similar to: OT: Using Sipsak to reboot a Snom phone < -a nswered my own question

Displaying 20 results from an estimated 8000 matches similar to: "OT: Using Sipsak to reboot a Snom phone < -a nswered my own question"

2006 Mar 15
0
OT: Using Sipsak to reboot a Snom phone
I have a custom sip reboot message I am transmitting to a Snom 200 to reboot it. The Snom gets it, it says "200 OK" but it doesn't reboot. I have turned off the challenge-reboot option in the Snom. When I modify the "Event:" directive from "reboot" to "snom-reboot" the phone yields "Bad Event" and when it is just "reboot" the phone
2005 Sep 30
2
OT: SIPSAK usage
I'm using sipsak to send messages to Snoms in my subnet. At work, works fine: sipsak -M -O desktop -B "foo" -s sip:1001@192.168.1.220 -H 192.168.1.46 displays "foo" on the Snom display On my home LAN (AAH 1.5, Snom 190 3.60s, switched 100, no VLAN, no routing) the same command (modified for my LAN) always yields: (type: 3, code: 3): from 192.168.171.8 at the console
2006 Mar 09
3
OT: Snom 320, displaying text on the scree n from *
try "sipsak -M -O desktop -B "foo" -s sip:<user>@<registrar> -H <ip of registrar>" the trick is to specify the "-O desktop" parameter + the "-H <ip of registrar>" parameter. Sipsak fakes the host-header of the registrar so that the Snom thinks it is coming from your Asterisk server, then lets the message through to the
2005 Sep 23
0
RE: SNOM 190 '486/Busy here' after upgrade to re 3.60s
AHA! # 1 is the case! Seems the user was fooling around with the phone after the firmware upgrade. Shame that that setting couldn't be locked out. Thanks to Mr Tahir and Mr Stredicke for their spot on responses. -----Original Message----- From: Usman Tahir [mailto:Usman.Tahir@snom.de] Sent: Friday, September 23, 2005 12:34 AM To: ColinA@landmarkmasterbuilder.com Cc:
2006 Jun 20
2
Snom 360 doesn't register after reboot
Hi, I'm trying my new Snom 360 phone (6.2 firmware) and I'm seeing that it doesn't register with the Asterisk 1.2.9.1 server after a reboot. I need to click "Re-register" in the web interface. I set: - Support broken Registrar: On - RTP Encryption: Off Any help? -- Domenico Viggiani
2008 Dec 11
0
SNOM Red LED on DND or unregistered Phone
Hello, I have BLF working on Snom phones. Ringing state (blinking) or "on the phone" state (solid) are working well. So the buttons are configured as "BLF" in the Snom webinterface. Now I would like to add another state for unavailable or dnd. In fact I would like to turn the LED red in the case the phone is not registered or the user pushed the DND button. So I though
2006 Mar 09
2
OT: Snom 320, displaying text on the screen from *
Hey all, First of all, thank you for the help I've gotten on this list in the past. Very helpful, and I apprecaite it. Now, what I'd like to do is send a message to my snom 320s. I'd like to have the message display regardless of what the phone is doing. I have been trying SMS, or the sipsak method on the wiki but I have had no luck thus far. Does anybody have this working,
2007 May 10
1
SNOM 360 Rejecting Calls
Does someone know coincidentally the cause for the error message specified in the Subject? The following scenario: Snom 360 behind one rout (wiederrum on a DSL line with static IP address hangs). The Snom has a private IP, routs accomplishes NAT. STUN and ICE are activated, as SIP haven 5060/udp are firmly used. Detailed packages passed on on haven 5060/udp of rout to the Snom. The telephone
2005 Jun 04
3
SNOM extension lights programmable, eg. based on asterisk variable setting?
