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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521 views

Avaya 9630 for SIP

Is there any way to make the 9630 Desktop Headset work just as a SIP IP telephone, without the extensive need for additional Avaya equipment. All of our infrastructure is Cisco and we may receive ...
2
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1answer
1k views

Adding more than one IP to sip.conf tcpbindaddr and udpbindaddr (Asterisk 1.8)

The defaults in the sample sip.conf under Asterisk1.8 contain udpbindaddr=0.0.0.0 and tcpbindaddr=0.0.0.0. I want to bind the incoming [foo] extension to udp:192.168.1.1/255.255.255.0, the outgoing ...
1
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1answer
146 views

SIP - linksys and no voice

I have a PC and a Linksys SPA3102. I have connected them directly with an etthernet cable (pc address 192.168.0.10, linksys address 192.168.0.1). An analog phone device is connected to linksys ...
3
votes
4answers
992 views

Asterisk Intrusion Prevention

Let me start of by saying that I'm a noob, and what I've figured out so far has only been by stumbling my way through it. I have Googled around and the solution may be out there already, but it was ...
2
votes
0answers
2k views

sending faxes over sip with t38modem and hylafax

I would like to send (and not to receive) faxes. AFAIK, t38modem + hylafax should work fine. After configuring hylafax to look for ttxT38-1,ttxT38-2,ttxT38-3, I'm running the following: # t38modem ...
2
votes
1answer
801 views

Lync 2010 SIP Trunk Providers

I'm playing around with a small Lync implementation in my home office and would like to try replacing my 3cx server with it. Currently I use callcentric as my VoIP provider, but based on another ...
2
votes
0answers
527 views

How can I record a call transfer with freeSwitch?

I am using freeSwitch to terminate calls that originate on the local network from an IVR system. I have it working with several different voip terminators, and I can record sessions successfully, ...
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2answers
942 views

Lync Server 2010 with Hosted VoIP PBX

We just deployed Lync Server 2010 in our organization and it is working great so far. The next step for us is to enable enterprise voice so that we can replace our telephones with service that is ...
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2answers
4k views

Asterisk Incoming Calls Don't Route to Extensions

We have just setup Asterisk 1.6 with FreePBX 2.8. We configured SIP Trunks with our Polish SIP Provider (a total of 3 trunks) in the system. Using a softphone registered to the only extension in the ...
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votes
1answer
884 views

SIP configuration with a Sonicwall firewall

I am working with a partner to enable SIP between a local SIP server and 2 Polycom phones located across a WAN connection. The server is located behind a Cicso ASA with SIP translation enabled, and ...
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votes
5answers
4k views

Command line SIP dialer

Is there a simple command line SIP dialer for unix which can connect to SIP server, make a call and play some media file (wav/mp3)? In ideal I would look like this: sip-dailer +1xxxxxxxxxx ...
13
votes
2answers
701 views

not able to register sip user on red5server, using red5phone

I start the red5, and then i start red5phone i try to register sip user , details i provide are username = 999999 password = **** ip = asteriskserverip And I got -- Registering ...
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votes
1answer
366 views

problem with red5phone unable to connect

i'm looking at red5 and red5phone to make it works. i am getting an error netconnection: Status: 1 rtmpconnection #phone _onStatus NetConnection.Connect.Failed netconnection: Status: 0 connection ...
0
votes
1answer
208 views

Test-service on Internet for testing incoming INVITE

I am trying to set up Asterisk at home. I think I'm having trouble configuring my firewall, so that inbound traffic is accepted, but I am not sure. I got the idea that, perhaps, there is a service ...
0
votes
1answer
304 views

Lost Unified Messaging (Exhcnage 2007) from Call Manager after external SIP Provider

ServerFault, After some really great trouble-shooting sessions with help from some people on Experts-Exchange... I've managed to get my 2621xm connected to BroadVoice's SIP Service through Call ...
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1answer
573 views

RAS server using SIP/h323?

We currently have a few modems (using standard POTS lines from the telco) connected to a server 2003 box running RRAS for remote access connections (mainly remote data pushes). We also have a Cisco ...
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vote
2answers
3k views

What is a SIP 'Gateway' and how is different from a SIP Proxy/Registrar?

