voip

voip

AudioTrack Latency

Dial -Out with Asterisk Sip - circuit is busy error

Display arabic correctly on ip phones

Calling short code number (support services) with Asterisk GSM Gateway

Why asterisk server updates the status of clients after one minute

How to do scalability testing for a JTAPI application

route/redirect voip to landline with asterisk and Linksys SPA2102 or 3102

Kamailio text file CDRs

Using prio qdisc for voip [closed]

what do i need to develop an asterisk script to auto create and assign extensions to clients who have joined my network

Is it correct to use voiceXML as a tool in this scenario

Cisco call manager api to conference a device

Is there a industry interface standard for TTS and ASR?

Is GSM6.10 audio format block or stream based?

How to generate call to a mobile using asterisk

Is it possible to have users log in and out of handsets?

remote dial using SIP client

Asterisk dynamic user agents on PSTN

Can't get libvlc to play sdp file

How to deduct customer balance when has more than one call connected at a time

RFC 4474 and special case of REGISTER methods

asteriskwin32 and WS

How to call telephone number through asterisk trunk from external application talking through prophecy?

SIP to PSTN gateway connection from asterisk?

Asterisk: Unsupported SDP media type in offer: video 0 RTP/AVP 0

H.460.18 and NAT

linphone code compilation

Modem toggle in PAP2 in dial plan?

Difference between SIP and H.323

Is Red5Phone Full Duplex VoIP Application? [closed]

how to use SNMP on Linksys PAP2T? [closed]

gnugk, openmcu and tandberg VC Unit

Voip callerid handling

How to calculate effective time offset in RTP

calling a sip phone

How do VoIP connections differ with and without gatekeeper?

Connect a PC to PSTN network

Freeswitch mod_java installation problem

Associate Ext. Number with IP addr on H.323 voip call

DCS in IP telephony communication standards

2 way voice as in GTalk

Possible to change SIP message via proxy?

VoIP with Godaddy Server [closed]

VOIP provider in the UK

IP Telephone call to RTP Stream

Whats the best architecture for CDR integration to a Asterisk based Application

voice encryption on fly in mobile phones

RTP packet combining

SIP Callee does not get notification that call ended

"RFC 2833 RTP Event" Consecutive Events and the E "End" Bit

How can I mux/demux RTP media from one stream?

VOIP: How to Create a Web app to make Call like skype or msn? [closed]

VoIP and SIP, where to begin [closed]

Does anyone know any service similar to Tropo?

Answer modem using voip

Simple sip based client interaction... Any Ideas

Need advice on VOIP call trunking and termination

Structure of a Voice Chat application (Client/Server)?

Asterisk Manager Api events

How do VoIP services connect to landlines? [closed]

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