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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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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 able to communicate with the reception. However,…
Disney Ganbaatar
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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 port 5060 to communicate with the Swyx. They can…
Venix
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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 calls when we can not for some reason. To redirect…
Adonist
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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: Ether01: ISP Connection(Internet/WAN) Ether02: SIP…
Johan Marais
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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 is disabled and system is not behind NAT. What could…
bluewhale
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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 password) into my soft phone settings I am able to…
Tono Nam
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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,tcp or even simply transport=udp) the phone can no…
fileinsert
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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). Note that everything works fine if: The call is…
Santiago Palladino
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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 dialler server. How would I set this up in Winbox? Will…
JoseB
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Load balancer + CPU load manager to handle SIP connections?

I no longer have a landline and subscribed to a VoIP provider to have a landline phone number. I prefer to use a softphone (EyeBeam, the commercial version of XLite), but there are times when a call comes in while a big download/compiling is under…
OverTheRainbow
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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 our location, but the PBX is located elsewhere.We've…
sbrattla
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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 calls to the inside over IPv4-only network. This is…
TJJ
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How to configure Gizmo5 on Trixbox/Asterisk

Does anyone know how to configure Gizmo5 for use on an Asterisk PBX? Specifically Trixbox if possible. For the life of me I just can't get this to work...
Ivan
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Is there a stable cross-platform softphone? Or a Linux-only one?

We use an Asterisk based solution as our PBX, and although we've found hardphones work flawlessly, we have yet to find a reliable softphone. The basic requirement is that it allows multiple calls and transfers. I don't have a comprehensive list, but…
Ivan
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Asterisk with softphone and webphone

I have a problem with Asterisk. I wrote a WebPhone, so I should add to my peers some config details: encryption = yes avpf = yes icesupport = yes dtlsenable =yes dtlsverify = no dtlssetup = actpass dtlscertfile = valid path to…
Matt
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