Questions tagged [sip]

Question 1

I am running HOMER 7 using Docker Compose with the following components: sipcapture/heplify-server sipcapture/webapp PostgreSQL 11 (Docker) The system is working correctly and SIP data (...

Question 2

We have been experiencing a nasty elusive problem with the GXP1628 IP-phones for several years now. At some moment whole network segment (all devices connected to the same hub) becomes completely ...

Question 3

I’m using the following Grandstream UCM devices and running into a performance issue: UCM6301 (supports up to 75 concurrent calls) UCM6302 (supports up to 150 concurrent calls) When I make just 15 ...

Question 4

I’m working with a Grandstream UCM6302 device and need some help configuring the network settings correctly. Here’s my setup: I have currently enabled "route" mode in my device, WAN port is ...

Question 5

I try to dockerize my asterisk setup but ran into a problem. The problem is: Twinkle registers successfully, however, initiating a call from twinkle does initially succeed, but after ~20 seconds, ...

Question 6

I have sip traffic which needs to flow bidirectionally between server A and server B on port 5060. I have tried running netcat on both the machines Netcat from machine A to B: ========================...

Question 7

Friends, I'd like to run a custom program whenever a SIP call is received, similar to common call monitoring UIs. I do not need any (G)UI for this. I imaginge something like: /usr/bin/the-imaginary-...

Question 8

This is not a question, I solved it. I set up Asterisk trunks between offices, with encryption, some offices do not have fixed IP and login in to the central server. That all went well. Anybody needs ...

Question 9

If I use a SIP provider that uses SIP REGISTER for connections (instead of using an IP address), the outgoing registration originating from my side opens a connection to the SIP provider, and they use ...

Question 10

There is an ATS provider with SIP phones. It provides phones via UDP, as I understand, giving sip server, login and password for each internal call-line. I want to write a site with browser calls ...

Question 11

I have the following problem: PC1 - eth0: 192.168.188.55 eth1: part of br0: 10.147.20.69 tap1: part of b0: 10.147.20.2 -> connected to PC2 on eth1: i have a trunc sip device ...

Question 12

Greetings fellow system engineers! I'm new to the forum and I do have a rather strange problem. Please be aware that I am encountering the following issue as a hobbyist / power-user. I do have some ...

Question 13

I have planned to implement Asterisk SIP server for testing eMTA calls. I don't have eMTAs so I decided to start with Linux Soft client and then, when I will have an eMTA and physical access to the ...

Question 14

I have been trying to install Asterisk-18.10.1 version on my ubuntu(20.04.4) running inside VM. I was able to maintain connection from GoTrunk SIP endpoint and Zoiper as softphone. Followed https://...

Question 15

i have the possibility to redo our Server Infrastructure - i need your advices and best practices to design a good foundation for future expansion. As we are a quite small company with a very limited ...

Question 16

We are using coTurn server to allow our SIP appliance work reliably between client apps and teleconferences. During our test's we have found that one type of calls is not working at all. But at first ...

Question 17

I'm trying to forward a specific incoming header to the other leg of the call, but can't figure out how to pass the value of the header in the incoming leg to the pre-dial handler [addheaders] exten =&...

Question 18

I have an Asterisk server (15.5, FreePBX) with three SIP trunks from different providers configured, two of them are working fine while the third for every call keep sendind the invite despite the ...

Question 19

We are running a FreeSwitch-instance at host A, that is placing outbound calls to a 3CX-instance running at host B, using the SOFIA module. Everything is working fine except for the hangup: The BYE ...

Question 20

Some times the webrtc transport connection is stablished but when I observe in chrome://webrtc-internals the dtls session in that transport it stays stucked in “connecting“ and the remote certificate ...

Question 21

I have a Ubiquiti Dream Machine (UDM) which is part of a project that is replacing a network topology with a different one. This involves both Internet traffic and VoIP (Asterisk). During the phased ...

Question 22

I am setting up some new switches and VLANs and I am getting trouble with our pre-existing Asterisk VoIP set-up. Most calls work ok. Some get just one-way audio. I tried to narrow it down to this ...

Question 23

I'm making a VoIP application and I have trouble to make it working properly. On each side there are SIP clients. In my office, we use 2 differents boxes to access internet. The first one is like a ...

Question 24

I have integrated a Freeswitch instance with an external SIP server and calls are working without any issues. But now I need to change the SIP server which only supports tel: URI. In the case of ...

Question 25

We have Cisco Expressway-Edge devices handling videoconferencing traffic with the outside world. This all goes through a Checkpoint firewall. The intention is that the inside endpoints can initiate ...

