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Questions tagged [sip]

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

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1answer
27 views

LG Nortel 8540 with normal SIP trunk [closed]

I have an LG Nortel 8540 and want to configure it to use a regular SIP trunk or a PBX like Asterisk or 3CX Is this possible without Microsoft's own Office Communicator 2007 server? The official ...
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0answers
6 views

'Make' command after 'configure.py' never ends [WINDOWS]

I have been trying to install the SIP module to install PyQt4 for Python 2.7 on Windows 10. I downloaded the newest SIP package, I extracted (on a subfolder of 'My Documents') and executed the ...
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0answers
48 views

PJSIP PJSUA python no sound but call OK on Raspberry

I have an Asterisk server on one Rasp and on another I would like to make a SIP client. To do this, I installed PJSIP, and I found this little code from their tutorial to make a phone call. I ...
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1answer
15 views

How to block a pcap pumped by tcpreplay from one server by iptables on the other?

I am new to stack exchange. What I am trying is: I am pumping traffic from one server by the following command: tcpreplay -i ens3 ~/daniel/sipdump.pcap from server 1 And I am capturing the traffic in ...
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1answer
74 views

Asterisk sip.conf in MySQL database

I'm able to include a phone inside a database for realtime usage. So, this code (from /etc/asterisk/sip.conf): [phone] type=friend username=phone secret=12345 host=dynamic disallow=all allow=g729 ...
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0answers
14 views

Correlating multiplex RTP streams to SIP SDP in VoIP

I'm trying to understand more about RTP multiplexing, i.e. where a single RTP session defined by its 5-tuple is used to send more than 1 RTP media streams. I've read that this could happen in cases ...
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0answers
20 views

Zoiper SIP to Digital Ocean Asterisk: Nothing happens on any test call

So i have a Zoiper app on cell phone and I'm trying to make a test call on a new Asterisk server with Digital Ocean. It connects and registers just fine, and seems to at least acknowledge I'm trying ...
3
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1answer
7k 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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0answers
165 views

How to setup Asterisk NAT on docker for Windows

I need to setup a docker container running an Asterisk PBX. The problem is that I receive the wrong ip in sdp replies. In fact Asterisk is using the container internal IP. The second problem is that ...
3
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4answers
2k views

Video chat software for Ubuntu [closed]

Is there any video/audio chat program available for Ubuntu 9.4 apart from Skype ?
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1answer
562 views

Cisco SPA112 ATA only rings half the time

I have a Cisco SPA112 ATA Unit which I have set up to connect to an online SIP account (no LAN Asterisk servers in use) and I have this issue where sometimes the unit will not ring when someone calls ...
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1answer
489 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 v1.10.8, ...
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0answers
73 views

On a PBX (Asterisk) determine which phones are connected

I am using Asterisk as my PBX. On my sip.conf I used to have this option enabled: [general] ... qualify=yes more info about qualify can be found in HERE. Thanks to that option I was able to tell if ...
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1answer
224 views

(ringinuse disabled) (Invalid) has taken no calls yet

I had a freepbx server in place. I setup a queue with static agents and ring all strategy. Now, suddenly no body in the queue will ring once the call routed to the queue. I check the queue and found ...
6
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2answers
23k views

Which one VoIP SIP audio codec should I choose to high quality calls?

The quality of VoIP calls depends on the codec used for the transmission and on the bandwidth of Internet connection. I want to use best optimized codec for my Internet connection. I have to select ...
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0answers
614 views

How do you fix “No Trust List Installed” error on a Cisco 7942G VoIP Phone?

I have a 7942G. I'm trying to connect it to Asterisk. Looking at Settings -> Status -> Status Messages, I see this: Error Updating Locale SEP############.cnf.xml No Trust List Installed I'm only ...
1
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1answer
371 views

twinkle configuration for Sangoma

I'm able to run AsteriskNOW virtualized on my local system. Also, I'm using Anveo on two different SIP hard phones. Anveo works fine, indicating that the network handles SIP fine. Now, how do I ...
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1answer
1k views

Asterisk initiate call through sip notify message

I have a Yealink ip phone on my desk and I want to remote control it so that I can place the calls from my computer instead than by typing the numbers directly on the phone. This link explains how to ...
2
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2answers
3k views

SIP providers with local number portability? [closed]

Anyone know of a good voip provider that will allow me to port my existing phone number to their service and be cheap? I'm looking to spend no more than $15 a month; am I in dreamland? Also, I've ...
0
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1answer
790 views

Cisco IP Phone to SPA3102 to PSTN

I've just picked up a Cisco 7940G and configured it with SIP firmware (8.2). I'd like to use this as my office phone, communicating with POTS/PSTN via a Linksys SPA 3102 VoIP gateway. In theory, this ...
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2answers
938 views

How to set up a home SIP Server/Proxy for multi ring?

I have a sip account which only allows one device to be registered. When i'm at home I want incoming calls to be able to ring on multiple devices. All of these devices are connected to the local ...
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2answers
207 views

Converting a P.O.T.S line to VoIP and simultaneously ringing a extension over VPN

Anyone with knowledge please help. I have analog phone service + internet at SITE 1. I only have internet service at SITE 2. When the phone rings at SITE 1 I would like the PC or an attached phone ...
0
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1answer
43 views

Ekiga under Windows: registered but calls fail

I have a VoIP account with sub-accounts. Sub-accounts are assigned 3-digit extensions and I can call between sub-accounts using regular desktop phones connected to my ATA. Now I am trying to set up ...
3
votes
2answers
4k 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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2answers
292 views

How to setup a voip server connected to a ISP voice line

My home fibre bradband ISP provides me with a Huawei HG8240 optical network terminal. This has one ethernet port and one phone port. The phone port is active, but unused at the moment. The ethernet ...
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1answer
110 views

How will a call get routed with incorrect LRN in SIP request?

