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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2answers
934 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 ...
-1
votes
0answers
9 views

Cannot find Kamailio settings in Elastix 3

Kamailio Server is added to Elastix 3. I installed Elastix but cannot find any setting of Kamailio in the web interface. Where can I find the settings? Should I install Kamailio and Siremis ...
0
votes
1answer
920 views

Asterisk Real time does not perform register with SIP TRUNK provider

I am using astersik real time (dynamic). I have entered my SIP trunk details into the sippeers tables. However, the sip trunk does not perform a register with the SIP TRUNK providers servers as it ...
1
vote
0answers
15 views

Ucce 8.5 ICM - How to extract SIP Header in ICM script

We are collecting the sip headers from CVP Ops console. We are setting variables using the mid(call.sipheader,Startlocation,Length) If the string length changes, the values are not correct. Is ...
1
vote
1answer
6k views

Asterisk and SIP behind NAT

I am trying to Setup an Asterisk-Server to accept calls from a client in an other Network. The Server and the client are behind an NAT. I have already activated STUN on the client, but I am still ...
1
vote
0answers
32 views

Connecting P2P over NAT?

I started to explore the option of connecting with other using a p2p connection, so I coded a simple socket program in JAVA for android devices in which the user in which the users can share simple ...
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votes
0answers
55 views

Asterisk : SIP/2.0 401 Unauthorized : same phone , different results

First of all I have already taken a look at similar questions such as : http://serverfault.com/questions/392979/asterisk-sip-2-0-401-unauthorized or Asterisk 401 Unauthorized when trying to register ...
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votes
1answer
56 views

How to have dynamic extensions in Asterisk

I have recently swapped VoIP providers. So far so good, but I have ran into one little snag (and I'm honestly not really sure how to ask / Google this). With my old provider I could have inside my ...
0
votes
1answer
200 views

Linux based asterisk server, SIP softphone able to hear audio but not send audio to others

Running Ubuntu 14.04 LTS and Asterisk 11.7.0. The server is set up for SIP phones to use our SIP trunk via voip.ms/didlogic (two different services because we like international rates on didlogic, but ...
2
votes
2answers
10k 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 ...
0
votes
1answer
26 views

Make an authenticated SIP call via URI

I'm trying to redirect a phone number to a SIP address that's only accessible when authenticated. Given the following, how would the SIP URI look like (if possible) ? SIP Server: sip.example.com Auth ...
0
votes
2answers
108 views

Asterisk call should be with peer, but gets trapped and stays local

I'm working with Asterisk and Vicidial, trying to place outbound calls through a SIP "trunking" provider. The calls never connect, instead I hear either demo-instruct.gsm or invalid.gsm playing from ...
0
votes
1answer
33 views

how do I route Asterisk dials from serverA to serverB?

I'm trying to make an Asterisk server connect to a SIP provider (which offers PSTN origination and termination). I've been reading the great Definitive Guide, but am having a conceptual problem with ...
0
votes
3answers
99 views

The benefits of a proxy server before Asterisk

We are busy to develop an app to let users call eachother using webRTC. On this moment we have audio working, but video not. Now I hear that a proxy server can be the solution for this problem. But it ...
0
votes
0answers
52 views

Asterisk Auto dial out issues

I'm trying to make auto dial out in asterisk. I have created call file in /tmp directory and moved the file to var/spool/asterisk/outgoing/. It's not working. Please help me to resolve this issue. ...
0
votes
2answers
91 views

Asterisk WaitForSilence NEVER detects silence

I am trying to use my dialplan to play recordings with WaitForSilence to make sure it wait until the person is done speaking or the message is left on voicemail. However, it doesn't seem to wait for 5 ...
0
votes
2answers
98 views

Asterisk new PJSIP driver security option

I have a working asterisk environment, but I get a lot of unwanted traffic, like sip scanners of people who even try to call as a guest. I'm using res_pjsip, the configuration is stored in ...
0
votes
1answer
39 views

Asterisk Dialplan Siptrunk dial user input

I have an SIP Trunk on my Asterisk box. So my Asterisk box has one public number and 100 internal numbers. external number: 0030XXXXXXXX internal numbers: 2000, 2001, 2002, 2003 ... and so on I ...
1
vote
1answer
83 views

why do all softphones fail to authenticate after receiving 401 unauthorized from Asterisk?

