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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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Asterisk Sip Server and "Screen Sharing" function

I'am newbie in asterisk, i followed the tutorials on the asterisk wiki, and installed the latest version (13) of the asterisk server. I was able to setup the voip now i wonder, there is a function im my SIP client - "screen sharing". Is it require…
Raziel
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Hot to integrate sipJs and freeswitch?

I am trying to integrate sipjs with freeswitch. below is the config I am using var config = { // Replace this IP address with your FreeSWITCH IP address uri: 'sip:1002@***.***.1.170', // Replace this IP address with your FreeSWITCH IP address //…
JohnN
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Linux Redhat inbound NIC routing

I am running a SIP application on a Redhat Linux server with 4 nics on separate LANs. SIP runs on UDP port 5060. When I send a inbound packet to port 5060 NIC0, the packet gets delivered to the application. When I send a packet to port 5060 to the…
Parmalat78
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Asterisks Dialplan Extensions

I'm playing around with a very simple asterisks setup. My asterisks server is 192.168.1.10 (Ubuntu 12.04), my 2 sip clients are 192.168.1.20 and .21 (both using ubuntu 12.04 and Jitsi as a sip client). I have 2 users (user1 and user2) who can…
l0sts0ck
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ASA is not forwarding packets to egress interface

We are seeing some very strange behaviour from our Cisco ASA 5505 running 9.1(2) We have a SIP PBX inside our network. It's got a bit of an odd configuration, it listens for inbound SIP requests for our trunk on UDP/60052. So in our ASA, I have a…
Mark Henderson
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One-way audio problems with SIP over AT&T

I have a customer who has an VOIP PBX connected to a Level 3 fiber connection. He has offices all across the country using different ISPs. Two of those offices use AT&T, both in different states. One is T1, the other is DSL. For the past week, every…
pooter03
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Asterisk 11.* TLS Configuration

I have installed asterisk 11.12.0 on CentOS (64-bit). When enabling the TLS support i get the following error WARNING[7620]: tcptls.c:673 handle_tcptls_connection: FILE * open failed! i tried disabling the TLS option on the server but still get this…
john
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Dual WAN setup in Redhat/Centos Linux

I want to setup a Linux Server running SIP (freeswitch) applications with 3 NICs as follows: 2 of the NICs are WAN connections and the 3rd is a Private network for routing requests to other hosts on the same subnet (and does not figure into the…
Shekster
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Cisco ASA 5500 - SIP ports other than 5060

Is it correct that the SIP inspection in the ASA 5500 firewalls only kicks in for traffic on port 5060? There is some hint at this, while not 100% definitive, on Cisco Docs -…
nepdev
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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 service number -> remote asterisk accepts call from pstn…
longbow
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Allow specific SIP connection FreePBX

Hey I am using FreePBX with firmware: 5.211.65-14 and service pack 1.0.0.0 On this asterisk server I have everything up and running, but inbound phone calls might be rejected: WARNING[56522][C-00000ba4]: Ext. s:6 @ from-sip-external: "Rejecting…
BonifatiusK
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sip unreachable after changing routes using linux router

SIP server was unreachable after changing routes. I have 2 ISP ,ISP 1 is my default and ISP 2 is my back-up. I have a dialer behind linux router linux. Linux router has 3 ethernet cards. Router network config root@intellipatient:~# ifconfig -a eth0 …
serutnev
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SecAst: Failing to Ban IP ''

To Generation D: This was the setup at time this issue was observed: secast-1.0.1.0-x86_64-ub12 on Ubuntu 12.04.4 Server LTS with Asterisk 11.10.2. The following events were captured and observed in the /var/log/secast after leaving seacast (build…
Elyod
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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 (Ext. 3126) on the remote subnet. The WAN IP on the…
morleyc
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Iperf UDP listening on used port

I get WARNING[9157] chan_sip.c: Retransmission timeout reached on transmission error on asterisk and trying to troubleshoot the problem. Now asterisk is listening on 5060 port: udp 0 0 0.0.0.0:10766 0.0.0.0:* …
gilbertasm
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