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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asterisk peer with SIP provider through proxy

I'm trying to make a asterisk server connect to a SIP provider (which offers PSTN origination and termination). I've been reading the great Definitive Guide, but it doesn't seem to cover the case ...
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248 views

can't register a soft phone to asterisk11

I have a VM (on oracle vbox) running Fedora17. I've installed asterisk 11 on it from sources. I've followed the wiki for installation (https://wiki.asterisk.org/wiki/display/AST/Creating+SIP+Accounts) ...
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2answers
928 views

Why are SIP calls via my server silent?

I have FreeSWITCH SIP server up and running. It has public IP and sits behind 1-to-1 NAT (it's Amazon EC2 instance actually). I can connect to it, make a call to other endpoint (namely, my android ...
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1answer
243 views

Provisioning SIP Phones over the internet

I have a few SIP Phones that are located of site and connect to my PBX over the internet to make calls. For some reason one of these phones has become unprovisioned. In my office phones get ...
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538 views

Asterisk dialplan context and label clarifications

I have been learning Asterisk dial plan for the past week.I have written down a simple IVR system with two levels of menu and an exit option.I have used concepts from different tutorials on the ...
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1answer
572 views

Configuration of a Cisco IP Phone 7916 and 7975g

Hi everyone, I would appreciate some help configuring my Cisco IP Phone 7975. In fact the phone works well with the SIP firmware (version SIP75.8-4-1ES2) and an Asterisk Server. Now I would like to ...
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241 views

LYNC - SIP “413 - Request entity too large” error to SUBSCRIBE requests

I'm currently trying to federate two Lync domains with one custom SIP proxy in the middle. The problem is that the front end server will reply "413 - Entity too large" to subscribe requests coming ...
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116 views

SIP me@domain.com vs (123) 456-7890

I'm trying to understand SIP, and one thing that keeps confusing me is phone numbers and dialing plans vs SIP URI's like me@domain.com. What is the difference? How do they coexist/work together? For ...
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72 views

What do I need to go from having a domain to being able to take a sipuser@mydomain.com call?

(Sorry if this question is a little basic or should go on a different SE site. Just let me know.) I'm trying to learn about SIP. The problem is that I'm at the point where I don't know enough to try ...
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2answers
2k views

How can I make Asterisk keep track of dynamic SIP agent statuses?

I am setting up a new server using Asterisk 1.8.11-certified4. In testing, we're seeing that agents dynamically logged into the queue will receive a second queue call as a call-waiting when ...
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1answer
205 views

Routing SIP between subnets

I've got the following configuration : 2 subnets (we'll call A and B) with IP phones in both. Internet access behind a NAT for each of them, and a routed VPN is set up between those 2 subnets. I am ...
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495 views

Lync 2010, Kamailio, & Trixbox 2.6.23 (Asterisk 1.4)

I'm having an issue trying to connect Lync 2010 phone calls with our trixbox PBX. I've gotten to the point where Kamailio seems to be functioning properly and acting as a bridge between TCP traffic ...
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4answers
12k views

How to stop registration attempts on Asterisk

The main question: My Asterisk logs are littered with messages like these: [2012-05-29 15:53:49] NOTICE[5578] chan_sip.c: Registration from '<sip:912@xx.xx.xx.xx>' failed for '37.75.210.177' - ...
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1answer
7k views

Asterisk SIP/2.0 401 Unauthorized

I'm running into a funny little issue with Asterisk 10.3, but it seems to be applicable to 10.4 as well. The server running Asterisk was relocated from a VPS to dedicated hardware, and now only 1 of ...
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0answers
299 views

sip endpoint receives direct sip calls but doesn't receive redirects from the registrar

I have an SIP endpoint, which I wrote, that receives sip invites, processes them, replies, and sets up the rtp session just fine when contacted directly. i.e. sip:user@[actual ip address of ...
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2answers
3k views

