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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How to connect a custom SIP Server to CUCM (Cisco)

We have a SIP Server (Kamailio) and would like to connect it to the Cisco Unified Communications Manager (CUCM). The goal is to have users registered on Kamailio who could call users registered on the ...
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7 views

Asterisk SIP Retransmission timeout after a month of server uptime

This Asterisk server runs fine behind NAT with a dozen of SIP registrations and the call volume of a few hundred calls per month. But then, after about a month of uptime, out of the blue the SIP ...
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0answers
6 views

FreePBX - How can I prevent external callers from transferring and using feature codes?

I'm running FreePBX v12. I recently noticed that I can call into a DID, answer the phone, and the external caller can transfer themselves using ##. This seems like a pretty big hole, and I'd like to ...
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1answer
12 views

FusionPBX Failed(405) on Cisco 303 IP phone

Since yesterday, my phone started throwing a Failed(405) error when trying to register to a FusionPBX server. I have tried resetting to factory settings and reinstalling the FusionPBX server on ...
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1answer
38 views

How to use conntrack to allow SIP traffic

I have the following problem: Device (eth0)----> SWITCH(trunk)+VLAN120 ---> (PC1) +VLAN200 ---> (PC2) I am able to ping from PC1 to PC2 which are in ...
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0answers
10 views

SIPp configuration (SIP/RTP packet replay)

Just today found sipp and would like to use it to replay a packet capture of a video SIP call, made with Wireshark. If it works, next, I will try to replay captures in-parallel, with some delay, for ...
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1answer
11 views

MCU packet capture

I am trying to do a packet capture of a video conference hosted with CISCO [Codian] MCU. The instructions for capturing SIP packets with Wireshark are available. I also configured MCU to accept SIP ...
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0answers
13 views

after 487 Requst terminated there is no ACK

We have a weird problem. After a non successfull call (like a very long number and such), the phone (Yealink-any) gets "call time out" and then loses it registration. I ran wireshark and noticed that ...
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1answer
23 views

Dial 0 for operator not working externally

We have a Deployment with Lync and UM enabled with Exchange, both 2013. When we dial the number of a person in a response group and cannot reach him/her we get the Voicemail auto attendant, also we ...
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1answer
42 views

Can I use Kamailio Sip Server to handle 500 calling? [closed]

I need a Sip server to build a VIOP server to establish voice call between 500 Android an desktop sip clients. Can Kamailio handle this or I need an Asterisk server too?
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56 views

Kamailio cannot connect to remote Mysql Server

I installed Kamailio 4.0 on a Ubuntu-Server 14.4 and a MySQL Server on another Ubuntu-server using Virtualbox. 192.168.0.131: MySQL Server 192.168.0.141:Kamailio 4.0 I changed the address ...
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1answer
48 views

Can't get voice after changing Asterisk UDP port

Test environment consists of a CSipSimple client, Asterisk server and a router with NAT. Formerly the equipment used UDP port 5060 and everything was good. Then I changed port number to 15000 in ...
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0answers
22 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 ...
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0answers
45 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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0answers
71 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
75 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
274 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
34 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
42 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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0answers
70 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
152 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
141 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
46 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
158 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
118 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
102 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
159 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
88 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 ...
4
votes
0answers
471 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
votes
2answers
164 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
votes
1answer
125 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 ...
0
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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
191 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
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1answer
83 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 ...
1
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1answer
265 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 ...
1
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1answer
293 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
votes
2answers
310 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
63 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
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1answer
156 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
2answers
152 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
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0answers
17 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 ...
0
votes
1answer
488 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
556 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
107 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
58 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
904 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
209 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
83 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
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0answers
242 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
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1answer
904 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 - ...