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Questions tagged [sip]

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

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On a PBX (Asterisk) determine which phones are connected

I am using Asterisk as my PBX. On my sip.conf I used to have this option enabled: [general] ... qualify=yes more info about qualify can be found in HERE. Thanks to that option I was able to tell if ...
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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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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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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
37 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
171 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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155 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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734 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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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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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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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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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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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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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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154 views

(ringinuse disabled) (Invalid) has taken no calls yet

I had a freepbx server in place. I setup a queue with static agents and ring all strategy. Now, suddenly no body in the queue will ring once the call routed to the queue. I check the queue and found ...

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