similar to: Rebooting GS phone thru sip_notify

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.

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

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

Hi guys, I'm having a really strange problem, which I'm pretty sure has only appeared since my last upgrade (1.2.12.1) . It's about NAT and Qualify. I'm using Asterisk to register with some external SIP providers. However, they're always marked as UNREACHABLE, when they weren't before! A typical debug looks like this: hera*CLI> sip reload Reloading

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, 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)

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, I have an * server on a routable (public) IP address and a sip client behind NAT using a Grandstream phone. He is connected through a bi-directional satellite so he has a bit of latency involved. Usually I can dial this extension and them to me. But I keep getting a registration failed message. I have other sip clients not on a satellite and they don?t get these time outs. So I assumed it

Hi. I've found a problem when I pickup a remote sip phone with *8. There're both budgetones 102 and are both in the same group. When one sip phone is ringing, I can pickup the call from another sip phone, but the first one keeps playing a loud busy signal... that don't go away until I receive another call or go off hook and then on hook on the first phone. I think that could be a

hi@all I have the following problem: With asterisk 1.09 the grandstream's registers fine with both ports, with version 1.2.1 (the newest port on freebsd) I get "Unauthorized" SIP messages from the 2nd port. The ports are configured identically, the only difference is the sip and rtp port. On the first port the sip port is 5060 on the second 5062. The rtp on the first 5004 on the

Hey Steve and everyone, I looked at the configuration, and unless I am missing something I don't think they are configured # ztcfg -vv Zaptel Configuration ====================== Channel map: 0 channels configured. In the zapata.conf file, it is the sample version, but I didn't notice anything in there that related to what you said. Or is it in a different file or location? I am

Hi All, I;m using asterisk 1.4 with FreePBX and a Grandstream 4108. I am noticing a delay calling in and out via the FXO, but calls to local extension are ok. What i noticed when i used ngrep is that, it sends invite but got no response from the server, send another invite but got no response again, then again until it finally gets it. but if you will notice on the 2nd ngrep, the asterisk

We are having many successes with Asterisk and starting to get the hang of it. But, I am still having problems getting my Budgetone BT100 (firmware 1.0.4.50) to work fully. I can receive calls, but cannot make them. We have the latest version of Asterisk, Fedora Core 3, Digium TDM400P with one FXO and one FXS card configured and working well. We have a PSTN line going into the Digium card,

Hi, I recently encountered a weired situation where my budgetone stopped working. My network is like this: Asterisk on Public IP ----------- ADSL NAT Router ----- GS01, GS02, GS03 on Internal IP We have an Asterisk server running with a public IP address, which serves as the master PBX. On a remote site, we have 3 Budgetones all having internal IP addresses assigned by the ADSL NAT router. The

Hi- I'm at my wit's end trying to get a Grandstream BT-101 POS to register on my * server. Asterisk version is 1.2.1. GS Firmware is rev 1.0.6.7. Basically, I've setup the phone following the instructions at voip-info.org, and it registers for about 10 seconds, then after receiving the SIP NOTIFY from the * server, goes into "flashing display" mode, which indicates some

Hello All, I have gone thru all the resources I could find on google on asterisk + iconnect and managed to get outgoing calls working. However, I cannot get incoming calls to work at all. With the sip debug on, I can see that something is happening everytime a call is received from iconnecthere, but I get an invalid tone on the caller side. The call never rings anywhere on the asterisk. Would

Hello, When i have a "Hangup" in my dialplan (extensions.conf) the RFC says to terminate the session is to send a BYE request to UA. After tracing the traffic on port 5060 UDP i recognized that my asterisk is NOT sending a BYE request to it's peer, so the peer doen't know to end the session and continues to send RTP packages to me. Does anyone know how to fix this? Here's

Hi, I am using SIP extensions connected to the PSTN with the CAPI Channel driver. All works fine except that one of the sip phones keeps ringing when the caller hangs up before extension is answered. The phones are grandstream 100, though we get the same behaviour using other phones (X-lite, Kphone). It behaves the same regardless of whether the incoming call is from a SIP extension or an

I just downloaded the newest version from CVS(Tuesday@~7pm) and I am getting an error whenever I call the asterisk box. I cannot here any audio on the budgtone. This works fine with my pingtel phone and my sip 7960. Also if I call my Skinny 7960 it rings but I get that same error when I pick up. When the skinny phone calls the Budgtone it works fine. I have 2 budgtone phones and it does this on

A previous poster mentioned the same thing, with no response: http://lists.digium.com/pipermail/asterisk-users/2004- December/080161.html Fresh asterisk 1.0.5 install on FC3, started with "make samples", nothing fancy. It's so bland, I'm surprised the list isn't full of people having the same trouble. I have several Uniden UIP200 phones and a single Grandstream BudgetTone