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.

Filter by
Sorted by
Tagged with
1
vote
1answer
62 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 -> ...
0
votes
0answers
48 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 ...
0
votes
1answer
56 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 ...
1
vote
1answer
43 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 ...
0
votes
1answer
300 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 ...
0
votes
0answers
27 views

Format results of dhclient custom options

I have CentOS 7.7 server that runs dhclient to get sip server information. but I can not format the results properly. option sip-server code 120 = array of unsigned integer 8; Results of dhclient....
0
votes
0answers
41 views

Can a sip client (Android phone) make calls directly to a sip gateway (with FXO ports)?

Can a sip client (Android phone) make calls directly through a sip gateway (with FXO ports)? I am trying to use and Android phone to call through a Grandstream HT813
0
votes
1answer
31 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 ...
0
votes
2answers
194 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 ...
0
votes
0answers
147 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 ...
0
votes
0answers
18 views

Unknown packets with 0x6a614b00 as their payloads in a SIP call traffic

During analyzing a SIP call traffic (pcap), I see some packets with 0x6a614b00 (4bytes) as their payloads and responses with 0x00000000 as their payloads. Does anyone know what are these packets sent ...
0
votes
0answers
32 views

how is MOS calculated in RTCP

want to find out how the MOS value is calculated in RTCP packets. different places give different details, like it is just a sum, where you start with an R value and using a formula, factor in delay, ...
0
votes
0answers
70 views

NO DATA on HOMER 7 with HEPlify agent

I have a newly installed HOMER 7 server on a Centos 7, with HEPlify agent on a RHEL 7, and I don't see any SIP DATA displayed in the web interface. HOMER 7 server is UP and running, HEPlify agent ...
-1
votes
2answers
211 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 ...
0
votes
1answer
50 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 ...
0
votes
0answers
55 views

Freeswitch + Asterisk servers over VPN

I have two offices which are connected via site-to-site VPN with static routing, VPN works great without any issues. Calls from Asterisk to Freeswitch works great, but calls from Freeswitch to ...
0
votes
1answer
70 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'...
0
votes
0answers
329 views

IPTABLES port forwarding to SIP UDP not working

I have a Asterisk server set up to trunk with Bandwidth.com. The SIP server seems to work fine on the internal network, a 10.0.0.0/8 network. The central router's IP is 10.0.0.1 and the PBX is 10.0.3....
0
votes
0answers
128 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 ...
0
votes
0answers
50 views

Add internal phone number to MS Teams user (Direct Routing)

I have configured MS Teams for Direct Routing via an SBC and local SIP trunk provider. However, I only have a single inbound DID. How do I allocate that DID to an auto-attendant and have it redirect ...
0
votes
0answers
99 views

opensips + asterisk integration issue

I am trying to conigured opensips as registar and asterisk for media services using this doc: https://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration so far everything went well ...
1
vote
0answers
458 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 ...
0
votes
1answer
74 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 ...
2
votes
1answer
149 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 ...
0
votes
1answer
108 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 ...
3
votes
1answer
208 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 ...
1
vote
0answers
39 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-...
0
votes
0answers
119 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, ...
0
votes
1answer
390 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 ...
1
vote
1answer
30 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 server, DNS ...
1
vote
1answer
74 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 ...
1
vote
0answers
548 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 ...
0
votes
1answer
146 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 ...
0
votes
1answer
157 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 (...
0
votes
1answer
330 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 ...
1
vote
1answer
379 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 ...
1
vote
1answer
526 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 ...
1
vote
1answer
1k 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. ...
0
votes
0answers
53 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@...
1
vote
1answer
524 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 ...
1
vote
1answer
65 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 ...
0
votes
1answer
157 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 ...
0
votes
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 ...
0
votes
2answers
6k 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 ...
0
votes
1answer
101 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:...
0
votes
1answer
112 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 ...
0
votes
1answer
1k 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: ...
0
votes
1answer
6k views

SIP registration timed out

I am facing trouble in registering asterisk to sip trunk. When i do >sip show registry, it shows SIP request is send but never gets response back. I cannot even ping sip.flowroute.com. My firewall is ...
4
votes
4answers
23k views

Register asterisk to sip trunk

I want to register my asterisk server to a SIP trunk. I have added following piece of code in my sip.conf and extensions.conf sip.conf [general] register => myusername:mypassword@sip.flowroute....
0
votes
1answer
677 views

How can I setup Asterisk as a Voicemail server for CallManager?

I have an existing Cisco CallManager 8.5 environment which services users needing voicemail, but I have no Unity licenses. How can I use Asterisk to service voicemail?

1
2 3 4 5 6