similar to: OT: SIPSAK usage

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

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

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

Forgot on the Snom 200 it won't reboot if under the Memory tab in the web interface, Connections > 0 then remote reboot is not possible. Manually cycling the power allows the phone to be rebooted by Sipsak remotely. HOWTO: Reboot a Snom with Sipsak Checklist: 1. Under Advanced in the web interface, is Network Identity set to 5060? 2. Under Advanced in the web interface, is Challenge for

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

On Fri, 20 Feb 2015 08:46:13 -0500, Andres wrote: > A "sip set debug on" will give you more info on why you are getting the > 404. It probably has to do something with your context/dialplan. on tleilax: tleilax*CLI> tleilax*CLI> sip set debug on SIP Debugging enabled tleilax*CLI> on doge: thufir at doge:~$ thufir at doge:~$ sudo sipsak -vv -s sip:devries at

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 No Yes 123

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

This is a bit off topic, because I 'think' it isn't an Asterisk problem. However I'm not sure and anyhow I'm hoping someone may recognize the symptom. We moved offices a month ago. Our trusty SNOM190s (all between 3 and 5 years old) were packed up for the move, then unpacked a couple of weeks later. On unpacking them and connecting them to the new network, several of

Hi all, I'm trying to connect one phone to a remote asterisk server via openvpn. First of all, I put the vpn server on the box hosting asterisk and the vpn client on another box, both with public ips. Then I set the client ip as my phone IP gateway and the remote pbx ip as the registrar and outbound proxy. I see in the phone log register packets are sent but nothing in return. Asterisk

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,

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

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

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]

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

On 2/20/15 2:29 PM, thufir wrote: > On Fri, 20 Feb 2015 08:46:13 -0500, Andres wrote: > > >> A "sip set debug on" will give you more info on why you are getting the >> 404. It probably has to do something with your context/dialplan. > > on tleilax: > > tleilax*CLI> > tleilax*CLI> sip set debug on > SIP Debugging enabled > tleilax*CLI>

Hello. I've been going through the archives, but can't discern the state or future direction of IAX on the SNOM100. The most recent image appears to be from September 2002. There was a message on this list stating that SNOM was coming to visit Digium last April with the intention of adding IAX support themselves. For a while there was reference to the I100E on the asterisk and/or

I have a whack of Snom 360's. Occasionally, *some* of them, prompt the user, on the screen, for the registration password. You enter it, everything's OK. Next day, same thing. This is like on 5 or 6 phones out of a lot of 120. It's *always* the same phones. I haven't drilled down to things like firmware rev yet, since I ordered them all as one lot, but I'm wondering if anyone

What you can still to is setting the port on the phone to port 5060 - just as a little dirty workaround until there is a better solution available. CS -----Urspr?ngliche Nachricht----- Von: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] Im Auftrag von Jason White Gesendet: Montag, 29. Oktober 2007 10:01 An: Asterisk Users Mailing List -

Some of you might already know that we are releasing a new phone, snom 360. To make the phone well-known and stable, we have made a soft phone version out of it and offer it for trial or private use for free (for more details, see the license conditions). There are only few limitations to the phone. First of all, the audio subsystem will work only work with an acceptable quality if you are using