Voip Server

Hi all,
Is it possible to connect Voip server via blackberry 8300 .
if possible, How to connect and make calls through it. Any samples...
With Regards
Karthik.J

you can probably opt for wireless lan solution for mobile access and VPN for remote access
http://www.cisco.com/en/US/products/hw/routers/ps272/products_configuration_guide_chapter09186a008022b30b.html
http://www.cisco.com/en/US/netsol/ns340/ns394/ns171/ns125/networking_solutions_sub_solution_home.html

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E72 update of VOIP server

i have problem in voice chat or voice call in skype. I want to ask all of you that plzz tell me how to updat VOIP serv. Some days before i had voip ver. 1. But now my voip has updated to 2.02. Tell me the way to install it.

Did you installed Fring? This happens when you have Skype or Nimbuzz installed, and you install Fring afterwards. Nimbuzz and Skype uses VoIPAudioServer 1.00, but Fring uses 2.x. This libraries are mutually incompatible. If you want to solve this problem, uninstall Fring and Skype, and install Skype again. I don't know any other method to revert back to 1.00

Device compatibility of Avaya VoIP Server with NEC Softphone/IP Phone

For NEC is UNIVERGE SV8500 and NEAX7400 IMX, and for AVAYA I still don't know which avaya media server can replace those PABX from NEC.
Do you have any suggestion?

Hi,
I'd like ask about device compatibility between avaya and NEC. Currently, I have a VoIP network infrastructure of NEC as a backend & end device.
Later I intend to replace the NEC with Avaya as a backend, but still using NEC softphone/IP phone as end devices.
Is Avaya media server compatible with NEC device?
Thanks,
This topic first appeared in the Spiceworks Community

How to make a good voip server +applet ?

Here is my applet wich I made http://84.244.8.225/test.html
It works a bit
Here the code:
http://84.244.8.225/javamic4.txt
Any ideas/suggestions to make it better...
(for example how to make a good server (windows) ?)
Thx!
Edited by: FrederikPot on Sep 24, 2007 3:01 PM
Edited by: FrederikPot on Sep 24, 2007 3:03 PM
Edited by: FrederikPot on Sep 24, 2007 3:07 PM

Hello Thanks for your replys and sorry for being so fussy about this.
I have now seen that one of the options for exporting from Premiere 6.5 is Adobe MPEG encoder. So Do you guys think it would be a good idea to do it with that, Or Would it be better AVI or DV AVI compressor from Premiere before making the DVD?
I have the Nero Vision 4.9.6.6 But I may be able to look for another one if you think it is better...
Thanks a lot for your advises...

Loosing contact with voip server

When the screensaver goes active on my phone it seemes to drop the contact with the sip server. When looking on the icons of the phone it seemes to have contact, all the icons that shuld be on the display are ther. I can make calls but don't recive them.
When I have made a call it works to recive calls untill the screensaver goes on again.
Anyone know if there are any solution to this?

According to System Status there have been problems with Connect during last week. Who can tell the effect of....
Incident #2013110601 Resolved
11:09 PM - 11/6/2013
We have resolved the incident affecting the following:
Performance
Adobe Connect - Meeting (NA3, NA4, NA5, NA7, NA10, NA11)
Incident #2013110601 Reported
6:22 AM - 10/26/2013
We are investigating a reported incident affecting the following:
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Link cocomo with a SIP VOIP-server (Asterisk)

any ideas/suggestions about communicating between cocomo and
a sip server (asterisk based) ?

