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voip-info.org

Created by: system,Last modification on Wed 20 of Aug, 2008 [10:06 UTC] by spamblock

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 & 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, but please read the How to add information to this wiki page and the Posting Guidelines before you post.

NEWS


News Resources


Getting Started


Connecting Phones to VOIP


Connecting VOIP to the PSTN and Cellular Networks


PBX and Servers - VOIP PBX and Servers

Please post new/other servers here, because they will be removed.
  • Asterisk: Open Source PBX
  • Bayonne: Open source PBX
  • FreeSwitch: Open source cross platform (*nix, Win, OSX) multiprotocol (SIP, IAX, Googletalk, Jabber, XMPP, WOOMERA/H.323, PRI/T1/E1) softswitch.
  • CallWeaver: Vendor independent, cross-platform open source IP-PBX, originally derived from Asterisk.
  • 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)
  • YATE - Open Source Linux/Windows GPL Telephony Server and Client (has support for SIP, H.323, IAX2, E1/T1, voicemail), H.323 - SIP translator.
  • more...

VOIP Misc.


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


VOIP Websites: Other VOIP websites on the Internet


Suggestions and Questions


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222

333with skype

by damianmontero, Friday 01 of August, 2008 [03:05:10 UTC]
I guess you're talking about how to get your Skype on your USB key or USB Drive?

I've seen some things online that show you hi.
222

333Re:

by batikova, Tuesday 29 of July, 2008 [15:31:16 UTC]
I´m sorry, but what is the reason to public news if not to announce news about your product? There is also a comparison of ME and SIP client and a clear manual right in the end, which I don´t find generic at all. Moreover 2N has no need to copy someone else´s articles. The article was written by a student of CTU and maybe it doesn´t look like as news for the first time because of catchy headlines but I think it can be considered as news for our customers.

What about these products? Are they not reffering to specific product to draw traffic?
http://asterjet.googlepages.com/voipinternational
http://www.indafon.com/signin

How do you imagine the right article not reffering to any producer?

222

333

by spamblock, Tuesday 29 of July, 2008 [13:07:55 UTC]
News is for news, let's agree on that. First two guidelines clearly state:
  • Ask yourself if the entry you are going to post is newsworthy, that is to say, does it look like a headline in a newspaper/magazine
  • Do not post advertisements disguised as news

The only reason for your posting is to draw traffic to the 2N website. The description is very generic and if one visits the actual page you're only referring to a scenario made possible by another vendor, Nokia. There is nothing newsworthy to that, as the SIP support has been there for some time and is fully documented on http://www.voip-info.org/wiki/index.php?page_id=3363.
222

333to spamblock

by batikova, Monday 28 of July, 2008 [15:05:14 UTC]
Hi,
could you, please, be more specific then "that type of ´news´"? What is wrong with the news about SIP client in the mobile phone
and with manual how do that? At least, when you are so brave to erase someone´s links, be so brave and provide some information
about yourself and don´t choose "private" mode.

Thank you for your reaction or email info, so we can discuss each other´s opinion.
222

333configuration d'asterisk-stat-2.0.1

by mettichi, Thursday 24 of July, 2008 [06:55:57 UTC]
salut les camarades:
svp j'ai téléchargé la verison 2.0.1 d'asterisk-stat.tar.gz mais j'ai pas aucune idéé sur sa configuratio pour la faire marché correctement
est ce que j'ai besoin de créer une base de donnée mysql?
svp j'ai besoin de votre aide
222

333connexion entre deux serveurs asterisk

by mettichi, Thursday 24 of July, 2008 [06:52:08 UTC]
salut :
j'ai besoin de votre aide j'arrive pas à connecter deux serveurs asterisk. le premier serveur à l'@IP 192.168.1.203 les extensions vont de 100 à 199.
le deuxieme serveur à l'@IP 192.168.1.67 les extensions vont de 200 à 299.
malgré que ma configuration est correcte je rencontre j'arrive pa à appeler depuis un serveur vers l'autre.
voici ma configuration:
1er serveur à l'@IP: 192.168.1.67
iAX.conf

server1
type=friend
user=server1
secret=server1
host=dynamic ; Nous obtenons l'adresse IP lorsque l'autre PBX s'enregistre
context=incoming_training_centre_calls
auth=md5 ; Securiser l'authentification
disallow=all
allow=g729
trunk=yes
qualify=yes ; Nous activons le trunking

extensions.conf

outgoing_training_centre_calls
exten => _1XX ,1,Dial(IAX2/server2:server2@server1/${EXTEN:2})
exten => _1XX ,2,Congestion ; En cas d'echec une tonalite de congestion est utilisee

incoming_training_centre_calls
exten => _2XX ,1,Dial(Zap/1) ; Appels provenant du centre de formation
                             ; diriges vers le telephone du telecentre


deuxieme serveur à l'@IP: 192.168.1.203*
IAX.conf

server2
type=friend
user=server2
secret=server2
host=dynamic ; Nous obtenons l'adresse IP lorsque l'autre PBX s'enregistre
context=incoming_telecentres_calls
auth=md5 ; Securiser l'authentification
disallow=all
allow=g729
trunk=yes
qualify=yes

extensions.conf
outgoing_telecentres_calls
exten => _1XX.,1,Dial(IAX2/server1:server1pass@server2/${EXTEN:2})
exten => _1XX,2, Congestion
incoming_telecentres_calls
exten => _2XX.,1,Dial(SIP/202)

voici le message d'erreur quand j'appelle:

