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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209 views

What is a SIP B2BUA, and how is it different from a UA?

I've been reading some stuff SIP related, and I'm confused, about what is a SIP back-to-back user agent(B2BUA) and a 'normal' user agent? Can anyone explain the differences? From what I read I ...
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2answers
3k views

Asterisk - inbound calls from SIP DID trunk “rejected because extension not found”

I have a DID with DIDforSale that's pointed at my Asterisk server. When I call it from my landline phone I get the AT&T disconnected line recording. The Asterisk CLI shows the error message: [Oct ...
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2answers
882 views

Asterisk DID incoming rules =s

So I have a 10 way SIP trunk from a UK provider. Calls in and out work fine, but as I have multiple numbers for different purposes I set up the inbound rules for each number but they never work, only ...
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1answer
111 views

SRV records not populating correctly

I have a lync box with a SIP SRV record that isn't working for a significant portion of clients, despite their ability to resolve the FQDN of the edge server. What could cause inconsistent resolution ...
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1answer
263 views

Can a Cisco AS5350 be used as a Session Border Controller (ip-to-ip)?

And how can it be configured as such? We don't need transcoding or anything like that, just something between our switch and the internet.
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1answer
399 views

SIP ( Asterisk ) CoS solutions with no VLAN's?

I have a Asterisk box actually two running pure asterisk. No GUI i.e. FreePBX, TL etc. The asterisk box is connected to a an EX4200 VC Stack, the EX4200's then connect to our headquarters via a Qwest ...
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1answer
1k views

What would cause SIP traffic to be seen going into a switch but not coming out?

Background I have been struggling to get my SIP phones to register behind a brand new router and switch in our brand new office. Our PBX is hosted offsite. I have worked with our provider to attempt ...
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2answers
182 views

Routing in OpenSIPS

How to modify under OpenSIPS1.6.4 the default routing to switch the From: field in every instance of incoming SIP packets to a predefined string? According to the documentation the $fd pseudovariable ...
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2answers
1k views

Trixbox: external SIP with no sound

I have a trixbox server and every works find except the external SIPs. Inside net all sound goes fine, but if I use a SIP phone outside the net, I can connect, I can receive calls but I there is no ...
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1answer
403 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 ...
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1answer
793 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 ...
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1answer
130 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 ...
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4answers
720 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 ...
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0answers
1k 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 ...
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1answer
712 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 ...
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0answers
357 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
861 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
2k 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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0answers
933 views

SIP through VPN has one-way audio

I'm struggling with a SIP installation with a remote user. THe phone boots up, provisions itself from the (remote) TFTP server, performs SIP registration with the PBX, and I can make calls to the ...
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1answer
734 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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5answers
2k 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 ...
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2answers
640 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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1answer
341 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 ...
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1answer
193 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 ...
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1answer
287 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
483 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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2answers
1k 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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1answer
1k views

STUN server for windows

Where can I find a STUN server for windows? Open source would be preferred.
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1answer
849 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
510 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
765 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 ...
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2answers
422 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
655 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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3answers
3k 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
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1answer
822 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
207 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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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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2answers
94 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 ...
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1answer
1k 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
8k 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
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1answer
192 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
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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 ...
1
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1answer
942 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
8k 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 ...
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2answers
321 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
347 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
117 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 ...