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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6 views

ISP SIP account on own Asterisk machine

I'm new to VoIP and SIP, have experience in Linux and servers in general just not SIP servers and such. I was curious about experimenting a little with SIP and VoIP. So I ordered a SIP account from ...
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0answers
14 views

Asterisk dimensioning hardware on AWS and find out how many concurrent calls a server can handle? [on hold]

I am looking for Suggested Hardware to install FIAF(asterisk) AWS EC2 c1 and m1 instance. I would like to know how many calls it can handle concurrently at the same time? OS: Centos Codec:...
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6 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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11 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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1answer
32 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
33 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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3answers
39 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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69 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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1answer
32 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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60 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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66 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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145 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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152 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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1answer
62 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
113 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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votes
1answer
20 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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2answers
44 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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0answers
42 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
691 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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1answer
29 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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0answers
50 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
46 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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162 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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1answer
71 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
144 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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2answers
767 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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41 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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votes
1answer
81 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
217 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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votes
2answers
184 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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66 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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128 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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163 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
199 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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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 "...
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0answers
276 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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votes
1answer
82 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
1k 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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votes
1answer
36 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 ...
0
votes
1answer
292 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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votes
1answer
169 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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373 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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votes
1answer
760 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
73 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
121 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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1answer
212 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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votes
1answer
792 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
221 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
81 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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2answers
566 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 ...