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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20 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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19 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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1answer
38 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 ...
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1answer
109 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 ...
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1answer
21 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 ...
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1answer
25 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 ...
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34 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. ...
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2answers
80 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 ...
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3answers
70 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 ...
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1answer
28 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 ...
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2answers
63 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 ...
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2answers
73 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 ...
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1answer
67 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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2answers
82 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 ...
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0answers
53 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 ...
3
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0answers
334 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 ...
2
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2answers
116 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? ...
6
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1answer
88 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 ...
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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 ...
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1answer
129 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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1answer
57 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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1answer
161 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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1answer
190 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. ...
2
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2answers
123 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
56 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
100 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
106 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
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 ...
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1answer
349 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
440 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
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0answers
92 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
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
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0answers
717 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
187 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
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0answers
72 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
208 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
727 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
210 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 ...
3
votes
0answers
145 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 ...
0
votes
0answers
66 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 ...
0
votes
0answers
107 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
1k 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 ...
0
votes
2answers
1k 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: ...
0
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1answer
163 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 ...
0
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0answers
104 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
125 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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2answers
500 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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0answers
61 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
354 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
452 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 - ...