The Session Initiation Protocol (SIP) is an IETF-defined signaling protocol widely used for controlling communication sessions such as voice and video calls over Internet Protocol (IP). The protocol can be used for creating, modifying and terminating two-party (unicast) or multiparty (multicast) ...

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140 views

Does anyone know a good Website comparing all the PBX/IP telephone solutions?

I'm looking for a good website dressing a comparative analysis of the most popular telephony solutions resellers (Cisco, Avaya, Siemens, Microsoft, etc...). I already found a very nice work comparing ...
2
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1answer
1k views

FreePBX: Basic config?

I've just successfully set up an Asterisk+FreePBX install on an Amazon EC2 instance per this guide: http://voxilla.com/2009/10/15/voxillas-freepbx-in-a-cloud-step-by-step-1457 (I've also assigned it ...
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8answers
14k views

How can I simulate a modem over a VOIP connection?

I have a hardware device I need a server to dial into at regular intervals. The problem is I no longer have a POTS line or a modem in any of development computers, and all my production servers are ...
2
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2answers
446 views

VOIP Codec Preference

Do any of you have a preferred codec for VOIP traffic? I guess this is another case where the answer varies depending on use case, equipment, topology, etc... I'm trying to find an optimal codec to ...
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2answers
1k views

Ways to monitor SIP termination on an asterisk server

I have a nagios setup which ensures that SIP is responsive on my Asterisk server, that's straight forward. My question is, what kind of possibilities are there that the Asterisk server can actually ...
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1answer
387 views

Can i have a Asterisk IP PBX Server Behind ISA 2000

Is it a simple procedure to configure ISA Server 2000 to allow an Asterisk IPPBX connect to SIP provider. On asterisk forums they say the ISA has difficulties handling SIP, softphones that i have ...
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2answers
2k views

Asterisk terminating outbound call when picked up, sends 'BYE' message

I'm running Asterisk 1.6.1.10 / FreePBX 2.5.2.2 and I've got an outbound trunk setup. Everything use to work fine until recently (perhaps due to upgrade to FC12 or other things I'm not sure). Anyway ...
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2answers
123 views

What switch spec would i need to support 50 SIP extensions

Were a call centre and thinking of moving our phone system to a voip based setup. It seems most advice advises keeping this network seperate from the standard LAN, what switch specification would be ...
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6answers
2k views

What bandwidth would be required for 55 VOIP Lines And what type of Internet connection

In a call centre environment, what bandwidth would be required for 55 lines? This would be on SIP protocol, would we be best to use G729 codec as i know that our sip provider supports this? What type ...
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9answers
65k views

How can I check whether port 5060 is open in centos?

How can I check whether port 5060 is open in centos? How can I test if my linux has real a real IP address and I set no iptables blocking rules or is there any tools which I can run in my linux so my ...
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vote
1answer
96 views

Load balancer + CPU load manager to handle SIP connections?

I no longer have a landline and subscribed to a VoIP provider to have a landline phone number. I prefer to use a softphone (EyeBeam, the commercial version of XLite), but there are times when a call ...
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4answers
7k views

Workaround for blocking SIP port (5060) by ISP

I found out my ISP is blocking outgoing SIP port (5060) at home. I have a remote Linux server that I can use to listen on different port than 5060 and do the forwarding for the traffic. Not sure what ...
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3answers
262 views

Cheap VoIP Phone Recommendation [closed]

I'm looking for an inexpensive corded VoIP phone for my home office that supports SIP. I've been using an old Cisco 7905G with SIP firmware but I'm concerned about security considering that it's ...
2
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2answers
1k views

SIP and NAT routers?

SIP was not built with NAT routers in mind, and I'd like to get to the bottom of this issue to check what needs to be done on all devices so it works with NAT routers, and understand in what context ...
0
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1answer
393 views

Virtual PBX for automated attendant and SIP clients [closed]

I have a task to make our own virtual PBX for automated attendant and SIP clients. Scenario is following: We have our SIP account given from our voice carrier now I want to share these lines with ...
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3answers
8k views

How to connect a Fritz!Box as a SIP client to Asterisk

My planned setup is as follows: Analog Fax machine -> Fritz!Box -> Asterisk -> Telephone network My problem is, how do I convince the Fritz!Box to connect as a SIP client to my Asterisk instance?
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8answers
19k views

One way audio on SIP phone on one call direction.. where to look next?

Got 2 SIP gateways (ZultysMX250). Sip trunk between them, they are in separate locations, different subnets 192.168.XX and 192.168.YY connected by IPSec VPN. Some phones work perfectly. But mine... (...
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5answers
1k views

Skype-SIP gateway

Does anyone know if you can setup a Skype-to-SIP gateway on your network? I tried Uplink, which works wonders, but only on Windows and with Skype on the same machine... It would be awesome if you ...
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1answer
1k views

How to configure Gizmo5 on Trixbox/Asterisk

Does anyone know how to configure Gizmo5 for use on an Asterisk PBX? Specifically Trixbox if possible. For the life of me I just can't get this to work...
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5answers
999 views

Is there a stable cross-platform softphone? Or a Linux-only one?

We use an Asterisk based solution as our PBX, and although we've found hardphones work flawlessly, we have yet to find a reliable softphone. The basic requirement is that it allows multiple calls and ...