Session Initiation Protocol is one of the protocols used for controlling sessions (signaling) in internet telephony.

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is this failed SIP call due to firewall or NAT?

I'm trying to find out why this call failed, with: <--- SIP read from UDP:63.247.69.226:5060 ---> SIP/2.0 100 Trying and then Everyone is busy/congested at this time (1:0/0/1) full debug ...
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SIP with Huawei LTE cube E5180

I have recently obtained a Huawei LTE cube E5180 for connecting my home WLAN to the Internet. As a newby to SIP I have also noticed that the LTE cube comes with some SIP support. It allows the ...
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23 views

How do I troubleshoot a Cisco IP phone with no sound? [on hold]

In provisioning the Linksys branded Cisco IP phone described below I think there's an error with regards to the dialplan in the hard phone, or perhaps ports, preventing sound. Admittedly, I have no ...
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24 views

How is a SIP invite sent?

The telnyx Data API tab describes a SIP request: Requires IP Authentication to be setup through the portal and associated with LRN under Telephone Data Tab Send a SIP Invite to lrnlookup....
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3answers
56 views

how do I provision a hard phone for getonsip?

In provisioning a Cisco SPA 942 hardphone, branded Linksys, how do I finish the setup for getonsip (or onsip)? SIP Address: foo@getonsip.com Username: foo Domain: getonsip.com SIP Password: ...
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SIP server name Defaults to $VERSION for Cisco SPA 942 hard phone

In reading the fine manual for a Cisco SPA 942 IP phone, branded as Linksys: what is, in particular, meant by: SIP Server Name Server header used in responses to inbound responses. Defaults to $...
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18 views

Unable to find Ubuntu Installation

I just installed Ubuntu via USB on my Mac, everything seems to be OK. After installation completed, I restarted and it boots on OS X automatically. I thought that will be a no problem, because I would ...
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22 views

Trying to run a REGISTER with 401 scenario but everytime SIPP is treating the RE-REGISTER as a new request

This is the UAC script: <?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd"> <scenario name="UAC Basic Registration"> <label id="0" /> <...
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48 views

How to make international phone calls from my computer' through my SIP server?

I'm currently working for a scientific company, and I'm the only programmer here, and the company wants me to use one of their servers to setup a SIP server, and somehow be able to make calls to ...
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21 views

SIP forwarding via DNS. Possible?

Let's say I have an address sip:john.doe@example.com that I would like to use for Internet calls, but my SIP provider is Callcentric and I need to route such calls to sip:17775551234@callcentric.com. ...
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40 views

Which folders are affected by System Integrity Protection?

I have trouble writing files with sudo to /Library under OSX v10.11.3. It seems the folder is affected by SIP. Which folders fall under SIP?
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97 views

I need to spoof the source IP of a packet

I'm running some tests for a security project and here is the scenario. I have a SIP server (SBC) that accepts calls from a list of IP addresses. I have a SIP INVITE packet and I want to send to that ...
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58 views

linphone: remove/hide “default identity”

Linphone automatically generates a computer-local SIP identity called "Default Identity". "Real" SIP accounts you have at a VoIP provider need to be created as "Proxy accounts": The main window ...
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26 views

Can't add 81.23.228.129 to iptables because of SERVFAIL - any way round this?

I need to allow the IP addresses of proxy.sipthor.net through my firewall: nslookup proxy.sipthor.net Address: 85.17.186.7 Address: 81.23.228.129 Address: 81.23.228.150 Add these to iptables: /...
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27 views

Can't complete opensips 'make menuconfig' on Xubuntu 15.04

I'm trying to compile and install Opensips-2.1.2 (or 1.11.6). I've installed lots of libraries to make it run (especially ncurses). Now when I try to run following command: make menuconfig I get ...
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34 views

What is: Keep port forwarding of the Internet router enabled for telephony

When using a fritzbox with attached phones in bridge mode behind another router, manufacturer avm suggests to enable the option Keep port forwarding of the Internet router enabled for telephony, ...
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35 views