When I attempt to connect to the Asterisk server from CSipSimple, over a Cisco router, on the same network, Asterisk logs show the attempted connection: <-------------> [Mar 23 20:01:34] ...
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votes
0answers
68 views

Asterisk using PORT NUMBER in FROM FIELD of SIP calls

I have an annoying situation with Asterisk 13. I setup a SIP trunk that works OK to receive calls. However, I can't make calls even though my SIP provider is working (I can place calls with a regular ...
0
votes
2answers
123 views

SIP Trunk for VOIP behind a SonicWALL

I appreciate this question is quite "out-there" but - has anyone had any success with VOIP over SIP behind a SonicWALL. We are having an issue with one-way audio inbound. Essentially, the person ...
3
votes
0answers
401 views

WebRTC on standalone asterisk - no audio

After struggling with Asterisk for WebRTC for a few weeks now, I decided to put my problem on this forum. My Problem is as follows: Im not getting audio from WebRTC to WebRTC clients. I work in a LAN ...
1
vote
1answer
208 views

Asterisk: execute shell command on server when call is accepted on SIP extension

I'm trying to configure asterisk to execute shell command for incoming calls - but only when the call is accepted. I've managed to setup extensions.conf so that command is executed when new call comes ...
9
votes
9answers
51k 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 ...
2
votes
2answers
136 views

nmap results to find open ports for SIP

I suspect that a firewall, or other security, on either the router, or on tleilax or doge is causing a problem with SIP calls. How do I establish that the connection is allowed and not being blocked? ...
3
votes
4answers
951 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 ...
4
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 ...
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votes
1answer
57 views

What type of switch should I get to implement a site with around 140 desktop machines, plus around 20 SIP phones? [closed]

We're moving to a new office and we will be needing to setup a network closet. I'm estimating that we will have around 140 ethernet ports and will be using CAT6a cabling for 1Gbps connections. I'm ...
0
votes
1answer
69 views

Hot to integrate sipJs and freeswitch?

I am trying to integrate sipjs with freeswitch. below is the config I am using var config = { // Replace this IP address with your FreeSWITCH IP address uri: 'sip:1002@***.***.1.170', // Replace ...
6
votes
1answer
98 views

What is “Cisco STG” and why would it dynamically replace a wildcard certificate on port 5061?

I have a lync client that is connecting to a Lync Edge server on port 5061. I get an invalid certificate error when connecting. When I run wireshark, during the TLS setup, and inside the ...
1
vote
1answer
234 views

Asterisk behind NAT sets wrong Contact Header

I'm using SIP with asterisk 13.1.0 behind a statically configured NAT. The servers private_ip differs from the public_ip, where I can reach it. I've already set these options in the sip.conf file. ...
0
votes
1answer
159 views

Asterisk Sip Server and “Screen Sharing” function

I'am newbie in asterisk, i followed the tutorials on the asterisk wiki, and installed the latest version (13) of the asterisk server. I was able to setup the voip now i wonder, there is a function im ...
0
votes
1answer
11 views

Does SIP offer a mechanism to initiate a stream and returning an error?

Imagine you are browsing to a HTTP URL that does not exists. You'll get a 404 which often contains an explanatory body. Now imagine you are calling someone using VoIP with SIP and use a non-existent ...
1
vote
2answers
346 views

Can I set up an “involuntary” conference call with Freeswitch?