Asterisk: Authenticated SIP trunk

Let's say I have an Asterisk system with a bunch of connections: there are phones (who register itself with *) and providers (who wish to establish SIP trunks to put a lot of calls over, with ...
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1answer
259 views

NAT setup for an SIP Registrar/Proxy

I am currently trying to get the following scenario to work. Warning: I am a software engineer--not a network admin. I have various SIP endpoints (sip based video servers) on a LAN. On a Windows ...
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1answer
112 views

Setting up office phone system (small business)

I've been asked, within six months, to deploy a phone system. The exact wording was "a phone on every desk." They want inter-phone transfers, external calling, and even software IP phones for our ...
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3answers
3k views

Cannot connect softphone as a FreeSwitch Extension

Having successfully configured and maintained few Asterisk based installations, I have now been provided a task to configure FreeSwitch SIP server. ISO downloaded from ...
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6answers
4k views

How to record SIP traffic / calls for future auditing?

We have a VOIP (SIP?) phone system and have to record all calls for specific phones in the company. It may be required at a future date to listen to these calls for auditing. These phones happen to ...
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1answer
930 views

How to pass/process extension numbers while calling Freeswitch from outside SIP gateway?

Suppose I have Freeswitch, which has SIP gateway configured. So, the local users of this Freeswitch can call outside via <extension name="russia"> <condition field="destination_number" ...
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3k views

Setting up a SIP NAT proxy

We currently run a VoIP server using an upstream providers SIP proxy for our clients who are behind NAT. We now have the problem that we re ending the relationship with the upstream provider, and will ...
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4answers
5k views

Asterisk + SIP 404 not found

I want to make a small Asterisk server in my house. I installed asterisk on my Ubuntu and I use 2 computers, in order to connect to one another. when I connect I can see in Wireshark that registrar ...
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331 views

Asterisk 1.8, Dedicated SDSL but SIP Provider Inbound Channel heavily delayed?

We have a dedicated Asterisk 1.8 server running as a front end SIP server for authenticated and internal users only, using SDSL 2MB/2MB, in the UK. The provider of our internet also is the provider of ...
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4answers
3k views

Is it possible or advisable to virtualize a PBX system? How would one go about this?

I'm totally new to the world of VoIP and we are looking to move from our current provider to a solution we host ourselves, mainly because the current service is so unreliable. Unfortunately I know ...
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379 views

Asterisk IVR to Extension slow

I have a IVR set to pickup our PSTN lines and tell the user to enter the extension OR to press 1 for a representative. If the person enters an extension it transfers within 1 second , but if they ...
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1answer
308 views

Does the Mitel 3300 support direct SIP Trunking?

We have a Mitel 3300 PBX and we're looking to replace our PRI and BRI lines with SIP Trunking. I've seen that the 3300 supports SIP, but does it support direct SIP from a prospective ISP? Thanks
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497 views

how UAC should handle SIP 183 Session Progress

I have the following scenario: 2 UAC are trying to talk, via a remote SIP server (openSER/Kamailio 3.1.3) = client infrastructure. The UAC software was developed over a local test infrastructure ...
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1answer
1k views

Sip configuration for Elastix, for Terrasip (Provider) with Multiple DIDs

I have just purchased several DIDs with Terrasip I have configured my Elastix (freepbx) SIP trunk according to their suggestion: Please follow this template configuration. (valid for outbound/inbound ...
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1answer
221 views

how is the voip network working when server is outside nat and clients are inside nat?

I want to install an asterisk on a server with a public ip. The sip clients are behind different NATs. How is the signaling and the rtp packets (the voice) working in this case? Will rtp be peer to ...
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2answers
108 views

how to associate an ip with a name?

I have installed an asterisk on my private network(so it s behind a router ). Is there anu way to set the asterisk on a public ip though it's installed in a server with private ip? As Ive read it ...
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304 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
4k 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
1k 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
120 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
331 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
456 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
2k 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
194 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
2k 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
487 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
990 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
142 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
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4answers
918 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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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 ...
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1answer
780 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
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0answers
481 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
927 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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1answer
842 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 ...