similar to: Which GUI for call screening ?

Dave McGuire writes: >> http://wiki2.dovecot.org/PasswordDatabase/ExtraFields/AllowNets >> >> then setup fail2ban to manage extrafields > > Now that's a very interesting idea, thank you! I will investigate this. If you don't expect yor firewall to handle 45K+ IPs, I'm not how you expect dovecot will handle a comma separated string with 45K+ entries any

Am 02.03.2015 um 11:34 schrieb Joseph Tam: > Dave McGuire writes: > >>> http://wiki2.dovecot.org/PasswordDatabase/ExtraFields/AllowNets >>> >>> then setup fail2ban to manage extrafields >> >> Now that's a very interesting idea, thank you! I will investigate this. > > If you don't expect yor firewall to handle 45K+ IPs, I'm not how

Hi, I would like to customise an end user application like Centiles's callpad software (http://www.centile.com/solutions-applications-callpad.php ). Its purpose is to allow users to set or read various personal phone-related parameters (call history, voicemail settings, conference, ...) instead of using phone keys combinations. Are you aware of any software that could be used for this ?

Can someone refresh my memory how to backlist caller ID in asterisk 1.8? I had it working in ver. 1.4 but in 1.8 it changed. -- Joseph

Hi, Let's say you have an Asterisk server running. Which parameters would you check to improve service continuity ? I was thinking of : - telco lines status (make sure every is up) - registered hardphones - config files backup (compare live and saved configuration files, if files differ, notifies the administration team) - systems variables (disk and CPU) - log files (trigger an alarm for

Hi, Has anyone met any success, installing localized (ie non-english) menus within SIP firmware enabled Cisco 7941 ? Those phones seem to be trying to download localized menus from Cisco Call Manager but as they are managed by an Asterisk server, I'm looking for a workaround. Any advice ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, we had a running configruation where asterisk passed the phone number and the ddi to the pstn (i.e. 595-431) Now only the rootnumber arrives: 5950 I do not know, what to do. I tried to use callingpres (now i am just hiding every number, because 595-0 is no valid extension..) but that did not worked. > Protocol Discriminator: Q.931 (8) len=44 > Call Ref: len= 2 (reference

Hi, I succesfully install spandsp chan_misdn and digium card. the rxfax works fine and I get the fax result by email. I would like to do the same using a Patton gw + zaptel but I can't receive fax anymore, the call comes in from ISDN in the Patton gw, patton sends it to asterisk, asterisk run a macro to make a tif file using rxfax, the tif file is correctly created but with a 0 size the call

Hello, Beside Cisco 79x1-GE, I'm not aware of any Gigabit SIP Phone. Did I miss something ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070612/b9b701b3/attachment.htm

Hi, Is anyone aware of a SIP hardphone with Busy Lamp Fields supporting 2 colors (or more) ? This could be very useful to support extended presence, for instance. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090519/0b8f1b62/attachment.htm

2009/3/17 Olivier <oza-4h07 at myamail.com> > > > 2009/3/16 Olivier <oza-4h07 at myamail.com> > > Hi, >> >> I'm rather new to this domain so I may be doing stupid things without >> being concious of that. >> >> I've got a Patton MATA I'm trying to setup as T.38 fax adapter. >> Whenever I connect a fax machine (Dell

Hi, As many remember, almost one year this Skype for Asterisk extension program was announced. Has anyone tried it ? Is there any available pricelist ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090627/37b93684/attachment.htm

Hi, I feel I've read a thread about this previously but I couldn't find it. Is there way for an Asterisk server to check if a sip phone is forwarded without bothering phone's user ? I was thinking of some Alert-Info option that would let the phone reply with a 302 Moved Temporarily or 182 Queued message and not let the phone ring or display anything on its screen. So that, you could

Hello, LLDP is more and more available on various network elements (endpoint, switches, ...). It seems to ease network configuration. Do you have any experience with it ? How would you rate LLDP ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091124/fce6307c/attachment.htm

2009/12/4 Olivier <oza-4h07 at myamail.com> > Hello, > > Has someone successfully used this QUEUE_VARIABLES() function (in > 1.6.2-rc7) ? > I tried to use it as I'm using SIPPEER() but without success. > > A previous question about it remainded unanswered ( > http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/224466). > > Regards > How can

2009/1/27 Olivier <oza-4h07 at myamail.com> > > 2009/1/27 Olivier <oza-4h07 at myamail.com> > > Hi, >> >> I carefully followed instructions in README file lasting with : >> /root/register >> ... blabla >> asterisk -r >> CLI> restart now >> >> Then asterisk -r fails with : >> # asterisk -r >> Asterisk

Hi, I need to replace digital handsets in offices where there cabling is appareantly not Ethernet-compliant. Today's usage is to press a key to toggle between private ou public line before issuing an outgoing call. Are you aware of a DECT handset (to overcome cabling limitations) that mimic this line-key behaviour ? For instance, acceptable behaviours would be to dial number string and press

Hi, New to Kubuntu and Linux, I'm looking for a syntax-enabled text editor with which I could easily edit Asterisk config files. It seems Kate provide this type of service but I couldn't find anything specific to Asterisk (unlike vim) What's your advice ? Best regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi,

Hi, In exim4, I can see lines such as : mainlog.9:2008-03-12 08:53:28 1JZLmC-0000E7-0A <= root at foo.com U=root P=local S=43802 id=Asterisk-0-123413860-4174-2662 at ipbx-bs-60200 In my voicemail.conf, I see : ; If you need to have an external program, i.e. /usr/bin/myapp called when a ;externnotify=/usr/bin/myapp ; If you need to have an external program, i.e. /usr/bin/myapp called when a