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Kinda confused by this... I have a Cisco 7960 configured with a couple SIP extensions configured on the phone. Just trying to dial one extension from the other on the same phone, but when I do, I get: -- Remote UNIX connection -- Executing Dial("SIP/2001-ffd4", "SIP/2002") in new stack -- Called 2002 -- Got SIP response 486 "Busy here" back

I've read about Fax to Email, but is there such a beast as email to fax? If not, what do people use to take care of outbound faxing?

I'm looking to install Asterisk on an EM64T Dell 1850. PERC raid 1, 1GB ram, single 3Ghz Xeon. Any red flags or anything I should know? Should I bother installing a 64 bit OS? (gentoo-amd64)? Does asterisk work in 64 bit mode? Should I turn hyper threading off? Etc?

Hello, I'm trying to set up a conference room. When I dial it's extension, I get an audible error saying "Not a valid conference room, please try again" followed by a disconnect. I've got debug sip peer 1001 (my X-Lite client) and I see this in the logs: (I'm pretty sure it has something to do with ztdummy, but I dunno... I have the port installed, but I

Hi folks, I've been a trixbox user for a few years now but I'm thinking about jumping ship. Trixbox is great, but it's missing two features out of the box which are really important to me: outbound faxing (hylafax) and imap voicemail. I see no indication that they will be included anytime in the near future, so I have a choice to make - I've looked around at Elastix,

...# meetme.conf [rooms] conf => 97531,24680 # extensions.conf [conf] exten => 1,1,Answer exten => 1,2,Wait(1) exten => 1,3,Authenticate(5447847) exten => 1,4,MeetMe(97531,Mas,24680) exten => 1,5,Playback(vm-goodbye) exten => 1,6,Hangup() exten => 2,1,MeetMe(97531,Ms,24680) [jlixfeld@trek://~ ]$ kldstat Id Refs Address Size Name 1 5 0xc0400000 5e16d8 kernel 2 4 0xc231e000 2f000 zaptel.ko 3 1 0xc234f000 6000 wcfxo.ko 4 1 0xc2355000 a000 wcfxs.ko 5 1 0xc235f000 2000 ztdummy.ko [jlixfeld@trek://~ ]$

Ok, I have a 7960 that's plugged into my laptop. my home network is wireless so I don't have a switch anywhere to plug the phone into directly. I'm running siproxd on my OS X laptop and I can make outbound calls from the 7960 fine (I guess I don't have the phone configured to register inbound calls via SIP), but the phone isn't registering to the asterisk box via siproxd

Anyone managed to get this to work? What's the recipe?

This link (http://blogs.digium.com/2008/10/13/asterisknow-15-beta-available-more-coming-soon/ ) seems to indicate that in order to upgrade AsteriskNOW v1.5 from Asterisk 1.4 to 1.6, it's as easy as installing an upgrade package. Does anyone know where to find that upgrade package? If it doesn't yet exist, what is the process for upgrading?

.... bristuff: music on hold but no dialoptions tT defined. (Thomas Winter) 24. Re: [France CID] Does Zaptel support ETSI FSK? (Vincent) 25. Re: G729a codecs + Asterisk 1.4.11 (bilal ghayyad) 26. Re: [France CID] Does Zaptel support ETSI FSK? (Ira) 27. Voicemail playback on iPhone (Jason Lixfeld) 28. NAT traversal packet loss measurement (Yitzhak Bar Geva) 29. Re: Voicemail playback on iPhone (Ron Stephan) 30. Re: NAT traversal packet loss measurement (Matt Riddell) ---------------------------------------------------------------------- Message: 1 Date: Mon, 22 Oct 2007 13:25:20 -0...

An EXT3 filesystem technically isn't supposed to require an fsck, correct? Well, as per my last email re: EXT3 Crash?! I ran an fsck on a couple of my LVMs. It said they were clean, but then I decided to be 100% sure and force an fsck. Error after Error after Error after Error. I wound up losing almost half the data on my LVM due to all the errors. What's the real deal? Am I doing

I realize this may be less of an Asterisk question and more of a... well... everything but asterisk, but still relating to asterisk question. I was looking for a Click to Dial/Web Dial solution and I found AsteriDex. I'm looking for something I can use on the road where I can hit an internal Click to Dial/Web Dial page from my cell, tap on a number and have it bridge a call between

I'm having some issues getting app_voicemail_imapstorage to talk to my IMAP server. From imapstorage.txt, I've got the voicemail.conf configured properly, but if I leave a voicemail for extension 9999, I see no indication that the module is trying to reach the IMAP server. What am I missing? # voicemail.conf [general] imapserver=172.16.17.2 [default] 9999 =>

...# meetme.conf [rooms] conf => 97531,24680 # extensions.conf [conf] exten => 1,1,Answer exten => 1,2,Wait(1) exten => 1,3,Authenticate(5447847) exten => 1,4,MeetMe(97531,Mas,24680) exten => 1,5,Playback(vm-goodbye) exten => 1,6,Hangup() exten => 2,1,MeetMe(97531,Ms,24680) [jlixfeld@trek://~ ]$ kldstat Id Refs Address Size Name 1 5 0xc0400000 5e16d8 kernel 2 4 0xc231e000 2f000 zaptel.ko 3 1 0xc234f000 6000 wcfxo.ko 4 1 0xc2355000 a000 wcfxs.ko 5 1 0xc235f000 2000 ztdummy.ko [jlixfeld@trek://~ ]$

I've recently switched my * server from FreeBSD to Gentoo using the same configs from FreeBSD on my Linux machine, except the new Linux machine is running 1.0.3 where the old machine was running 1.0.2. Whenever I try to dial into one of my DIDs, I get this in the debugs and the call gets dropped. Jan 18 13:35:15 WARNING[5735]: channel.c:270 ast_best_codec: Don't know any of 0xf800

I've got a simple MeetMe conference configured using Asterisk 1.0.3 on Gentoo. I'm using zaprtc for timing from the bri-stuff package. extensions.conf exten => 37455,1,NoOp(Drill Squad Conference) exten => 37455,2,Monitor(wav,drillsquad-37455,mb) exten => 37455,3,MeetMe(37455,pMs) Now, when I enter the conference as the first call, the MusicOnHold plays, but it's blasting

I'm using * 1.0.3 on Gentoo 2004.3, zaprtc from bri-stuff for timing. When I put a caller on hold, the volume of the hold music in the callers ear is extremely loud. I'm using the default entry from the musiconhold.conf: default => quietmp3:/var/lib/asterisk/mohmp3 Volumes with a called or calling party are fine, it's just the hold music volume that seems to be way off

I have a 7960 desk phone and I'm running x-lite on my laptop. They are both behind a NAT box so they would appear to * as being from the same IP. I'm trying to make them ring at the same time but appear to everyone as one extension. Is it possible to have them both register to * as the same extension or should I just configure them both as different extensions and have my dial plan

I'd like to provide the ability for a friend to conduct interviews using an asterisk conference and then email them to him when done. Kinda like a voicemail. There doesn't seem to be one single hook to be able to do this so I'm wondering what other people have used to jam this together and if they could possibly provide some config examples. Thanks in advance!

Was using this monitor line to get soxmix to mix test-in.wav and test- out.wav into test.wav. exten => 1200,1,Monitor(wav|/tmp/test|m) When I start the conference, the * console shows this: monitor executing ( nice -n 19 soxmix "//tmp/test-in.wav" "//tmp/test- out.wav" "//tmp/test.wav" && rm -f "//tmp/test-"* ) & /tmp shows test-in.wav,