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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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
720 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
707 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
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1answer
471 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
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1answer
589 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 ...
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1answer
270 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 ...
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1answer
23 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 ...
0
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1answer
41 views

VoIP: No audio, local IP in SIP header

I'm trying to get SIP trunking to work on my Bintec Elmeg hybird 130j with sipgate.de as trunking Provider. So far, call signalisation works fine, so I can call outside phones and also get called ...
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1answer
148 views

SIP: No matching Codecs / 480 Temporarily not available

I'm installing a SIP Phone in a VoIP environment. There are 2 System Phones which work flawlessly (same manufacturer as PBX), and the third phone can be called, but it can't call the other 2 phones. ...
4
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0answers
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 ...
4
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0answers
220 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 ...
2
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91 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
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0answers
354 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
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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
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0answers
605 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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0answers
54 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
90 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
208 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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0answers
394 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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0answers
188 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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528 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 ...
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0answers
353 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 ...
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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 ...
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11 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 ...
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votes
0answers
85 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 ...
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0answers
22 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 ...
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0answers
24 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 ...
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67 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 ...
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0answers
36 views

GOIP-1 does not forward SMS messages to Asterisk

I have a GOIP 1 channel gateway and followed this tutorial: https://github.com/tangaza/Tangaza/wiki/Connecting-a-GoIP-GSM-VoIP-gateway-to-an-Asterisk-AGI For me receiving SMS-es only works if the ...
0
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0answers
23 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
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138 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
44 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
70 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 ...
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votes
0answers
99 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 ...
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0answers
11 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
158 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
32 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
206 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
127 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
0answers
166 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
0answers
176 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
0answers
69 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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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
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0answers
1k 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
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0answers
323 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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0answers
335 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
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0answers
797 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 - ...
0
votes
0answers
1k views

Avaya-Asterisk call drops immediatly via SIP (sip:anonymous@anonymous.invalid)

I'm trying to call from Avaya to Asterisk (11.5.0) in SIP. Avaya calls to Asterisk, then Asterisk rings my softphone but when I answer the call (in the softphone), call drops immediately. This is the ...
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0answers
53 views

Routing SIP client thru WAN router

I may ask again: I have following setup: SIP Client on LAN can make calls, etc thru PBX (is a closed device). I want the same for the remote SIP client. Can I configure all in the WAN router ? May ...
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0answers
2k views

Not able to register via SIPclient on asterisk server?

I'm unable to register a SIP client on my Asterisk server. What changes do I need to make to my Asterisk server configuration to fix this? Here are my configuration files for Asterisk. sip.conf ...