I am trying to set up a SIP/RTP public announcement infrastructure. Basically there are several slave user agents that are configured to answer automatically, and a master UA which should be able to ...
2
votes
2answers
196 views

The benefit of different SIP and SMTP domains?

I was wondering if there was an advantage of having a different Lync SIP domain name from the Exchange SMTP domain name? I could see the disadvantages, but I'm trying to understand the real world ...
0
votes
1answer
122 views

Asterisks Dialplan Extensions

I'm playing around with a very simple asterisks setup. My asterisks server is 192.168.1.10 (Ubuntu 12.04), my 2 sip clients are 192.168.1.20 and .21 (both using ubuntu 12.04 and Jitsi as a sip ...
0
votes
0answers
61 views

Linux Redhat inbound NIC routing

I am running a SIP application on a Redhat Linux server with 4 nics on separate LANs. SIP runs on UDP port 5060. When I send a inbound packet to port 5060 NIC0, the packet gets delivered to the ...
0
votes
2answers
125 views

ASA is not forwarding packets to egress interface

We are seeing some very strange behaviour from our Cisco ASA 5505 running 9.1(2) We have a SIP PBX inside our network. It's got a bit of an odd configuration, it listens for inbound SIP requests for ...
0
votes
0answers
16 views

SIP UAC crash during a call

If a SIP UAC crashes in the middle of a SIP call, what is the expected behavior of the UAS and the remote UAC? Assume that a single UAS connects both UACs in the call. Is this scenario discussed in a ...
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 ...
0
votes
6answers
12k views

How can I stop SipVicious ('friendly-scanner') from flooding my SIP server?

I run an SIP server which listens on UDP port 5060, and needs to accept authenticated requests from the public Internet. The problem is that occasionally it gets picked up by people scanning for SIP ...
0
votes
1answer
408 views

One-way audio problems with SIP over AT&T

I have a customer who has an VOIP PBX connected to a Level 3 fiber connection. He has offices all across the country using different ISPs. Two of those offices use AT&T, both in different states. ...
0
votes
0answers
491 views

How to set up kamailio proxy server and route calls to Twilio?

i am trying to route all calls to twilio through kamailio proxy. with my config file, call gets connected and automatically drops after about 30 seconds. This is because ACK sent to twilio for 200 OK ...
0
votes
0answers
101 views

Kamailio doesn't restart with snmpstats module

I have Kamailio 4.2.0 running in a RedHat Entreprise Linux Server and I wanted to add some KAMAILIO-MIBS in order to monitor the machine. I went to the ...
0
votes
0answers
803 views

Asterisk 11.* TLS Configuration

I have installed asterisk 11.12.0 on CentOS (64-bit). When enabling the TLS support i get the following error WARNING[7620]: tcptls.c:673 handle_tcptls_connection: FILE * open failed! i tried ...
1
vote
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 ...
0
votes
0answers
201 views

VLAN setup Cisco SG300 and SIP gateway

Hopefully someone can shed some light here. I'm quite new to networking and not too sure if I'm doing the right thing here. We are setting up a new VOIP phone system with SIP gateway for our company. ...
0
votes
0answers
81 views

Asterisk SIP calls between remote L2L VPN sites

I am running into an issue trying to set up another remote L2L VPN site with VoIP running SIP and Asterisk. Each remote Site (site 1 and site 2) Can talk to the Asterisk server and any phones on the ...
0
votes
0answers
230 views

Dual WAN setup in Redhat/Centos Linux

I want to setup a Linux Server running SIP (freeswitch) applications with 3 NICs as follows: 2 of the NICs are WAN connections and the 3rd is a Private network for routing requests to other hosts on ...
0
votes
1answer
559 views

Configure SIP Trunk from Lync 2013 to Cisco Call Manager v6

I'm implementing Enterprise Voice for Lync, and I have to get Lync to work with a antiquated Cisco Call Manager version 6. The SIP trunk has been created but I get this error on my Lync servers: A ...