search for: voicemailmain2

I would like to access VoiceMailMain2 skipping extension and password prompting if calling from a resource that has a mailbox defined. What variables can I use to retrieve the calling channel & calling extension (if it exists)? Here is what I'm trying to accomplish (of course ${CallingResourse.MailBox} is not a real way to ret...

Has anyone browsed through the source code and made a list of menu option for VoiceMailMain2? Or know of some user documentation hiding in Internet land some place? If not there well be soon. Ho hum.

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 'vm-incorrect' -- Playing 'vm-password' -- Incorrect password '...

Hi guys, I'm running Asterisk-0.5.0 and accidentally stumbled on this problem while in the VoicemailMain2 application: If you login to it, or even if you call it w/ 's<extension>' to skip the login and press an '8' near the beginning (and possibly at any point, I'm not sure), the channel seems to lockup, even if the handset is hungup, the channel remains frozen in that sta...

Hi! I've been trying to use the Voicemail (and Voicemail2) applications with an SIP Phone (X-Lite, for those who cares), when I use g.711(a/u) codec, it works perfectly with inband (it detects the whole mailbox (in my case 10007)), but not with rfc2833 (in this case, it only detects 107 as the mailbox number). With gsm codec, the inband doesn't work, I guess that's due to the

...ng with voicemail2. I'm having two minor problems that I can't seem to find discussed in the archives. 1) New message 0 in mailbox 7606. New voice mail message count seems to start with 0 for the first new message instead of 1. Any tricks to fix this? 2) When listening to messages with VoicemailMain2, the time stamp is in GMT and not corrected for the local time zone offset. But the email that voicemail2 sends has the correct time. I added "|tz=eastern" to the end of the mailbox definitions in voicemail.conf, but that did not seem to fix the problem. Any thoughts on these two probl...

...oicemail2(u1000) exten => 1000,102,Voicemail2(b1000) exten => 1000,103,Hangup should be.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000@sip) exten => 1000,102,Voicemail2(b1000@sip) exten => 1000,103,Hangup and.. exten => 8500,1,Wait,1 exten => 8500,2,VoiceMailMain2(${CALLERIDNUM}) exten => 8500,3,Hangup should be.. exten => 8500,1,Wait,1 exten => 8500,2,VoiceMailMain2(${CALLERIDNUM}@sip) exten => 8500,3,Hangup sip.conf entry.. [1000] type=friend insecure=np username=1000 secret=secret nat=no host=dynamic canreinvite=no context=sip mailbox=1000...

...und [app_voicemail2.so] => (Comedian Mail (Voicemail System)) Sep 23 16:25:43 WARNING[-1610571284]: pbx.c:2304 ast_register_application: Already have an application 'VoiceMail2' Sep 23 16:25:43 WARNING[-1610571284]: pbx.c:2304 ast_register_application: Already have an application 'VoiceMailMain2' Sep 23 16:25:43 WARNING[-1610571284]: loader.c:334 ast_load_resource: app_voicemail2.so: load_module failed, returning -1 == Unregistered application 'VoiceMail2' == Unregistered application 'VoiceMailMain2' Sep 23 16:25:43 WARNING[-1610571284]: loader.c:429 load_modules:...

I have a question, I am unclear on how to park a call. I know that you are supposed to be able to press "#" and then transfer the call to extension 700. However, * doesn't seem to be graping the dtmf. I am using dtmfmode=inband. Asterisk is in the media path as well. Thanks in advance Justin

...for getting mailboxes from DB (for Vonage) * Segmentable mailboxes, allowing you to truly multihost voicemail for multiple companies * Reads voicemail.conf only on startup and reload I'd really like people to help bang on it by replacing VoiceMail and VoiceMailMain in your config with VoiceMailMain2 and VoiceMail2 and try to make sure that it all works nicely for you. Thanks! Mark

I am having problems retrieving voicemail from outside the asterisk system. My extensions.conf is configured as follows: exten => 7900,1,VoiceMailMain2(s${CALLERIDNUM}) exten => 7900,2,hangup exten => 7902,1,VoiceMailMain2 exten => 7902,2,hangup exten => 7999,1,dial(sip/7999,20) exten => 7999,2,voicemail2(u7999@incoming-pri) exten => 7999,102,voicemail2(b7999@incoming-pri) exten => 7999,103,hangup I have the phone call 7900 t...

