Questions tagged [sip]

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

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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 200 OK ...
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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 ...
Paul's user avatar
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SIP plugin for Trillian

I like using Trillian for chats (Skype, Google Talk, ...) but I cannot add a SIP Account to make calls with Trillian directly. Google always tells me to download the SIP Plugin for Trillian, but this ...
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getting pbxes, sip2sip, ipkall all working

I was able to make a call from my landline to my ipkall number and have it ring on my android device running sipdroid. Just once. Since then, when I call my ipkall number, I get "The person at ...
user12468's user avatar
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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 ...
Oliver's user avatar
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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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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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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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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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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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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 ...
Paul's user avatar
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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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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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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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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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SIP ALG detector

I am trying to create a detector for SIP ALG. I really just need to know what the BEST method of testing this is so I can write a program to test whether SIP ALG is enabled or disabled. Thanks
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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 ...
Paul's user avatar
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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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Android SIP / VoIP with doorbell: View video before answering

I am currently using the 2N Helios IP Vario doorbell device. It can be used via SIP protocol, but can also communicate with the 2N Android app directly, without an additional server, which I would ...
user2941747's user avatar
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pfSense allowing inbound sip phones, but not sip trunk

I am currently trying to get an inbound trunk working on a remote Asterisk machine. The ip phones are remote and connecting just fine through port 5060 (I can make internal calls to voicemail, other ...
akester's user avatar
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Consumer Router/Firewall qualities which are compatible with IP phones

I use a SIP phone from home over an internet connection to connect to a PBX in the internet. The internet connection is 50 Mbps down / 10 Mpbs up. Empirically I have tested several routers and found ...
steampowered's user avatar
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MacOS IVR SIP client

I'm looking for a SIP client that provides IVR functionality on MacOS X. Is there something like that? Max
bot47's user avatar
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Connecting VM guest with Docker containers running on the same host

Summary I'm a backend developer with a limited networks knowledge. I'm in a situation where I need to test a SIP system consisting of several components (various SIP and media elements, SIP UAs) ...
IvanR's user avatar
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Put extensions in some logical groups, so that only extensions that have same group can call each other in FusionPBX

I need to use FusionPBX as SIP registar (Session Initiation Protocol) or SIP server, to route SIP calls to appropriate SIP accounts (extensions). Currently every extension has possibility to call any ...
Svaba1012's user avatar
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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 ...
flyrelay's user avatar
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1 answer
190 views

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 ...
Suxsem's user avatar
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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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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 ...
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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 ...
Amarundo's user avatar
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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 ...
catfood's user avatar
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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 ...
Nicholas Saunders's user avatar
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1 answer
780 views

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 ...
Petepf's user avatar
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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 ...
Ranoo Hashem's user avatar
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122 views

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 ...
HypeWolf's user avatar
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1 answer
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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 ...
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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 ...
Debiprasanna Mohanty's user avatar
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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 ...
tpgalan's user avatar
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1 answer
305 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 ...
user776948's user avatar
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1 answer
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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 ...
edtech's user avatar
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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 ...
Sven's user avatar
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1 answer
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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 ...
Thufir's user avatar
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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 ...
Daz's user avatar
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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 (...
RiddleMeThis's user avatar
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1 answer
164 views

SIP forwarding via DNS. Possible?

Let's say I have an address sip:[email protected] that I would like to use for Internet calls, but my SIP provider is Callcentric and I need to route such calls to sip:[email protected]. ...
Ωmega Δ's user avatar
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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 ...
redkha vindo's user avatar
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1 answer
1k 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, ...
user2818782's user avatar
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1 answer
994 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. ...
Paul's user avatar
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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 ...
Clyde's user avatar
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1 answer
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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, ...
MotoFan's user avatar
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