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Questions tagged [sip]

The Session Initiation Protocol (SIP) is an IETF-defined signaling protocol widely used for controlling communication sessions such as voice and video calls over Internet Protocol (IP). The protocol can be used for creating, modifying and terminating two-party (unicast) or multiparty (multicast) sessions. Sessions may consist of one or several media streams.

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VOIP, How to setup on Vicidial?

I am runing a vicidial system on a computer , so far so good. I have done basic configure like creating campagnes, users, users group, phones. And if i make a call inside the network it works. But now ...
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No RTP engine was found. Do you have one loaded? Asterisk-18.10.1

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://...
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Server Layout for Web-, API-, Database-, VPN- and SIP Server

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 ...
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Lack of audio for SIP call propagated from IPv6 network when using coTurn server

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 ...
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Working on video configuration suitable for webrtc and sip/udp user agents. One way audio and rtcp timeouts

WebRTC-WebRTC call works well, but SIP-SIP and WebRTC-SIP no: [Oct 21 18:27:22] DEBUG[59]: pjproject: <?>: icess0x7f66780b0c98 Controlled agent timed-out in waiting for the controlling agent to ...
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Status puplication on asterisk

Is it posible to make asterisk publish user agent statuses like online, dnd, etc..? Then i change status on Linphone client i get folowing warning: [Oct 22 12:30:47] WARNING[46]: res_pjsip_pubsub.c:...
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Forwarding SIP headers with asterisk (PJSIP)

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 =&...
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how can I check which process is sending SIP and RTP traffic to which server on windows

My company uses a web based softphone to receive or make calls to customers. but that softphone platform is provided by a third party. Now I want to figure out, our web based softphone will connect to ...
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VoiP SIP trunk to Cisco RV345 and SG200-50 Switch epic Fail

Im having some serious trouble with VoiP phones that are sip trunked over the internet. I.E. we do not have an On Premises PBX. Every thing from the provisioning to the calls is done over the internet ...
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Setting up Asterisk + webrtc

This question has already been asked here. However based on a comment there, I am posting it here. I am trying to set up Asterisk to work with webrtc.. On the client side I am using sipML5. This is ...
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asterisk not recognizing answer from sip trunk

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 ...
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How to hangup Freeswitch/Sofia SIP-Calls at a 3CX correctly?

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 ...
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Openstack Neutron - preserve SIP source port

currently I'm running freeswitch servers on Openstack Ussuri with Neutron linuxbridge. The problem is that the SIP source port got changed from 5080 to random port when sending INVITE to the carriers. ...
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WebRTC Grandstream UCM6510

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 ...
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Routing traffic for specific port range

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 ...
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VoIP one-way audio, only when call initiated from one side

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 ...
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VoIP and NAT (and blocked ports)

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 ...
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tel: URI support in Freeswitch(for outgoing calls)

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 ...
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SIP traffic and firewall rules

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 ...
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SIP Phone behind NAT iptables

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 --------------- ...
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Configure siproxd to use behind a openvpn connection

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 -> ...
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Yealink T48S upgrade/downgrade issues

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 ...
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Forwarding separate Twilio menu options to separate FreePBX inbound routes

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 ...
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Asterisk: Using Queue() in the h extension

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 ...
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For SIP trunks, is it possible that DTMF talk-off could mute a call?

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 ...
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VoIP over VPN (FritzBox) – can register, can call, but no sound

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 ...
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Asterisk WebRTC outgoing call delay

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 ...
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Block FaceTime/Whatsapp video

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 ...
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netsed transparent proxy and server on the same system

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 ...
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linux with 2 nic for SIP and iptables rules

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'...
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FreePBX scalability with DigitalOcean

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 ...
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FreeSWITCH Dual Stack IPv4/IPv6

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 ...
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Capture the VOIP calls receive to a sip phone that send Notificantions to an API on a call

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 ...
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2 votes
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Asterisk skips first DTMF

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 ...
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SIP call connection is p2p? [closed]

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 ...
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3 votes
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Hyper-V Virtual Switch not recieving any packets

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 ...
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Provisioning a Unidata ICW-1000G WiFi phone

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-...
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Freeswitch/fusion 'ghost' calls bypassing F2B/domain acl

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, ...
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Error with simpl5 on call to freeswitch (onGetUserMediaError)

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 ...
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One Way Audio making "internal" SIP calls

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 ...
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Replacing IP address of connections from AWS towards SIP

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 ...
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Skype for Business calls behind pfSense dropping after five seconds

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 ...
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Grandstream GXP2170 XML provisioning for 7-digit dial plan

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 ...
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linux port forwarding between two nics [closed]

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 (...
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Multiple Asterisk Servers Behind Single NAT

I'm debugging an issue we're having with VoIP SIP registration. After looking at packet dumps, I think the issue results from 2 subnets accessing the same web service and the router not knowing how to ...
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Using Asterisk as a gateway to Provider

I have 2 sip servers on different LANs. Freeswitch and another is Asterisk. Asterisk sits on a VPN with a provider and who has provided DIDs. All users register on Freeswitch. How can I route calls ...
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iptables blacklist empty User-Agent sip tcpdump

I scan using tcpdump for user-agents that hit my phone server. I get interesting results. i add them to a list that protects alot of my servers out there. This works good. The problem is I get ...
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2 votes
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How to enable asterisk Call Completion on Busy Subscriber (CCBS)?

I cannot enable Call Completion on Busy Subscriber (CCBS) on asterisk, witch is part of the Call Completion Supplementary Services (CCSS), as is the Call Completion on No Response (CCNR) feature. ...
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Configuring multiple interfaces with a gateway each

I am trying to configure a SIP networking interface. My goal is to siproxd my internet-interface (enp3s0) to the SIP ethernet device (enp4s0). I have edited my /etc/network/interfaces like so: root@...
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Connect one SIP phone to multiple Asterisk Servers

I am working on a project where there are multiple apartment complexes that share the same reception desk, it's just one of the buildings. There are VOIP devices in those apartments that have to be ...
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