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 ...
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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 ...
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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 ...
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route incoming asterisk sip calls - fake auth rejected
we are trying to handle incoming sip_calls on our asterisk server, but somehow we always end up getting either 403 or 603 (which should be the same?)
what should happen if it works:
user calls ...
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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 ...
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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 ...
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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 ...
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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 ...
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SIP Trunk for VOIP behind a SonicWALL
I appreciate this question is quite "out-there" but - has anyone had any success with VOIP over SIP behind a SonicWALL. We are having an issue with one-way audio inbound.
Essentially, the person ...
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How can Asterisk voicemail handle call holding?
Given an Asterisk PBX system version 20 or above. Given that a user calls another user, the call is missed and is sent to voicemail. While the caller is directed to voicemail, the caller receives ...
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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 ...
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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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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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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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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 ...
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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 ...
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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 ...
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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 ...
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Command line SIP dialer
Is there a simple command line SIP dialer for unix which can connect to SIP server, make a call and play some media file (wav/mp3)? In ideal I would look like this:
sip-dailer +1xxxxxxxxxx /path/to/...
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Asterisk dialplan context and label clarifications
I have been learning Asterisk dial plan for the past week.I have written down a simple IVR system with two levels of menu and an exit option.I have used concepts from different tutorials on the web....
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Configuration of a Cisco IP Phone 7916 and 7975g
Hi everyone,
I would appreciate some help configuring my Cisco IP Phone 7975.
In fact the phone works well with the SIP firmware (version SIP75.8-4-1ES2) and an Asterisk Server. Now I would like to ...
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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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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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SIP INVITE packet has WAN address rather than call manager LAN IP
I am using SIP between two subnets (192.168.3.0/24 and 192.168.30.0/24) each connected via VPN.
I have a call server on 192.168.3.100, and two phones 192.168.30.118 (Ext. 3128) and 192.168.30.119 (...
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How to have dynamic extensions in Asterisk
I have recently swapped VoIP providers. So far so good, but I have ran into one little snag (and I'm honestly not really sure how to ask / Google this).
With my old provider I could have inside my ...
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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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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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How can I simulate a modem over a VOIP connection?
I have a hardware device I need a server to dial into at regular intervals. The problem is I no longer have a POTS line or a modem in any of development computers, and all my production servers are ...
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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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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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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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How to use conntrack to allow SIP traffic
I have the following problem:
Device (eth0)----> SWITCH(trunk)+VLAN120 ---> (PC1)
+VLAN200 ---> (PC2)
I am able to ping from PC1 to PC2 which are in ...
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Kamailio cannot connect to remote Mysql Server
I installed Kamailio 4.0 on a Ubuntu-Server 14.4 and a MySQL Server on another Ubuntu-server using Virtualbox.
192.168.0.131: MySQL Server
192.168.0.141:Kamailio 4.0
I changed the address of ...
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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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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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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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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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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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Unregister SIP UAC message
I've looked so much on the internet, but I could not find a any SIP unregister example, and when I search RFC 3261,3665 the word does not even appear, perhaps I'm searching for the wrong phrase. I ...
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How to install Asterisk with SRTP support?
I'm rather new to Asterisk, and I need my server to support WebRTC. As far as I know, Asterisk version in Asterisk Now is compiled without SRTP support, which is necessary for WebRTC.
So, I try 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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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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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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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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Security, hardening and NAT for RTP and SIP
I decided to create a VOIP server and it has gotten more complicated then I thought it would. So in order to make my server more secure, I keep it behind a pfSense appliance. I use IP filtering to ...
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