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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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Custom ENUM/NAPTR DNS Server

I need to configure custom DNS server to handle ENUM requests for VoIP calls I have custom built DNS server in Java that handles ENUM requests for my company's VoIP services. ENUM is used to dynamically route the calls and it uses own routing logic,…
Matthias
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Networking issue with asterisk server

I replaced an IVR machine for incoming call after going down. It is running asterisk 1.4.23 on ubunutu 10.04 I decided to put the server behind iptables because my server was under brute force attack. eth0 is my private card and eth1 is the public…
meda
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How to route/forward/connect incoming SIP call through other SIP provider to a phone number?

I have 2 VOIP SIP accounts. One is for a landline (incoming). The other should be purely outgoing and allows for calls to my mobile phone. Basically I want to connect the landline to my mobile phone. With VOIP on my mobile phone, call quality is…
Andreas Reiff
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Fortigate restrict traffic through one external IP

I've got a fortigate 400A at a client's site. They've got a /26 from British Telecom, and we're using 4 of those IPs as a NAT Pool. Is there a way to say that traffic from 172.18.4.40-45 can only ever come out of (and hence go back into) x.x.x.140…
Tom O'Connor
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can't register a soft phone to asterisk11

I have a VM (on oracle vbox) running Fedora17. I've installed asterisk 11 on it from sources. I've followed the wiki for installation (https://wiki.asterisk.org/wiki/display/AST/Creating+SIP+Accounts) to the letter. The ip on the VM machine running…
Tom Klino
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Why are SIP calls via my server silent?

I have FreeSWITCH SIP server up and running. It has public IP and sits behind 1-to-1 NAT (it's Amazon EC2 instance actually). I can connect to it, make a call to other endpoint (namely, my android device to my pc and vice versa) and signals are send…
zencodism
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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.Can someone confirm if the IVR below is correct?…
liv2hak
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What do I need to go from having a domain to being able to take a sipuser@mydomain.com call?

(Sorry if this question is a little basic or should go on a different SE site. Just let me know.) I'm trying to learn about SIP. The problem is that I'm at the point where I don't know enough to try it myself to learn and sometimes I don't know…
Azendale
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Setting up office phone system (small business)

I've been asked, within six months, to deploy a phone system. The exact wording was "a phone on every desk." They want inter-phone transfers, external calling, and even software IP phones for our travelling sales guy. If this wasn't all enough I…
Pinam
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How to pass/process extension numbers while calling Freeswitch from outside SIP gateway?

Suppose I have Freeswitch, which has SIP gateway configured. So, the local users of this Freeswitch can call outside via
user102132
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Asterisk + SIP 404 not found

I want to make a small Asterisk server in my house. I installed asterisk on my Ubuntu and I use 2 computers, in order to connect to one another. when I connect I can see in Wireshark that registrar ok. here is the output of sip show peers…
Uriel Frankel
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how UAC should handle SIP 183 Session Progress

I have the following scenario: 2 UAC are trying to talk, via a remote SIP server (openSER/Kamailio 3.1.3) = client infrastructure. The UAC software was developed over a local test infrastructure using Asterisk, where it was possible to establish a…
hovanessyan
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Sip configuration for Elastix, for Terrasip (Provider) with Multiple DIDs

I have just purchased several DIDs with Terrasip I have configured my Elastix (freepbx) SIP trunk according to their suggestion: Please follow this template configuration. (valid for outbound/inbound traffic) Outgoing trunk name…
abutbul
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how to associate an ip with a name?

I have installed an asterisk on my private network(so it s behind a router ). Is there anu way to set the asterisk on a public ip though it's installed in a server with private ip? As Ive read it ispossible using externip and externhost. My second…
mee
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SRV records not populating correctly

I have a lync box with a SIP SRV record that isn't working for a significant portion of clients, despite their ability to resolve the FQDN of the edge server. What could cause inconsistent resolution of SRV records or an inability to resolve SRV…
Couradical
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