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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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Adding more than one IP to sip.conf tcpbindaddr and udpbindaddr (Asterisk 1.8)

The defaults in the sample sip.conf under Asterisk1.8 contain udpbindaddr=0.0.0.0 and tcpbindaddr=0.0.0.0. I want to bind the incoming [foo] extension to udp:192.168.1.1/255.255.255.0, the outgoing [foo] to tcp:192.168.3.3/255.255.255.0, the…
tarkaköpönyeges csecsfogdosó
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sending faxes over sip with t38modem and hylafax

I would like to send (and not to receive) faxes. AFAIK, t38modem + hylafax should work fine. After configuring hylafax to look for ttxT38-1,ttxT38-2,ttxT38-3, I'm running the following: # t38modem --no-h323 -u T38modem --sip-no-listen --ptty…
Akasha
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Lync 2010 SIP Trunk Providers

I'm playing around with a small Lync implementation in my home office and would like to try replacing my 3cx server with it. Currently I use callcentric as my VoIP provider, but based on another answer on serverfault, it looks like CallCentric won't…
Joshua Dale
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How can I record a call transfer with freeSwitch?

I am using freeSwitch to terminate calls that originate on the local network from an IVR system. I have it working with several different voip terminators, and I can record sessions successfully, except when the call gets transferred. When I do a…
Eric Z Beard
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Asterisk SIP digest authentication username mismatch

I have an asterisk system that I'm attempting to get to work as a backup for our 3com system. We already use it for a conference bridge. Our phones are the 3com 3C10402B, so I don't have the issue of older 3com phones that come without a SIP…
Matt
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Router gets disconnected once I terminate my SIP application

Here is an interesting one, I have a SIP VoIP application which is able to register to the PBX server, and I can invite and see the user call on the callee end receiving an Invite, and on the caller end I see the Ringing response... now here is…
TacB0sS
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FreeSWITCH Dual Stack IPv4/IPv6

I'm currently trying to understand how I can enable my freeSWITCH to talk both IPv6 as well as IPv4. Currently, I thought it was going to be easiest to first create a set-up which works on IPv4 and then switch the IPv4 address for an IPv6…
Xabre
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Capture the VOIP calls receive to a sip phone that send Notificantions to an API on a call

This is whats I'm trying to achieve I want a SIP phone or (is there is any other way I'm open too) to take the calls it receives from a VOIP provider take the information related to the call when someone calls e.g: Call ID, The call it self(maybe)…
Luis Javier Davis
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Provisioning a Unidata ICW-1000G WiFi phone

Apologies if this is off-topic here (if so, is there a better place?). I just bought a Unidata ICW-1000G WiFi phone (because I prefer to use just WiFi and not to use DECT). I'm attempting to do auto-provisioning with it by using the instructions in…
James Youngman
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One Way Audio making "internal" SIP calls

I have just moved our phone system over to SIP using BT Voice Cloud and having a problem with one way audio when making "internal" calls. The original router (Vigor 2860n+) is the DHCP server, DNS server for the network and I need to keep that…
Gavin
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Replacing IP address of connections from AWS towards SIP

We have installed a PBX on AWS and connected it to our on-prem Router via VPN. My on-prem router is connected to the SIP provider via a physical connection with another on-prem MUX device (device given by sip provider). All connections are working…
vichar
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Skype for Business calls behind pfSense dropping after five seconds

I'm having a problem with Skype for Business (Lync) disconnecting behind a pfSense firewall after exactly 5 seconds. It's an odd problem, and I've opened a ticket with Microsoft, but thus far, that has proven fruitless. Here is the…
C Hamm
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Asterisk terminating outbound call when picked up, sends 'BYE' message

I'm running Asterisk 1.6.1.10 / FreePBX 2.5.2.2 and I've got an outbound trunk setup. Everything use to work fine until recently (perhaps due to upgrade to FC12 or other things I'm not sure). Anyway the setup does not appear to have issues…
vo
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Using Asterisk as a gateway to Provider

I have 2 sip servers on different LANs. Freeswitch and another is Asterisk. Asterisk sits on a VPN with a provider and who has provided DIDs. All users register on Freeswitch. How can I route calls to the provider through Asterisk and back, I tried…
Ben Kabale
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iptables blacklist empty User-Agent sip tcpdump

I scan using tcpdump for user-agents that hit my phone server. I get interesting results. i add them to a list that protects alot of my servers out there. This works good. The problem is I get blank or empty user-agents. how do i block that? here…
user2630659
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