 bandwidth 1024 bandwidth receive 20480 ip address dhcp ip nat outside ip virtual-reassembly duplex auto speed auto auto discovery qos crypto map mymap max-reserved-bandwidth 80 service-policy output QoS-Priority-Output crypto map mymap 10 ipsec-isakmp set peer 1.2.3.4 default set transform-set ESP-3DES-SHA match address 110 qos pre-classify ! fa8 is my connection to the internet. Voice traffic goes over a VPN ("mymap") to the SIP server. That's why I specified "qos pre-classify" which I believe is the way to classify traffic over the VPN. However even when I ping a public IP while saturating upload bandwidth, the latency is exceptionally high. Is this configuration correct? Are there any suggestions that might make this work for my setup? Thanks in advance.

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• QoS for Cisco Router to Prioritize Voice and Interactive Traffic

  - by TJ Huffington

  I have a Cisco 891W NATing Voice and Data to the internet over a 10mbit/2mbit connection. Voice traffic gets degraded when I upload large files. Pings time out as well. I tried to configure a QoS policy but it's basically not doing anything. Voice traffic still degrades when upload bandwidth gets saturated. Here is my current configruation: class-map match-any QoS-Transactional match protocol ssh match protocol xwindows class-map match-any QoS-Voice match protocol rtp audio class-map match-any QoS-Bulk match protocol secure-nntp match protocol smtp match protocol tftp match protocol ftp class-map match-any QoS-Management match protocol snmp match protocol dns match protocol secure-imap class-map match-any QoS-Inter-Video match protocol rtp video class-map match-any QoS-Voice-Control match access-group name Voice-Control policy-map QoS-Priority-Output class QoS-Voice priority percent 25 set dscp ef class QoS-Inter-Video bandwidth remaining percent 10 set dscp af41 class QoS-Transactional bandwidth remaining percent 25 random-detect dscp-based set dscp af21 class QoS-Bulk bandwidth remaining percent 5 random-detect dscp-based set dscp af11 class QoS-Management bandwidth remaining percent 1 set dscp cs2 class QoS-Voice-Control priority percent 5 set dscp ef class class-default fair-queue interface FastEthernet8 bandwidth 1024 bandwidth receive 20480 ip address dhcp ip nat outside ip virtual-reassembly duplex auto speed auto auto discovery qos crypto map mymap max-reserved-bandwidth 80 service-policy output QoS-Priority-Output crypto map mymap 10 ipsec-isakmp set peer 1.2.3.4 default set transform-set ESP-3DES-SHA match address 110 qos pre-classify ! fa8 is my connection to the internet. Voice traffic goes over a VPN ("mymap") to the SIP server. That's why I specified "qos pre-classify" which I believe is the way to classify traffic over the VPN. However even when I ping a public IP while saturating upload bandwidth, the latency is exceptionally high. Is this configuration correct? Are there any suggestions that might make this work for my setup? Thanks in advance.

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• QoS basics on a Cisco ASA

  - by qbn

  Could someone briefly explain how to use QoS on Cisco ASA 5505? I have the basics of policing down, but what about shaping and priorities? Basically what I'm trying to do is carve out some bandwidth for my VPN subnets (in an object-group called priority-traffic). I've seen this Cisco QoS document, however configuring shaping and priority-queue don't seem to have any effects in my test. A full download of the linux kernel from kernel.org will boost a ping to a server via VPN sky high. Policing has been successful in passing this test, although it doesn't seem as efficient (I cap non-vpn traffic at 3 of my 4.5 megabits of bandwidth). Am I misunderstanding the results of the test? I think there is some simple concept I'm not grasping here. EDIT: Here is my config thus far (I have 4.5 megabits of bandwidth): access-list priority-traffic extended permit ip object-group priority-traffic any access-list priority-traffic extended permit ip any object-group priority-traffic access-list priority-traffic extended permit icmp object-group priority-traffic any access-list priority-traffic extended permit icmp any object-group priority-traffic access-list non-priority-traffic extended deny ip object-group priority-traffic any access-list non-priority-traffic extended deny ip any object-group priority-traffic access-list non-priority-traffic extended permit ip any any priority-queue outside queue-limit 440 class-map non-priority-traffic match access-list non-priority-traffic class-map priority-traffic match access-list priority-traffic class-map inspection_default match default-inspection-traffic policy-map type inspect dns preset_dns_map parameters message-length maximum 512 policy-map global_policy class inspection_default inspect dns preset_dns_map inspect ftp inspect h323 h225 inspect h323 ras inspect rsh inspect rtsp inspect sqlnet inspect skinny inspect sunrpc inspect xdmcp inspect sip inspect netbios inspect tftp policy-map outbound-qos-policy class non-priority-traffic police input 2500000 police output 2500000 class priority-traffic priority service-policy global_policy global service-policy outbound-qos-policy interface outside

