Welcome to the VOIP Wiki - a reference guide to all things VOIP
This Wiki covers everything related to VOIP, software, hardware, service providers, reviews, configurations, standards, tips and tricks and everything else related to voice over IP networks, IP telephony and Internet Telephony. However, the Wiki is primarily for information, not for advertising.
Your contributions are welcome, please read the How to add information to this wiki page and the Posting Guidelines before you post.
This Wiki covers everything related to VOIP, software, hardware, service providers, reviews, configurations, standards, tips and tricks and everything else related to voice over IP networks, IP telephony and Internet Telephony. However, the Wiki is primarily for information, not for advertising.
Your contributions are welcome, please read the How to add information to this wiki page and the Posting Guidelines before you post.
NEWS
- 2010-09-06 - Kamailio (OpenSER) SIP Router Masterclass - advanced training in Berlin, Germany, Nov 8-12, 2010
- 2010-09-06 - 10 EUR startup balance for your free hosted callshop billing software
- 2010-09-03 - 9 Years SIP Express Router (aka SER) - a walk through the development of the open source SIP proxy and application server
- 2010-09-03 - Telesis A.S. announces XApi (Xymphony API Project) for software developers (released under simplified BSD license)
- 2010-09-03 - Codec2 low bit rate open source speech codec V0.1 Alpha Released.
- 2010-09-01 - PBX-In-A-Box Releases New Cisco IP Phone XML Services for General Use Setup your Service URL to http://www.pbxinabox.com/ciscoservices.php
- 2010-09-01 - New VXI* VoiceXML browser 5.1 for Asterisk released! by I6NET
News Resources
- Software Releases - Check here for recent VoIP-related software releases.
- VoIP Services - Check here for news on Voip Services.
- VOIP Event Calendar - Check here for news on VOIP Events, Tradeshows, Training and Conferences
- Older News - Check out older news articles here
Getting Started
- What is VOIP? The very basics.
- Free VOIP Publications: Magazines and Newsletters free to qualified subscribers about VOIP related products
- Training: Seminars, tutorials, on-line classes. Check here for recent.
- VOIP Consultants: Finding help.
- VOIP Service Providers - VOIP service providers.
- DID Service Providers - VOIP call origination service providers.
- VoIP Practical Guide - VoIP Resources for Your Home and Business.
- How to start a VOIP Business
Connecting VOIP to the PSTN and Cellular Networks
- Cheapest ATAs and Service - For Calling PSTN Numbers, or receiving phone calls from PSTN Numbers
- ENUM - Translating E164 numbers to VoIP addresses
- Failover Switches - Automatically or Manually Switch PSTN Interfaces on failure for reundancy
- FXS-FXO Converters - Convert an FXS interface to an FXO interface
- Phone Numbers - Dozens of test and information numbers you can call for free.
- PSTN Gateways - VOIP to PSTN gateways (also known as: Media Gateways)
- Routing calls using a free international calling service - How to use two-step dialing to save on costs.
- VOIP GSM Gateways - VOIP to GSM gateways
PBX and Servers - VoIP PBX and Servers
Popular choices - please do not alter this list, add new entries here- Asterisk: Open Source PBX
- CallWeaver: Vendor independent, cross-platform open source IP-PBX, originally derived from Asterisk.
- FreeSWITCH: Open source cross platform (*nix, Win, OSX) multiprotocol (SIP, IAX, Googletalk, Jabber, XMPP, WOOMERA/H.323, PRI/T1/E1) softswitch.
- Kamailio: Flexible and powerful open source GPL SIP ( RFC3261 ) server with integrated TLS (former OpenSer)
- Sip Express Router: High-performance, configurable, free SIP ( RFC3261 ) server
- sipXecs: SIPfoundry's sipXecs Project - The SIP PBX for Linux (L-GPL) - Utilizes FreeSwitch, OpenFire and OpenACD
- 3CX Phone System: Windows PBX with free and commercial versions
- Yate - open source softswitch/PBX, with E1/T1, ISDN, SS7, RBS, analogics, IAX, H.323, SIP, MGCP, Jingle (GoogleTalk), for Windows, Linux and BSD's
- more...