Question 26

I have a SIP Phone on my LAN. The outcall work's but incomingcall not. When I call the SIP phone with my mobile phone, it goes straight to the voice mailbox. This is my network Phone --------------- ...

Question 27

I need to configure a proxy-sip. I discovered siproxd, but I am not able to use it. My network is as follows: PC-Windows -> Use MicroSip -> 192.168.1.10 PC-Linux -> Use siproxd -> ...

Question 28

So I’ve been using the T48S on the Skype for Business firmware for a while now (I purchased it second hand on eBay and it came with version 66.9.0.45) I originally flashed it to 66.81.0.70 lync ...

Question 29

I have a phone number with Twilio that goes to a large Twiml application. Within the Twiml application, there are three certain menu options that forward the caller to a phone. Until now, the calls ...

Question 30

I'm trying to get users to be able to record a message, hangup and have the call continue and dial Queues and playback the recording. I've gotten most of the way there, but right now when I call ...

Question 31

My company sells/supports Allworx SIP PBXs. We have a weird edge case with one of our customers who use PRI for dial-tone. On outbound calls, if a user happens to receive an SMS on their iPhone, and ...

Question 32

We have a small company network here (192.168.178.x) and a private network in home office (192.168.2.x). These networks are both running a FritzBox router and are connected via VPN trough FritzBox to ...

Question 33

I run an Asterisk 16 installation and a WebPhone based on SIP.js. Unfortunately, I often don't hear the first few seconds when I call someone. But everything is fine with incoming calls. The Asterisk ...

Question 34

I want to avoid video calls in my network, mainly from whatsapp and FaceTime. Ideally, the phone shouldn’t even receive the ring, but it might not be possible as the ring is certainly sent ...

Question 35

I want to use netsed to alter the incoming SIP traffic (UDP port 5060) on a PBX server which is running on a linux system (debian 10 stretch). In the first step I simply tried to set up the phone to ...

Question 36

I have a linux machine with 2 NIC, on one where the public ip of my bridged modem is setup On the other one with local lan enp7s0: WAN IP enp6s0: 192.168.0.1 I setup my iptables rules, but it doesn'...

Question 37

So please excuse my loose use of the PBX / VoIP terminology as I have just recently been introduced to these services. Essentially I was tasked with building a solution with FreePBX (without GUI and ...

Question 38

I'm currently trying to understand how I can enable my freeSWITCH to talk both IPv6 as well as IPv4. Currently, I thought it was going to be easiest to first create a set-up which works on IPv4 and ...

Question 39

This is whats I'm trying to achieve I want a SIP phone or (is there is any other way I'm open too) to take the calls it receives from a VOIP provider take the information related to the call when ...

Question 40

I configured an asterisk server to receive calls from one sip trunk and then dial out through another (my VoIP provider). Both trunks are configured with dtmf mode SIP INFO. The thing is: When I ...

Question 41

I have a small home server (a ARM64 Rock64 4GB RAM) and am wondering if I want to install a sip server and/or a xmpp server. In the sip case I would like to know if it can handle many active users ...

Question 42

I have a virtual machine set up with two network cards, one card is connected to a virtual switch for connection to the main network, the second is connected to another external port on the host ...

Question 43

Apologies if this is off-topic here (if so, is there a better place?). I just bought a Unidata ICW-1000G WiFi phone (because I prefer to use just WiFi and not to use DECT). I'm attempting to do auto-...

Question 44

We have been receiving 'ghost' calls from non-existent extensions. I've run into this before on asterisk systems and usually just configured the sip profile to disable guest/anon calling. However, ...

Question 45

I have been working on setting up an HTML5 client (sipml5 by doubango: https://www.doubango.org/). The infrastructure of my setup is shown below: Server 1: sipml5 client, served through ngnix and ...

Question 46

I have just moved our phone system over to SIP using BT Voice Cloud and having a problem with one way audio when making "internal" calls. The original router (Vigor 2860n+) is the DHCP ...

Question 47

We have installed a PBX on AWS and connected it to our on-prem Router via VPN. My on-prem router is connected to the SIP provider via a physical connection with another on-prem MUX device (device ...

Question 48

I'm having a problem with Skype for Business (Lync) disconnecting behind a pfSense firewall after exactly 5 seconds. It's an odd problem, and I've opened a ticket with Microsoft, but thus far, that ...

Question 49

I have a need to be able to provision a dial plan that can prepend the area code to a dialed 7 digit number on Grandstream GXP2170 (config files are very similar between the GXP21XX models) on ...

Question 50

I have a vicidial server running on suse. I have installed two nic cards. One nic connects to my local network and the other connects to the firewall. The config is as below: Nic 1 : 192.168.2.21 (...