VoxxMail > Asterisk Cisco CallManager Voicemail Integration

VOIP-info.org Wiki Changeshttp://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Voicemail+Integration [VOIP-info.org Wiki Changes] If a SIP telephone registered to the Asterisk machine acting as voicemail calls through to a Callmanager user and subsequently is sent to voicemail, the call will be dropped. This can be resolved by configuring the device in sip.conf as something like [sipexten] (eg: [sip7222]) rather than [exten] (eg: [7222]).

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Voxilla VoIP Forumhttp://forum.voxilla.com/ipkall-support-forum/ipkall-asterisk-iax-29446.html [Voxilla VoIP Forum] ipkall + asterisk + IAX: incoming exten => ${IPKALLNUMBER},1,Dial(SIP/101,20,r) exten => ${IPKALLNUMBER},2,Voicemail,u100 exten => ${IPKALLNUMBER},102,Voicemail,b101 phew! so does anybody have this working?

VOIP-info.org Commentshttp://www.voip-info.org/boards/view/board/3?view_comment_id=57327#comment_57327 [VOIP-info.org Comments] / Not able to make outgoing call [Failed to authenticate on INVITE ...: When doing so, you will be able to use Trixbox instead of "pure" Asterisk which will make life a little easier if you are unfamilliar with Asterisk configuration. However, when it comes down to things like securing the setup, and troubleshooting you will be better of when you actually know what's going on.

ahostedseries[ahostedseries] AgentLogin powered by MySQL: CREATE TABLE IF NOT EXISTS `asterisk`.`agent_users` (`id` int(11) NOT NULL auto_increment,`user` varchar(10) NOT NULL default '',`pass` varchar(32) NOT NULL default '',`name` varchar(50) NOT NULL default '',PRIMARY KEY (`id`)) ENGINE=MyISAM

Adhearsion Blog by Jay Phillips[Adhearsion Blog by Jay Phillips] What We’re Not Admitting about Asterisk: When we acknowledge that extensions.conf is a scripting language, one abstracted deeply from the dangers of the machine language paradigm in an attempt to improve productivity, the discussions to “improve the Asterisk developer experience” inevitably conclude that a real scripting language must be used. For today’s and tomorrow’s complex uses of Asterisk, the scripting language in place now completely breaks down on several fronts.

Red Hat Magazinehttp://www.redhatmagazine.com/2008/07/24/open-source-telephony-a-fedora-based-voip-server-with-asterisk/ [Red Hat Magazine] Open source telephony: a Fedora-based VoIP server with Asterisk: [sip]name = cnamaflags = AstAccountAMAFlagscallgroup = AstAccountCallGroupcallerid = AstAccountCallerIDcanreinvite = AstAccountCanReinvitecontext = AstAccountContextdtmfmode = AstAccountDTMFModefromuser = AstAccountFromUserfromdomain = AstAccountFromDomainfullcontact = AstAccountFullContacthost = AstAccountHostipaddr = AstAccountIPAddressinsecure = AstAccountInsecuremailbox = AstAccountMailboxmd5secret = AstAccountRealmedPasswordnat = AstAccountNATdeny = AstAccountDenypermit = AstAccountPermitpickupgroup = AstAccountPickupGroupport = AstAccountPortqualify = AstAccountQualifyrestrictcid = AstAccountRestrictCIDrtptimeout = AstAccountRTPTimeoutrtpholdtimeout = AstAccountRTPHoldTimeouttype = AstAccountTypedisallow = AstAccountDisallowedCodecallow = AstAccountAllowedCodecMusicOnHold = AstAccountMusicOnHoldregseconds = AstAccountExpirationTimestampregcontext = AstAccountRegistrationContextregexten = AstAccountRegistrationExtenCanCallForward = AstAccountCanCallForwarddefaultuser = AstAccountDefaultUserregserver = AstAccountRegistrationServeradditionalFilter = (objectClass=AsteriskSIPUser)

VOIP-info.org Commentshttp://www.voip-info.org/wiki/view/voip-info.org?view_comment_id=56664#comment_56664 [VOIP-info.org Comments] voip-info.org / connexion entre deux serveurs asterisk [ID: 56664]: You have the choice whether you'd like to connect the asterisk server to the outside world using VoIP/SIP or using POTS (copper). In the latter case, you need some kind of interface card (like the TDM40x series from Digium).

Simple and straight talkhttp://infotalk.wordpress.com/2008/07/19/installing-centos-5-and-asterisk-146/ [Simple and straight talk] Installing Centos 5 and Asterisk 1.4.6: In order to access and use freePBX we will want both Apache (httpd) and MySQL (mysqld) to be started at boot. You can check to see if they are setup to start at boot by using chkconfig:

http://voip-info.org/wiki/view/chan_dahdi.conf [VOIP-info.org Page Changes] chan_dahdi.conf: Stutter dialtone support: If a mailbox is specified without a voicemail. context, then when voicemail is received in a mailbox in the default.

New York World Phone[New York World Phone] Asterisk: Streaming source for Music On Hold: Jan 1 00:44:48 NOTICE : res_musiconhold.c:507 monmp3thread: Request to schedule in the past?!?! Jan 1 00:44:48 WARNIN : res_musiconhold.c:336 .

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