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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.

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.

The SIP protocol is an Application Layer protocol designed to be independent of the underlying Transport Layer; it can run on Transmission Control Protocol (TCP), User Datagram Protocol (UDP), or Stream Control Transmission Protocol (SCTP).It is a text-based protocol, incorporating many elements of the Hypertext Transfer Protocol (HTTP) and the Simple Mail Transfer Protocol (SMTP).

Source: wikipedia

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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 another call and answers it, putting the first call…
Charles Langlois
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Atxfer does not transfer

I have two agent A and B. Agent A has a call online and wants to transfer the call to agent B. I already have axtfer => *2 configured in the features.conf file and I have alltransfer set to YES on my platform with Magnusbilling, but I still can't…
drop sip
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Rocky Linux 8 Blocking SIP traffic

I am having trouble with Rocky Linux blocking SIP traffic from an external LAN. If I send a SIP message from external LAN, it reaches the server but the message doesnt get passed to the application. If I send a SIP message from local LAN, it reaches…
FredBloggs78
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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 that same connection to send calls to me. I…
JMain
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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 ability. As I understood, searching the web, I can't…
Ngdgvcb
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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 amount of knowledge but seemingly not enough. I am…
xevoryn
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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 equipment play with eMTAs. I installed Asterisk on…
Oleksandr Znachkov
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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://github.com/GoTrunk/asterisk-config/tree/dynamic-ip…
xenon
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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 customer base (<100) security and desaster…
Kasi-Laubfrosch
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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 => addheader,1,Verbose("Setting header") exten =>…
Pownyan
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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 correct answer from the trunk. The trunk was working…
Spuria
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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 peer from the grandstream never arrives , what…
Leonel Franchelli
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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 introduction of the new topology, I have the…
pgr
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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 home network and it is quite not restricted. On…
Bastien Matthai
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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 --------------- eth1.100|iptables NAT (Debian buster)|eth0.100…
Surfinside
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