Hi all, I've programmed on a SNOM function key the destination "**77" which in my extension plan reprograms/toggles an asterisk DB variable which I use in the extension plan for some call routing decisions. I would like the SNOM extension light to permanently reflect the current toggle status of my application logic/asterisk DB variable. Is something like that possible to do in
2010 Oct 23
1
SipSak: Send SIP OPTION with password
Hello, I'm trying to use SipSak to check if my Asterisk server is available/running with the following : sipsak -vv -s sip:username at sip.domain.tld -c sip:username at sip.domain.tld --password guessthis --hostname XX.XX.XX.63 The SIP OPTION is received by Asterisk as follow : OPTIONS sip:username at sip.domain.tld SIP/2.0 Via: SIP/2.0/UDP
2020 Sep 09
1
Info 16.13.0 with SNOM FW 8.7.5.35
Hello, I found a problem with latest Asterisk 16.13.0 and SNOM3xx phones (EOL) running the above FW. Incoming calls are no more working, we get error 404 despite the fact that broken registrar is on for the account. Previous FW for this phones don't have the problem. At this time I open a ticket at SNOM support, don't know if I should also open an issue at jira. Regards -- Daniel
2015 Feb 19
0
sipsak: 404 error
Hi, I **think** that I have user of thufir101, because I get a 200 response below, but I also get a 404. It seems to depend on how I send the ip address/fqdn? tleilax*CLI> tleilax*CLI> sip show users Username Secret Accountcode Def.Context ACL Forcerport 201 password 201 default No Yes
2005 Jan 03
6
SipSak: error: this FQDN or IP is not valid: voicegw
Hi, I've tried to use SIPSAK to understand the troubles i'm having about sending my voice to the person I've called (extension), after doing this tests below I always got this error "error: this FQDN or IP is not valid: voicegw". This could cause problems (namely audio problems)? Best regards, Helder voicegw:~# sipsak -C empty -a password -s
2015 Feb 20
0
sipsak 200 for a user, but 404 for a different user...why?
On 2/20/15 6:15 AM, thufir wrote: > What's the difference between user "123" and "devries"? Based on the > output here, they seem the same..? > > tleilax*CLI> > tleilax*CLI> sip show users > Username Secret Accountcode > Def.Context ACL Forcerport > 201 password 201 > default
2009 Jun 08
0
SendText and sipsak
Hi, Following advice in voip-info.org, I could successfully send text to a remote SIP endpoint using sipsak and this command : # sipsak -M -v -s sip:7530 at 192.168.100.123 <sip%3A7530 at 192.168.100.123> -B "Lunch time" warning: ignoring -i option when in usrloc mode timeout after 500 ms timeout after 1000 ms timeout after 2000 ms timeout after 4000 ms timeout after 4000 ms
2015 Feb 20
0
sipsak 200 for a user, but 404 for a different user...why?
On 2/20/15 3:20 PM, thufir wrote: > On Fri, 20 Feb 2015 15:07:35 -0500, Andres wrote: > >> This is showing nothing so I don't think your test message even made it >> here. I think it looped in the 'doge' server. > > I was wondering the same thing :) > > > in tleilax, I looked in /var/log/asterisk/messages and see: > > [Feb 20 15:13:19]
2005 Jan 12
1
SNOM 190 Configuration with Asterisk
Anyone have a suggestion for a configuration example using Asterisk and SNOM 190 SIP phone? I have read both sets of documentation and for the life of me, I can't get the phone to register and work. I can use a IAX softphone and it works perfectly. It is just the SIP thing I guess. Anyone have conf examples they can share? Thanks
2005 Sep 01
3
Snom 360 and hints
PH> I am setting up a snom 360, and the lights come on OK when the mapped PH> user makes an outgoing call, but when the user takes an incoming call PH> the light does not come on. PH> I do not want to install the bristuff patch if possible. PH> (although I can see that with the devstate command I can make the lights PH> do whatever I want) First, ensure that the 360 has
2003 Oct 13
1
newbie: need help configuring asterisk and snom
Hi all, I have been struggling desperately to get * work together with my snom100 for days on end, but I am not making any progress... Of the entries marked *#) I'm still not sure what it does; so far I have on the snom in "SIP/lines" -user name - empty *1) -account - Conrad -registrar - 192.168.200.83 -action - "None" *2) in
2015 Feb 20
2
sipsak 200 for a user, but 404 for a different user...why?
On Fri, 20 Feb 2015 15:07:35 -0500, Andres wrote: > This is showing nothing so I don't think your test message even made it > here. I think it looped in the 'doge' server. I was wondering the same thing :) in tleilax, I looked in /var/log/asterisk/messages and see: [Feb 20 15:13:19] VERBOSE[3661] chan_sip.c: [Feb 20 15:13:19] <--- SIP read from UDP:192.168.1.3:38154