similar to: catch when no voicemail configured

Hi all, What's the best method to send automon call recordings (*1) to the voicemail box of the Asterisk user? Do you have to trap hangups, etc, or is there some global variable that can be set? Thanks! S.

> I'm using asterisk 1.2.9.1 and mysql 3.23, asterisk add-ons 1.2.3. > All seems to work normally with realtime voicemail, reads vmbox > parameters from the db fine. When I try to change the password, > asterisk operates normally, "enter new password" ok, "re-enter new > password" ok, "password has been changed" > > There are no entries in

I tried your extension definition as suggested: exten => *98,1,Verbose(0,${CHANNEL(peername)} calling voicemail) same => n,VoicemailMain(${CHANNEL(peername)}@VoiceMail,s) same => n,Hangup But there was no change in the prompts asked, ie. the voice first asked for 'mailbox', and then 'password' as before. The prompts are not removed. Please clarify what you mean by the

Hi, I would like to enable mysql handling of voicemail boxes ... following that tutorial http://www.voip-info.org/wiki-Asterisk+voicemail+database so I modified the makefile of /apps directory to include USE_MYSQL_VM_INTERFACE=1 and copied mysql-vm-routines.h in the /apps dir, set up the db and some boxes in the table, also edited the voicemail.conf file. Everything compiles just fine, then

I've setup the voicemail that auths against the mysql db. Now, everything works ok, except voicemail() calls fail with Sep 26 18:09:34 WARNING[157070336]: app_voicemail.c:1517 leave_voicemail: No entry in voicemail config file for '' all my users are in 'sip' voicemail context, but adding context to it: voicemail(@sip) doesn't help.. while if I put a vmbox # to it, it

I replied to a post of mine a few days ago asking of anyone uses mailboxexists(). I haven't received any replies. Perhaps few use it or perhaps the reply was overlooked. I thought I'd post the question one last time before giving up on it for now... Thanks! -Michael

While setting my first couple IP phones, I set their voicemail buttons to an extension that runs VoicemailMain. exten => 8500,1,Wait(1) ; voicemail exten => 8500,2,VoicemailMain ; exten => 8500,3,Hangup ; I would like to be able to pass the mailbox number allowing each phone to go in directly but I'd rather tno have

asterisk-users-bounces@lists.digium.com wrote: > Vadim Berezniker wrote: > >> That's not a solution, but just a workaround. >> 1.2.1 has a bug where it always uses an empty context when searching >> for a mailbox when using realtime config. >> At around line 546 of apps/app_voicemail.c there is a line that says >> var =

Hey all I have been beating on this all weekend long. Any feed back would be appreciated. We stood up a 11.6 system. We tested everything we could think of. We moved over to it and all seemed to be working good than a customer told us that they were not hearing our vociemail greetings. When we call into the system and it drops to voicemail we just get a beep no greeting played. We checked

I have SER running with Asterisk, both on seperate servers. If I call another SIP number from my SIP phone SER looks up the phone number to see if it's online. If it's not online it forwards the call to Asterisk. How do I configure the extensions.conf file so that calls being forwarded to Asterisk destined for VoiceMail do not conflict with normal outbound calls destined for the PSTN?

Hi All, I'm using asterisk 1.2.9.1 and mysql 3.23, asterisk add-ons 1.2.3. All seems to work normally with realtime voicemail, reads vmbox parameters from the db fine. When I try to change the password, asterisk operates normally, "enter new password" ok, "re-enter new password" ok, "password has been changed" There are no entries in the mysql.log setting the

Guys. Since ARA is not available in stable 1.0.9 I was wondering how to use externpass in voicemail.conf to update mysql based voicemail password. My question is, can externpass send parameters to the called file? I couldnt find any examples on the wiki so, does anybody have any tips on this? Thx !

If anyone could help me with this, I'd appreciate it! I've got an Asterisk deployment where I'd like to use an existing external Octel voicemail system. I've been trying to define an extension that if the call isn't answered in a few rings, to dial our external voicemail number. That voicemail system works by seeing the CALLED number and routing the call to the

Hi folks, i'm trying to configure my extensions.conf as small as posible and for that reason i'm using macros. The problem is that maybe I have a misunderstood the concept for the directive "mailbox" in sip.conf. Under my knowledge configuring the mailbox directive to the mailbox I want would be enought to leave an retreive messages in that voicemail box. Of course it seems to

There is a patch on Mantis (http://bugs.digium.com/view.php?id=4371) Which includes several features. 1. Support for central voicemail server(s) with remote server notification via IAX In other words, this patch allows you to configure an Asterisk server as a central voicemail server and to send out voicemail notification to remote Asterisk servers who can then pass the notification on to

I'm having a problem getting my queue to function as it should. After 20 seconds or so, it should prompt the user with a message "thanks for holding..... press # to leave a message or stay on the line to continue holding". I set up the "context" in the queues.conf file, so if a user presses a digit, they should be able to leave. But I get a SIP BUSY message. Here are my

Hi, In voicemail.conf.sample, you can read this: format=wav49|gsm|wav ; WARNING: ; If you change the list of formats that you record voicemail in ; when you have mailboxes that contain messages, you _MUST_ absolutely ; manually go through those mailboxes and convert/delete/add the ; the message files so that they appear to have been stored using ; your new format list. If you don't do this,

Hi everybody In advance sorry for my bad english and if my problem was already exposed (I didn't find any tips in the mailing list archive. Bad luck) I have some questions about asterisk 1.6 release : 1) how can I do a n+101 priority jumping if a SIP canal is busy ? I read that the general parameter "priorityjumping" is depreciated in the 1.6 release and I already try the

Greetings, Running CVS HEAD about 3 weeks old, I have been beating my head trying to get this to work properly.. Or at least figure out what's going on. Maybe I have done things wrong... I have created a 'catch all' extension at the end of our last context where all phones & voicemail extension exist. This catch all is included in all and works quite nicely except when voicemail

Hi. Has anyone had any success in implementing a callback or ringback function in Asterisk? I've had a look at the callback-voicemail example on voip-info.org http://www.voip-info.org/wiki/view/Asterisk+tips+callback However it won't quite work for me. I need it for local SIP users which most of them don't have voicemail. If one SIP user calls another SIP user and the second user is