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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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 ...
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5 views

SIPp timewait vs pause

Regarding scenario files, what is the difference between timewait and pause? There is nothing in the SIPp documentation outlining this unfortunately. Is timewait ignorant to the parameter: -...
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1answer
24 views

VOIP phone working intermittently

I'm having problems with my Polycom VVX400 phones. It's the first time i've configured them to work with Aircall.io, our SIP provider. I have them registered just fine and I can make calls, but when ...
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1answer
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 ...
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2answers
19 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 ...
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7 views

Outbound Rule not working AsteriskNow

I am trying to set up outbound routes for UK numbers (both landline and mobile). UK numbers follow this convention: 0[12378]000000000 However, I cannot make outbound calls on my AteriskNow server. I ...
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2answers
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 ...
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1answer
21 views

freeswitch received audio delayed for 20-30 seconds

I manage to make a very simple configuration for basic phone to phone calls using FreeSwitch, When i Calls A -> B (and B answered), A can hear B instantly, but B cannot hear A, after waiting for 20-30 ...
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1answer
30 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 ...
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46 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 ...
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39 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 ...
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47 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 ...
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0answers
137 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 ...
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1answer
52 views

Phone won't register with Asterisk server

I'm trying to register a Grandstream phone on our asterisk server. We've done this tens of times, have documented how to do this, and recently it doesn't work anymore... I've tried with several phones ...
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0answers
127 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 ...
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2answers
639 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. ...
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0answers
71 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. ...
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1answer
19 views

Why does scp congest my network?

I observe a weird behaviour when I copy files to a remote machine using scp. Take for example this scp command with no special options: scp my.phar remote:apps/ Copying works flawlessly, however my ...
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1answer
557 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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2answers
572 views

What is a SIP B2BUA, and how is it different from a UA?

I've been reading some stuff SIP related, and I'm confused, about what is a SIP back-to-back user agent(B2BUA) and a 'normal' user agent? Can anyone explain the differences? From what I read I can'...
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2answers
40 views

Asterisk One Way Audio or No Inbound Calls

I have an Elastix server running in Amazon Web services, but I can't seem to resolve an issue that is causing either one way audio or no inbound calls to ring. If I set my SIP extensions to "Nat = ...
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2answers
11k views

Asterisk SIP/2.0 401 Unauthorized

I'm running into a funny little issue with Asterisk 10.3, but it seems to be applicable to 10.4 as well. The server running Asterisk was relocated from a VPS to dedicated hardware, and now only 1 of ...
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0answers
37 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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1answer
28 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 extension....
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1answer
79 views

Softphone not connecting Asterisk

I am trying to connect my X-Lite with a remote Asterisk server. I had installed that on a Ubuntu VPS server (14.04) following this tutorial: https://www.callcentric.com/support/device/asterisk/1_8 I ...
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0answers
45 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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40 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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9answers
65k views

How can I check whether port 5060 is open in centos?

How can I check whether port 5060 is open in centos? How can I test if my linux has real a real IP address and I set no iptables blocking rules or is there any tools which I can run in my linux so my ...
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1answer
64 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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142 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 ...
1
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1answer
135 views

Asterisk with softphone and webphone

I have a problem with Asterisk. I wrote a WebPhone, so I should add to my peers some config details: encryption = yes avpf = yes icesupport = yes dtlsenable =yes dtlsverify = no dtlssetup = actpass ...
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1answer
2k 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
40 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 ...
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2answers
226 views

What is SIP trunking?

Can someone explain to me in plain English what SIP trunking is, please? I've read about it on Google, but I don't really grasp it yet. Does it allow a VoIP call to be placed outside of the LAN? So ...
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1answer
2k 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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203 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+) ...
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2answers
166 views

i need a sip softphone that would on incoming call open an url containing the caller number

The idea is that : a customer call, I automaticly open the page of his accout in my CRM. On the many SIP softphone on ubuntu (ekiga, linphone,...), the only one that i could make do that is twinkle. ...
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64 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 SIP.URI('sip','...
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0answers
121 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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2answers
181 views

Does a SIP trunk relay the RTP stream or just set up the call?

Suppose I have a SIP PBX like Asterisk and a bunch of phones registered to it, and outgoing/incoming calls are handled through a SIP trunk. Do the RTP streams go directly between the phones and the ...
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0answers
156 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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2answers
967 views

Lync Server 2010 with Hosted VoIP PBX

We just deployed Lync Server 2010 in our organization and it is working great so far. The next step for us is to enable enterprise voice so that we can replace our telephones with service that is ...
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0answers
17 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 "...
1
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2answers
2k views

Asterisk DID incoming rules =s

So I have a 10 way SIP trunk from a UK provider. Calls in and out work fine, but as I have multiple numbers for different purposes I set up the inbound rules for each number but they never work, only ...
2
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1answer
1k views

asterisk/freeswitch in nat/no-nat setup

my current setup - i use bunch of sip hard-phones around few offices. all devices have two sip accounts configured - one on internal sip proxy [for calls between the branches], another - at 3rd party ...
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0answers
259 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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1answer
81 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
992 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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1answer
34 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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1answer
155 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?