I am seeing strange behavior that I don't understand. Voicemail2 and voicemailmain2 work fine if I call from a sip phone directly connected to *, but if I call either of them from an analog line on the other side of a sip gateway, voicemail seems to ignore digits. If I am recording a message and press #, nothing happens except that it records the tone onto the message and I can...

...re and goes to voicemail but in my case >> it >> transfers to the main voicemail instead of the users voicemail. >> >> Here is what my SIP extensions look like in the extension.conf file >> >> exten => 3957,1,Dial(${Theresa},20,Tt) >> exten => 3957,2,VoicemailMain2(u${TheresaVM}) >> exten => 3957,3,Hangup >> exten => 3957,102,VoicemailMain2(b${TheresaVM}) >> exten => 3957,103,Hangup > > Change the above from VoicemailMain2 to Voicemail and it will work > as expected. > > The 3,Hangup isn't required... remove it. 1...

Dear all, I have a question regarding the configuration of *. I have 3 port FXS, and 2 port FXO. I have 4 users that use analog phone connected to FXS (I have 3 phones). I need to limit the user's capability (user A can call International, user B can call long distance, etc). I want to implement the password say to call , he/she needs to puch 9(for outgoing call)2-4 digits password,then

...For the dutch, it is customary that a user creates their own message which includes 'please leave your message after the tone' or similar, so the generic message is undesirable (or should be override-able). Is there something in the apps I've missed that allows this already ? - In voicemailmain2 there is no option in the menu that allows creating your own messages (in fact, option 3 is defunct). Is this in the coming, or am I missing more stuff ? Thanks! Met vriendelijke groet, Florian Overkamp ObSimRef BV (http://www.obsimref.com/)

Ok, so I figured out my problem with my pt480s. But, now I have a few more. 1. When I dial into the voicemailmain or voicemailmain2 application, the phone and * start talking adsi, but then the phone tells me "programming download canceled, services is full.", but my services list isn't full, only "Asterisk PBX" occupies slot 2, slots 1, 3 and 4 are available. Any ideas? I have tried erasing all the ser...

Hi, There is any possibility to dial a specific extension and then enter in your own mailbox (the one defined for that specific SIP phone) without asking for the exxtension number but only for the password? I want to be the same extension for all phones, not a specific one for each of them. It is possible to have a time stamp in the recorded message? I want to know when the message has been

Hi, Try as I might, I can't get hangups detected on a Zap channel with loop start lines. So, after someone leaves a voicemail and then hangs up, Asterisk doesn't know it, exits VoicemailMain2, and loops back to the corporate greeting, tying up the line even though the outside caller has hung up. Therefore, I've added the following hideous hack - er, code - to voicemail2.c. It starts right after the call to play_and_record() in leave_voicemail(). if (res != '#' &&amp...

...one have any idea why this is happening! It have checked everything but running out of options! [app_voicemail2.so] => (Comedian Mail (Voicemail System)) == Parsing '/etc/asterisk/voicemail.conf': Found == Registered application 'VoiceMail2' == Registered application 'VoiceMailMain2' [app_transfer.so] => (Transfer) == Registered application 'Transfer' [codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator) Illegal instruction thanks hallian _________________________________________________________________ The new MSN 8: advanced junk mail protect...

I've noticed something strange when saving a new message to the new messages folder. The message number gets incremented (message0000 becomes message0001), then voicemailmain2 thinks that there are no new messages to be played, but the MWI stays on, and the end user tries in vain to retrieve a message that can't be played.