Hi Leonard,
I was interested in doing something similar, and used Ribbit Flex SDK to make calls directly from my Flash app to a phone. It worked OK (although connecting the call was rather slow), but if the Flash app user did not wear a headset then the person on the phone would have an echo (as the sound from my laptop speakers was being fed back into the mic). This is because the Flash Player does not have Advanced Echo Cancellation built in (Skype, Adobe Connect and others have AEC built in).
If you'd like to vote for AEC to be built into Flash Player, the Adobe Bug is written up here (currently has 161 votes, open over a year but no comment from Adobe). This is a shame, as I think with AEC lots of people could write some killer apps that used the Flash Player.
thanks
Mark

VoIP connection to company server

Hi there,
When reading the specs of the IPhone 3G I was interrested in its capability to connect to VoIP servers. I noticed there are lots of apps around which let it connect to skype like services. In other words: services for which you need an account and you place calls via the Internet.
This is not what I am looking for.
What I need is a replacement for the Nokia E series.
With the Nokia E series I can make a VoIP profile connecting to our company VoIP server. The profile becomes an intergral part in the phones methodes of placing and receiving calls.
This way I can place and receive calls through our company VoIP server when in range of our Wifi network. When out of range it switches to the network of the telecom provider (like Vodafone).
Is this posible with Apple IPhone?
Peter

Seems it is called ISip nowadays.... looks good.
Now all I need is an IPhone to actually test it .... thanks

Voip application for windows phone 8.1 fails store submission

A voip application with the new 8.1 architecture based on a separate process for the voip server is not accepted by the store.
Voip architecture is based on 8.1 sample: https://code.msdn.microsoft.com/windowsapps/ChatterBox-VoIP-WP81-64921ad4
Errors during submission:
3011: The package is missing RpalManifest.xml. Update it and then try again.
1028: The native API api-ms-win-core-processthreads-l1-1-2.dll:ExitProcess() isn’t allowed in assembly doubango.BackEndServerHost.exe. Update it and then try again.

Update. I am currently working with the sample team to update the sample. We found the problem to be in the way the BackEndServerHost VIsual studio
project was created. While we work on re-creating that from scratch, testing, and re-publishing I wanted to post what we have done so far in case anyone
wants to move forward.
First there are a couple bugs in the sample, maybe you worked around those already in your development but in case not here they are.
In the backendaudio.cpp BackEndCapture::InitCapture I changed the code:
if (SUCCEEDED(hr))
WAVEFORMATEX format = {};
if (SUCCEEDED(hr))
FillPcmFormat(format, m_pwfx->nChannels, m_pwfx->nSamplesPerSec, m_pwfx->wBitsPerSample);
m_sourceFrameSizeInBytes = (format.wBitsPerSample / 8) * format.nChannels;
hr = m_pDefaultCaptureDevice->Initialize(AUDCLNT_SHAREMODE_SHARED, 0x88140000, 1000 * 10000, 0, &format, NULL);
The m_pwfx was already retrieved via GetMixFormat earlier, just use that. Don’t create a custom one.
In the BackEndAudio::Stop routine change the order of cleanup:
if (m_pDefaultRenderDevice)
m_pDefaultRenderDevice->Release();
m_pDefaultRenderDevice = NULL;
if (m_pDefaultCaptureDevice)
m_pDefaultCaptureDevice->Release();
m_pDefaultCaptureDevice = NULL;
if (m_pCaptureClient)
m_pCaptureClient->Release();
m_pCaptureClient = NULL;
The next step is to basically delete the BackEndServerHost project and create a new one with the following steps.
1) Add a new C++ Dirext X App (Windows Phone) to the solution and name it VoipBackendServerHost NOTE: using "." in the name causes problems, so don't use the same name as the original sample.
2) Remove the Common and Content Folders
3) Remove App1Main.cpp and App1Main.h
4) Contents of app.cpp should be:
#include "pch.h"
#include "App.h"
#include <windows.applicationmodel.core.h>
#include <wrl.h>
#include <string>
#include <stdexcept>
using namespace ABI::Windows::ApplicationModel::Core;
using namespace ABI::Windows::Foundation;
using namespace Microsoft::WRL;
using namespace Wrappers;
HRESULT __cdecl MyGetActivationFactory(_In_ HSTRING activatableClassId, _COM_Outptr_ IInspectable **factory);
class GetCustomClass : public RuntimeClass<RuntimeClassFlags<RuntimeClassType::WinRtClassicComMix>,
IGetActivationFactory,
CloakedIid<IAgileObject >>
public:
IFACEMETHODIMP GetActivationFactory(_In_ HSTRING activatableClassId, _COM_Outptr_ IInspectable **factory)
return MyGetActivationFactory(activatableClassId, factory);
private:
HMODULE m_hMod;
[Platform::MTAThread]
int main(Platform::Array<Platform::String^>^)
HRESULT hr = Initialize(RO_INIT_MULTITHREADED);
if (FAILED(hr))
throw std::runtime_error(std::string("Failed to Initialize(RO_INIT_MULTITHREADED), HRESULT: ").append(std::to_string(hr)));
// Scoping for smart pointers
ComPtr<ICoreApplication> spApplicationFactory;
hr = GetActivationFactory(HStringReference(RuntimeClass_Windows_ApplicationModel_Core_CoreApplication).Get(), &spApplicationFactory);
if (FAILED(hr))
throw std::runtime_error(std::string("Failed to GetActivationFactory(RuntimeClass_Windows_ApplicationModel_Core_CoreApplication), HRESULT: ").append(std::to_string(hr)));
ComPtr<IGetActivationFactory> spGetActivationFactory = Make<GetCustomClass>();
spApplicationFactory->RunWithActivationFactories(spGetActivationFactory.Get());
Uninitialize();
5) Right click and bring up Project properties. go To Linker->Input.
6) Set the Configuration and Platform dropdowns to change all configurations all platforms at the same time
7) Choose Edit for Additional Dependencies and set to this:
$(SolutionDir)$(PlatformTarget)\$(Configuration)\BackEnd\PhoneVoIPApp.BackEnd.lib
WindowsPhoneCore.lib
RuntimeObject.lib
8) Under Cofniguration Properties->General Set the Target Name to PhoneVoIP.BackEnd.HostServer and Output directory to $(SolutionDir)UI
9) APply the settings. Then click OK
10) Build the Proxy Stub
11) Build the Agents
12) Build the new Project
13) Expand the UI (Windows Phone Silverlight 8.1) Project and find the PhoneVoIP.BackEnd.HOstServer.exe and remove it.
14) Right click UI and Choose Add Existing and add the PhoneVoIP.BackEnd.HostServer
This should hopefully get you moving forward. Please test and make sure your application is working. The end result should pass WACK now. Please let me know if there are any issues. It is a lot of steps and I hope I didn't miss any.
Bret Bentzinger (MSFT) @awehellyeah

