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Session Initiation Protocol is one of the protocols used for controlling sessions (signaling) in internet telephony.

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transfer/bounce phone calls from xiaomi redmi 4x / SIM 1 to samsung note 3 / SIM 2 [on hold]

I would like to know if I can transfer the phone calls that I get from the SIM 1 to the SIM 2 using the VOIP technology. The SIM 1 is on the xiaomi red mi 4x and the SIM 2 on the samsung galaxy note 3....
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How do you fix “No Trust List Installed” error on a Cisco 7942G VoIP Phone?

I have a 7942G. I'm trying to connect it to Asterisk. Looking at Settings -> Status -> Status Messages, I see this: Error Updating Locale SEP############.cnf.xml No Trust List Installed I'm only ...
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28 views

Radvision VC240 - SSH and HTTP access

I've searched all the superficial web looking for the default login and password for the Radvision VC240, a videoconference terminal, for access via a SSH client. Need to find out if the VC240 stores ...
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2answers
103 views

Converting a P.O.T.S line to VoIP and simultaneously ringing a extension over VPN

Anyone with knowledge please help. I have analog phone service + internet at SITE 1. I only have internet service at SITE 2. When the phone rings at SITE 1 I would like the PC or an attached phone ...
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1answer
36 views

Ekiga under Windows: registered but calls fail

I have a VoIP account with sub-accounts. Sub-accounts are assigned 3-digit extensions and I can call between sub-accounts using regular desktop phones connected to my ATA. Now I am trying to set up ...
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1answer
144 views

twinkle configuration for Sangoma

I'm able to run AsteriskNOW virtualized on my local system. Also, I'm using Anveo on two different SIP hard phones. Anveo works fine, indicating that the network handles SIP fine. Now, how do I ...
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2answers
94 views

How to setup a voip server connected to a ISP voice line

My home fibre bradband ISP provides me with a Huawei HG8240 optical network terminal. This has one ethernet port and one phone port. The phone port is active, but unused at the moment. The ethernet ...
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611 views

Asterisk initiate call through sip notify message

I have a Yealink ip phone on my desk and I want to remote control it so that I can place the calls from my computer instead than by typing the numbers directly on the phone. This link explains how to ...
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42 views

SIP registration not converting to “YES”

I am trying to configure VoIP phone which is of grandstream GPX-1625. I have successfully set and configured the IP address in it, but the issue is SIP registration is not converting to "YES".I am ...
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21 views

Do VNCs (ie TightVNC, UltraVNC) and the like conflict with X-Lite (SIP)

We have a very small office consisting of 4 pcs with X-Lite installed in each for receiving calls? Since last 2 weeks, we usually encountered X-Lite "Account Failed to Enable" on every PCs from time ...
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20 views

Can't figure out why SIP Server invite isn't getting through to SIP client

We're using a SIP client application and SIP server for a Call Centre solution. While most of the calls come through successfully, quite a few drop out intermittently and we're trying to figure out ...
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1answer
59 views

How will a call get routed with incorrect LRN in SIP request?

I was reading about frauds in telecom. I read about an LRN fraud in which the fraudster would change the LRN in the SIP invite to a cheaper destination. How will the call get routed with an incorrect ...
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0answers
40 views

Asterisk: “Every one is busy” when using encryption

I am trying to setup a secure voip connexion using Asterisk server and Blink for client side with tls+srtp feature. sip.conf : [general] bindaddr=0.0.0.0 allow=all tlsenable=yes tlsbindaddr=0.0.0....
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1answer
28 views

IPSec doesn't hide IAX2 informations

I am trying to setup a secure truncked IAX2 connexion with IPSec between routers: But when I make sniffing attack with wireshark, all call's informations are showen ! IPsec is well configured and ...
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211 views

How to Asterisk server sizing?

I want to replace my current physical VoIP/SIP telephone gateway with Asterisk inside a virtual machine. Since I use a VoIP/SIP setup the dedicated telephone jacks/sockets (including main lines) are ...

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