similar to: Does any one uses PortSIP VoIP SDK?

Hi, I'm setup the Asterisk 1.4.33 and try test it with the PortSIP SDK( www.portsip.com), but seems the video does not works. When I make the call from PortSIP SDK Demo to GrandStream GXV3140, it's working fine if no video codec selected. If make call with H.264 codec, the PortSIP got "503 service unavailable" response from Asterisk, why? Thanks -------------- next part

Hi all, I have a project for the 3G related, AMR and AMR-WB support. I'm using the client develop suite from the PortSIP(http://www.portsip.com), as their said support the AMR, AMR-WB with RFC4867. Now I have to setup a SIP server/SIP PBX in our Lab for test, does the Asterisk support these codecs and RFC4867 ? If no, there has any plugin to support this ? Also, any other Server/PBX which

Hi all, I'm new to VoIP, now we have a project that needs a PBX with client APPs. In our team we have argument for choosing PBX. By so far, we have following candidates: A: Open source 1) Asterisk PBX (http://www.asterisk.org) (with longest history that almost every one knows it, now the last version using the PJSIP stack) 2) FreeSwitch (http://www.freeswitch.org) (A lot people

Hi BJ, Further explanation about the 3rd party ecosystem question this morning Cory Andrews from VoIP supply was on the Voip-Users conference call last week. I asked the question - how much of VoIP Supply revenue is product versus applications - he said we don't sell any services such as ITSP hosted Asterisk so I replied that wasn't what I was thinking of and gave the example of Snap

Dear NUT developers, As you all know, Eaton is NUT biggest corporate fan, supporting the project through many contributions. Now, Eaton would like to provide its customers with a way to build their own applications to interface with Eaton devices, using the NUT code-base. To do this, in a smart and efficient way (ie limiting additions and modifications to the bare minimum), we (Frederic and

The solution you choose should be based on many factors which should include your business requirements, team's experience, your budget, growth expectations and more. You can choose Asterisk or Freeswitch as a platform and start building on that - but it is not simple and being new to VoIP you are likely to make mistakes. The "do-it-yourself" approach will some money initially, but

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, I made some filters on tc to match the VoIP traffic based on RTP packet RFC 1889. The only patern I found was the rtp version and the payload type wich identify the codec. The u32 rules are: - - match ip protocol 0x11 0xff -> UDP - - match u8 0x80 0xc0 at 28 -> RTP version 2 (two fists bits ''10'' of the byte) - -

Hi I am using ogg-vorbis win32 SDK 1.0 [ VC++ 6.0 , Windows XP ] I am trying to use the function "vorbis_analysis_headerout() ; " When I called the function in a stand alone small C progam it worked , but when I tried to use the same function as part of some big project it failed saying "Access violation in Vorbis.dll " and the application closed. Any clue ??? Thank you,

lol - yep when news of this first broke I thought thats actually a very good idea to have implemented, though it sounds the way Trixbox implemented it may have been unsecure. Maybe someone else can come up with a better way of implementing this. If the data was all randomised there's no harm in doing this; some basic infomration like; Hours of uptime Reboots Number of extensions Number of

The scenario: This is all done SIP with a VOIP provider (have to register to single IP) We have two inbound DID numbers / Accounts. We have to register each individually with the VOIP provider. I'd like inbound from each registered account (DID) to be able to come into a unique PEER or dialplan context. What matters is that the inbound call lands in the context of my choice. I've been

Hello All, I've been trying for some time to get Asterisk to register with a remote SIP gateway. I?ve recently managed to configure an SJ Phone to work with W2000 so know the configuration parameters work correctly. Asterisk doesn't authenticate properly and I notice that the authentication request appears different to SJPhone's. Do any tools exist to enable me to check these

We currently use asterisk 1.4.x with two Zaptel cards connected to POTS lines. So we make "outbound" calls from their softphones (using ulaw format), which go over a dedicated DSL line to the asterisk server in our office, which then converts the calls to POTS. This all works fine, assuming there aren't any unusual problems. It sounds as good as POTS on both ends. However, we

I need to decide on the best way to add a voip SIP or IAX client to a website. I'm thinking that I'd like it to be inline, like an aplet, on the page. I've got some asterisk servers running to connect up to, so the real challenge is finding an easily integrated open source client. Any suggestions from those who know? Jason

There's an error trying to build a "hello world" project out of the box when selecting LLVM from the list of compilers in iPhone SDK: http://arrelid.com/archive/2009/01/15/xcode-iphone-sdk-llvm-gcc-fixing-what-apple-forgot This is on XCode 3.1.2, iPhone SDK 2.2.1 . The blogger suggests what looks like a hack -- is there a proper solution to this? Is llvm-gcc just missing a header

hi i need to install the api sdk for xen cloud platform can u pls tell me how to install the api sdk and, the there procedure how to develop my own program using python , can u pls tell me the steps for the sample progam -- R.G.BABUKARTHIK _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

Hello there, I'm having a problem compiling llvm with vc++ 2005 and windows sdk6.1. It seems that the Platform SDK supplied with vc++2005 uses PSTR for the first param of PENUMLOADED_MODULES_CALLBACK64 typedef (dbghelp.h) and the windows sdk 6.1 (also windows sdk 6.0a that ships with vc++2008) uses PCSTR. The _MSC_VER macro guard for ELM_Callback(...) kinda kills the possibility of using

I updated the win32sdk build environment. To summarize: 1.) Added a "mkmak.bat" script to create, install and run an automation macro to perform makefile export from the command line. 2.) Updated makesdk.bat to utilize "nmake" and generated makefiles for performing builds. 3.) Added "sdk\build" which contains a complete build environment for building all the

Ok, so I've taken hold of this, and since this appears to be the only thing holding up 1.0.1, I have done what I think closes the loop...Based on the conversations here on the list I have made the following changes to the vc6 project files : - consistent runtime linking. They are ALL now using the Multithreaded static runtime. Hopefully this makes the generated DLLs the most portable. -

The Win32 SDK directory layout seems to need a small update in my eyes. Here's why: 1. Debug symbols get overwritten Problem: When the project files in the vorbis/win32 build, all temporary files of vorbis, vorbisfile and vorbisenc go into Dynamic_Debug (for the dynamic debug build as an example). The problems arise, when someone wants to debug the code. Since Visual C++ puts the symbols

How about this: http://www.debenhamsuffolk.co.uk/t/SDK-22-08-03.exe [1654 kB] Its all now in one directory (even the dlls). And I added a small doc saying how to set up MSVC 6 for vorbis. Need someone with MSVC 7 to say where the options have moved to (as they no doubt have :) ... Who wants this by the way? --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: