similar to: Bug or feature : how to customize SIP REFER from dialplan

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

Hello, In sip.conf, each peer/friend/user entry gathers several parameters such as type, canreinvite or mailbox. How can you specifically access to mailbox value from dialplan ? I know how to access custom parameters (ie setvar=FOO=value) but I don't know to access standard parameters. I'm specifically concerned to access to mailbox's value (from a given entry) but would be

Hello,

Hi, Your colleague has forwarded his incoming calls to his secretary. How do you call the feature allowing you to circumvent your colleague call forward to make your colleague's phone ringing ? Best regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070326/20c37b72/attachment.htm

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

Hi, Digging on this case : 2009/5/26 Olivier <oza-4h07 at myamail.com> > Hi, > > In my sip.conf, I've got : > [general](+) > ; register=>tcp://trunk4ipbx:password at 192.168.100.129<trunk4ipbx%3Apassword at 192.168.100.129> > register=>trunk4ipbx:password at 192.168.100.129<trunk4ipbx%3Apassword at 192.168.100.129> > > When

Hi, Currently, it seems impossible to use the output of SayNumber application as an input to Read application. So, if you want to develop an IVR with something like "You've got 23 messages. Type 1 to listen to the first one. Type 2 to leave", you must parse this message into 3 pieces and want for the last one play to start listening of user input : Playback ( "You've

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

Hi, I'm using asterisk 1.6.1 and AEL2. I'm trying to find the best way to write my own custom functions ? At the moment, I'm using this pattern (extensions.ael) : context foo { 123 => { &myfunc(123456); NoOp(${GOSUB_RETVAL}); }; macro myfunc (arg) { Return (${arg}); } 1. First, I keep getting warnings like Warning: file /etc/asterisk/extensions.ael, line

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

We use ssh (RedHat 2.5.2p2-5) heavily in non-interactive mode, for managing servers from central controllers, and transferring applications/ data around networks. Very occasionally we've seen the situation where the ssh client and server are both stuck in select, both selecting on only the tcp socket of the connection, and with no timeout. No children of sshd remain (even as zombies), and it

HP-UX 11 is looping on SIGCHLD/sigchld_handler2() when exiting a protocol 2 session apparently because we don't call wait before reinstalling the handler. Any thoughts on this issue or how to address it? serverloop.c from latest snapshots: void sigchld_handler2(int sig) { int save_errno = errno; debug("Received SIGCHLD."); child_terminated = 1; signal(SIGCHLD,

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

2009/1/27 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