similar to: Problems with Hangup Detection in VoiceMail2.

Many thanks to Martin Pycko and Mark Spencer. Mark's suggestion below was correct: "Maybe it's stuck trying to send the e-mail notification. If you take the e-mail address out of /etc/asterisk/voicemail.conf does that speed it up?" Indeed it did! The problem turned out to be a 60second delay while invoking mail, caused by a mis-configuration of my hostname and

We've been testing (aim:frziegler and aim:end1r) the Voicemail2 app for a few days now, based on a CVS build from Monday, 5/12/03-23:15. Works good! Thanks Mark! We seem to have found a bug in the MWI (Message Waiting Indicator) logic. By simply creating msg0000.txt files in both structures, e.g.: for extension 4000: voicemail1: /var/spool/asterisk/vm/4000/INBOX/msg0000.txt

asterisk 1.4.32 have zapata.conf soft link to chan_dahdi.conf to use flash operator panel < 2.0 (from extensions.conf) exten=> 304,1,ChanSpy(Zap/4|q) exten=> 304,2,hangup There is no entry ChanSpy(Zap/41) in extensions.conf On dialing 304 and Zap/41 is in use this happens: [Jul 1 18:24:47] VERBOSE[14447] logger.c: -- Executing [304 at flash:1] ChanSpy("Zap/31-1",

The voicemail and voicemail2 applications both invoke sendmail as /usr/sbin/sendmail -t. However sendmail is invoked using popen(3), the input being terminated when the pipe is closed and not when a . is entered on a new line. While it looks like the code at the moment can't generate a single line with a '.', it might be prudent to apply the following patch which tells sendmail to

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.

A few weeks ago Brad posted his patches to the mailing list: http://www.universaltime.org/~brad/vmail/ But I can't find his email address... does anyone happen to have his address. I hope he would be willing to see if Mark wanted to add those options to the CVS. I think you need to fill out a disclaimer and post it to bugs.digium.com.... bkw

Does anybody have any reason why I should *not* permenantly replace app_voicemail with app_voicemail2? If so, speak now or forever cvs update -D "8/18/2003". Mark

Hi, In testing the Budgetone we have noticed something strange with DTMF and Voicemail. When we set the Budgetone for RFC2833, and connect to voicemail, the detected DTMF digits do not correspond with what we pressed on the phone. For example user=1001, password=1001 is detected as: Incorrect password '1111000000111' for user '111000000111' (context = <any>) Any idea

Hi all: Somebody knows the mysql table structure for VoiceMail2 application? Thanks in advance, Gus

Created new message, not a response with new Subject: line -- check Plain text mode set in Outlook -- check Good day all, I just recently upgraded to 0.5.0 and subsequent CVS releases. After the upgrade, I've noticed that voicemail2 has differing opinions on what time it is, assume this is because I haven't set the TZ per user in voicemail.conf. Anyone have an example as to how to

Using CVS update from 11:00 CET today * crashes at this point. == Parsing '/var/spool/asterisk/voicemail/default/2201/INBOX/msg0000.txt': == Parsing '/var/spool/asterisk/voicemail/default/2201/INBOX/msg0000.txt': Found Sheriff*CLI> Disconnected from Asterisk server -- Dave Cotton <dcotton@linuxautrement.com>

Hi all: This is a warning that appears when * loads. WARNING[1074494336]: File app_voicemail2.c, Line 2889 (load_module): SQL init VoiceMail2 tries to bring up VM SQL service? Regards, Gus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031014/3bad51d0/attachment.htm

>> Wouldn't that break everybody's dialplans where they would have to >> replace all occurrences of Voicemail2 with Voicemail and all >> occurrences of Voicemailmain2 with Voicemailmain? > > No, we would register with both names. Is it necessary (with reasonably current cvs) to make any changes in the *.conf files to use Voicemail2, or is that happening

I'm using the stable branch. Is voicemail or voicemail2 deprecated? TKS Paul pmahler@signate.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040502/1b0ab572/attachment.htm

I figured out what was going on with the lack of/stuck on stuttered dial tone. Apparently, there are two voicemail directories being referenced: /var/spool/asterisk/voicemail/default, and /var/spool/asterisk/voicemail/local. The sip phones were using /var/spool/asterisk/voicemail/local to dump VM messages into, yet the MWI looks at /var/spool/asterisk/voicemail/default. Does anyone know why

Here are a few things I would like to see .. 1. In addition to time/date stamps, store/read the caller id info with the voicemail messages. 2. Have the ability to configure the system to ignore and delete messages left by a caller that are 3 seconds or less (maybe make this configurable) Not sure but that would cut alot of hangup calls out of your voicemail box. I can't think of much more

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

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

I have the MusicOnHold feature working great when called from ATA-186 extensions. It's pretty cool. However, when I call from a BudgeTone-100 phone, no music is heard -- instead it continues the ringing feedback and acts like the call is unanswered. At the same time, I can call from (multiple) ATA-186 extensions and hear music as long as I like. How can I debug this? As far as I can tell,

Hi, I am having a problem with (Asterisk is crashing) with a Fritz card PCI / chan_capi. Receiving Calls from PSTN works, but outbound calls make asterisk crash (Speicherzugriffsfehler/Segmentation fault). The crash occurs upon dialing with the other phone not even ringing. I hereby ask if somebody reading this list can confirm or disprove my issue. Does anbody run a recent asterisk 2.6 with