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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27 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 ...
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6 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 ...
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0answers
11 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 ...
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0answers
34 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. ...
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2answers
27 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 ...
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0answers
45 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 ...
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1answer
27 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 ...
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2answers
49 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 ...
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0answers
13 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 ...
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1answer
99 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. ...
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0answers
245 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 ...
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0answers
45 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 ...
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votes
1answer
47 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 ...
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0answers
272 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 ...
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0answers
124 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. ...
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0answers
51 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 ...
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0answers
162 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 ...
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1answer
387 views

Cisco ASA 5500 - SIP ports other than 5060

Is it correct that the SIP inspection in the ASA 5500 firewalls only kicks in for traffic on port 5060? There is some hint at this, while not 100% definitive, on Cisco Docs - ...
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0answers
123 views

sip webrtc call to softphone 488 Not Acceptable Here

I'am using resiprocate stack on the server, it's working fine for softphone-softphone, browser-browser and softphone-browser calls, but when I'm trying to call from browser to softphone it failes with ...
2
votes
0answers
119 views

RTP analysis - Discerning ptime (packetization time) for a given VoIP packet capture

I would like some help on the subject of an automated way of discerning the average packetization time (ptime) of a VoIP call's packet capture. The reason I am not depending on the value in the SDP ...
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0answers
54 views

Snom sip phone strange behavior

Didn't know where to ask this question, if someone can point me out the correct place, i'll be grateful I will be grateful, if someone will be able to help me with the following, I have around 200 ...
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0answers
73 views

Combining Elastix And Zultys PABX

have anyone integrated Zultys MX250 with Elastix? I got some problem with it. Before, my PABX is Elastix, now i want to upgrade using MX250 but keep Elastix as gateway for MX250. But i can't call out ...
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votes
1answer
768 views

route incoming asterisk sip calls - fake auth rejected

we are trying to handle incoming sip_calls on our asterisk server, but somehow we always end up getting either 403 or 603 (which should be the same?) what should happen if it works: user calls ...
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2answers
735 views

Allow specific SIP connection FreePBX

Hey I am using FreePBX with firmware: 5.211.65-14 and service pack 1.0.0.0 On this asterisk server I have everything up and running, but inbound phone calls might be rejected: ...
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1answer
122 views

sip unreachable after changing routes using linux router

SIP server was unreachable after changing routes. I have 2 ISP ,ISP 1 is my default and ISP 2 is my back-up. I have a dialer behind linux router linux. Linux router has 3 ethernet cards. Router ...
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0answers
67 views

SIP/2.0 400 Sip message body is shorter that is indicated by the content length

During SIP communicating between my application, Freeswitch, and nextiva trunk, I am getting the following error message from 208.73.146.95 (Nextiva server): SIP/2.0 400 Sip message body is shorter ...
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2answers
101 views

SecAst: Failing to Ban IP ''

To Generation D: This was the setup at time this issue was observed: secast-1.0.1.0-x86_64-ub12 on Ubuntu 12.04.4 Server LTS with Asterisk 11.10.2. The following events were captured and observed in ...
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votes
2answers
293 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 ...
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votes
0answers
32 views

udp not working with sipp

I am doing some end to end testing using SIPP on RHEL 4.2. Both my client and server machiens are running RHEL 4.2. When I try to issue the command : sipp -t tn -sn uac <server_vip> -i ...
0
votes
1answer
232 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 ...
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1answer
312 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 - ...
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votes
0answers
165 views

Iperf UDP listening on used port

I get WARNING[9157] chan_sip.c: Retransmission timeout reached on transmission error on asterisk and trying to troubleshoot the problem. Now asterisk is listening on 5060 port: udp 0 0 ...
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votes
1answer
672 views

Asterisk server running on loopback address instead of local ipv4 address

I installed Asterisk on my local Ubuntu 12.04 machine. After starting the asterisk [asterisk -c] I ran the command netstat -nlpa | grep asterisk and the output shown is this -> tcp 0 ...
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0answers
146 views

Homogeneous cluster versus heterogeneous cluster

By homogeneous cluster I mean a cluster of nodes that all have the same services running. I am trying to set up a cluster of nodes that offer a set of services. The cluster can respond to HTTPS ...
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0answers
74 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 ...
0
votes
1answer
651 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 ...
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0answers
147 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 ...
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votes
1answer
81 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 ...
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votes
0answers
77 views

Connect Cisco IP phone on two Asterisk sip servers

I have a cisco ip phone with sip firmware v9.3.1. I want to connect two lines on two seperate Asterisk sip servers. In section i can only define USECALLMANAGER (in node) and not the IP of an ...
0
votes
1answer
449 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 ...
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0answers
195 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
1answer
83 views

Cannot access Asterisk sip from remote location

I've used this tutorial to configure my iptables so I can access asterisk sip from outside of my server. My iptables -L looks like this Chain INPUT (policy ACCEPT) target prot opt source ...
0
votes
0answers
96 views

opensips sever running fine and UDP connection cannot register but TCP connections works fine

Im a newbie to opensips and Centos/Linux. Recently installed opensips sever which running smoothly on TCP connections, but users cannot register on UDP connections. here my Log, have a look. Feb 26 ...
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votes
0answers
400 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 - ...
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vote
1answer
3k 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 ...
0
votes
1answer
5k views

Asterisk 401 Unauthorized when trying to register sip clients

Trying to register a sip client to my asterisk server often (just about 90% of the times, not always, weirdly) results in 401 Unauthorized errors. This is the config for one of the extensions: [11] ...
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1answer
47 views

Authentication for SIP

I am reading about authentication protocols for SIP. I found that all protocols transfer the id/Username during the mutual authentication phase, directly with the message. Is it necessary to have the ...
1
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1answer
210 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
6k 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 ...
0
votes
2answers
365 views

asterisk - use a different “SIP CoS mark”

On my asterisk server, when i do a sip reload, i get the message "Using SIP CoS mark 4", followed by a registration time out. I need to make that statement say "Using SIP CoS mark 5". How do I ...