Why is my sip software calling 105

I have a very bad sip provider "vbuzzer" but there prices are cheap ,they don't have an app for android (there is one in the store but i am pretty sure it is not official ,and a scam) and their ...
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57 views

SJPhone worked. now says “Service Unavailable”

I have been using SJPhone on Win7 for about a month for testing a SIP service and everything worked flawlessly out of the box. Today I'm getting a popup saying service unavailable when I try to ...
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61 views

Cannot see VOIP calls in Wireshark 2.0.0

I received a pcap dump from a site, running Cisco JTAPI, who's VOIP traffic I need to analyse. I use Wireshark's VOIP calls analyser via the menu: The conversations display fine in Wireshark v1.10....
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57 views

Linksys SPA3102: receive PSTN calls on sipphone, route outgoing calls to voip provider

I'm having problems to configure my Linksys SPA3102 according to my needs. I have a sipdiscount.com account, a PSTN line, a SPA3102 router (just bought it) and a sip phone. Until now I used my sip ...
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177 views

How to send a SIP message with a batch program

I must be missing something. I can't seem to find how to send an instant message via SIP using a simple batch program to which I can pipe the content of the message. I know how to do it ...
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230 views

SIP address for groups in Lync

For opening the chat window with Microsoft Lync, we can call either Lync SIP:user@company.com or SIP:user@company.com However I am looking for single command for open a group chat. It ...
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79 views

no sip account work with any application but linphone account work with all, so, what to do?

I want to use sip to call my friends and family so first I installed jitsi on windows7 64-bit laptop then I registered with linphone which work perfectly, but, I want another separate account from ...
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471 views

Asterisk SIP calls ending after 2 minutes using Twilio's trunking service

I have an Asterisk server hosted on DigitalOcean that is having calls drop after exactly 120 seconds using Twilio's trunking service. It is notable that directly connected softphones do not drop their ...
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817 views

How to call a PSTN number from a soft SIP Phone through Asterisk

I am getting started with Asterisk. All I want to do is to call a PSTN number from a SIP softphone. I have gone through the latest edition of the book: Asterisk, the Definitive Guide and done the ...
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3answers
467 views

Jitsi and Telekom Germany - outgoing calls “Call failed - Not Acceptable”

I am trying to configure Jitsi to operate as a softphone for my Telekom Germany VoIP-connection. So, I went to the website of Telekom Germany, which provides the following details for connecting: ...
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233 views

Make one way SIP video call

Using Asterisk, CSipSimple and MicroSIP I was able to make a video call, but to make it work both parties should click on "video" and video will be transferred in both directions. I do not like this. ...
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64 views

Do I need STUN?

My router has a "real" static outer IP address. Behind router's NAT there is an Asterisk server. Router is configured to forward 5060 UDP port to Asterisk machine. Do I need STUN server assuming that ...
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355 views

SIP: The other party can't hear me but I can hear them when using Asterisk

Please read carefully all the question and do not rush to a conclusion. === Test environment for point-to-point calls, without Asterisk === City1: A router with NAT and WiFi and 5060 UDP port ...
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2answers
2k views

Failed to authenticate SIP peer

I am trying to connect an SIP peer using Zoiper to my asterisk server. The peer is a soft-phone on my server. I have the following config for the peer: [201] disallow=all allow=alaw host=192.168.1....
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48 views

What's Lax and Lax2 protocols?

Are LAX and LAX2 communication protocols like SIP? What are these? Some telephone systems support them beside SIP but I cannot find information on google
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99 views

Why do Ekiga touchtones only work with the RFC2833 setting?

I have Ekiga acting as a softphone for my VOIP account. I made a call to an automated switchboard which required me to "press 2 to continue". I attempted to press "2" on the keypad pane but the DTMF ...
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2answers
251 views

How to get and install g723 codecs for sip applications?

I install an ubuntu desktop in my company network (windows network) that use SIP on software phone and hard SIP phones. I use Jitsi that is a good client for SIP and works well with others software ...
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225 views

No audio after SIP reinvite

I am trying to get a Gigaset C470 IP to work on my Internet connection. I have figured out the correct configuration for the device and my router (a pfSense sitting between the DSL modem and the LAN) ...
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62 views

Ekiga softphone shows as “registered” — but is it actually?

