similar to: voice mail indicator on phone

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040106/eae682d3/attachment.htm -------------- next part -------------- Hello Whenever I receive voicemail on CISCO or SNOM or Analog Phones (Scitec), I would like to have some kind of indication in terms of beep sound or blinking voicemail indicator....Could you please tell me the way to enable

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

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

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

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.

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 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

On 27.12.18 at 18:14 Joshua C. Colp wrote: > On Thu, Dec 27, 2018, at 1:07 PM, Michael Maier wrote: >> Hi! >> >> I just want to say, that 13.24.1 doesn't fix the problem described in >> the posts above. > > You're going to need to file an issue[1] with traces and actual configuration. > > [1] https://issues.asterisk.org/jira > Before I'm

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

Hi! I just want to say, that 13.24.1 doesn't fix the problem described in the posts above. Regards, Michael

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

Hi! I have an automated dialout via a call file to a mobile. Can I detect when the call is not answered but forwarded to the mobile operator voicebox? I would like to stop the dialout if this is the case. TIA, Mike

I was able to get Asterisk working with the demo on FreeBSD 5.3 without crashing, but not the music on hold, so I just have that disabled for now, but I'm ready to get some IP hardware working. So I picked up a Cisco 12sp+ IP phone (mistake?) and am having difficulty finding any truly helpful instructions / troubleshooting to get this configured to work with asterisk. If I could just get

We have a couple of Asterisk boxes with one being the main system with everyone's voicemail and the other a slave used merely to link a couple of remote phones to the main system using IAX. How can one propagate message waiting indication from the main system to the remote phones? g.

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

I am attempting to get fax detection to work. I am using a NETjet-s card under ISDN4Linux. Asterisk does not seem to be detecting the fax tone. I have tried following as a test: [MainMenu] exten => s,1,Answer exten => s,2,DigitTimeout(3) exten => s,3,ResponseTimeout(5) exten => s,4,Background(Welcome) exten => s,5,Background(MainMenu) exten => fax,1,Dial(Zap/1,,d) [FaxTest]

Hi all I noticed that most caller are quite confused by the standard voicemail announcement text. Especialy as the number read is the 'internal' number. Callers often hang up because they think having called the wrong number when they hear the announcement. Is there a way (like in many other PBXes) that the VoiceMail user could record his own announcement? (like, hello, this is the

Hello everybody. Is there a way how to setup asterisk to notify caller's phone? Example: I have some numbers and names in asterisk database ( cidname, cidnum), and I want to display the name of person on my phone ( which has no addressbook, but can display chars ) which I am calling to be sure that I have dialed the right number. Thank you for any answer. Andrej