Recently I started looking at SIP implementation for a future work. I was reading (Googling) about what SIP means and how to go about implementing a end-to-end SIP enabled VoIP network. What I did not ...
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vote
1answer
2k views

STUN server for windows

Where can I find a STUN server for windows? Open source would be preferred.
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1answer
2k views

How do I configure the SIP source interface on a Cisco router so that it sticks?

I manage a cisco router acting as a SIP gateway. In order to get it to register to the SIP provider, the connection needs to come from the right IP address. This is done with the following lines in ...
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2answers
537 views

How different is “Skype Connect” from “Skype for Asterisk”

We already use "Skype for Asterisk" channels to connect Skype to Asterisk PBX. Skype has a new offering called "Skype Connect" (or formerly "Skype for SIP"). What are the differences?
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2answers
2k views

Asterisk Server with SIP Account Routing to Cell Phones

I'm having trouble finding the exact documentation to do the following: I have a SIP Account. I want my Asterisk Server on a VPS somewhere in the United States to accept the credentials of the SIP ...
0
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2answers
780 views

Firewall issue with multiple SIP PROXY / REGISTRAR servers

We have a pair of Internet-facing SIP PROXY/REGISTRAR servers (for resilienced and load-balancing). When a SIP phone registers, it will be handled by one of the REGISTRAR servers (round-robin DNS) - ...
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2answers
868 views

SIP Server with dynamic public IP address and dyndns

Is it possible to run SIP server with dynamic public IP address and dyndns? The question is what happens when IP address is changed and DNS is not yet updated.
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votes
3answers
4k views

Unregister SIP UAC message

I've looked so much on the internet, but I could not find a any SIP unregister example, and when I search RFC 3261,3665 the word does not even appear, perhaps I'm searching for the wrong phrase. I ...
2
votes
1answer
1k views

asterisk/freeswitch in nat/no-nat setup

my current setup - i use bunch of sip hard-phones around few offices. all devices have two sip accounts configured - one on internal sip proxy [for calls between the branches], another - at 3rd party ...
2
votes
1answer
226 views

Router gets disconnected once I terminate my SIP application

Here is an interesting one, I have a SIP VoIP application which is able to register to the PBX server, and I can invite and see the user call on the callee end receiving an Invite, and on the caller ...
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vote
0answers
41 views

add presence for a remote user to a legacy telephone system?

we have a small call center that uses an old nortel phone system with analog lines. one of our sales people works from home so her calls do not go through the phone system. this creates a problem at ...
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vote
2answers
104 views

SIP server (“gateway”) for joining accounts

I have a phone supporting 2 accounts, but I need 4 accounts. Is it possible to install some sort of SIP server/gateway/proxy (on a linux server), that would register those 4 accounts and I would be ...
1
vote
1answer
2k views

Asterisk, IAXModem & Hylafax how-tos?

I'm trying to set up Asterisk and IAXModem to send faxes via T38 (Yes, I know I'm swatting a fly with a Buick...) However, since I'm trying to do something so small with a product so large, I'm having ...
2
votes
2answers
11k views

Asterisk SIP digest authentication username mismatch

I have an asterisk system that I'm attempting to get to work as a backup for our 3com system. We already use it for a conference bridge. Our phones are the 3com 3C10402B, so I don't have the issue ...
1
vote
1answer
212 views

What is SIP trunking?

Can someone explain to me in plain English what SIP trunking is, please? I've read about it on Google, but I don't really grasp it yet. Does it allow a VoIP call to be placed outside of the LAN? So ...
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1answer
1k views

T38modem manuals? [closed]

Are there any T38modem users here? I'm trying to figure out how to call T38modem with SIP. I've got everything except the --route option for receiving. I know my own phone number, but I am not sure ...
3
votes
1answer
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
votes
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 ...
4
votes
8answers
12k 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
votes
2answers
407 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 ...
5
votes
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 ...
0
votes
1answer
370 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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vote
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 ...
0
votes
2answers
119 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 ...
3
votes
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 ...
9
votes
9answers
54k 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 ...
1
vote
1answer
93 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 ...
1
vote
4answers
5k 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 ...
3
votes
3answers
254 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
votes
2answers
990 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
votes
1answer
390 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 ...
0
votes
3answers
7k 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
16k 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... ...
7
votes
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 ...