I was reading about frauds in telecom. I read about an LRN fraud in which the fraudster would change the LRN in the SIP invite to a cheaper destination. How will the call get routed with an incorrect ...
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1answer
116 views

Using a VoIP router behind a DrayTek 2760 router

I'm struggling with the following problem: I use a DrayTek Vigor 2760 as router 1 to access internet by using a VDSL line. The DrayTek behaves as a Modem and DHCP server with IP 192.168.1.1. My ...
1
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1answer
41 views

IPSec doesn't hide IAX2 informations

I am trying to setup a secure truncked IAX2 connexion with IPSec between routers: But when I make sniffing attack with wireshark, all call's informations are showen ! IPsec is well configured and ...
2
votes
3answers
4k 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 ...
0
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1answer
114 views

Keeps detecting SELinux as enabled even though I disabled it (When trying to install FreePBX)

I was following this guide to install FreePBX on CentOS 6.9 (minimaslist), everything worked, until the last step to install FreePBX: ./install -n from /usr/src/freepbx, I keep getting the follwing ...
1
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1answer
145 views

Default port for calls with Asterisk

I have AsteriskNOW 11.9.0 with FreePBX 2.11.0.43. I moved Asterisk to a non-standard port number, but now I can call extensions only if I specify the port number explicitly in a callee's number, for ...
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0answers
105 views

SIP based door entry and phone system

For a residential home, we've allowed for a door entry system, where we'd like to the following: Push button at door entry panel calls all intercom stations (handsets) - audio only From any intercom ...
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1answer
40 views

jitsi account “name” or “default identiy” sticks and cannot be removed

how do I change or remove where it shows a "name" above offline to the right of the silhouette of the user and below the files tools help menu? I removed all accounts. I'm only using Jitsi for SIP ...
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0answers
144 views

Testing my GNU Ring installation

I've installed GNU Ring on my system, but, frankly, none of my friends use it (none that I know of anyway), so I don't have anyone to call. I'm planning to invite someone to call me up on Ring, but I ...
1
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1answer
347 views

PPPoE connection type with VoIP service

Two VoIP ATA adapters is used for VoIP service, adapters are in different offices, but within same VoIP service networks. The PPPoE connection type is used in ATA for VoIP service: is it possible to ...
1
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1answer
1k views

Reroute all internet traffic through a tinc tunnel

I want to route all internet traffic (including udp for sip) through a tinc tunnel (https://www.tinc-vpn.org/). To set up the tinc tunnel is quite straight forward. After starting the tinc daemon on ...
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1answer
624 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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0answers
184 views

Unable to accept incoming calls on Polycom VVX 410

I'm using a Cisco DPC3848 cable modem running in bridge mode. I have a microtik CRS125-24G-1S-2HnD router behind it. I can make outbound calls on it no problem, but if I try to make an inbound call ...
1
vote
1answer
669 views

Why SIP-ALG in routers and modems?

We currently are using a NetSapiens based VoIP system and have extreme problems with a lot of consumer grade routers and modems due to SIP-ALG not being disabled or not being able disable. So if SIP-...
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0answers
799 views

Installing Python module sip 4.19.1 on Windows 7

I've got a problem with installing SIP 4.19.1. What I have done by now: - installed Anaconda (for Python 3.6) with compiled SIP 4.18, - pip install sip "Successfully installed sip-4.19.1" - and ...
3
votes
2answers
6k views

What SIP client for Windows does include g729 codec? [closed]

I'm looking for a, preferably free, multi-SIP client for Windows that does support g729 codec.
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1answer
616 views

Kill a listening port for a specific IP

I have a SIP server with multiple network interfaces. (virtual) When I execute the command netstat -nlput it shows the all listening IP and ports as below. root@DUO-CALLSERVER004:/etc/init.d# netstat ...
2
votes
2answers
2k 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 ...
3
votes
3answers
2k views

Is there any use of SIP on top of TOR

Does it make any sense it terms of perfomance and latency?As far as i know TOR is a transports layer protocol trying to encapsulate the cells with onion routing in order to enhanse privacy.Since sip ...
3
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0answers
989 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 has ...
2
votes
1answer
109 views

Setting my ISP's modem to let me call via my landline when I'm not home, using its internal hidden SIP server

My ISP offers to his VDSL users the ability to use any smartphone as a cordless home phone, using an official app. They provide no documentation or source code, but someone managed to de-obfuscate ...
2
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2answers
2k views

What does it mean to 'unlock' a VOIP adapter?

I'm thinking about buying VOIP adapter for home use. I already use SIP software on my PC and Mobile, but i think it would be easier to just pick up the phone at any time and call. And I could recycle ...
1
vote
1answer
146 views

Set sip/tel URL/LSHandler back to OS X Mac Continuity/Wifi Calling instead of Skype

I recently installed skype for business and it hijacked my Mac's ability to make calls via my iPhone (Continuity) or Wifi Calling. I wanted to know how to set it back, but I figured it out before i ...
2
votes
1answer
1k views

SIP address for group chat 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 can be ...
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0answers
182 views

Linphone on Windows Server 2012, with multiple users

Im going to explain everything first. We have Windows server 2012, where our users are connecting to (lets say around 60-70 per server). After Linphohe installation and first start of the application (...