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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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27 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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21 views

Allowing Skype for Business calls to FreePBX and PSTN

I have this setup: Skype for Business (3 Frontend servers) FreePBX (Integrated with Sip Trunk provider) FreePBX and Skype for Business are integrated as well and calls are working from FreePBX, ...
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1answer
42 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
19 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 ...
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20 views

How do I add static video to a call?

Problem: my conferencing software's SIP bridge only supports calls with audio and video. Now we can all agree that this is broken – in fact I already emailed my opinion about this to their support – ...
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1answer
41 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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175 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
69 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
69 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
117 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
54 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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113 views

Asterisk/SIP - server with two NICs

We are trying to setup our Asterisk server with two NICs on it. One NIC is used to connect to a local network (which has access to internet and other sip trunks over internet) and other NIC is a ...
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22 views

No Audio on VOIP Phones

I have a Grandstream UCM6108 Server on my internal (school) network and 30~ Grandstream Phones. The server and phones are not on the same VLANs but are routed. Over the holidays something somewhere ...
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1answer
240 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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0answers
477 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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42 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
291 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
56 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
135 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
932 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
3k 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
56 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
54 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
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: ...
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1answer
3k 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 ...
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4answers
13k 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....
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1answer
455 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?
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550 views

Use asterisk as client to act as a soft phone

I have an account with vonage and I am able to make phone calls from my computer entering the following information into my soft phone (zoiper): When entering that information (domain,usrename and ...
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0answers
1k views

Asterisk times out and terminates connection after 6400ms due to 'no response'

I have a SIP trunk set up with Twilio for outbound calls. Twilio-FreePBX and then my test device is the simple X-Lite from CounterPath. I can make an outgoing call from X-Lite. My cell phone rings ...
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2answers
2k views

Asterisk client behind NAT - UDP registration does not work

I have some clients connected to my Asterisk server behind a NAT device. When using TCP everything works OK. If I change the client and server config to use UDP (from transport=tcp to transport=udp,...
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1answer
279 views

HP 4120 phone and Asterisk

I got a task to set up a cheap PBX for HP 4120 phones. I want to try Asterisk, but I can't find any info on connecting HP 4120 to any SIP PBX. So here's the question: are HP 4120 compatible with ...
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1answer
506 views

Call flow explanation

In the book "Understanding SIP" they say that only for responses for INVITE an ACK is sent, but in this call flow there is ACK for BYE also. Can someone please explain this phenomenon to me? ] ]
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2answers
266 views

Polycom SoundPoint IP 4000 won't power on. How to isolate power supply?

I have a Polycom SoundPoint IP 4000 SIP conference phone that has sat unused in a box for a number of years. Today, I pulled it out of the box and powered it on with the conference power supply only ...
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698 views

No sound in XLite softphone calling Asterisk PJSIP channel bound to callcentric

When calling from an XLite softphone to a Callcentric number which has an Asterisk PJSIP channel registered, we cannot hear anything at all on the softphone (though the call is indeed established). ...
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1answer
991 views

Getting PJSIP with TLS to work with Twilio SIP Trunking on FreePBX

I'm wondering if someone can help me debug this problem I'm having. I'm trying to get secure trunking setup between my FreePBX server and Twilio using the PJSIP stack. Unencrypted trunking works fine ...
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1answer
695 views

Configure asterisk for blink and linphone

I have setup an Asterisk server (14.0.2) on Ubuntu 14.04. I can get sound from Twilio using ulaw and from Zoiper (no STUN or ICE). In each case the asterisk server plays gsm files. I can't get any ...
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1answer
279 views

Configuring Dual Interfaces for for two external IPs on Ubuntu

I am trying to connect a second ISP connection to my server for our SIP Phones. eth0 has our default 25Mb service on 192.168.1.x, I've configured eth1 for the new 6Mb service to be 10.10.1.x Both ...
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0answers
247 views

Forward all traffic from interface to internal IP address

We've had a new SIP trunk setup and connected to our Mikrotik router, I need to forward all traffic coming from int ETH2 to an internal IP address of a VM (VM Host is connected to Eth5) for our ...
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1answer
380 views

Connect Asterisk system to SIP provider with Asterisk dynamic realtime

Is it possible to connect an Asterisk system to a SIP provider with Asterisk dynamic realtime (ARA)? In other words, is it possible to configure a table (or perhaps view) such as sippeers for family ...
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1answer
157 views

Using freePBX as a sip client

I'm trying to find a way of setting up freePBX to somehow use an external sip account (username, password and domain) and not a trunk to make inbound and outbound calls. I was wondering if anyone knew ...
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1answer
82 views

Lync 2013 SIP Trunk with credentials?

We have a Lync 2013 installation and are currently in the process of testing SIP functionality. Our current SIP trunk requires a username and password to authenticate. I can't find an awful lot of ...
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0answers
83 views

ISP SIP account on own Asterisk machine

I'm new to VoIP and SIP, have experience in Linux and servers in general just not SIP servers and such. I was curious about experimenting a little with SIP and VoIP. So I ordered a SIP account from ...
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1answer
185 views

VOIP phone working intermittently

I'm having problems with my Polycom VVX400 phones. It's the first time i've configured them to work with Aircall.io, our SIP provider. I have them registered just fine and I can make calls, but when ...
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1answer
434 views

freeswitch received audio delayed for 20-30 seconds

I manage to make a very simple configuration for basic phone to phone calls using FreeSwitch, When i Calls A -> B (and B answered), A can hear B instantly, but B cannot hear A, after waiting for 20-30 ...
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3answers
325 views

FAX on VoIP line does not work

My provider switched me to an IP-based line. Now my analog ISDN (G3) Fax does not work anymore. Is there any way I can use the conventional Fax with a SIP connection? My Router supports T.38 Fax ...
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1answer
369 views

Getting SIP, RTP and NAT right : who is doing what?

I'm a little confused about who (PBX vs. phones) is doing what in our IP telephone setup. We have our phone system hosted externally. This means that we've got physical IP telephones installed at ...
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0answers
640 views

Kamailio as TCP-only SIP register and relay

I have a rather complicated setup in which I'd like to run a SIP server. In general, I want to run an IPv6-only Kamailio SIP-server on internal network and have outside SIP-clients be able to make ...
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0answers
274 views

Linphone iOS behind CGNAT sets wrong Contact Header

We are using Linphone client on iOS over Verizon network. Our client sends wrong IP address on Contact header at 200 OK message as a response to INVITE message from Asterisk. SIP trace Our client's ...
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1answer
4k views

Hacker bypassing iptables

(moved from SO) I have iptables protecting a sip server. It blocks all IPs except ones I specifically opened, and it seems to work for almost everyone. I have tested from lots of ip addresses that ...
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1answer
1k views

Phone won't register with Asterisk server

I'm trying to register a Grandstream phone on our asterisk server. We've done this tens of times, have documented how to do this, and recently it doesn't work anymore... I've tried with several phones ...