search for: stevej456

Hi list, There are many Polycom experts on this list -- hopefully someone has a solution. With *several* versions of Asterisk 1.4.x, doing a reload of Asterisk causes the Polycom 601 phones to start dumping these messages to the CLI. -- Incoming call: Got SIP response 500 "Internal Server Error" back from 192.168.2.x They continue on until we force the devices to reboot from the

Hi everyone, We have a particular user on our Asterisk 1.4.x system who always listens to his voicemail messages via email. - Is there some way to send the voicemail ONLY to email and not retain them on the phone? - Alternatively, can the voicemail system only keep, say, just the last 10 messages (as backup in case of email delivery failure or a message getting deleted in email accidentally

Is there an Asterisk CLI> command to show a list of calls in progress (for all channels: Zap/SIP/IAX2 etc). "Restart when convenient" waits until the system is idle, but is there an obvious way of seeing what's going on at the moment? Thanks, Steve

The DNS for www.voip-info.org seems to be non-responsive. Is there a mirror of this invaluable resource site? Tx, Steve dig www.voip-info.org ;; Got SERVFAIL reply from xxx.xxx.xxx.xxx, trying next server ; <<>> DiG 9.4.1-P1 <<>> www.voip-info.org ;; global options: printcmd ;; Got answer: ;; ->>HEADER<<- opcode: QUERY, status: SERVFAIL, id: 61402 ;;

Hello all, I have one Asterisk 1.4.21 system connected to a North American POTS line. Normally hangup detection works fine, and Asterisk hangs up properly if you are talking to a caller and they hang up; but occasionally a call comes in (typically from a US telemarketer) where the caller hangs up just as voicemail recording is starting, and you get a voicemail recording of dialtone (then

Hi everyone, How can I add the equivalent of: exten => s,n,SIPAddHeader(Alert-Info: Ring Answer) in a .call file? This is to support paging to Polycom phones... Thanks for all info! Steve

Is anyone using fixed Park buttons (some of the ones on the left side of the screen) on a Polycom phone? Here's what I mean: - Call is received and parked, by the user pressing an unlit park button (e.g. 701) and the call is parked there. - The call can be picked up at any other extension by pressing the flashing park 701 button. - Once the call has been picked up, the 701 park slot is idle

Anyone want to talk briefly about one-button call parking/pickup on Asterisk with Polycom phones? Does anyone use it or know to do it? On many phone systems there are 2 or 3 park buttons, and you can park a call onto an unlit park button, and then the light flashes. You can go to any other phone, and press the park button with the flashing light to pick up the call. Super easy from the

Hello list, How can you obtain the default mailbox for a SIP extension (as stored in sip.conf and shown with "sip show peer <ext>")? Is there a function to extract it? Why? Some extensions have shared mailboxes and others do not and I don't want to duplicate logic, just use the extension's default mailbox as coded in sip.conf. sip.conf ---------- [100] mailbox=100

When you press *1 in Asterisk (1.6.2.7) to start/stop call recording, the console CLI> shows: > User hit '*1' to record call. filename: wav,auto-1273791789-103-5551212,m Is it possible to play a sound to back to the person who pressed *1 to indicate to them that recording has actually started or stopped? Something like "Recording" / "Record Off", or else sounds

Is it possible to use an Asterisk feature code to transfer a call to a specific extension? For instance, if you take a call, and the caller wants to go to a conference, it would be nice to use a feature code for this, rather than going through a longer transfer sequence. e.g.: - You have a meetme conference: [conferences] exten => 21,1,NoOp(MeetMe Conference) exten => 21,n,MeetMe(50,pM)

I want to set up a conference call to be recorded automatically, so I'd like the recording to start when the second caller joins the conference (one caller already there). The recording would continue until the last user hangs up. How can you determine how many are already in the conference bridge? [conferences] exten => 66,1,Answer exten => 66,n,Wait(1) exten =>

A very nice feature of another conferencing system that I've used is that the admin/moderator can press a star code to MUTE ALL OTHER USERS on the conference. This is great if you have several people on the call and one of the people puts the call on hold (and so the music/advertisement/your call is important/etc) message starts, or someone's cellphone handsfree unit in their car is

Hi all, What's the best method to send automon call recordings (*1) to the voicemail box of the Asterisk user? Do you have to trap hangups, etc, or is there some global variable that can be set? Thanks! S.

On Tue, Apr 20, 2010 at 10:49 AM, Steve Edwards <asterisk.org at sedwards.com> wrote: > On Tue, 20 Apr 2010, Tilghman Lesher wrote: > >> On Tuesday 20 April 2010 11:05:07 Steve Johnson wrote: >>> I wanted to force a hangup of a SIP to SIP call from the Asterisk CLI> >>> prompt, and found references on using the command "soft hangup >>>

Hello asteriskers, I wanted to force a hangup of a SIP to SIP call from the Asterisk CLI> prompt, and found references on using the command "soft hangup <SIP/channel>", but as you can see below, the "soft hangup" command does not seem to exist, and there is no mention about it in the UPGRADE*.txt documents. Can anyone shed light on what would replace "soft

Hi List, We have a remote asterisk SIP phone at the cottage. I'd like it to have minimal privileges when it first registers with Asterisk. Ideally it should be in a restricted context. Dialing any number would intercept the call and tell the person to log on. This way, if the phone was stolen or someone got into the cottage, we wouldn't have a bunch of surprise charges on our phone

Can anyone comment on the DSCP quality of service settings on your Asterisk server? The network we're setting up has data on the default VLAN, Asterisk server and phones on VLAN 4, and we're using Polycom phones with a PC hooked up to the phone's pass-thru port. What iptables settings are you using on the Asterisk server for DSCP? What are your Polycom DSCP settings? We're using