similar to: OT: Snom 320, displaying text on the screen 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

Hi folks, For the longest time, I've had a problem changing or modifying ACLs from my window clients. Whenever I tried, I'd get this in the logs: [2004/07/29 12:36:26, 0] smbd/posix_acls.c:create_canon_ace_lists(823) create_canon_ace_lists: unable to map SID S-1-5-21-1292428093-651377827-xxxxxxxxx-1333 to uid or gid. I could change the ACLs using getfacl/setfacl, btw. After a

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 all, I'm running Asterisk 1.0.0. I am a customer ( and supporter ) of voicepulse. For me, it works perfectly, but one of my customers noticed a small problem: During a conversation, when the otherside isn't talking, it's almost like the mic turns off. Not that big of a deal I know, and the more I think about it, the more this seems a voicepulse issue. But in the off

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

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

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

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

Is there a traceroute equivalent in the VoIP world? I would like to see the route a call takes after it gets to the gateway. Basically showing all the hops until it reaches it's destination or PSTN termination. -Dave

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

[root@asterisk asterisk]# asterisk -V Asterisk SVN-branch-1.2-r8632M I was wondering if there was some documentation I was missing on the '*' key and queues. I have my features setup to use *x, where x is a #, but these features don't work for calls coming in from a queue. As soon as the '*' button is hit, the call is disconnected. I have a vague memory of reading about

Hi, Is it possible to get the XML config off of a Linksys SPA-941 or 942 phone? I've tried http://[ip address]/admin/spacfg.xml however that file doesn't appear to exist. Thanks.

Hi All, I was wondering, what tools are readily available out there in Asteriskland for me to use in stress/load testing asterisk box I have in the lab. I want to observe how my box holds out under heavy/light/medium load. Thanks, Jeng ___________________________________________________________ Want ideas for reducing your carbon footprint? Visit Yahoo! For Good

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 all, I'm researching IP phones for a new office setup. We will need 30 phones. I have read the wiki and the polycom site for the phones, but I still had some questions about real world experience with these phones. -According to the documentation, the 500 series ( and 600, according to a polycom rep ) have built in hubs. Has anybody noticed performance issues in this setup, when

Is there a method from the dialplan that I can turn on a voicemail indicator on a polycom phone. Like a blinking light or something. Then I would also need to turn it off. Is there a way to do that? Jerry

Anyone have a good tool(s) to use for simulating a bunch of calls? Benchmarking or stress testing? I only need SIP protocol, and do appreciate any replies...I realize I could google it, but I am looking for opinions as well. Sherwood McGowan -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

>Date: Sat, 11 Oct 2003 22:07:49 -0700 >To: asterisk-users@lists.digium.com >From: John Todd <jtodd@loligo.com> >Subject: RE: [Asterisk-Users] SIP / IAX over satellite > >[post re-ordered chronologically] > >>-----Original Message----- >>From: asterisk-users-admin@lists.digium.com >>[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Tilghman