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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44 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 ...
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20 views

How do I redirect SIP server VOIP ports (such as 8025) to a subdomain such as sip.mysitename.com, instead of the SIP server IP address?

After searching, I haven't found an answer to this, but I may not be using the correct terminology. Corrections welcomed. We have a VOIP server in the cloud, with an IP address, and a custom port. ...
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18 views

Skype for Business 2015 - Screen Sharing Fails in Groups of more than two people

When a user creates a meeting with more than two users, nobody in the meeting is able to share their screen. This only happens with this user, let's call him John Smith. His machine has been re-imaged ...
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1answer
56 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 ...
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45 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 ...
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87 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 ...
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23 views

How to detect noise from RTP and clear the voice

Currently, I am developing a VOIP system using Mumble and SIP. I can send the SIP stream to the mumble server, after converting the RTP stream to PCM and the voice quality is fine in mumble client. ...
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26 views

Is it possible to connect hardware VoIP phone to the PBX behind IPSec VPN?

I have a hardware VoIP phone, a PC with VPN Client, a PBX (FreePBX) over that VPN connection. I have a softphone that works pretty well through that IPSec connection. Can I use named PC as some kind ...
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36 views

Make port range on home network externally accessible through VPN tunnel?

First off, I hope this is the right place to ask this. If it's not, I apologize, and if you have any suggestions on places where this would be on-topic I would greatly appreciate it. Background: I am ...
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488 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 --------------- ...
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151 views

Low audio volume on sip calls

Recently a new business moved into our premises and we added them to our FreePBX switchboard and ported their numbers to our VOIP provider. All the routing IVRs and everything has been set up. They ...
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195 views

Monitor Active Calls on Mediant 800 SBC via SNMP

Currently trying to get the stats for 'Active Calls' on a Mediant 800 SBC. I'm going through the SNMP reference guide (https://www.audiocodes.com/media/14595/mediant-800-gateway-and-e-sbc-snmp-...
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138 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 -> ...
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49 views

Cseq RFC comparison and understanding

I have a below call flow, wherein I have some confusion related to CSEQ..... A --INVITE CSEQ-1-> B A <--180 CSEQ-1 --- B A --PRACK CSEQ-2-> B A <--183 CSEQ-1 --- B A --PRACK CSEQ-3-> B ...
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298 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 ...
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48 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 ...
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1answer
493 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 ...
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1answer
117 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 ...
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2answers
675 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 ...
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1answer
630 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 ...
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2answers
705 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 ...
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1answer
93 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 ...
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1answer
109 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'...
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212 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 ...
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932 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 ...
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1answer
82 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 ...
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1answer
164 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 ...
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1answer
162 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 ...
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1answer
406 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 ...
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79 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-...
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166 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, ...
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1answer
595 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 ...
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1answer
34 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 ...
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1answer
90 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 ...
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714 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 ...
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1answer
183 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 ...
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1answer
215 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 (...
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1answer
474 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 ...
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1answer
724 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 ...
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1answer
696 views

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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1answer
2k views

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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56 views

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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1answer
626 views

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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1answer
72 views

Does this follows the SIP standard?

We are currently using a Swyx PBX which connect using SIP to a OneAccess box (which translate ISDN <-> SIP). We have trouble to make it works. Here's the deal: The Box uses the SIP protocol on ...
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1answer
175 views

Shoretel Connect: Create SIP Trunk Backup for Analog Trunk on SG90

I've worked with some VoIP Products before, mostly Asterisk (base version, build from source) management for a small ISP doing in-network VoIP solutions for customers. My current position hasn't ...
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2answers
2k views

Centos 7 Sofia-sip dependency for Janus sip gateway

I trying to install Janus WebRTC Gateway on Centos 7. But when I run sudo yum install sofia-sip-devel I have this message No package sofia-sip-devel available. I found this answer in Janus google ...
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2answers
7k views

Asterisk redirect calls to external number

We have a phone system created on pure Asterisk (no web GUI) and there are the usual Day/Night mode. In the day mode it calls a ring group. However we have a call centre that we use to pick up our ...
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1answer
123 views

SIP SUBSCRIBE requests fail

I have the problem, that my client can login with a password and call out, but does not receive calls. When i troubleshoot it via tcpdump, it tells me, that REGISTER works, but SUBSCRIBE does not. 21:...
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1answer
153 views

How does buffering work in a chain of Asterisk servers?

I have two Asterisk servers in a chain like this: SIP client (G.729) <-SIP-> Asterisk 1 (converting G.729 to G.711) <-IAX2-> Asterisk 2 -> terminated to analog line. When no buffering was ...
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1answer
2k views

Mikrotik SIP trunk routing

I have a Mikrotik RouterOS v6.36 with my network as follows (I am a newbie to networking in general and have been able to get stuff working half way. I just don't know what to look for.) Router: ...

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