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[trixbox - The Open Platform For Business Telephony] No audio on some inbound calls via queues with static agents.. Help!: Inbound calls come via an IAX trunk, that itself works and the server is external so no NAT, etc to worry about. Most inbound trunk calls work fine, some though that enter the queue seem to get the ivr menu, go into the queue, get hold music then when the agent answers the call, the caller gets blank/no audio at all, and the agent hears background noise from the caller but cannot conduct the phone call, they have to hangup at that point.

[My Blog] Blog: 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]).

[Voxilla VoIP Forum] AsteriskNow w/Linksys SPA3102 trunk registration issue: voicemail.conf - 1234=4242,Example Mailbox,root@localhost 6000=6000,Cordless,user.com,user-pager.com,tz=pacific. One thing I wanted to add is if I execute sip show registry the output is: 192.168.3.7:5060 asterisk 120 request .

[pkgsrc.se] pkgsrc.se | The NetBSD package collection: See the sample voicemail.conf for more details * The voicemail externnotify script now accepts an additional (last) parameter containing the number of urgent messages in the INBOX. * The Jack application now has .

[trixbox - The Open Platform For Business Telephony] Remote Extension Showing as Unreachable: Connected to Asterisk 1.4.22-2 RPM by vc-rpms@voipconsulting.nl currently runnin g on trixbox1 (pid = 2618)

[asteriskneeds.com] Vicidial On Trixbox: If you use the vicidial.sql from the above link, you will not need to load anything into your database at all.  If you use the asterisk flat config files above, you will not need to add any of Matt Florell’s additions to your Trixbox dialplan - they are all included in the configs you download here.

[trixbox - The Open Platform For Business Telephony] no Internal call | trixbox: I use trixbox 2.6.1, I have 3 extensions (1xx) register. I can do an external call call by my trunk but there no possibility to call from extension 1xx to an another extension 1xx, all extension are on the same context (from-internal)

[VOIP-info.org Wiki Changes] Asterisk Cisco CallManager Integration - voip-info.org: And configure asterisk exactly the same as above for Call Manager 3.2, except for voicemail to work add: dtmfmode=rfc2833. Into your sip.

[Comments for Nerd Vittles] Nerd Vittles » Asterisk on Steroids: The Orgasmatron Installer ...: One of the age-old limitations of Asterisk@Home and now trixbox was the inability to make a full disk backup of your PBX so that it could be restored after a catastrophic event, man-made or otherwise. Tom King solved all of that with his implementation of Mondo Rescue for PBX in a Flash systems.

[Narisa.com Community Blog List] Narisa.com -> Asterisk 1.4.22-3:   ZapHangup        <none>           Hangup Zap Channel

[computing] computing » Blog Archive » SPA3102 and FreePBX HOWTO: sip show peerssip show peersName/username Host Dyn Nat ACL Port Statuspstn/pstn 192.168.7.50 5061 OK (11 ms)100/100 192.168.7.50 D N 5060 OK (7 ms)[...]4 sip peers [Monitored: 4 online, 0 offline Unmonitored: 0 online, 0 offline]bell*CLI> sip set debug peer pstnsip set debug peer pstnSIP Debugging Enabled for IP: 192.168.7.50:5061bell*CLI>

[mimicIT] mimicIT » Blog Archive » Asterisk Configuration with debian: Edit a file: /etc/asterisk/extensions.conf. Change the value TRUNK: TRUNK=SIP/telasip-gw. exten=>_2403961450,1,Dial(SIP/201,30,r). and add the dial plan. You can delete from the context and after, everything. .... debian :~#cd /var/spool/asterisk/voicemail/default/1234/INBOX. And copy the file you want as a message. debian:~#cp -a msg000.gs /usr/share/asterisk/sounds/greet1.gsm. debian:~#asterisk -vvvvvvvvvvvvvc ...

[Comments for Nerd Vittles] Nerd Vittles » Googlified Messaging: Asterisk's New Best Friend: I have been using this service for the past 2 years by just creating a ringroup in FreePBX (the number of the ringroup is, of course, 411) and adding the 18006446411# to the extensions to be dialed. As we have the capabilities of making free calls to tool-free numbers in the US by using a number of different ways, then the call is always free and it has prevented me from paying the $0.75 that you get charged if you dialed to 411 directory assistance of any of the Ma.

[undertoe's blog] Experience the Love of Ubuntu 8.04 & FreePBX 2.5 | chemlab blog: check the manual that corresponds to your MySQL server version for the right syntax to use near ‘”˜0000-00-00 00:00:00”², `clid` varchar(80) NOT NULL default ”, `src` varcha’ at line 1

[VoIP Wonders] VoIP Wonders » Blog Archive » Configuring 3CX phone system with ...: You’ll notice that the wizard auto fills in the SIP proxy addresses, these may be different for your area. Since this option is dimmed out, you need to leave the default addresses, click Next, then click Back, now you’ll notice they are not dimmed anymore!

Reflected tags on Technorati: Blog, Voicemail Conf, VoxxMail