Home Hub 3 Security exposure allowing VOIP (SIP) t...

OK its a bit tech but BT don't seem interested but it worries me!
I have a HH3 configured with with Firewall set to Default (to block unsolicited incomming traffic), DMZ disabled (so no default routing of inbound connections) and UnNP off (So no dynamic port opening stuff). Also, the internal DHCP server has been disabled and the internet network is not on the default IP range.
This configuration should block ALL inbound traffic. I.E. traffic that originates from the public Interenet. Packets that are replies to data sent out, will be allowed through.
I have an internal VOIP server running Asterisk which connects to SIPGate on the Internet and SIP phones in the house.
My Asterisk server is logging inbound SIP connections that have their source IP address as the Public IP address of the HH3. Somehow, inbound SIP connections are getting through the HH3 and then hitting my Asterisk server. The SIP connection then attempts to call an number in Israel (00972592653787) a few times (with different call prefixes). The Asterisk server is correctly configured to drop these calls. BUT it should never get them in the first place!
So the question is, why is the HH3 allowing these connections through in the first place?
And what else might it be letting in that I haven't spotted?
PS - Latest firmware running 4.7.5.1.83.8.94.1.11 (Type A) 29/12/12
Anybody seen this or know the reason?
I'll guess not, so just say hello so I know you've seen the post :-)
Richard.

