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voip-info.org

Created by: system,Last modification on Thu 29 of Jul, 2010 [19:32 UTC] by jgg

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.


NEWS



News Resources



Getting Started



Connecting VOIP to the PSTN and Cellular Networks


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
  • sipX Solution Summary: SIPfoundry - The SIP PBX for Linux (L-GPL)
  • 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



Markup Languages

  • Basic call routing and rules for UA's or VOIP serversCPL
  • 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


VOIP Events and Conferences


Business Services


Resources



Suggestions and Questions



Don't Know How To Wiki But Want Your Information On This Site??


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222

333Pak/Egypt/BD available in Aggressive Rates (Summer offer)

by symphonictel, Thursday 29 of July, 2010 [21:17:18 UTC]
Dear All,

This is Syed Ali, Manager Sales at Symphonic Tel. Let me introduce my company first, Symphonic Tel is now become an emblematic slogan in the world of VoIP as we cope up heavy amount of traffic competently for Asia and Africa region. We are aware that lot of companies are working under this VoIP’s umbrella but our professional staff & their skills make the contrast clear. Currently we have good capacity for Pak Premium CLI, Egypt Premium & Non CLI, Bangladesh & India CLI/ Non-CLI. Kindly let us know if you are interested in any of these routes. We can offer you good competitive rates and make a good business deal out of it. Also let us know your target rates and capacity requirements.

All routes are completely FAS free with good ASR & ACD. We have 24*7 proactive NOC.

Now payment made easy! Symphonictel accept payments through Bank wire & Western union on prepay & postpay basis. Hope to Hear from you soon.

Best Regards,
Syed Ali,
Manager Sales,
Symphonic Tel.
Email/IM: sales@symphonictel.com / syed.ali@symphonictel.com
222

333Re: How can I do VOIP call from my Mobile?

by alice27, Thursday 29 of July, 2010 [13:12:40 UTC]
On Android you can use IMSDroid which is free and open source
222

333PHP VoIP ?

by rfresh, Wednesday 28 of July, 2010 [05:20:54 UTC]
I found this forum and hope its the right place to post my question. I am looking for a company or service that will allow me to use PHP to make outbound phone calls. This is not for sales or marketing. My business allows schools to txt msg and email parents and I would like to add phone calling capability as well. Is there a service out there that will allow me to write PHP code to make outbound VoIP calls?

Thanks

222

333unconsistent calling serice

by dmuwazi, Tuesday 27 of July, 2010 [08:28:43 UTC]
Hallo Admin, am a new student of asterisk, ss7 and sangoma and am having a challenge in that the calls keep going on and off. when I run asterisk -r and get into its command line mode am seeing this kind of message and I have failed to interprete it from google

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 
222

333How to connect a Cisco Router with PRI module to Asterisk

by adnanshahid, Wednesday 21 of July, 2010 [17:53:45 UTC]
Hi All,
  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. 


222

333define volumn for conference

by martin-hermann, Thursday 27 of May, 2010 [09:49:00 UTC]
Hi, in my public.xml, i have created a conference and basic functions are OK, which includes a PIN. My problem is, that the volumn of the sound files, which are played (PIN and music) is too low. Is there a way, to increase the volumn for that?

thanx Martin
222

333New Powerful LCR platform RouteNGN

by gentelsupport, Thursday 29 of April, 2010 [15:20:28 UTC]


Powerful LCR and Routing Solution

RouteNGN is a powerful, cost-effective and easy-to-use, high-capacity routing solution that provides maximum opportunities for flexibility.

RouteNGN is compatible with any RFC-compliant SIP agent such as a switch, gateway, and/or SBC, and it enables you to leverage your existing assets, both people and equipment. All your network devices talk to a single, centralized LCR and you interface to RouteNGN via any internet-enabled computer with an easy-to-use interface.

RouteNGN also has APIs for integration with other systems including billing and LERG databases. The transparent components work with any OS, across multiple locations. Easily add jurisdictional routing to route US domestic traffic by State (inter-intra), LATA, or OCN.

222

333the softswitch renovation: embeded hardware softswitch

by allywll_sylvia0403, Thursday 22 of April, 2010 [01:47:02 UTC]
hardware-based carrier-grade softswitch system. http://www.allywll.com/Product_in.asp?ProductID=44


222

333Re: Best VOIP configuration for 10 lines

by allywll_sylvia0403, Thursday 22 of April, 2010 [01:28:10 UTC]
if it is need control under about 10 line. The device with 8FXS and 8 smart lines may be can help you? 8FXS is for SIP account with RJ11 which can transfer if the call busy,
while the 8 smart line is for the analog telephone(PSTN line), which can keep your normal PSTN number for call in, and if the internet off it can choose the PSTN automatically。

Skype：yiyi0716
222

333Basic 2 way comms program/emulator

by gregoryj, Monday 19 of April, 2010 [10:57:38 UTC]
I'm currently working on a training system, part of which is emulating the comms between 2 team members. We need to replicate the user interaction with a comms terminal (Box with buttons and dials). I have already created the emulated equipment GUI, I just need something to handle the the back-end comms between the student emulation and the instructor. I'm an experienced C++ programmer so I'm happy to get dirty with an API/SDK if anyone can suggest anything. I've seen a few APIs such as asterix and freeswitch, but they're overkill for the simple communications I need to emulate.

Many thanks for your suggestions!