Protocols - the language of VOIP
- IP Protocols COPS, ENUM, H.323, IAX, IMS, LTP, Megaco, MGCP, PINT, RTP, SCCP, SCTP, SDP, SIMPLE, SIP, STUN, T.37, T.38, TRIP, TURN
- ITU protocols SS7, ISUP
- OSP, PacketCable MRCP
- Encryption Protocols ZRTP
Markup Languages
- Basic call routing and rules for UA's or VOIP servers CPL
- IVR Presentation and dialog management: VoiceXML, CallXML
- Call control / conferencing / call routing: CCXML
- IVR / Speech recognition definition: SRGS
- IVR / Speech synthesis definition: SSML
- IVR / prompting / recording / conferencing / DTMF / Voice: CallXML
Traditional Telephone Network
- Analog Telephone Information
- PBX features
- PSTN Interface Hardware
- RESPORG: Toll Free 800 Number Programming
- Telecom Dictionary
- Telco Engineering Information
- Telephone History
VOIP Events and Conferences
- Astricon
- AstriEurop The Asterisk European Exhibition
- Asterisk-Tag.org Annual German conference on Asterisk and related topics
- ClueCon Annual conference on open source telephony development
- Global VoIP and Telephony-related events
- Training and Conferences - Check here for recent Training and Conferences
- VoiceCon Annual conference on IP Voice Communication.
- Voice Peering Forum on routing, interconnection and peering of Web2.0 & VoIP networks
- VoIP business related events
- VOIP Event Calendar — List of upcoming VOIP related events, Conferences, Trade Shows, Training, etc.
- VoIP Today Asterisk and VOIP related events, Conferences, Trade Shows, Training, etc.
Business Services
Resources
- Zonetel Forum is a free IT community that welcomes, shares, and contributes knowledge about telecommunications such as Asterisk, VOIP, DID Number, Grandstream, Skype and Sangfor related topics.
- Phone System Tech Forum
- Twitter VoipUser Directory: Twitter VoipUser Directory
- VOIP IVR Usage: Real world examples of business VOIP usage
- VOIP Websites: Other VOIP websites on the Internet
- Voip Forum: Free voip forum for the voip community.
Suggestions and Questions
- How to add information to this wiki
- Suggestions: Put your requests and suggestions here
Don't Know How To Wiki But Want Your Information On This Site??
- Your contributions are welcome, please read the How to add information to this wiki page and the Posting Guidelines before you post.


Comments
333how to using speex lib to cancel echo in IP phone?
I am developing an IP Phone on ARM(samsung 6410; CPU:667MHZ), and use lib mediastreamer.2.6.0 for audio
communication, but now the echo occur while talking. In the lib mediastreamer.2.6.0, I find a filter can
do echo Cancellation by calling lib speex. I active it to filter the collected voice, but the telephone
receiver can't hear any voice. Maybe I setup parameters wrong, the parameters are as follow:
int ec_tail_len = 100;
int ec_delay = 20;
int ec_framesize = 0;
if (use_ec) {
stream->ec=ms_filter_new(MS_NEW_SPEEX_EC_ID);
ms_filter_call_method(stream->ec,MS_FILTER_NEW_SET_SAMPLE_RATE,&pt->clock_rate);
//inec_tail_len = 100;
//ec_delay = ;
if (ec_tail_len!=0)
ms_filter_call_method(stream->ec,MS_ECHO_CANCELLER_SET_TAIL_LENGTH,&ec_tail_len);
if (ec_delay!=0)
ms_filter_call_method(stream->ec,MS_ECHO_CANCELLER_SET_DELAY,&ec_delay);
if (ec_framesize!=0)
ms_filter_call_method(stream->ec,MS_ECHO_CANCELLER_SET_FRAMESIZE,&ec_framesize);
}
Expecting anybody can give me some advices to cancel the hateful echo.