Executing 102@internal:1 Dial("SIP/vente-b761ee68", "IAX2/server2:server2@server1/2") in new stack
Jul 23 14:17:56 WARNING4790: chan_iax2.c:8815 iax2_request: Unable to create translator path for unknown to ulaw on IAX2/server1-16384
   — Hungup 'IAX2/server1-16384'
Jul 23 14:17:56 WARNING4790: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'IAX2' (cause 0 - Unknown)
 == Everyone is busy/congested at this time (1:0/0/1)
   — Executing 102@internal:2 Congestion("SIP/vente-b761ee68", "") in new stack
 == Spawn extension (internal, 102, 2) exited non-zero on 'SIP/vente-b761ee68'
   — Executing 200@internal:1 Dial("SIP/vente-b761ee68", "IAX2/server2:server2@server1/0") in new stack
Jul 23 14:18:01 WARNING4791: chan_iax2.c:8815 iax2_request: Unable to create translator path for unknown to ulaw on IAX2/server1-16385
   — Hungup 'IAX2/server1-16385'
Jul 23 14:18:01 WARNING4791: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'IAX2' (cause 0 - Unknown)
 == Everyone is busy/congested at this time (1:0/0/1)
   — Executing 200@internal:2 Congestion("SIP/vente-b761ee68", "") in new stack
 == Spawn extension (internal, 200, 2) exited non-zero on 'SIP/vente-b761ee68'
   — Executing 202@internal:1 Dial("SIP/vente-b761ee68", "SIP/vente|20") in new stack
   — Called vente
   — SIP/vente-0a1752b8 is ringing
 == Spawn extension (internal, 202, 1) exited non-zero on 'SIP/vente-b761ee68'
   — Executing 203@internal:1 Dial("SIP/vente-b761ee68", "SIP/commercial|20") in new stack
Jul 23 14:18:15 WARNING4793: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
 == Everyone is busy/congested at this time (1:0/0/1)
   — Executing 203@internal:2 VoiceMail("SIP/vente-b761ee68", "3000@default") in new stack
   — <SIP/vente-b761ee68> Playing 'vm-intro' (language 'fr')
 == Spawn extension (internal, 203, 2) exited non-zero on 'SIP/vente-b761ee68'
   — Executing 101@internal:1 Dial("SIP/vente-b761ee68", "IAX2/server2:server2@server1/1") in new stack
Jul 23 14:18:21 WARNING4794: chan_iax2.c:8815 iax2_request: Unable to create translator path for unknown to ulaw on IAX2/server1-16386
   — Hungup 'IAX2/server1-16386'
Jul 23 14:18:21 WARNING4794: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'IAX2' (cause 0 - Unknown)
 == Everyone is busy/congested at this time (1:0/0/1)
   — Executing 101@internal:2 Congestion("SIP/vente-b761ee68", "") in new stack
 == Spawn extension (internal, 101, 2) exited non-zero on 'SIP/vente-b761ee68'
   — Executing 103@internal:1 Dial("SIP/vente-b761ee68", "IAX2/server2:server2@server1/3") in new stack
Jul 23 14:18:25 WARNING4796: chan_iax2.c:8815 iax2_request: Unable to create translator path for unknown to ulaw on IAX2/server1-16387
   — Hungup 'IAX2/server1-16387'
Jul 23 14:18:25 WARNING4796: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'IAX2' (cause 0 - Unknown)
 == Everyone is busy/congested at this time (1:0/0/1)
   — Executing 103@internal:2 Congestion("SIP/vente-b761ee68", "") in new stack
 == Spawn extension (internal, 103, 2) exited non-zero on 'SIP/vente-b761ee68'
   — Executing 104@internal:1 Dial("SIP/vente-b761ee68", "IAX2/server2:server2@server1/4") in new stack
Jul 23 14:18:30 WARNING4797: chan_iax2.c:8815 iax2_request: Unable to create translator path for unknown to ulaw on IAX2/server1-16388
   — Hungup 'IAX2/server1-16388'
Jul 23 14:18:30 WARNING4797: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'IAX2' (cause 0 - Unknown)
 == Everyone is busy/congested at this time (1:0/0/1)
   — Executing 104@internal:2 Congestion("SIP/vente-b761ee68", "") in new stack
 == Spawn extension (internal, 104, 2) exited non-zero on 'SIP/vente-b761ee68'
Jul 23 14:26:17 NOTICE4300: chan_iax2.c:8645 __iax2_poke_noanswer: Peer 'server1' is now UNREACHABLE! Time: 1
Jul 23 14:33:08 WARNING4305: chan_zap.c:6685 handle_init_event: Detected alarm on channel 4: No Alarm

please help me!!!



222

333What about the other projects?

by anthm, Monday 21 of July, 2008 [21:48:37 UTC]
Please add links to the menu on the left for the other projects, Asterisk is not the only Open Source VoIP app.
FreeSWITCH http://www.freeswitch.org
sipX http://sipx-wiki.calivia.com/index.php/SipX#sipX_-_The_SIP_PBX_for_Linux
CallWeaver http://www.callweaver.org
YATE http://yate.null.ro/pmwiki/
Bayonne http://www.gnu.org/software/bayonne/
OpenSER http://www.openser.org/
222

333how to configure the sflphone account?

by zwh, Friday 18 of July, 2008 [07:29:47 UTC]
i have been install a sflphone client,but i cannot use it ,how i configure the sflphone account?and how i use
the sflphone to call somebody esle?please tell me the way of using it.thanks very much!
222

333Help for Call Recording

by seamaster, Wednesday 16 of July, 2008 [14:23:54 UTC]
Hi,

our customer need to record some non-crypto (rtp) and crypto (srtp) phone call in a VoIP asterisk network; Is there a solution for this issue?

Thanks
222

333Help with Polycom 430 SIP and Switchvox

by marshalr, Thursday 10 of July, 2008 [21:36:45 UTC]
New to game. Need guidance on how to reconfig Polycom and set up as different extension in Switchvox.