similar to: Destroying a SIP extension doesn't destroyvoicemail box?is this a bug?

Hi all, I'm using Asterisk@home 2.5, and i've done: 1-Create a SIP extension. 2-Leave there a Voicemail message 3-Remove SIP extension Then I've create another SIP extension but with the same number of the above one. I found imediately a voicemail message in my voicemail box. Is this a bug? Am I doing something wrong? Best regards, Marco Mouta

Hi all, I've a Planet VIP-150T VoIP Hardphone connected to Asterisk. When I'm in a call and i press Hold button, the other party starts listening Music on Hold but then when i press the button again to get the call back it doesn't work! I've checked asterisk CLI: -- Stopped music on hold on Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 --

Try again with : samba-tool ldapcmp dc5.$(hostname -d) dc3.$(hostname -d) DNSFOREST As in dc5.your.dns.domain.tld ... Whats the result.? If it fails, please tell os your: OS? Content of /etc/hosts /etc/resolv.conf /etc/nsswitch.conf /etc/samba/smb.conf > -----Oorspronkelijk bericht----- > Van: samba [mailto:samba-bounces at lists.samba.org] Namens Mike > Ray via samba >

Have joined a new DC to an existing AD and am trying to transfer the fsmo roles to the new DC with a view to retiring the old DC. The forestdns role transfer failed with the error: ERROR(<class 'AttributeError'>): uncaught exception - module 'samba' has no attribute 'drs_utils' all the other roles transferred successfully. The console output is below: root at

samba-tool dbcheck --cross-ncs --fix Yes, should be possible, but i normaly do that after i do the following. search for : CN=58eba604-07e5-4c5d-a104-9e6f4907248f And CN=16b8c008-6c59-4b65-9f1b-530751904a75 In _msdc.dom.tld. Verify which GUID is removed, you can see that, then remove the old server GUID. Run : dig CNAME 58eba604-07e5-4c5d-a104-9e6f4907248f._msdcs.ad.fvg.lnf.it dig CNAME

Hi, My output of smbstatus is bugged somehow in the Locked files section. For every line with a non ascii character, the whole string from the character to the end (included) is removed. Hence the line is attached to the next one. Here is a simplified example Locked files: Pid ... SharePath Name Time --------------------------------- 8910 ... /path

Because of Plain text mode, my output example is split across several lines and hence is totally meaningless. If you want to see the example, follow this link : http://download.caoba.fr/r3tb7hxb2pl5/ I marked in red the places where the bug appears. ------------------------- Santiago DIEZ Quark Systems & CAOBA 23 rue du Buisson Saint-Louis, 75010 Paris ------------------------- On Thu, Feb

i create inbound confi my confi is: [incoming] exten=> 18888246463,,1,Dial(SIP/8003,60,rT) exten=> 6463,1,Dial(SIP/8003,60,rT) exten=> 18888246463,,n,Wait(5) exten=> 18888246463,,n,Hangup but y calling and send this error in my CLI: [Apr 15 09:58:48] NOTICE[26985]: chan_sip.c:14383 handle_request_invite: Call from '101396_procall' to extension '8888246463' rejected

---------- Forwarded message ---------- From: Marco Mouta <marco.mouta@gmail.com> Date: Feb 15, 2006 1:58 PM Subject: Which ATA device do you recommend? To: asterisk-users-request@lists.digium.com Hello, I'm developing a Voip Solution for a client, which ATA SIP do you recommend? there are some ATA devices fully tested with Asterisk? I hope that Asterisk experient users could give me

I am getting a log message of Failed to authenticate on INVITE to ... after months of a system working. I have changed nothing... What can cause this. I did some searching and tried setting in sip.conf (canreinvite to both yes and no - made no difference) by default I had no entry at all when this started happening. I am using sip phones, grandstream, cisco combination and all running sip.

I wish to know if there is a way to show a graphic file with 640X480 when starting syslinux bootloader withou showing the boot prompt. I want to show the graphic, then, after 4 seconds, just boot the linux, with no prompt. I tryed to make this in the syslinux.conf : prompt 0 DISPLAY boot.msg timeout 40 in the boot.msg: ^Xsyslogo.lss But it doesn't work. The timeout timer don't work.

So one of my employees just got married and is taking her husband's last name. As soon as her business cards run out she will be using the new last name for business as well. So for example is there a way to make it so that if someone goes to the directory to find her extension and they dial DOE it matches her or if they type SMI it matches her. Alternatively, can I make a new voicemail

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix

Hello Marco, On 7/27/07, Marco Mouta <marco.mouta at gmail.com> wrote: > > hi, > > The VoiceMail<http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2>application uses > */usr/sbin/sendmail* to mail voicemail messages to users. This can be any > sendmail-compatible MTA. In practice you can use Sendmail<http://sendmail.org/>, > Postfix