similar to: sip.conf 's tonezone option working ?

Hello list, Hope you all doing fine! I've tried to use the 'alias' directive in the indications.conf file but apparently it doesn't work.... It looks like maybe this feature was removed, because old sample for the indications.conf file have example using the alias parameter, but newer samples don't have it anymore.... also I couldn't find any ticket saying this parameter

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, I've installed a new Asterisk 17.0.0 on a Debian Buster system. This Asterisk instance is run by asterisk user (and group). I've got: # ls -l /etc/asterisk total 68 -rw-r--r-- 1 asterisk asterisk 501 nov. 18 19:12 asterisk.conf -rw-r--r-- 1 asterisk asterisk 135 nov. 18 18:57 cdr.conf -rw-r--r-- 1 asterisk asterisk 684 nov. 18 18:57 cdr_custom.conf -rw-r--r-- 1 asterisk

On Mon, Apr 25, 2016 at 11:11 AM, Olivier <oza.4h07 at gmail.com> wrote: > > > > 2016-04-25 18:14 GMT+02:00 George Joseph <gjoseph at digium.com>: > >> >> >> On Mon, Apr 25, 2016 at 10:00 AM, George Joseph <gjoseph at digium.com> >> wrote: >> >>> >>> >>> On Mon, Apr 25, 2016 at 9:29 AM, Olivier <oza.4h07

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

On Mon, Apr 25, 2016 at 10:00 AM, George Joseph <gjoseph at digium.com> wrote: > > > On Mon, Apr 25, 2016 at 9:29 AM, Olivier <oza.4h07 at gmail.com> wrote: > >> Hello, >> >> I've just discovered PJSIP 's support of set_var setting in pjsip.conf. >> Is this setting also supported in pjsip_wizard.conf ? >> On a fresh 13.8.2, it

Hello, How should I configure Asterisk (13.7.0) to get persistently PJSIP SIP messages in a log file and not in console ? I would expect adding "debug=yes" in pjsip.conf to produce the same output as "pjsip set logger on". Am I understanding correctly ? Best regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

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, 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

Hi, With 1.6.1.7-rc2, doc says: select*CLI> -= Info about function 'SPRINTF' =- [Syntax] SPRINTF(<format>,<arg1>[,...<argN>]) [Synopsis] Format a variable according to a format string [Description] Parses the format string specified and returns a string matching that format. Supports most options supported by sprintf(3). Returns a shortened string if a format

Hello, I've just discovered PJSIP 's support of set_var setting in pjsip.conf. Is this setting also supported in pjsip_wizard.conf ? On a fresh 13.8.2, it doesn't seem but I may have missed somthing. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

2018-06-05 15:27 GMT+02:00 George Joseph <gjoseph at digium.com>: Thank you very much, George for replying. > > > On Tue, Jun 5, 2018 at 3:35 AM Olivier <oza.4h07 at gmail.com> wrote: > >> Hi, >> >> After a long discussion with a friend, I would like to ask here: >> >> 1.According SIP RFCs, is possible/recommended to have different values in

Hello, Could you help me to summarize what is SIP Early Media useful for ? I was thinking of: - Passing error messages to caller, - Custom ringing tones to caller. Did I miss something ? Best regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160203/33dec62b/attachment.html>

Hi, I'm seeing a trend in which SIP devices such as Yealink SIP phones (with v72 firmware), are dropping support of SNMP in favor of "HTTP eventing" if may call this as such : when configuring the SIP device, you can define a couple of HTTP URL which triggered when some event occur (end of boot, on hook, ...). How do deal with those devices ? Do you still try to monitor them with

I know that this subject has been treated in the past! As a matter of fact reading some old messages about compiling zaptel I made a couple of tests after the first compiling failure to understand why I can't compile on a specific machine, but I do not know how to handle the results. The machine has SUSE 9.3, and an updated kernel (2.6.11.4-21.7-default; as shown below). YAST (the graphical

Hi, After a long discussion with a friend, I would like to ask here: 1.According SIP RFCs, is possible/recommended to have different values in >From and P-Asserted-Id fields ? For instance, From field showing 123456789 and P-Asserted-Id showing 987654321 (beside privacy considerations) ? 2. When Bob forwards to Cory a call coming from Alice, would expect Diversion/History-Info header to

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 >>

Hello, How does extensions.lua compares to extensions.conf or extensions.ael on stability, performance and features ? Would you recommand extensions.lua as an easy/easier way to access memcached, redis or equivalent ? Thoughs ? Comments ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

2017-02-16 14:27 GMT+01:00 Joshua Colp <jcolp at digium.com>: > On Thu, Feb 16, 2017, at 09:11 AM, Olivier wrote: > > Hello, > > > > I'm currently testing a so-called VQ RTCP-XR feature from a a SIP > > hardphone. > > > > When a phone has enabled this feature, it would send a SIP PUBLISH to its > > SIP Server letting this server dispatch to