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• WRTU54G-TM router with 3rd party firmware; Can custom firmware include stock binary portions?

  - by dlamblin

  I've been doing a lot of reading online about the Linksys WRTU54G-TM router model that I now own. It seems getting a custom firmware onto it is not a problem. But no one is talking about retaining the Voip features (yet). So far they're all disappointed that it's not a SIP machine and used GSM over IPSec. Personally I don't care about using it with non-t-mobile. If I take the original firmware, shouldn't I be able to extract it, and it's SquashFS image, and then move all of the t-mobile specific binaries for enabling the calling features over to a custom firmware installation (maybe OpenWRT)? You might ask why, and the reason is, that if I do this I could retain my calling features, which I do want, and ssh to the router and use it to run additional software, as any OpenWRT router could do. Does anyone know if this can be done, and how the firmware's binaries could be gotten at and installed correctly? Update I have found someone working on 3rd party WRTU54G-TM firmware. I am still interested in my second part of the questions, that is can't the stock firmware images be pulled apart and have the close-source, if any, binary kernel modules moved into another more flexible custom firmware?

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• WRTU54G-TM router with 3rd party firmware; Can custom firmware include stock binary portions?

  - by dlamblin

  I've been doing a lot of reading online about the Linksys WRTU54G-TM router model that I now own. It seems getting a custom firmware onto it is not a problem. But no one is talking about retaining the Voip features (yet). So far they're all disappointed that it's not a SIP machine and used GSM over IPSec. Personally I don't care about using it with non-t-mobile. If I take the original firmware, shouldn't I be able to extract it, and it's SquashFS image, and then move all of the t-mobile specific binaries for enabling the calling features over to a custom firmware installation (maybe OpenWRT)? You might ask why, and the reason is, that if I do this I could retain my calling features, which I do want, and ssh to the router and use it to run additional software, as any OpenWRT router could do. Does anyone know if this can be done, and how the firmware's binaries could be gotten at and installed correctly? Update I have found someone working on 3rd party WRTU54G-TM firmware. I am still interested in my second part of the questions, that is can't the stock firmware images be pulled apart and have the close-source, if any, binary kernel modules moved into another more flexible custom firmware?

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• How to send T.38 from a mac?

  - by Brian Postow

  I'm trying to set up a fax-server on a macintosh. I have Hylafax, and we're going to use an internet FOIP fax provider (Haven't decided who yet, that may be another question). The problem is how to get from Hylafax to T.38. I know of two options, but I'm not sure how to decide between them: T38modem Advantages: It's only one extra program, and i know that I can compile it for the Mac. (well, At least I can get the H323 version working on a Mac) Disadvantages: It is mostly undocumented and seems to be supported only by one guy in Russia. IAXModem/Asterisk Advantages: It's well known, and well supported. We can pay for support. It presumably does the T38 with SIP correctly, so we don't have to worry about it. Disadvantages: It's two separate programs. While I know how to get Asterisk on a mac, I'm not sure about IAXModem. (It's sourceforge, and linux, but compiling things for a mac isn't always straight forward...) It's also mostly undocumented. Do these seem like an accurate listing of the pros/cons? Anyone have any other suggestions? thanks.

