similar to: Is adding "sip show username" easy ?

Hi, I would like to originate my first call from CLI. As I'm new to this, I'm wondering if it's possible. When I type "originate" from CLI, I've got this : " There are two ways to use this command. A call can be originated between a channel and a specific application, or between a channel and an extension in the dialplan. This is similar to call files or the

"In computer software standards and documentation, the term deprecation = is=20 applied to software features that are superseded and should be avoided.=20 Although deprecated features remain in the current version, their use = may=20 raise warning messages recommending alternate practices, and deprecation = may indicate that the feature will be removed in the future. Features = are=20

hy Hope everyone is fine, I have one issue coming in asterisk , What i am doing is i am generating a callback if some one calls at a specif access number on asterisk, Asterisk sends a busy signal to the calling party that he received a request from party and then sends the call back to the person from where asterisk received a request but in From field as you can see below astrisk is sending the

Is there anyway to get CDR to show just the answered calls. Not by exporting to a spreadsheet and editing. We have ring groups and queues and CDR shows everything as calls received. Even if it's multiple extensions ringing it shows them as multiple calls received. This seems kind of goofy.

Hi, In ASTDB, I've got a rather long list of entries like: /FamilyA/Key1 Value1 /FamilyA/Key2 Value2 /FamilyA/Key3 Value3 ... Instead of sending several DBGet queries (and parsing every response), I'm wondering if a single "database show" or "database show family" query could be implemented. Alternative if to use ssh ("asterisk -rx "database show

The Asterisk.org development team has announced the release of Asterisk 1.2.31.1, 1.4.22.2, 1.4.23.1, and 1.6.0.5. These releases are available for immediate download from http://downloads.digium.com/. This update for Asterisk includes a security fix for chan_iax2. Please see the associated security adivisory for more details: http://downloads.digium.com/pub/security/AST-2009-001.html These

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:

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

2009/3/9 James Sneeringer <jsneerin at gmail.com> > On Mon, Mar 9, 2009 at 11:13 AM, Olivier <oza-4h07 at myamail.com> wrote: > > Anyway, whenever I'm typing make menuselect, app-fax is greyed out as in > my > > opinion, spandsp libriaries have not been found. > > > > Maybe, I should have typed something like (as suggested > >

Hi, To a large extend, Asterisk's /etc/asterisk/*.conf configuration files conform to a format such as: [section1] key1=value1 key2=value2 [section2] key1=value1 key2=value2 ... To increase coherence when running custom-made application in Perl, Java, PHP, ...) and Asterisk on the same platform, I'm wondering if could extend a bit Asterisk's config files instead of duplicating data

Hello, I?m trying to get started with asterisk/SIP so I was trying the demo that is provided in the extensions config file, but the call isn?t ?answered? by my server when I try calling the number that I registered at my SIP provider. I?ve registered with register => John.Doe:MyPass:MyUser@my-sip-provider in sip.conf and if I use ?sip debug? I can see the call is coming in but then nothing

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, 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, 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