search for: 4h07

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

On Tue, Sep 6, 2016 at 1:55 AM, Olivier <oza.4h07 at gmail.com> wrote: > Hello, > > After testing "pkill -SEGV -f /usr/sbin/asterisk" on Debian Jessie > platform, I've got several questions : > > > 1. When I issue a "cd /tmp; asterisk -cvvvvvvvvvvvg -U asterisk -G > asterisk" command, and then i...

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,...

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

2016-09-02 20:40 GMT+02:00 George Joseph <gjoseph at digium.com>: > > > On Fri, Sep 2, 2016 at 9:34 AM, Olivier <oza.4h07 at gmail.com> wrote: > >> Hello, >> >> I had a recent case where Asterisk stopped due to a segfault. >> This reminded me that being sure that whenever such issue occurs, it's >> useful to have a core file or various data at hand to analyze and exchange >&g...

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

...tected as a NIC and all configuration is done using a web interface. Then you configure FreePBX a new trunk to use the berofix IP and that's it! This way the mISDN and other channel drivers' burden is skipped and works very well for me so far! On 26 May 2015 at 16:52, Olivier <oza.4h07 at gmail.com> wrote: > > > 2015-05-26 12:17 GMT+02:00 Lukasz Sokol <el.es.cr at gmail.com>: > >> Hi, >> please whoever has some expertise in choice of BRI ISDN cards, >> please restore my faith in community support :) >> >> (on private email I can...

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:

...Regards, > Marcelo H. Terres <mhterres at gmail.com> <mhterres at gmail.com> > IM: mhterres at jabber.mundoopensource.com.brhttps://www.mundoopensource.com.brhttps://twitter.com/mhterreshttps://linkedin.com/in/marceloterres > > > On 12 June 2017 at 08:07, Olivier <oza.4h07 at gmail.com> <oza.4h07 at gmail.com> wrote: > > Hello, > > I'm a faithful reader of this mailing list, for several years now. > > Lately, I'm receiving emails asking me to re-enable my list > subscription due > to "excessive bouncing". > > W...

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 doesn't seem but I may have missed somthing. >> >...

...hufir On Mon, 16 Jan 2017, Olivier wrote: > Thinking over my previous, I wonder if sipsak could be used to send > outgoing SIP NOTIFY messages. > Would both Asterisk and sipsak be able to share networks resources ? > > Thoughts ? > > 2017-01-16 14:10 GMT+01:00 Olivier <oza.4h07 at gmail.com>: [..] >> >

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 >> From and P-Asserted-Id fields ? >> For instance, From field showing...

Hello, Le lun. 6 janv. 2020 à 19:01, Olivier <oza.4h07 at gmail.com> a écrit : > May I add I could successfully (if pjsip show transports has any meaning) > add a PJSIP TLS-transport with: > > [transport-tls] > type=transport > protocol=tls > bind=0.0.0.0:5061 > cert_file=/etc/asterisk/keys/asterisk.crt > priv_key_file=/e...

Hello, I'm load testing a new Asterisk 13 system (Debian Stretch, packaged asterisk). One system writes CDR though an ODBC connection to a local Postgres database over the LAN. When sending 50 new calls per second with SIPp, I'm seeing one system outputs : taskprocessor.c: The 'subm:cdr_engine-00000003' task processor queue reached 5000 scheduled tasks again. This [1] thread

...ncorrect but I don't believe you can hint on a mailbox like > that. > > I've always used custom device states and dialplan logic for my shared > voicemail boxes that are not being watched directly by a endpoint natively. > > On Mon, Nov 20, 2017 at 5:07 AM, Olivier <oza.4h07 at gmail.com> wrote: > >> Hello, >> >> I'm trying to supervise an existing Voicemail box with a BLF button on >> Debian's asterisk 13.14.1 system. >> >> I mostly found this [1] document. >> I added in a context a line like: >> exten = *7...

2016-09-06 17:48 GMT+02:00 Tzafrir Cohen <tzafrir.cohen at xorcom.com>: > On Tue, Sep 06, 2016 at 06:37:52AM -0600, George Joseph wrote: > > On Tue, Sep 6, 2016 at 1:55 AM, Olivier <oza.4h07 at gmail.com> wrote: > > > > Where should core file be created when Asterisk is run as a daemon by > > > asterisk user and group ? > > > Is there a setting I can use to specify the directory used (so that we > can > > > make sure appropriate ownership is...

..."how to specify, with asterisk and systemd among others, where core files are dumped ?". I'll read again this thread advices and report back here. 2016-09-07 21:44 GMT+02:00 George Joseph <gjoseph at digium.com>: > > > On Wed, Sep 7, 2016 at 11:15 AM, Olivier <oza.4h07 at gmail.com> wrote: > >> >> >> 2016-09-06 17:48 GMT+02:00 Tzafrir Cohen <tzafrir.cohen at xorcom.com>: >> >>> On Tue, Sep 06, 2016 at 06:37:52AM -0600, George Joseph wrote: >>> > On Tue, Sep 6, 2016 at 1:55 AM, Olivier <oza.4h07 at gmail....

...u say of adding an extra input argument to have certificates built for a specific duration (default is 365 days and some may expect a different duration) ? Cheers [1] https://wiki.asterisk.org/wiki/display/AST/Configuring+Asterisk+for+WebRTC+Clients Le mer. 8 janv. 2020 à 10:04, Olivier <oza.4h07 at gmail.com> a écrit : > Hello, > > Le lun. 6 janv. 2020 à 19:01, Olivier <oza.4h07 at gmail.com> a écrit : > >> May I add I could successfully (if pjsip show transports has any meaning) >> add a PJSIP TLS-transport with: >> >> [transport-tls] >>...

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