similar to: The most efficient way to know SIP phones IP addresses ?

Hello All: Does the Asterisk support to insert an off the board transcoder for a call? Thanks, Charles ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping -------------- next part -------------- An

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

I have members with usernames. In the event that a new member requests an already-existing username, I''d like to automatically "increment" a next-best string: johnny johnny1 johnny2 Knowing RoR, my gut tells me there''s some elegant, concise way to do this, but I can''t think of it. Any advice? -- Posted via http://www.ruby-forum.com/.

-----Original Message----- From: MAKRO BAGAFORO, Melvin Sent: Friday, November 16, 2001 11:52 AM To: 'Ben Elliston' Subject: RE: samba error Hi Guys, I already downloaded the latest config.guess and config.sub files, './configure' command is now error free. However, I have another problem, samba won't start. Samba test results:

Hello, With an extensions.ael enabled system, I keep getting whatever I change into my "astup.call" file : [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:267 apply_outgoing: At least one of app or extension (or keyword message/pdu) must be specified, along with tech and dest in file /var/spool/asterisk/outgoing/astup.call [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:457 scan_service:

2008/12/9 Olivier <oza-4h07 at myamail.com> > > > 2008/12/9 Tilghman Lesher <tilghman at mail.jeffandtilghman.com> > > On Tuesday 09 December 2008 09:14:11 Olivier wrote: >> > Hi, >> > >> > In voicemail.conf: >> > ; Supported values: >> > ; 'filename' filename of a soundfile (single ticks around the >>

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, 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

Hi, I can't find a way to tailor DHCP/TFTP/HTTP environment so that brand new Aastra SIP phones can be auto-provisioned when config files are stored in a specific TFTP subdirectory instead of TFTP root directory. For instance, TFTP root directory is /srv/tftp. When config files are stored in /srv/tftp, a new Aastra can find its config files. When config files are stored in /srv/tftp/aastra,

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 know this is a little OT but there are many Asterisk users of the excellent Siemens DECT/VOIP phones like the S685IP and 475IP and this is probably newsworthy for them. One of the biggest bug bears has been no mute function on the handset. When I woke up this morning, the handset told me there was a firmware update. I updated and then visited the web site to find out what had been

I have had a search for this, but didn't come up with any results, so maybe I am using the wrong terms, sorry if this is an FAQ. For those who want to forward their incoming voice calls to a mobile, it could be a cheaper option to call a mobile from another mobile on the same network. This probably wouldn't be useful for users in USA, Canada or Hong Kong as costs to call a mobile is

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

Hello everyone, this is my first mail to the list, please forgive me if some of my questions have been asked before. I'm currently thinking about a way to implement an efficient logging method for authentication results together with client connection info on my linux boxes. My aim is to circumvent expensive and delayed mechanisms like tail()ing syslog to get to the required info and be able

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

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