search for: sip_notifi

Good day all What is the file sip_notify.conf for Thanks Altus

Hello list, Does anybody managed to reboot GrandStream phone with sip notify <sip_notify.conf section> <peer> It seems that I need to send a sys-control Event but i suspect that's not enaugh.... my phone just answer me a CSeq: 102 NOTIFY..... Cheers Laurent

Hello, One common mean to remotely configure a phone is to send it some XML data using HTTP. Of course, this XML data is vendor specific but thanks to Asterisk multiple tools, it is quite easy to customize your dialplan to both build and send this specific XML data. I have just discovered one interesting capability from one phone vendor: getting XML data from incoming SIP NOTIFY messages instead

I would be very interested in using sipsak for something like this. What have you tried so far? -Thufir 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

Hi, Can you give me some councils of remotely rebooting sip phones in asterisk server? How to configure sip_notify.conf and sip.conf? Kind regards, Guan ; Reboot Polycom Phone Event=>check-sync Content-Length=>0 ; Untested (Reboot Sipura Phone) Event=>resync Content-Length=>0 ; Untested (Reboot GrandStream Phone) Event=>sys-control ; Untested (Reboot Cisco Phone)

With Asterisk 1.4 I could use commands like: /usr/sbin/asterisk -rx "sip notify reboot-snom mjc_home" to reboot a snom phone. Now, with 1.6, when I try that, I get: Unable to find notify type 'reboot-snom' Command 'sip notify reboot-snom mjc_home' failed. Do I need to add some magic to sip_notify.conf? I haven't quite figured out how to make it work. - Mike

Hi all, I've got a number of Grandstream phones and I'd like to be able to reboot them remotely, as I do my Polycoms... I've got this in my sip_notify.cfg: [grandstream-check-cfg] Event=>sys-control Doesn't seem to work. Any ideas? -- Take care and have fun, Mike Diehl.

Hi, list. We have a problem with loss peers after 'sip reload', our configuration: Asterisk 11.6-cert1, SIP realtime peers, sip.conf: - rtcachefriends=yes - rtsavesysname=yes - rtupdate=yes - rtautoclear=yes When we do 'sip reload' , peers are removing from available. Before `sip reload` : srv-pbx2*CLI> sip show peers Name/username Host

Hello, I noticed that changes to realtime sip peers are not applied until a 'reload'. A 'sip reload' does not make any changes to realtime sip peers. When changing for instance the mailbox-parameter in the realtime sip_buddies table, the change is not applied with a 'sip reload'. For every change there is a complete 'reload' necessary. Why does a 'sip

I have this in my dialplan... [general] static=yes writeprotect=no clearglobalvars=no [start] exten => 5000,1,Answer exten => 5000,n,Wait(1) exten => 5000,n,NoOp(${CALLERID(num)}) exten => 5000,n,Playback(tt-monkeys) which, when I dial 5000, executes this... == Parsing '/etc/asterisk/sip_notify.conf': Found -- Executing [5000 at start:1]

Hi there, this is the first time that I'm building an Asterisk-server. I have compiled Asterisk together with Zaptel on an CentOS 5.3-system. Zaptel is for later, when configuring the POTS-line. Now first internal communication with SIP. Thought it would go easier... I have 2 Grandstream IP-phones : BT-201 and GXP-1200. These are my settings : sip.conf : [root at asterisk asterisk]# cat

Hello everybody, I have setup asterisk real time extensions and its working pretty well. But the problem is when I am jumping between the contexts using the Goto statement in the database. I am getting a error = Parsing '/etc/asterisk/sip_notify.conf': Found -- SIP Seeding peers from Astdb: 'ezzibpo4' at ezzibpo4@210.211.246.47:5061 for 60

Hi All, I am having problems with some SIP peers. I seem to loose registration. If I reload SIP the registration comes back. They usually stay registered for about 2 days before they drop. The problem is not all of them drop usually just the list 2 in the list. The other strange thing is that the 2 the do drop their registration do not occur at the exact same time. It could be many hours

If I fire up asterisk it connects to my MSSQL server via ODBC just fine. However, if I issue a reload it unloads the ODBC.. then loads it again and I get an error... and keep getting it until I On a fresh start: Aug 26 15:43:14 WARNING[13818] cdr_odbc.c: cdr_odbc: table not specified. Assuming cdr Aug 26 15:43:14 VERBOSE[13818] logger.c: -- cdr_odbc: dsn is MSSQL-asterisk Aug 26 15:43:14

I am using GotoIf all over the place in 1.4.8 but for some reason, the following in my dial plan: ############################################################# exten => _1NXXNXXXXXX,1,GotoIf([${EXTEN} = "15554441212"]?100) exten => _1NXXNXXXXXX,n,Dial(SIP/provider1/${EXTEN},60) exten => _1NXXNXXXXXX,n,Dial(SIP/provider2/${EXTEN},60) exten => _1NXXNXXXXXX,n,Hangup exten =>

Running 11.13.1 on Fedora. This is a new install, but a copy of a previous - working -install. module load chan_sip Unable to load module chan_sip Command 'module load chan_sip' failed. SIP channel loading... [Oct 23 14:46:08] NOTICE[669]: chan_sip.c:31438 reload_config: Unable to load config sip.conf I don't think it's permissions: ls -ld /etc/asterisk /etc/asterisk/sip*

Through the optware-package I have installed Asterisk on an external USB. Further I have a Linksys WRT610N with DD-WRT v24 mega. I start asterisk with the following command : /opt/sbin/asterisk -c I get the following WARNING : root at DD-WRT:/opt/etc/asterisk# /opt/sbin/asterisk -c Asterisk 1.4.22.1, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer <markster at

Hello there ! I have successfully installed Asterisk on a normal server and on a Linksys WRT610N with DD-WRT running from connected USB-stick. This is not my first Asterisk-installation. I'm always finding a smaller and smaller platform to install Asterisk on. But now I encounter a problem. I'm using the same Grandstream GXP2010 with a of my installations. Every time registration goes

The Asterisk Development Team has announced the release of Asterisk 1.6.2.12. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.12 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

Hello, How to turn off native bridge in Asterisk. Is it possible ?? Bart -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031023/91dbe38d/attachment.htm