Is Ekiga actually registered? How would I know? I think Ekiga connects, or at least it says "registered": But, when I exit Ekiga, and use SFLphone, it fails to register: While sipsak shows a ...
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123 views

topology for an unmanaged switch (firewall)

I'm considering an unmanaged switch. My understanding is that many of these devices explicitly lack a firewall. In that case, what do you use as a firewall? I suppose one option would be to use a ...
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157 views

Not registering on Asterisk over VPN

I want to work in local area from home, I configured OpenVPN on my Asterisk server (it is only one Linux server in LAN), it works fine. I enabled forwarding on Asterisk server, and client now can ...
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1answer
360 views

Getting Asterisk PBX to do SIP URI dialling properly?

I've installed AsteriskNOW in a VM and I'm having a hard time getting calls from the PBX to head outbound to another SIP address on the SIP2SIP network I use for occasional testing purposes. However, ...
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1answer
377 views

Plivo SIP problems with NAT traversal

Ok, I've looked and looked, and can't seem to find anyone that's written about this problem. I have a simple office setup where there are two SIP phone extensions (Polycom 331) behind a Sonicwall ...
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1answer
306 views

Can Asterisk accept and send SMS?

Is it possible to send and receive SMS over Asterisk? I am using Asterisk for my communications and it works pretty good. I just wish that I could send SMS to my callers. thanks
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556 views

How to debug “SIP registration failed”?

My SIP phone, which I use from Turkey with a virtual UK number, has recently stopped working with "SIP registration failed" and the supplier of the UK number has told me the times of the last ...
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1answer
2k views

Android won't register SIP on FritzBox router

In the past I've used VoIP/SIP apps to call from my Android phone. Recently I got the upgrade to Android KitKat (4.4.x) and I've noticed that it includes SIP accounts natively (accessible through ...
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342 views

FreeSWITCH simple configuration for audio call between two computers

I have two computers and I want to make audio calls between them. Each one of them is behind a different NAT. I am an absolute beginner and I have no idea what I need to configure. After reading and ...
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1answer
80 views

tcpdump prints nothing while trying to listen communication with sip server

The following command sudo tcpdump -nqt -s 0 -A -i eth0 port 5060 works and prints incoming SIP messages. Now I want to see entire communication and entering the IP address of the SIP server: ...
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5k views

Failed to make SIP calls (Asterisk + PhonerLite + Zoiper + Blink)

I've installed Asterisk PBX on a VirtualBox virtual machine, then created 3 phone extensions: 101 - PhonerLite, 102 - Zoiper, 103 - Blink all of them are configured as "friends". All the clients ...
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1answer
216 views

Why isn't my router allowing my PAP2T to connect to my VoIP service?

My tech knowledge is middle-low: as you can see I'm not shy about getting into the interfaces for my router, but have no idea what the more complex settings mean. I got a new wireless router (D-Link ...
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2answers
168 views

What is SIP, in plain English? [closed]

I have tried reading the Wikipedia page for SIP but I can't follow at all (incidentally, it is currently marked as too technical). What exactly is SIP? I understand how it is a protocol for ...
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1answer
102 views

How to create a proxy to access sip which is IP locked

Here is what i want to do , I have a sip phone configured on my home system which only works on my static IP , Using zoiper soft phone. I want to be able to connect to my system remotely and be able ...
2
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2answers
1k views

What happens when two SIP clients log in at the same time

If two SIP clients are logged in to a service provider at the same, what happens when an inbound call comes and what happens if you try to make an outbound call? (This question is regarding SIP ...
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3answers
2k views

Why can't SSH tunneling be used for SIP VOIP encryption?

On http://www.voip-info.org/wiki/index.php?page_id=2716 it says "Note that SSH tunneling is not a viable method for VoIP media path encryption." Why can't SSH tunneling (as opposed to a VPN) be used ...