similar to: Determine UAS on remote SIP phones

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

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

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

Hello all, this may be slightly offtopic :-) I have some Cisco 7940 phones with SIP firmware, connected to an Asterisk 1.2.18-BRIstuffed-0.3.0-PRE-1y-g (HorstBox Pro with custom extensions.conf). On some of the phones, two lines are configured, one for business, one for private calls. When forwarding a line to another destination (e.g. to voicemail), we can't use the phone's own

I have the following setup: - UAs registered with SER/OpenSER - SIP peers (non cached), extensions, voicemail setup (not message storage) defined in Asterisk 1.2 using Realtime When a message is left in the user's mailbox, no Notify message is sent to SER. 1. If the SIP peer is defined in sip.conf with a host=ser.domain.com then the notfy is sent to SER. 2. If realtimecache=yes is set in

hello, can anybody help me with folloving problem I have asterisk with the public IP and two UAs (snom100, x-lite) in the same private network behind the same FW/NAT. All is working good, but whan I tried to establish call between these two UAs, first 10-15 second is nothing to hear and then is the quality terrible :( Can anyone tell how to get it work with normal quality ? best regards

I would like to determine through an AGI script if a specific SIP extension has DND on or off. I know that if the SIP client dialed *78 or *79 it is usually enough to just do a: database show dnd to fetch the DND status from the database. However, not all clients dial *78 or *79 (or whichever feature code is defined for DND). Some softphones such as SJPhone have a DND button. When pressed and

Hello! I'm facing ReInvites as caller from UAS despite configured session-timers=refuse (which can be seen in the SIP trace) always after 900s. (The behavior is the same if session-timers is set to accept). This just happens with one provider (German Telekom to callee at kabelbw). - The incoming ReInvite is answered immediately by asterisk (Status 100 / Status 200 - 0.02s). Media stream

Hi, Is there ANY way for me to see the status of the Polycom DND buttons in the Asterisk hints? I`m using the BLF buttons to see the status of other people`s lines, and DND should logically be somehow reflected (I don`t care as much about Polycom showing the BLF button as DND, but I do care about Asterisk hints showing it in the CLI). Right now, a Polycom phone on DND shows up as being

I am beating my head against a problem where queue calls offered by Agent channel to a SIP UA cannot be REFER transferred if the target UA/extension hasn't accepted the call. If the members of the queue are SIP channels, this is not a problem. I suspect chan_agent isn't flagging the bridge from Zap/n -> SIP/n properly, or this is by design. The following line is what is spoken before

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

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

Hi, I've noticed that when I set a phone on DND (phone-side DND, meaning it rejects calls with a busy status, SIP 486 response code I believe) the queue keeps on trying the phone over and over again. This creates issues in terms of CDR entries - in a scenario where there is only one phone on DND, and a delay between attempts of 1 second, the queue will attempt to ring the single phone

Hi, I'm looking to disable rejecting calls from my call center employees. They are using Polycom phones. Is there a way to either disable the reject/DND features on the Polycom phones (don`t think so) or have the Asterisk PBX ignore "Got SIP response 486 "Busy Here" back from 12.23.34.45" response from specific phones/SIP registrations and just keep on ringing?

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]

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

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