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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: No ringback on specific provider combination

please help me out with this one: PROBLEM DESCRIPTION No ringback when calling cellphones using VoWIFI/VoLTE Only happens when calling from a specific VoIP telephony provider to a specific cellphone ...
Giorgos Amoiridis's user avatar
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0 answers
37 views

Loss of SIP Header Values ​on Transfer

Closed. This question is not about programming or software development. It is not currently accepting answers. This question does not appear to be about a specific programming problem, a software ...
Mustafa Furkan Bilen's user avatar
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0 answers
84 views

How to Prevent FreeSWITCH from Adding Reason Header to SIP BYE and CANCEL?

I noticed that freeswitch adds a Reason header automatically to SIP BYE and CANCEL requests. I've successfully used the disable_q850_reason parameter to prevent FreeSWITCH from adding Reason headers ...
Mohamed's user avatar
0 votes
0 answers
169 views

ED137 SIP & RTP (VoIP) for Air Traffic Control Use with Cisco CUBE

I am working with an air traffic control client who is using a variant of SIP called ED137. There are 2 main uses, Ground to Ground (basically telephone calls) and Air to Ground to Air (radio traffic ...
user10021657's user avatar
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0 answers
117 views

Setup Kamailio to accept only register request with FQDN and block IP request

I have set up Kamailio and am quite new to the process, I'm able to register with FQDN and IP directly to Kamailio but my main setup is Kamailio as edge proxy to asterisk to allow register only with ...
Mohammad Sarshar's user avatar
1 vote
1 answer
51 views

One way audio, bad echo on asterisk trunks with PAP2T ATAs

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 ...
SeniorGeek's user avatar
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0 answers
258 views

Is using Outgoing SIP Registration the only way to allow incoming calls if you do not allow port forwarding or any new incoming connections?

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 ...
JMain's user avatar
  • 115
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1 answer
797 views

SIP Websocket to UDP proxy server

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 ...
Ngdgvcb's user avatar
1 vote
0 answers
135 views

how to allow sip traffic, from field's IP is modified by routers ip

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 ...
SHERIF OMRAN's user avatar
0 votes
2 answers
534 views

Grandstream wp820 not registering after router/modem switch

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 ...
xevoryn's user avatar
0 votes
1 answer
4k views

Asterisk shows No matching endpoint found

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 ...
Oleksandr Znachkov's user avatar
0 votes
1 answer
428 views

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://...
xenon's user avatar
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0 answers
27 views

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 ...
Kasi-Laubfrosch's user avatar
1 vote
0 answers
116 views

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 ...
Paweł Madej's user avatar
0 votes
3 answers
4k views

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 =&...
Pownyan's user avatar
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595 views

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 ...
Spuria's user avatar
  • 11
1 vote
1 answer
360 views

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 ...
Sebastian Schmitt's user avatar
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1 answer
119 views

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 ...
Leonel Franchelli's user avatar
0 votes
2 answers
1k views

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 ...
pgr's user avatar
  • 459
1 vote
1 answer
1k views

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 ...
pgr's user avatar
  • 459
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1 answer
779 views

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 ...
Bastien Matthai's user avatar
1 vote
1 answer
417 views

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 ...
uts9's user avatar
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1 vote
0 answers
180 views

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 ...
user602412's user avatar
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0 answers
2k views

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 --------------- ...
Surfinside's user avatar
1 vote
1 answer
342 views

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 -> ...
Victor Sanchez's user avatar
0 votes
1 answer
630 views

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 ...
XBF's user avatar
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1 vote
1 answer
113 views

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 ...
Moshe Katz's user avatar
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0 votes
1 answer
1k views

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 ...
Mattisdada's user avatar
0 votes
1 answer
246 views

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 ...
pooter03's user avatar
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0 votes
2 answers
2k views

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 ...
Kevin Lieser's user avatar
0 votes
1 answer
2k views

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 ...
Hativ's user avatar
  • 111
0 votes
2 answers
2k views

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 ...
Max13's user avatar
  • 141
0 votes
1 answer
208 views

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 ...
A. Fendt's user avatar
0 votes
1 answer
325 views

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'...
SmileMZ's user avatar
  • 131
0 votes
0 answers
376 views

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 ...
user282190's user avatar
1 vote
0 answers
2k views

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 ...
Xabre's user avatar
  • 153
1 vote
1 answer
169 views

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 ...
Luis Javier Davis's user avatar
3 votes
1 answer
349 views

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 ...
Roberto Neves's user avatar
0 votes
1 answer
329 views

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 ...
alexandre1985's user avatar
3 votes
1 answer
1k views

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 ...
reidi2000's user avatar
  • 131
1 vote
0 answers
162 views

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-...
James Youngman's user avatar
0 votes
0 answers
350 views

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, ...
Malaxes's user avatar
  • 71
0 votes
1 answer
893 views

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 ...
Husk Rekoms's user avatar
1 vote
1 answer
51 views

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 ...
Gavin's user avatar
  • 173
1 vote
1 answer
183 views

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 ...
vichar's user avatar
  • 11
1 vote
0 answers
1k views

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 ...
C Hamm's user avatar
  • 81
0 votes
1 answer
262 views

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 ...
Tony Bucci's user avatar
0 votes
1 answer
415 views

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 (...
McDee's user avatar
  • 1
0 votes
1 answer
951 views

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 ...
sleepisfortheweak's user avatar
1 vote
1 answer
2k views

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 ...
Ben Kabale's user avatar

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