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• Security for university research lab systems

  - by ank

  Being responsible for security in a university computer science department is no fun at all. And I explain: It is often the case that I get a request for installation of new hw systems or software systems that are really so experimental that I would not dare put them even in the DMZ. If I can avoid it and force an installation in a restricted inside VLAN that is fine but occasionally I get requests that need access to the outside world. And actually it makes sense to have such systems have access to the world for testing purposes. Here is the latest request: A newly developed system that uses SIP is in the final stages of development. This system will enable communication with outside users (that is its purpose and the research proposal), actually hospital patients not so well aware of technology. So it makes sense to open it to the rest of the world. What I am looking for is anyone who has experience with dealing with such highly experimental systems that need wide outside network access. How do you secure the rest of the network and systems from this security nightmare without hindering research? Is placement in the DMZ enough? Any extra precautions? Any other options, methodologies?

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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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• Blink build with Xcode failed

  - by Merci

  I found a GPL-ed SIP client for Mac, Blink. I'd like to build it from source since the binaries are only available as paid download. Just FYI i'm studying programming at university but have no experience in building complex application from source. After downloading the content of the repository i opened the Xcode project and tried to build on OS X 10.7, Xcode 4.2.1. Unfortunately the build fail with 1 error and many warnings Most of the warnings are like this: Attribute Unavailable: Custom Identifiers in Interface Builder versions prior to 3.2 The error message is: Apple Mach-O Linker (ld) Error Command /Developer/usr/bin/clang failed with exit code 1 preceded by the warning Apple Mach-O Linker (ld) Warning directory not found for option '-L/Users/Sergio/Downloads/Blink/devel.ag-projects.com/repositories/public/blink-cocoa/Distribution/Frameworks' I notice that in the list of required files i have this files missing: Dependencies/Frameworks libgcrypt.11.6.0.dylib libgcrypt.11.dylib libgnutls-extra.26.dylib libgnutls.26.dylib libgpg-error.0.dylib libintl.8.dylib liblzo.1.dylib libtasn1.3.dylib Dependencies/Resources lib Frameworks/Linked Frameworks Sparkle.framework Products Blink.app It should be possible to download these files somewhere. Unfortunately googling did not help. There's no documentation on the project site.

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• How to play WAV file through Network Paging Interface

  - by BGM

  In our building we have a Viking Paging ZPI-4 Interface for our intercom. The interface receives data from our Asterisk Phone system via a Cisco SPA112 Port Adapter which has it's own IP address on the network and converts digital into analog. Asterisk plays the "5" tone and then allows the user's voice to commence over the connection. Now, what I want to do is to play a wav file over this Viking Paging device using the Cisco Port Adapter. I know how to get Asterisk to do it, but I want to do this without Asterisk. I want some kind of program that can talk to the Cisco Port Adapter and then transmit the wav file into the Viking Paging Device. What kind of program do I need to get or make? Now, I found this link if it helps anyone with ideas. I also found this information, but I'm not sure how to apply it. I also found this, but it involves an arduino. However, I already have the analog-to-digital convertor, and the Viking will handle sending sound over the paging speakers. I just need to know how to send the wav file to the Viking via the Port Adapter. So far, I know my wav file should be formatted as 8bit mono, and I need to send the "5" tone to open the Viking Pager's channel. [update] I am trying to figure out if I can use VLC player to stream to the ipaddress of the Port Adapter. So far I'm not having success with that, and don't even know if it will work. Windows Media Player has a streaming option too. I am thinking that since the Cisco Port Adapter thinks it is a sort of phone, that the only way this can be done is via SIP.

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• How configure 2 Lan cards in Windows 7/8 pc one to connect to Internet and other to Local Network

  - by Maharshi Raval

      I am about to install a dedicated VOIP server in our office. It is a 3CX pbx system on Windows 7/8 machine. The environment currently is a Windows SBS 2011 with 8 client machines. I want to use a dedicated broadband connection for the PBX (3CX) box, but the box also needs to be accessible in the local network as we will be using IP Phones and software IP phones. How configure two network cards on PBX box, so that one will be always used to connect to our SIP host over the Internet and the other will be connected to local network accessible from other client pc to connect to the pbx system. It must be noted that currently the Windows SBS 2011 acts as the Primary Domain Controller and gateway for all the client machines.     I cannot use a load balancer as it will conflict and cause issues within the current setup of our SBS2011 as it is also our Exchange Server. Any input is much appreciated. thanks in advance

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