Part of a setup such as Asterix/SIPGate is the use of a STUN server. The purpose of STUN is to enable incoming connections without the need for configuration of firewalls etc. It is needed if you are to get any incoming calls, whether malicious or not. It is only when a call arrives at Asterix that a decision can be made as to whether it is acceptable. If you really don't need to accept any incoming calls to the server, you could remove the STUN configuration.
http://en.wikipedia.org/wiki/STUN

VoIP SIP Client

Now I know I'm not the only one seeking the ability to write a snazzy little Flash Application for SIP access. I am, however, the only one willing to start a neat little open source project to help, people like me, use SIP to its full potential.
So here it goes, I'd like (and hope) to write a SIP client for two projects. The first is a for-profit endeavor for the company I work for, and the second is a not-really-for-profit webservice I own that would have its customers benefit from on-site VoIP call. Since they both require the same thing, I'm willing to open it up as an open source project to allow it to continue growing.
The idea is this:
A list of contacts (unassociated with the SIP client) have phone numbers or VoIP extensions. Click on the call button will activate a snippet of javascript that will communicate with the Flash SIP Client, sending instructions to dial. On the bottom of the screen the flash client will begin dialing, or ask for permission access cam and mic then dial. The sip client must also be able to receive inbound calls being projected to it from my VoIP server. Do all the things needed for a SIP client to do, such as hold, conference, mute, answer, and hangup calls.
So I'm thinking the Flash Client will be a thin, virtually no interface, that will communicate with the onboard javascript. This thin-layer of flash would be invisible, to everyone except a developer maybe looking at trace information. Javascript would instruct the flash client to connect to SIP, establish calls, hangup, etc. Flash would also send instructions from the VoIP server to javascript such as incoming calls, text, sign off, messages... etc.
So what I'm asking you as the community here at adobe for is somehelp. I've been Googling and found not to much helpful in this area.
I know flash can communicate to javascript. I know Javascript can communicate to flash (Yahoo does it for their IM client). I know Flash can communicate with VoIP servers via sip. What I don't know is where to start writing this client. I herd that AIR has an API for such things, that maybe even flash has a SIP/VoIP API, where is it? If I have all this information I'll start a nice open source project on sourceforge, github, or something like that where I hope to get input and offer the very thing I'm looking for to people all over the net. Expand VoIP capabilities so we can truly see inexpensive solutions popping up over the net. With the advent of VoIP integrated telephony should not be an expensive effort.
P.S. - Nothing says that when you sign in that the Flash API and Javascript components don't communicate with each other through another window to keep the SIP client connected. Also nothing says that when the main windows closed it doesn't disconnect the sip client and the user goes on his/her merry way... This IS possible, I just need to put the larger peices together.
Let me know!

I am definitively interested. Will you contact me at radoslav <At> everestkc.net.
What about this: http://flashphoner.com/
Is this totaly open source or you need to pay to be able to check out the code?
Rad

Has anyone seen this voip soloution yet?

Has anyone seen this voip soloution yet advertised on Nokias website?
if you take a look at this page it looks like nokia and avaya are getting ready to release a new voip client that can work on series 60 and series 80 phones. it looks prett cool, but i have no idea if this will look at any voip server, or a companies voip server.
i am guessing that this will be worth a look once it is released, after all it seems the mail for exchange client is now working on Nseries phones which is cool.

Hello,
As I promised, we checked in AVAYA center with AVAYA engineer how E60 works with AVAYA SIP server. We found that there is a problem to begin usage of IP telephone service. E60 always sad that craeted profile "Not registered". And don't like to act with WLAN network although WEB-browser works fine in the same time with the same AP-7 and WLAN settings.
AVAYA took a timeout for identify what's a problem with E60. Some terms on Nokia setting don't correspond to AVAYA terms.
I hope that it will take not a lot of time for AVAYA.
I'll let you know about the results.
Andrey
6150, 8310, 2100, 6310, 6310i, 6230, 6600, 6630, 6700, E60, N80, E61, N93
N95, E61i, E71, E72,8600, 5800, N8

Nokia N82 VOIP problems

I have VOIP set up on my N82.
It works perfectly if I put the IP address of the VOIP server in, and remove it for the realm and proxy. However, I need to do this every hour or I get 'registration failed'
Any ideas? It's becoming really annoying!

