similar to: Getting to the last "old" voicemail message

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

Hi folks, we use meetme application with pin so when a customer joins he's prompted for his name. Then the voice say:"press one to accept the recording..." My question is, is it possible to cut off that request to"press one"? Thanks to all -- .:FaberK:.

I'm sure asterisk would make a great stand alone voice mail server. Basically I want to get rid of our voice mail system and replace it with *, but the problem is we use a cisco cluster with skinny clients. So I was thinking the way to contact a * server, would be through our 3640. But so far any attempt has failed. I am wondering if anyone has done something similar. Just want to verify the

Hi ALL, When i was accessing the voice message it suddenly goes dead and after that i couldn't able to retrieve the voicemessage again from my mailbox . This happens once in a while for any configured mailboxes I am using the following system configuration. asterisk 1.4.22.1 odbc storage of voicemail messages centos 5.2 64bit unixODBC-2.2.11-7.1 mysql-connector-odbc-3.51.12-2.2

In Asterisk 1.2.13 in app/app_voicemail.c, line 4700 ext_pass_cmd is checked to decide whether to use vm_change_password or vm_change_password_shell to change a user's password for his voicemail account. I wonder, what is the difference between vm_change_password and vm_change_password_shell - what is that shell? The only reference I found on the Internet was the following bug report:

Is there a way for asterisk's voicemail to send an email (including voicemail attachment) to multiple email addresses? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110408/8908db5f/attachment.htm>

Thanks for your work on xen packages, I have some question about 4.5 packages: I saw a debian/rules that disable blktap2 but debian/patches/tools-blktap2-prefix.diff is still present and used in series, if blktap2 is now disable I think is nomore needed. I saw also many other patches about prefix still present but for xen 4.5 time ago I saw a big series of patches about honor tools prefix,

On Fri, 2014-12-05 at 00:33 -0500, Stephen Oberholtzer wrote: > (I hope this is the right address to send this message to.) > > I've spent the last few days getting Xen 4.5.0 RC3 to form up Debian > packages, adapting the files from xen-4.4.1-4. > Would you guys like my patches? > It was a non-trivial exercise, especially because of commit 2c96ecb1, > which invalidated a

I've been banging my head on a brick wall for about an hour now trying to understand why the following doesn't work (which is even provided as an example in the distribution!). The goal is to create a voicemail-only extension not associated with a phone. I'd rather not have an extension dedicated to VoicemailMain(), so I would like the user to be able to hit '*' during

I am currently having issues when trying to change my password in Voicemail. I am not utilizing realtime because I believe there is a problem with MWI being sent to phones and realtime databases (this may have changed since I last used Asterisk). Whenever I try to change the password for the account, the asterisk program gives a Sig fault (signal 11). I do see the voicemail.conf.new file

On Wed, 2014-12-10 at 12:47 +0100, Fabio Fantoni wrote: > Thanks for your work on xen packages, I have some question about 4.5 > packages: > I saw a debian/rules that disable blktap2 but > debian/patches/tools-blktap2-prefix.diff is still present and used in > series, if blktap2 is now disable I think is nomore needed. Yes, this can probably go. > I saw also many other patches

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

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'd just like to verify what the correct operation of the timeout parameter is for the dial application. I'm not sure if I've encountered a bug or a configuration issue. When a sip phone is not responding to invites on an outbound call, the dial application still waits the duration of timeout before continuing with dialplan execution. I was under the impression that app_dial

Hi All- I am running into a small problem trying to implement voicemail Directory(). I'm sure it is a simple thing, but I can't figure out where the problem lies. I can get into the directory without a problem and can look up users by their last names, however I hit a snag when asterisk says "if this is the person you are looking for press 1 now". When I hit 1, the attendant

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

Hi all, i have asterisk 1.2.17 with sip tcp support and i am trying to connect asterisk with HiPath 4000 V.3.0 using SIP. I can see the registration from the HG3540. But when i try to place a call from Asterisk to HiPath, the call fails with SIP/2.0 603 Declined. The strange thing is that the first INVITE uses tcp and the response is a 100 TRYING, the next 7 INVITE are using udp and the respose

Hi All, Anyone know if theres a way to share a Cisco 7940 between hot-desk users? My phones get their setup via SIP .cnf files, that load at boot via tftp, so I'm assuming the configs a failry static. However if I want a phone to be hot-desked, I could have different users sitting there. Is there any concept of "logging on" in these environments? Cheers, Adrian

(I hope this is the right address to send this message to.) I've spent the last few days getting Xen 4.5.0 RC3 to form up Debian packages, adapting the files from xen-4.4.1-4. Would you guys like my patches? It was a non-trivial exercise, especially because of commit 2c96ecb1, which invalidated a huge chunk of the quilt patches. I thought it might save you some effort when 4.5.0 is finally

Is this a version that you compiled? If so I would suggest re-compiling it with the new kernel Cattab Cattabiani wrote: > That's my problem: i've recompiled the kernel > and the wine script doesn't run nomore. The error message is: > "/proc/sys/fs/binfmt_misc/register: > no such file or directory." > The directory is there but i can't write in that