similar to: Voicemail with NFS

Crikey. I can't get this to work! Allegedly, you can press 0 while in the voicemail greeting and be dropped to the 'o' extension. For some reason, I can't get it to work. The 'docs' aren't clear about what context the o extension should be in. The voip wiki says "the context for the voicemail box that we're looking for in the dialplan for the jump to the

Paste your extensions.conf section that is relevant. -Matthew ----- Original Message ----- From: "Greg - Cirelle Enterprises" <gcirino@cirelle.com> To: <asterisk-dev@lists.digium.com> Sent: Monday, December 27, 2004 4:32 PM Subject: [Asterisk-Dev] realtime voicemail > Let me clarify my last message. > > If I put in the wrong password I get polled > again for

Anyone know if a way to have voicemail files stored as mp3's? Thanks, Doug.

Ugh. We thought we'd fixed some problems by using regexten and DUNDi. Guess not. We have a configuration with three Asterisk boxes. Phones register with a single, primary asterisk box under normal conditions. For voicemail deposit, retrieval, we trunk the calls over to our asterisk voicemail server. However, the voicemail server now has no knowledge of the location details of the phones,

I'd love to understand what the function of the peer list returned by 'sip show peers' is for, especially when Realtime is used. If I start Asterisk with realtime enabled, a 'sip show peers' yields none. As each peer (phone) registers, or a call is made to the peer, Asterisk adds them to the list returned by 'sip show peers'. Correct? Apparently Asterisk doesn't

> -----Original Message----- > From: JR Richardson [mailto:jmr.richardson@gmail.com] > Sent: Saturday, June 17, 2006 2:30 PM > To: asterisk-users@lists.digium.com; Douglas Garstang > Subject: Voicemail with NFS (working, I think) > > I'm using a stand-alone VM server and exporting the VM files ro for > MWI function only. All my registration servers mount the remote

I'vce got three Asterisk systems here that I'd like to be able to place calls between with IAX. As usual, I've spent several hours playing with it, really getting nowhere. Asterisk is so mentally draining. Each system, pbx1, pbx2, pbx3, should be able to connect to every other. Do I need separate user/peers or can I combine them into a single user=friend for each system? if I place a

Does this model give you functioning mwi? > -----Original Message----- > From: Ray Jackson [mailto:ray@jacksonz.net] > Sent: Friday, January 05, 2007 3:17 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Voicemail personalised greetings using > DB/IMAPbackend? > > > Hi all, > > I am attempting to build a horizontally

Here's an interesting question: If I transfer a call from Asterisk system to another with IAX, is there any way I can get control back on the original system? Or.. do I lose control, and the dialplan has to continue on the new system? Scenario is we transfer calls to an Asterisk system that handles ACD queues. If the ACD queue times out, we want to send the caller to voicemail on another

Hi list, I'm with a little problem on codec negotiation between a cisco827 and asterisk. My sip.conf is like that: [general] port = 5060 bindaddr = 0.0.0.0 context = default amaflags = default allow=g729 allow=gsm allow=alaw allow=ulaw ;disallow=all and cisco like that: dial-peer voice 6 voip destination-pattern 0T session protocol sipv2 session target ipv4:<asterisk-ip>

I've been noticing lately what seems to be some distortian in the standard asterisk sound files, used for voicemail. These files are stored on the local Asterisk system. When Asterisk plays them, I can hear some cracles and pops. I'd never noticed these until recently. I did a little test. This sounds fine... exten => 1000,1,Answer exten => 1000,n,Wait,1 exten =>

Hi, I am designing a asterisk load balancing model as follow. There are 3 asterisks connected to a single DB and a single server storing all the configuration file and voicemail. Round Robin DNS will distribute the request to asterisks. DNS round robin ---+ asterisk1--------------------------+ DB and file server +---asterisk2-----------------------+

Has anyone heard anything about these guys? Anyone seen anything like this? http://www.orative.com/solutions.php It's seems very cool, basically uses GPRS as a digital overlay on your mobile phone for additional functionality such as presence and IM though I'm sure they have some other functionality (voicemail access, call announce etc) coming down the pipeline. Any thoughts,

While reading about how to maximize capabilities in asterisk i have read about SER and OpenSER. The sites do not explain to newbies (maybe that's on purpose) what are the benefits of using those products tied with asterisk (or is SER an asterisk replacement??) Can someone give me an idea of what's the usage for open(ser) and asterisk? is it for scalability? should I run it in the same

Boy, am I stuck... I'm officially ready to toss Asterisk out the window. I have to admit it isn't necessarily all the fault of Asterisk either. It just seems that every option I turn to suddenly ends in failure. I don't know if it's me that's bitten of more than I can chew with this project, or maybe Asterisk just isn't mature enough yet. Nothing complicated really....

I''ve got this: file { ''/opt/sugarsync/tomcat/tomcat-home/current'': ensure => "tomcat-$config[''tomcat_version_server'']"; where $config[''tomcat_version_server''] was set with extlookup (the yaml one), by loading: --- tomcat_config: tomcat_version_server: 6.0.20-1 tomcat_version_libs: 1.0-1

Anyone know if there's a way to match a dialplan extension, execute some code, say set a variable, and then continue with the dialplan? I want to set a variable when the dialplan flows beyond a certain context. This would be a great feature. Doug.

This is pretty ugly. I''m using puppet to install Oracle, ie an exec{} wrapped around: /u01/oracle_extract/linux.x64_11gR2_database/database/runInstaller -silent -responseFile /etc/oracle_response.rsp The problem is that the damn installer backgrounds itself and returns control to the shell. I tried putting the above command in a script, followed by a wait command, but that

Here we go with puppet 0.25 certificate problems again. I had a system where puppet was running fine. I reinstalled it. Running puppet on the client causes this: "Could not request certificate: Retrieved certificate does not match private key; please remove certificate from server and regenerate it with the current key". Fine... so I run ''puppetca --clean

All, I was looking at the new feature of puppet to manage switches, and it says: "A current limitation is that it isn’t possible to have 2 switches with the same interface name" Does this mean that if one switch has an interface called FastEthernet 0/1, that puppet can''t manage a second switch with an interface called FastEthernet 0/1? If so, that probably means you can