Best regards!
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Thanks
333unconsistent calling serice
Resetting CIC 127
Jul 27 11:12:45 WARNING23832: chan_dahdi.c:9489 ss7_linkset: RSC on unconfigured CIC 127
Jul 27 11:13:15 WARNING23832: chan_dahdi.c:9725 ss7_linkset: CGU on unconfigured CIC 98
Jul 27 11:14:05 WARNING23832: chan_dahdi.c:9715 ss7_linkset: CGB on unconfigured CIC 98
Jul 27 11:14:33 WARNING23832: chan_dahdi.c:9512 ss7_linkset: GRS on unconfigured CIC 65
Any information to help me interpret this information and any other logs I can look at to make it clearer where the problem is, will be greatly appreciated.
Note when I switch off asterisk and run an ss7linktest these are my results if this is of any help;
Link state change: ALIGNEDREADY -> INSERVICE
0 MTP2 link up
Len = 20 ff 80 11 81 02 40 00 00 11 a0 32 35 36 34 32 38 36 32 38 38
FSN: 0 FIB 1
BSN: 127 BIB 1
>0 MSU
ff 80 11
Network Indicator: 2 Priority: 0 User Part: STD_TEST (1)
81
OPC 1 DPC 2 SLS 0
02 40 00 00
H0: 1 H1: 1
11
Len = 25 ff 80 16 f1 63 42 03 0c 11 f0 a5 a5 a5 a5 a5 a5 a5 a5 a5 a5 a5 a5 a5 a5 a5
FSN: 0 FIB 1
BSN: 127 BIB 1
<0 MSU
ff 80 16
Network Indicator: 3 Priority: 3 User Part: STD_TEST (1)
f1
OPC 12301 DPC 611 SLS 0
63 42 03 0c
H0: 1 H1: 1
11
And for another signaling channel
Link state change: ALIGNEDREADY -> INSERVICE
0 MTP2 link up
Len = 20 ff 80 11 81 02 40 00 00 11 a0 32 35 36 34 32 38 36 32 38 38
FSN: 0 FIB 1
BSN: 127 BIB 1
>0 MSU
ff 80 11
Network Indicator: 2 Priority: 0 User Part: STD_TEST (1)
81
OPC 1 DPC 2 SLS 0
02 40 00 00
H0: 1 H1: 1
11
Len = 25 ff 80 16 f1 63 c2 03 0c 11 f0 a5 a5 a5 a5 a5 a5 a5 a5 a5 a5 a5 a5 a5 a5 a5
FSN: 0 FIB 1
BSN: 127 BIB 1
<0 MSU
ff 80 16
Network Indicator: 3 Priority: 3 User Part: STD_TEST (1)
f1
OPC 12303 DPC 611 SLS 0
63 c2 03 0c
H0: 1 H1: 1
11
Received MSU with network indicator of national_spare, but we are national
Len = 25 80 81 16 f1 63 c2 03 0c 11 f0 a5 a5 a5 a5 a5 a5 a5 a5 a5 a5 a5 a5 a5 a5 a5
FSN: 1 FIB 1
BSN: 0 BIB 1
<0 MSU
80 81 16
Network Indicator: 3 Priority: 3 User Part: STD_TEST (1)
f1
OPC 12303 DPC 611 SLS 0
63 c2 03 0c
H0: 1 H1: 1
11
333How to connect a Cisco Router with PRI module to Asterisk
I have a cisco router 2811 with PRI module in it. PRI will use only for voice traffic. On the other side
i am using asterisk. Both asterisk machine and router are connected to a switch. Can any one help
me out and share the configuration of both router and asterisk with PRI.
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thanx Martin
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