Questions tagged [sip]
Session Initiation Protocol is one of the protocols used for controlling sessions (signaling) in internet telephony.
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How to use baresip CLI to play an audio file
I'm trying to make my home automation system make announcement calls via SIP so that when an event occurs, the system would call to a specified number, play back an audio file and hang up.
I have ...
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In a GPON network, running wireshark captures SIP phone calls from other users
I plugged my laptop into my ONT and fired up Wireshark.
To my surprise a lot of SIP packects were present, and I could see phone numbers of other costumers, and the people they called. I didn't ...
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Asterisk confbridge, timeout only while waiting marked
I have just configured an asterisk server for my home doorbell.
The goal is to allow my doorbell to join a conference room but wait for the first user to also join in order to start talking.
I managed ...
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Microsoft Teams SIP URI to open a direct chat
I work in a mega-corp. For chat, we used Microsoft Lync, then Microsoft Skype for Business, and finally Microsoft Teams. For Lync and Skype, it was possible to create a SIP URI, like sip://$...
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Should a SIP ITSP provide the ability to call a sip identity other than an E.164 style number?
I am planning moving from a PSTN line to a SIP based VoIP service. I am trialing a service where they map a PSTN style telephone number to your SIP device. And if a caller dials the E.164 style ...
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How to configure directory lists for fleet of SIP phones (Cisco 7960 and Aastra 480i) connected to Asterisk?
So I work for a medium sized company running a FreePBX/Asterisk PBX server on-prem with a fleet of mostly Cisco 7960 and Aastra 480i deskphones. Right now, our Asterisk server also runs a TFTP ...
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Jitsi as a VoIP SIP softphone with asterisk - calls don't work
I have an asterisk server and I am trying to use Jitsi Desktop on Debian 11 as a SIP softphone (like Jami and Linphone) to connect to it and make calls.
Jitsi can register and connect to the asterisk ...
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Building VoIP Lan to Apartment Lan
I am quite new on network engineering (coming from programming background). Got stuck at this problem.
Example 1
I have an ASUS RT-AX86U AX5700 router and the building has a 2N Helios IP Verso ...
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Asterisk realtime sip.conf deny/permit
I have a realtime sip.conf using deny/permit rules. For example:
in sip.conf
[MyPeer]
type=peer
host=xxx.xxx.xxx.139
deny=0.0.0.0/0.0.0.0
permit=xxx.xxx.xxx.136/255.255.255.248 ;
in realtime i ...
Gökhan AKSOY
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Asterisk: how to get current sip.conf settings?
I'm not sure if sip general parameter (such as rtptimeout) is set. There is no mention of it in /etc/asterisk/sip.conf. But may be it is set somewhere else (in include files, for example). How do I ...
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Asterisk: how to end frozen calls
Sometimes my voip provider shuts down service without ending calls which are in progress (these calls are incoming to my server and they are coming to queue).
So calls are never gracefully finished, I ...
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Strange SIP register with false SIP Invite recall
I have here a smartphone app (DIVUS VIDEOPHONE app for iOS version 4.2.16) which makes a SIP REGISTER on an AVM Fritz!Box 7490 and then later should get a SIP INVITE. I just do not know... Does the ...
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How to restrict SIP traffic with tp-link TL-R600VPN router
I need to restrict incoming SIP/VoIP traffic from outside my network to a specific domain. I have a cloud SIP/VoIP system with 2 phones and I'm being inundated with fake/ghost calls (dead air when ...
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VoIP adapter unable to receive calls
I have my landline phone connected to a VoIP adapter, which works perfectly fine when I am making calls but doesn't work when I am receiving calls. The phone will ring like normal, but as soon as I ...
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Connect a music player in SIP voice call
I need to establish a normal Voip conversation between 2 persons and one of them, using a softphone in Windows, insert audio from a music player which can be controlled by the own user.
The voip ...
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How to use Jitsi as a softphone SIP client?
Wikipedia lists Jitsi features as including:
IPv6 support for SIP and XMPP
But where and how are SIP credentials configured?
So that I can troubleshoot an Alice and Bob SIP call:
sip diagram
Above is ...
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How to make asterisk call file as user account or extension
I am trying to figure out why the callerid works when I call with twinkle through asterisk but not when making a call file. The callerid with the call file is always unknown. Twinkle account is setup ...
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Using Zyxel Modem As A Simple Voip ATA Device
I have a Zyxel P-2602HWL modem/router which I want to use as a Voip ATA device only. In other words, I already have a router that I am using and I only want to use the Voip functionality of the Zyxel ...
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How to configure instant messaging in freepbx 14 & Asterisk 13
i have a trouble to configure instant messaging using freepbx 14 and asterisk 13 , i want that two sip clients can send and receive messages using their soft phones on smartphone and desktop , can ...
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Is there a way to monitor a GrandStream registration status?
I have a provider locked GrandStream device and I had some issues where the device would unregister itself during the night and I would lose my phone line.
I cannot afford to not notice when the VoIP ...
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Asterisk how to call by names instead of numbers
I haven't found anywhere clear sample how can I setup alphanumeric alias for internal numbers.
For example I am calling now to 504@ip.
How can I make it possible to reach 504 number by john@ip by ...
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How does a router know which device handles calls?
How does a home router know which connected device is meant to handle a VOIP call? Is there a specific broadcast that is sent out? My home setup uses an analog telephone adapter (ATA) box in bridge ...
user523752
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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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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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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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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 ...
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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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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 ...
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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 ...
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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 ...
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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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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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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 ...
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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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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 ...
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(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 ...
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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 ...
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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 ...
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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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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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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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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 ...
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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 ...
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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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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 ...
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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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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 ...
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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 ...
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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 ...
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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 ...
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