similar to: Attatching VM via email for more than one user

>From voicemail.conf: ; Maximum number of messages per folder. If not specified, a default value ; (100) is used. Maximum value for this option is 9999. ;maxmsg=100 ; Maximum length of a voicemail message in seconds ;maxmessage=180 ; Maximum length of greetings in seconds ;maxgreet=60 I would like to configure these parameters on a per mailbox basis using Realtime voicemail. I

Hi i have installed a new Asterisk server on Fedora. My first server use Asterisk 1.6.x with a MySQL CDR and realtime. I have a small problems, when i configure on the new server, the same information in MySQL, we have a error: [Jun 3 16:27:59] ERROR[3140] res_config_mysql.c: MySQL RealTime: Failed to connect database server SSI on myhost.myserver.com (err 2003). Check debug for more info.

Hello, I am struggling with what seems a common unresolved problem, changing the password from voicemailman when using a realtime engine (adaptive_odbc in my case, connected to mysql). I have seen messages dating back to 2007 with this problem and the last one was bug 5168, reported as closed, but without explaining the fix

Hi list, Can anyone who has successfully ran asterisk on a home router please give me the modell number as well as how they did it ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061207/92c7f946/attachment.htm

Good morning, I have a Polycom 601 with two side cars. I created a list of contacts in XML and it shows up on the side cars exaclty how I set it up in the XXXXXXXXXXXX-directory.xml file (in the order that I wanted it etc.). However when I hit the directories button and then contact directory I see the list in alphabetical order based on the last name. I want it to show up in this list as well in

I am having major problems with the first run of Snom 360s that rolled out last month. I am working with the US vendor and they in turn are working with Snom but I wanted to see of anyone else was using these or having issues with them. Issues: Speakerphone/Hands Free volume spikes up and down during a call. You have to manually set the volume during every call. This makes it totally unusable.

Does anyone know if there is a way to get a DB or any other means to see if I can see if a call is coming from a cell phone or not. If I am able to see if it is cellular or not is there any way to see aprox. what area the phone is in (I know this wont be simple but would it work if I have an agreeement with the cell phone companies) ? This is for the US. Thanks. Dovid -------------- next part

Hi List, I was thinking the other day that even with T.38 there are still some issues with faxing. I was thinking of a protocol that instead of just sending down the fax tones an ATA or "VOIP fax machine" would get the entire fax convert it into some sort of image and pass it down the line to the receiving end. I got the idea from RFC2833. Yes I know that fax machines send bit by bit and

Hi List, I have a client that has an interesting request. He wants to have people call in to a conference room and all be able to talk however they should not hear each other. There should be admin access to for one user to call in and be able to listen in to everyone that is talking (they may want this admin to be able to talk to). Any ideas ? Thanks. Dovid -------------- next part

Hello, When I tried to install chan_h323-1.0.1-module.i386 RPM i got these errors. Failed dependencies: libh323_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 libpt_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 But i found the same files in /usr/lib/libh323_linux_x86_r.so.1 /usr/lib/libpt_linux_x86_r.so.1 What to do for asterisk to detect the same

Are there any phones whose display can show queue statistics, ie: calls waiting, etc, on the phone itself without too much trouble with Asterisk? Especially while the phone is in use (on a call)?

Does anyone know if it is possible to change the packetization time in Asterisk ? I was told by a client of mine that adjusting this with using G729 can greatly lower the amount of bandwidth used. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070819/b0cc470f/attachment.htm

Hi List, I wanted to know if anyone knew of a way with asterisk to "switch the voice" of a caller from male to female or vice versa. Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070426/2d483875/attachment-0001.htm

Hi, I am currently playing with DD-WRT and I like it. I am looking for something that is more "SIP Aware". Anyone know one those that are out there ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071126/eb28ce44/attachment.htm

I know this has been asked before and I went over the wiki but I have not been able to come to a clear answer. 1) If I have SIP Provider ----> Asterisk -----> ATA and vice versa (ATA -----> Asterisk ----> SIP Provider) from what I understand if NO NAT is being used then asterisk just starts and stops the session however the RTP media stream will be passed directly from the SIP

Hi, i have some spare (read: Boss get's a new one every few month ;)) Android Phones laying around. Does someone know a way of using them as a mobile gateway for asterisk? I could not find any SIP-Gateway in the Market, and i don't think it's possible to use the GSM Audio directly with something like chan_datacard... Regards, Jay -------------- next part -------------- An HTML

Hi List, I am trying to call from my asterisk box (1.2.18) to and audiocodes MP114. I keep getting an error from asterisk of -- Got SIP response 415 "Unsupported Media Type" back from XXX.XXX.XX.XX. Both box's are set up to use G729. Anyone have a hint as to what it may be ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have a client that wants a phone system that will play sounds from a sleep machine. I tried using all different formats (GSM, WAV, WV49, MP3 etc.). Over SIP it was OK however with the PSTN it broke up from time to time. I assume this has to do with the fact that the PSTN is limited to 8khz. Is there something I am missing here or is this simply a limitation of the PSTN? Regards,

<RANT> Funny your concerned about copyrights and moral issues regarding the work of others. One question you may want to ask YOURSELF is: Why would I use as my email a copyrighted work followed by the name of the Company that owns the copyright??? asteriskdigum@yahoo.com, Come on!! Who are you trying to fool? Are you out for the fast buck, by having someone that thinks you work for

Hi all, I need to support this feature. When caller dial if the dial fail or no answer from the called number then play a music. So how to achieve that? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110509/c3bb9124/attachment.htm>