similar to: OT: Snom 320, displaying text on the scree n 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,

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

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

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

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

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

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

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

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

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

I know that there are some patches being worked on to cache realtime users that might ultimately fix this problem, but until then, here is a little script that brings back the MWI when using the excellent mysql realtime architecture with sip: http://www.cheapnet.net/~mike/asterisk/send_mwi.txt This script relies on sipsak utility found at http://sipsak.berlios.de/ Download, rename to

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]

Hi, I recently configured Linux HA for Asterisk service (using Asterisk resource agent downloaded from link: https://github.com/ClusterLabs/resource-agents/blob/master/heartbeat/asterisk ). As per configuration it is working good but when I include "monitor_sipuri=" sip:42 at 10.3.152.103" " parameter in primitive section it is giving me an errors like listed below; root at

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>

Hi all, I have SER and Asterisk set up together with ser handling user registrations and asterisk providing voicemail services. When I ring a phone and it doesnt answer after a designated amount of time, the request is forwarded to asterisk, and I can leave a message. Now, this may seem a ridiculous question but how can I listen to my message afterwards? I have read about a solution by Java

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

Hi, Searching this list archives, I couldn't find a definitive answer to my question : how to send SMS to Gigaset phones ? My goal is to send Alert SMS such as "This phone system will be stopped in 5mn for maintenance" to every terminal (SIP phones and Gigaset DECT phones). (So at the moment, I'm not looking for way to send SMS from handsets). I could successfully send 1 short

I think I'm able to connect with Ekiga, at least it reports "registered". Curiously, when I exit Ekiga and switch to SFLphone, it isn't able to connect with the exact same parameters; it just says "trying" and never resolves. I'm not able to test outside connectivity because of too many hops: thufir at doge:~$ thufir at doge:~$ sudo sipsak -vv -s sip:thufir

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

Hello, you can have a look on Homer http://sipcapture.org/ regards On 27/09/2016 10:39, Gholamreza Sabery wrote: > Hello, > > For service monitoring you can use tools like sipsak in combination > with Zabix or Zenoss. Also using Zenoss or Zabix you can monitor the > health of your servers. This way you have both top-down and bottom-up > monitoring. For monitoring call