similar to: Voicemail - remove option to save in different folders

Hi, I have Asterisk running with the VoiceMail. Using the latest CVS. I have my extensions.conf setup so that if a SIP caller dials *99 the VoicemailMain() as follows: exten => *99,1,Wait(1) exten => *99,2,VoicemailMain() A couple days ago I installed the MySQL/Voicemail support described at http://www.voip-info.org/wiki-Asterisk+voicemail+database Now for some reason

Hi, At first I would like to express how much I like Asterisk. Amazing product. I compiled Asterisk with mySQL support for CDR and Voicemail. Everything seems to be fine, I can see that Asterisk connects to mysql and logs CDRs. I can also see that the VOicemail app is also logged in, however I can not access any mailboxes. Similar messages to others, app_voicemail.c:3011 vm_execmain: Couldn't

Hello steve. Here is a patch I wrote for app_voicemail.c which does exactly as you describe. When the outgoing message is playing, if the listener hits the "*" key, they're prompted for a mailbox and password, whereupon they can check their voicemail as if they were using the internal phone. I found no other way of doing this. If you patch your app_voicemail.c, I have V1.44 from

Can not check voice mail-box externally. I'm trying to log-in externally (from PSTN line) to check my "voice-mail" so I created context to authenticate log-in ... exten => 7,4,Authenticate(01894546) exten => 7,5,DISA(4789|disa-access) Authentication works OK, I get inside dial none enter my mailbox extension but it doesn't accept my mailbox password even though it is the

Voicemail.conf in MySQL is not functioning where I get the following error from Asterisk messages log file: CLI debug output is as follows: Executing VoiceMailMain2("SIP/2205-3df0", "") in new stack -- Playing 'vm-login' -- Playing 'vm-password' -- Incorrect password '1234' for user '0' (context = <any>) -- Playing

If you have many old voicemail messages, to get to the most recent one, you have to keep hitting "6" until you reach the last one. It would be better if you could hit "4" from the first message to get to the last message and/or have a digit that takes you the first and last messages respectively. Anyone have any patches for this?

Hi All, I'm trying to test asterisk voicemail on recording my own unavailable message, busy message or temporary message. I was looking at the console and saw this message: app_voicemail store_file Memory map failed Then i looked at /var/spool/asterisk/.... there were no recorded greetings. what does the error mean? TIA Regards Ron

Hi, I think there is an essential option of the Voicemail application that is missing. I would like to suggest the implementation of a function to give the user the ability to either allow or disallow the recording of messages. If the ability to record a message is disabled, options u, s, and b must not be considered in order to avoid the playback of messages such as "Please leave your

Is there a way to specify multiple email addresses in voicemail.conf for a specific user? I seem to remember that it was possible, but can't remember the character to separate the email addresses. (I tried '&', but that didn't work...) later, PaulH

Hi All, I'm trying to use a php script to send voicemail e-mail so i can send custom e-mail message based on what mailbox. on my voicemail.conf i have mailcmd=/var/www/voicemail.php but when i tried to call an extension and goe to voicemail i'm not receiving the e-mail. but when i execute "php /var/www/voicemail.php" on the shell, i can receive the e-mail. how would i

Hi, I'm a Polycom user who have just been trying out the Aastra phones. I`ve run into a bit of an issue. I cannot configure the voicemail dialout correctly on one of the two phones I am using, even if the documentation seems to state that it should work in the same manner on both phones. I have both a 9143i and a 57i CT. I`ve set both up with provisioning from config files. Both

Searched the archives thoroughly... Can't find this specific problem... Simple setup with Asterisk on RedHat. No voice cards in the box, 2 SNOM 200 phones... Phones seem to work well, can leave VM, Message Waiting Indicator lights up but when I try to retrieve messages the call terminates and the following happens: -- Executing VoiceMailMain("SIP/520-a25e",

Hi All, I bought a DID number from VOxbone...this number could be dialed from any PSTN line and could be forwarded to any SIP server like asterisk server...Now I need to forward this number to my asterisk server so when a customer dial this number from his GSM or Land line PSTN number the call will be forwarde to my asterisk server and I need to play a wav file for example.. Can you please give me

Hi, I've looked high and low for any changes that streaming MoH needs on Asterisk 1.4 (.21), followed NerdVittle's article about it (http://nerdvittles.com/index.php?p=92) yet nothing worked. After creating dir stream/ and touch stream.mp3, here's my musiconhold.conf [stream] mode=mp3 directory=/var/lib/asterisk/mohmp3/stream stream =>

Dear All Can you please do me favor and let me know how can I convert *.wav files into 32 bit 44 KHz ? Please be informed that I have specific sound files in *.wav format that I converted them into *.gsm format with the aid of the following command : #sox FR00003.wav FR00003.gsm It got through but the voice quality is poor . I need to convert the original *.wav sound files (their file attribute is

Hello everyone I am trying to look for a free test call generator that will get me some stats like PDD, ASR and call quality etc on each route. As well as do test at every interval too If you know something like this please enlighten me. Sam -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi List, Does anyone have experiences to relate on the various Asterisk-based PBX appliances out there? Like the Aastra 160, Digium S844i, etc. Do the Epygi Quadro and Grandstream GXE also use Asterisk? Thanks, Femi -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi I've configured my asterisk and voicemail all works fine but I want to restrict call time to voicemail that is when user calls voicemail he can use voicemail system only for a max of 5 min that is after five minutes asterisk should disconnect the call. thanks Arun -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm running asterisk realtime, i had prob when a user does not unregister properly. I tested with SPA942 and a PAP2, when phone is registered, i call using the SPA using x-lite no problem, but when i unplugged the power, it does not unregister properly, so asterisk think SPA942 is still registered, when i call using x-lite, asterisk tries to call it.so it gets stuck at [Aug 11