similar to: VoiceMailMain skipping extension and password prompting

Hello, I recently started playing 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

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

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

Asterisk Users: I've been working hard on app_voicemail2 which is an enhanced scalability version of app_voicemail. Specifically, its features are: * Highly improved internal architecture (maybe someone else can actually code on it) * Foot print for getting mailboxes from DB (for Vonage) * Segmentable mailboxes, allowing you to truly multihost voicemail for multiple companies

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

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

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 Lee, You need to specify the VM context that you are using.. so using your examples.. extensions.conf entry.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(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

I'm using a Cisco 7960 with asterisk and any recording on the machine, be it voicemail prompts, time of day, echo test message, etc, is cut off for the first 1/4 to 1/2 second. I've tried setting the phone to gsm but it still happens.

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.

Hi, compiled 1.0 on macosx latest (10.3.5). compiled fine. when running, complains about voicemail2 module. Any hints? Marc. == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk CVS-HEAD-09/23/04-09:20:48, Copyright (C) 1999-2004 Digium. Written by Mark Spencer <markster@digium.com>

I have a situation where I need to know which Zap channel an incoming call is on, so that the call can be answered appropriately when a SIP phone displays the channel. These Zap calls are coming in over PSTN and don't have caller ID. As far as I can make out my SIP phones (WuChuan HOP-1002) display the user part from the SIP "From:" header as the second line on the display. If the

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't specify

Hi, I've been playing with Voicemail and Voicemail2 a bit for my users, and there are a few things I'm wondering about: - We can specify parameters to the mailbox (s, b or u) to select which prompts to play. However, if we specify 'b' or 'u' it plays that (customisable) message, but it also plays the voicemail instructions. For the dutch, it is customary that a user

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

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?

Hi, Every time I dial into my asterisk box i hear nothing but asterisk jittering. The following is an example of what I get on the asterisk CLI Thanks *CLI> DEBUG[81926]: File chan_sip.c, Line 3826 (check_user): Setting NAT on RTP to 0 DEBUG[81926]: File chan_sip.c, Line 4807 (handle_request): Check for res DEBUG[81926]: File chan_sip.c, Line 952 (find_user): Call from user

ser forwards a sip message with extension 99999996 to asterisk which plays my 'userisoffline' message and hangs up and should stop here but instead asterisk continues to process the match everything extension ._ and dials out which is not what I want... if I change the starting priority of the Dial app to a higher level than 3 asterisk stops after the hangup but then doesn't accept

Hi, First off, a big thanks to Digium (Mark, John, and Martin) for helping sort out a BellSouth config issue on our PRI. T100P working like a champ! Now it's back to tweaking the configuration on our SIP phones (7960s). The message_uri parameter in the phone's configuration file is working great. Dials comedian mail directly. Is there a way to let voicemail2 know what the incoming

Hey gang, There must be any easy solution for this but my mind is frazzled on compiling 2.4 with RTC as module. Bleh. Currently extension 9000 is our VoicemailMain(@company) line. Some employee's are complaining that the old system was better because you didn't have to enter your mailbox number and that instead the old system took you right to it. I figured there was something similar