TheTaZ wrote:
Hello,
I want to use E51's VoIP function but i don't have any "Internet Tel" pictogram.What can I do.Do I need to update my fireware or there is any program?.
Thanks in advance,
TaZ
What do you exactly mean by saying "i don't have any "Internet Tel" pictogram"?
The built-in SIP VoIP client is there, and it is at Rel. 2.3 level.
You first have to choose a SIP provider, sign up for his services, buy some credit, ask for his typical settings, add your credentials, and create a SIP profile.
You will then be able to define an Internet Telephone call profile.
To attain better control over parameters of the SIP profile that are not accessible to the user, it is recommended to install the proper SIP VoIP Settings Software:
http://www.forum.nokia.com/info/sw.nokia.com/id/d61bd0ec-1304-45dd-9283-63d631cb86b1/SIP_VoIP_Settin...
For the E51 - select the 2.x version (approx. 88KB)
Did you read the E51 Guide?

Nokia E61 Transfer VoIP Call Bug

Hi! I have an Asterisk 1.4 VoIP Server and found two ways to transfer a call ... Unfortunately both ways are not working properly ... 1. Automatic transfer When you have an active call and choose Options -> Automatic transfer and then enter a number you get the "Waiting for acceptance to transfer call" message. The call is not transfered until you hang-up. Question: Is this normal or should the phone hang-up automatically? 2. New call and then transfer When you have an active call you can choose Options -> New call -> Internet call. After entering the number ... a. the Standby application crashes b. the call gets transfered and the Standby application crashes and displays a light bubble with the text "call1". In both cases I can only restart the phone by pressing the power on button. Does anyone know the solution to my problems? TIA, Mike

Hi! I have an Asterisk 1.4 VoIP Server and found two ways to transfer a call ... Unfortunately both ways are not working properly ... 1. Automatic transfer When you have an active call and choose Options -> Automatic transfer and then enter a number you get the "Waiting for acceptance to transfer call" message. The call is not transfered until you hang-up. Question: Is this normal or should the phone hang-up automatically? 2. New call and then transfer When you have an active call you can choose Options -> New call -> Internet call. After entering the number ... a. the Standby application crashes b. the call gets transfered and the Standby application crashes and displays a light bubble with the text "call1". In both cases I can only restart the phone by pressing the power on button. Does anyone know the solution to my problems? TIA, Mike

Cisco ASA 5505 - Port Mapping

Hi, I'm new into IT and I was wondering if somebody could help me set up Port Mapping. Here's my scenario -
We have set up an Asterisk VoIP server that uses UDP port 5060 and another port range, and we want any public incoming connections destined for our Asterisk server on Port 5099 to be translated at the firewall to go to our Asterisk server on port 5060. I have been using ASDM 6.4 but theres no easy way to do this (as far as I know, and why I've came here looking for an answer).
We have currently just left port 5060 open to the public (so our home workers can use our phone system) but really want to get this sorted ASAP due to SIP Bots that look for ports like 5060 that are open!!
Any help would be greatly appreciated and if anybody needs anymore information just ask!!

Hi,
You need to have a NAT rule set for port-forwarding to make as per your requirement..... I will give you cli based configuration example....
If your ASA is running with pre-8.3 version:
static (inside,outside) tcp interface 5099 192.168.1.10 5060 netmask 255.255.255.255
If your ASA is running with post-8.3 version:
object network SERVER-01
host 192.168.1.10
object network SERVER-01
nat (Inside,Outside) static interface service tcp 5099 5060
Regards
Karthik

ASA 5505 Site-to-Site VPN dropping at end of lifetime

I have 4 ASA 5505's with Site-to-Site IPSEC VPN tunnels built between them. One of the tunnels stays up just fine but the other 2 drop at the end of the SA lifetime for a period of time equal to 10% of the SA lifetime.
Orignially, I had the the lifetime set to 1 hour and the tunnels would drop for 6 minutes. I changed the lifetime to 8 hours (480 minutes) and they dropped for 48 minutes. I've gone over the configurations and the only differences I can find is that the sites where the tunnel drops have the outside interface forwarded to an VOIP server and all ports but SIP blocked.

Can you post the configs?

Voip Server

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