search for: sip_notify

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

...of plain HTTP. This has an advantage with remote phones as you do not have to keep an HTTP connection alive to Asterisk box send XML content to registered phones. I have just tried this capability: it works fine but I am now facing an issue: for the moment, I must regenerate and reload a specific sip_notify config file, each time I need to send some specific content. If I am not mistaken, there is no Dialplan function or application able to send SIP NOTIFY, right ? Would you rate it as a welcome addition, either for SIP or PJSIP stack ? Regards -------------- next part -------------- An HTML attach...

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) Event=>check-sync Content-...

...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.

...Feb 16 14:30:20] DEBUG[1468]: res_config_mysql.c:497 realtime_multi_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sipusers WHERE name LIKE '%' AND callbackextension LIKE '%' ORDER BY name [Feb 16 14:30:20] DEBUG[1468]: config.c:1650 config_text_file_load: Parsing /etc/asterisk/sip_notify.conf == Parsing '/etc/asterisk/sip_notify.conf': Found [Feb 16 14:30:20] DEBUG[1468]: chan_sip.c:32383 reload_config: SIP reload_config done...Runtime= 0 sec [Feb 16 14:30:20] DEBUG[1468]: sched.c:546 ast_sched_dump: Asterisk Schedule Dump (12 in Q, 623646 Total, 30 Cache, 42 high-wate...

...Why does a 'sip reload' only checks the following files : [Jun 8 17:31:16] == Parsing '/etc/asterisk/sip.conf': [Jun 8 17:31:16] Found [Jun 8 17:31:16] == Parsing '/etc/asterisk/users.conf': [Jun 8 17:31:16] Found [Jun 8 17:31:16] == Parsing '/etc/asterisk/sip_notify.conf': [Jun 8 17:31:16] Found While a 'reload' checks : [Jun 8 17:32:01] == Binding sipusers to mysql/mydatabase/sip_buddies [Jun 8 17:32:01] == Binding sippeers to mysql/mydatabase/sip_buddies Could a sip reload also check the realtime bindings ?? Jonas. -------------- nex...

...... [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] Answer("SIP/5000-0a281f80", "") in new stack -- Executing [5000 at start:2] Wait("SIP/5000-0a281f80", "1") in new stack -- Executing [5000 at start:3] NoOp("SIP/5000-0a281f80", "192...

...ensions.conf [intern] exten => 210,1,Dial(SIP/BT201) exten => 211,1,Dial(SIP/GXP1200) Asterisk CLI shows me : asterisk*CLI> sip reload Reloading SIP == Parsing '/etc/asterisk/sip.conf': Found == Parsing '/etc/asterisk/users.conf': Found == Parsing '/etc/asterisk/sip_notify.conf': Found asterisk*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status GXP1200 192.168.4.211 5060 Unmonitored BT201 192.168.4.210 5060 Unmonitored...

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 -- Executing Goto("SIP/ezzibpo4-4636", "incoming-next,6069,1") May 19 05:00:04 NOTICE[6420]: pbx.c:1688 pbx_extension_helper: Cannot find extension '6069...

...sip reload Reloading SIP == Parsing '/etc/asterisk/sip.conf': Found == Parsing '/etc/asterisk/sip-register.conf': Found == Parsing '/etc/asterisk/sip-klavo.conf': Found == Parsing '/etc/asterisk/users.conf': Found == Parsing '/etc/asterisk/sip_notify.conf': Found sip show registry Host Username Refresh State Reg.Time 202.168.56.133:5060 61990xxxxxx 105 Registered Fri, 11 Apr 2008 15:16:15 sip.pennytel.com:5060...

...k Aug 26 15:44:19 VERBOSE[13818] logger.c: -- cdr_odbc: username is voip Aug 26 15:44:19 VERBOSE[13818] logger.c: -- cdr_odbc: password is [secret] Aug 26 15:44:19 VERBOSE[13818] logger.c: -- cdr_odbc: table is cdr Aug 26 15:44:19 VERBOSE[13818] logger.c: == Parsing '/etc/asterisk/sip_notify.conf': Aug 26 15:44:19 VERBOSE[13818] logger.c: == Parsing '/etc/asterisk/sip_notify.conf': Not found (No such file or directory) Aug 26 15:44:19 VERBOSE[13818] logger.c: > cdr_odbc: Connected to MSSQL-asterisk Anyone have any ideas? my cdr_odbc.conf says: [global] dsn=MS...

...0) exten => _1NXXNXXXXXX,n,Dial(SIP/provider2/${EXTEN},60) exten => _1NXXNXXXXXX,n,Hangup ############################################################# Results in this in my CLI: ############################################################## dalint1*CLI> == Parsing '/etc/asterisk/sip_notify.conf': Found -- Executing [12223334545 at out-personal:1] GotoIf("SIP/5556685598-b7dc7f90", "[18005558888 = "15554441212"]?100") in new stack -- Goto (out-personal,18005558888,100) -- Executing [18005558888 at out-personal:100] NoOp("SIP/5556685598...

...sterisk/sip* drwxr-x---. 4 asterisk asterisk 4096 Oct 23 00:34 /etc/asterisk -rw-r-----. 1 asterisk asterisk 3588 Oct 22 18:37 /etc/asterisk/sip.conf -rw-r-----. 1 asterisk asterisk 91033 Oct 23 00:28 /etc/asterisk/sip.conf.rpmnew -rw-r-----. 1 asterisk asterisk 790 Oct 23 00:28 /etc/asterisk/sip_notify.conf ps aux | grep asterisk asterisk 294 0.1 5.5 1076736 33364 ? Ssl 14:36 0:03 /usr/sbin/asterisk -f -C /etc/asterisk/asterisk.conf The sip module itself is loaded: module show like chan_sip Module Description Use Count chan_sip.so Sessi...

...t root 3594 Jan 13 2009 oss.conf -rw-r--r-- 1 root root 11732 Jan 13 2009 queues.conf -rw-r--r-- 1 root root 496 Jan 13 2009 rtp.conf -rw-r--r-- 1 root root 38583 Jan 13 2009 sip.conf -rw-r--r-- 1 root root 317 Jan 13 2009 sip_notify.conf -rw-r--r-- 1 root root 6695 Jan 13 2009 sla.conf Could I change something to point Asterisk to the correct modules.conf-file ?? Also there are no sip and/or iax-users : *CLI> sip show users No such command 'sip show users' (type 'help sip show' for other p...

...this is the output on the Asterisk CLI : NSLU2*CLI> sip set debug SIP Debugging re-enabled NSLU2*CLI> sip reload [Oct 3 15:06:34] Reloading SIP [Oct 3 15:06:34] == Parsing '/opt/etc/asterisk/sip.conf': [Oct 3 15:06:34] Found [Oct 3 15:06:34] == Parsing '/opt/etc/asterisk/sip_notify.conf': [Oct 3 15:06:34] Found NSLU2*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status NSLU/nslu (Unspecified) D 0 UNKNOWN 1 sip peers [Monitored: 0 online, 1 offline Unmonitored: 0 online, 0 offli...

...source is changed to resolve audio delay. (Closes issue #17404. Reported, tested by sdolloff. Patched by jpeeler) * Only do magic pickup when notifycid is enabled. A new way of doing BLF pickup was introduced into 1.6.2. This feature adds a call-id value into the XML of a SIP_NOTIFY message sent to alert a subscriber that a device is ringing. This option should only be enabled when the new 'notifycid' option is set, but this was not the case. Instead the call-id value was included for every RINGING Notify message, which caused a regression f...

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