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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1k 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 ...
3
votes
1answer
1k views

MS Lync on Terminal Server

Is there a way that I can install (hosted) Lync on our terminal server and when a user logs on, it automatically inserts their email address and the correct manual SIP settings so I don't have to log ...
2
votes
1answer
763 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
10 views

FAX on VoIP line does not work

My provider switched me to an IP-based line. Now my analog ISDN (G3) Fax does not work anymore. Is there any way I can use the conventional Fax with a SIP connection? My Router supports T.38 Fax ...
1
vote
1answer
22 views

Getting SIP, RTP and NAT right : who is doing what?

I'm a little confused about who (PBX vs. phones) is doing what in our IP telephone setup. We have our phone system hosted externally. This means that we've got physical IP telephones installed at ...
1
vote
1answer
765 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 - ...
1
vote
1answer
523 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 ...
1
vote
1answer
648 views

Serious Packet Loss on UDP using Amazon EC2

I am using ec2 to host a udp TURN server for SIP purpose. Anyway I get a lot of packet loss during voice call, so I did a udp performance test using iperf util. Result: [ ID] Interval Transfer ...
1
vote
1answer
289 views

Routing SIP between subnets

I've got the following configuration : 2 subnets (we'll call A and B) with IP phones in both. Internet access behind a NAT for each of them, and a routed VPN is set up between those 2 subnets. I am ...
0
votes
1answer
331 views

SIP.js connection to Asterisk 11.20 over WSS not working

I have successfully setup sip.js using a standard non secure ws:// to an asterisk 11 server using firefox 43. I can make and receive calls to another ff browser/hardphone. But this does not work with ...
0
votes
1answer
27 views

Creating Own DID Number

I have 2 Asterisk Servers installed in different machines. Lets say I have created 3 extensions: 101, 102 and 103 in the first server. Generally we can have a single call connected for one ...
4
votes
0answers
236 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 ...
3
votes
0answers
83 views

Hacker bypassing iptables

(moved from SO) I have iptables protecting a sip server. It blocks all IPs except ones I specifically opened, and it seems to work for almost everyone. I have tested from lots of ip addresses that ...
2
votes
0answers
92 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?
2
votes
0answers
369 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 ...
2
votes
0answers
2k views

sending faxes over sip with t38modem and hylafax

I would like to send (and not to receive) faxes. AFAIK, t38modem + hylafax should work fine. After configuring hylafax to look for ttxT38-1,ttxT38-2,ttxT38-3, I'm running the following: # t38modem ...
2
votes
0answers
644 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, ...
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vote
0answers
14 views

Linphone iOS behind CGNAT sets wrong Contact Header

We are using Linphone client on iOS over Verizon network. Our client sends wrong IP address on Contact header at 200 OK message as a response to INVITE message from Asterisk. SIP trace Our client's ...
1
vote
0answers
61 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
0answers
104 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 ...
1
vote
0answers
216 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 ...
1
vote
0answers
427 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 ...
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vote
0answers
202 views

How can I use the FLASH key of my IP phone to transfer a call (atxfer)

My IP phone has a FLASH key. This key sends the DTMF of Flash [16 in decimal] (http://www.voip-info.org/wiki/view/SIP+DTMF+Signalling). How do I set the atxfer parameter in the features.conf?
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vote
0answers
537 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 ...
1
vote
0answers
356 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 ...
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 ...
0
votes
0answers
14 views

Cisco ASA PBR and multiple ISPS for multiple SIP Providers

Have a question about policy based routing on my Cisco ASA 5515 running 9.5 and think despite it being available in this software version that I might be screwed. I have an asterisk PBX behind the ...
0
votes
0answers
4 views

Kamailio as TCP-only SIP register and relay

I have a rather complicated setup in which I'd like to run a SIP server. In general, I want to run an IPv6-only Kamailio SIP-server on internal network and have outside SIP-clients be able to make ...
0
votes
0answers
9 views

SIP Extension Registration Port 5060 and Port 1024 1026 1028 etc

First time here. I have a problem registering remote extension on a Multi tenant FusionPBX/FreeSWITCH. I have signed up with a FusionPBX/FreeSWITCH EC2 in AWS for training purposes. I have setup ...
0
votes
0answers
49 views

It is possible to use not NAT able VoIP Softswitch behind pfsense/siproxd

We are using diffrent VoIP Softswitches (asterisk/squire) behind a pfsense Firewall and connect to diffrent SIP Endpoints. The local softswitches only have a local IP and the Endpoints are public IPs. ...
0
votes
0answers
28 views

Kamailio CPU usage very high and not accepting new registrations

I am facing a very strange issue with Kamailio, our Azure VM is D2 configuration with 2 cores and 7 GB of RAM, Ubuntu 14.04 LTS, through-out the day we see high CPU usage from Kamailio, and ...
0
votes
0answers
35 views

What is needed to enable a passwordless SIP connection in Asterisk to accept inbound calls?

I have an Asterisk system with multiple users. Until now, where a user required the ability to place calls, we provided them with a SIP/IAX account with a username and password, plus a custom routing ...
0
votes
0answers
37 views

Kamailio 4.3 replecing opensips 1.11

We have a customized sip routing script in opensip 1.11 now for good performance and stability we are moving to kamailio 4.3. So here am facing some issues related to kamailio module and funciotns ...
0
votes
0answers
114 views

Vodafone SIP trunking

I came here to ask if anyone of you has experience with Vodafone Business SIP trunking in Germany or in general. Vodafone offer (german). The reason I ask is because I'm planning on setting up ...
0
votes
0answers
33 views

VoIP phones not registering on second zone - SonicWALL TZ210

Environment: Hosted VoIP SonicWALL TZ210 running SonicOS Enhanced 5.9.1.1-39o (latest) ADSL WAN connection The SonicWALL has two zones (subnets), X0 and X2, LAN and Voice respectively. Both are ...
0
votes
0answers
179 views

Asterisk recieves 403 on every outbound call after 5 minutes delay

I have a home server in my local network with Asterisk installed. It has been working without any problem for several month. I have just changed my router (ZyXel Keenetic Lite -> LP-LINK TL-R480T+) ...
0
votes
0answers
55 views

dns SRV lookup with sip.js

I am trying to use sip.js to make video calls using WebRTC. Our video engineer told me that we need to perform a dns srv lookup before connecting. Here is what I have: var uri = new ...
0
votes
0answers
93 views

Calls on new telephony Server (Vicidial) being “dropped” after 30 minutes being INCALL?

Hello everybody! Im in a weird situation with my Vicidial Cluster with the SIP Account 1. I have actually 3 Servers: 192.168.1.200 (All-In-One Server) 192.168.1.201 (Telephony 1) 192.168.1.202 ...
0
votes
0answers
132 views

PJSUA playing wave file on state “EARLY”

I'm trying to create a system, where a wave file is generated and then some of our employees are called by pjsua and the wave file is played to them. Everything seems fine, I am running the following ...
0
votes
0answers
13 views

How to recognize conference calls from individual simultaneous calls When we have Asterisk user agents?

I want to know how I can distinguish between conference call from some independent simultaneous calls when we have Asterisk agents? RFC 4353 says that in conference calls we have a call member with a ...
0
votes
0answers
224 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 ...
0
votes
0answers
40 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 ...
0
votes
0answers
283 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 ...
0
votes
0answers
143 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 ...
0
votes
0answers
70 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
0answers
1k 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
2k 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 ...
0
votes
0answers
339 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
0answers
373 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 ...
0
votes
0answers
856 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 - ...