-1
votes
0answers
11 views

Does exists a Sip server with different cost / profile based on number? [on hold]

Does exists a SIP server where it is possible to configure price per minute or second based on destination number ? For instance: 0041 12345677 -> 1 € per minute 0041 22222333 -> 2 € per minute ...
0
votes
2answers
31 views

Asterisk inbound trunk unauthorised without allowguest=yes

I am trying to configure an Asterisk (Elastix) box to receive SIP calls from a provider without requiring allowguest=yes to be enabled in sip.conf. Basically the SIP trunk provider uses multiple IPs ...
0
votes
1answer
95 views

SIP INVITE packet has WAN address rather than call manager LAN IP

I am using SIP between two subnets (192.168.3.0/24 and 192.168.30.0/24) each connected via VPN. I have a call server on 192.168.3.100, and two phones 192.168.30.118 (Ext. 3128) and 192.168.30.119 ...
1
vote
1answer
89 views

Freeswitch : Audio handshake failure 1 error while making call using Sipml5

I just installed FreeSwitch and successfully connected to server with user 1001. Details -> OS - Ubuntu 12.04 LTS 64 bits FS - 1.5.13b+git~20140614T114905Z~fc7a74905b~64bit OpenSSL - 1.0.1 chrome - ...
0
votes
0answers
30 views

How do I make Cisco phones use <user>@<domain> instead of <user>@<ip> in SIP messages

I am using Cisco 8000 and 9000 series IP phones and registering them to Opensips. The problem is that I want to communicate with the proxy using @ instead of @ so I can have multiple domains on one ...
1
vote
0answers
64 views

Werbrtc2sip crashing intermittently. Aborted (core dumped)

I am having issues with making webrtc2sip actually work. Webrtc2sip is now getting crashed intermittently during a call. I have seen this even when only one call is active in a queue. By the way, my ...
-5
votes
1answer
70 views

Normal phone number over ethernet, what is this service called? [closed]

Plug in phone to wall with ethernet cable Someone calls a toll free number Your phone rings What is the correct, vendor-neutral term for this service? Some terms out there are: Comcast calls ...
1
vote
0answers
70 views

freeswitch forbidden gateway error

i've stuck into a strange problem, i have Ubuntu 12.04 with static IP, installed Freeswitch with FusionPBX with easy install script from here http://wiki.fusionpbx.com/index.php?title=Easy_FusionPBX ...
0
votes
0answers
146 views

Soft phone not getting register to my asterisk server

I have config in my local lan asterisk for test. which is working fine. With the same OS version and packages on "AWS Cloud" I am setting as following. But it don't get register "error = sip 408 - ...
1
vote
1answer
88 views

Security, hardening and NAT for RTP and SIP

I decided to create a VOIP server and it has gotten more complicated then I thought it would. So in order to make my server more secure, I keep it behind a pfSense appliance. I use IP filtering to ...
0
votes
1answer
2k views

Cisco ASA 5505 - NAT or Port Forward for SIP / VoIP ver 8.4

I just had an NEC PBX installed that lets me use SIP trunks for VoIP services, My gateway is a Cisco ASA 5505 running 8.4 and I only have one public/static IP Addresses. So far, my trunks are ...
2
votes
0answers
45 views

How to set Mitel 5224 ringer volume in configuration?

We have several Mitel 5224 SIP phones, but can't seem to find out to make the ringer volume setting persist after it reboots. Can the ringer volume can be set somewhere in the XML configuration file?
1
vote
1answer
63 views

Which one VoIP SIP audio codec should I choose to high quality calls?

The quality of VoIP calls depends on the codec used for the transmission and on the bandwidth of Internet connection. I want to use best optimized codec for my Internet connection. I have to select ...
1
vote
1answer
140 views

Asterisk ignoring sip_domain part of URI

I am using Asterisk version 11.5. My problem is the following: I literally cannot make any outbound call. That's because, whatever I dial in any client, the sip_domain part gets ignored, it is ...
1
vote
3answers
308 views

Asterisk and SIP trunking, is this configuration possible?

I am new to asterisk and before I dive in, I just want to make sure that what I plan to do is possible/correct. My office will run an asterisk server and have both local and remote extensions. We ...
1
vote
1answer
580 views

VoIP Network Components: Difference between Signaling Gateway, Media Gateway, Softswitch [closed]

In my degree course of Software Engineering we had a lecture in Computer Networking about VoIP and the Session Initiation Protocol (SIP). There were many terms brought up for some network components ...
1
vote
2answers
102 views

What determines the length of RTP packets?

I have established a SIP session between two clients. I observed the RTP trace between them on wireshark. The 'length' column has a value of 172 for the RTP packets that flow from Client 1 to Client 2 ...
0
votes
1answer
213 views

What is the best option for SIP service redundancy?

From your practical experience, what is the best option for SIP service redundancy? a) one SRV record pointing to a hostname which has two A records b) two SRV records pointing to two different ...
1
vote
2answers
252 views

VoIP client permamently losing connection after ISP IP reset

I have a SIP client device connected to my ADSL router, so it is behind a symmetric NAT. My ISP resets the client's IP every 24 hours, it looks like this in the router log: Mar 26 12:36:56 daemon ...
0
votes
1answer
160 views

Custom ENUM/NAPTR DNS Server

I need to configure custom DNS server to handle ENUM requests for VoIP calls I have custom built DNS server in Java that handles ENUM requests for my company's VoIP services. ENUM is used to ...
0
votes
1answer
2k views

How to route/forward/connect incoming SIP call through other SIP provider to a phone number?

I have 2 VOIP SIP accounts. One is for a landline (incoming). The other should be purely outgoing and allows for calls to my mobile phone. Basically I want to connect the landline to my mobile phone. ...
1
vote
1answer
352 views

Asterisk Connection not working

I have installed Asterisk on VirtualBox by following the steps from here. Everything went ok until I got to navigate to an IP to configure Asterisk using FreePBX: 10.0.2.15 (Shouldn't be something ...
0
votes
1answer
562 views

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 ...
0
votes
1answer
255 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 ...
1
vote
1answer
386 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 ...
1
vote
1answer
108 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 ...
1
vote
0answers
432 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 ...
3
votes
4answers
8k 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' - ...
2
votes
0answers
246 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 ...
1
vote
1answer
201 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 ...
0
votes
1answer
108 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 ...
1
vote
6answers
3k 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 ...
0
votes
1answer
738 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" ...
3
votes
2answers
2k 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 ...
0
votes
3answers
3k 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 ...
1
vote
0answers
300 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 ...
6
votes
4answers
2k 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 ...
2
votes
1answer
209 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 ...
0
votes
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 ...
0
votes
0answers
222 views

Is it possible to split Asterisk SIP signaling and voice traffic?

How do I configure Asterisk to send SIP signaling to one IP address and the RTSP stream, voice traffic, to another address?
0
votes
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 ...
1
vote
2answers
905 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 ...
2
votes
1answer
804 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 ...
2
votes
1answer
714 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
364 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, ...
0
votes
2answers
862 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 ...
0
votes
1answer
289 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 ...
0
votes
1answer
870 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 ...
2
votes
1answer
834 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 ...
1
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 ...