similar to: Phones flashing but not ringing

Hi ?? I have been trying for several days get 3 Cisco spa508g phones (firmware 7.5.5) to work with asterisk 11.6 cert1 and sla. I can get the phones to all ring when an incoming call arrives, and I see the slatrunk working. However the blf function does not work. If one extension picks up the call the others do not show the trunk in use.? And as you might expect the hold and outbound dialing does

I understand that the Asterisk SLA implementation is somewhat different from most key systems and PBX systems. I also understand that in Asterisk, one does not put an SLA line on hold since it is just a MeetMe conference. However, is there any way to make the BLF flash when the answering party on the Asterisk system presses the hold key on their set and leaves the calling party alone in the

Greetings, We are currently testing a Sipura SPA-3000 as a gateway from our Asterisk system to a PSTN line for 911 access. We have a number of locations and want to place an SPA-3000 in each, connected to a PSTN line that will provide the correct ANI/ALI information to 911 for each location. It all works great, except for a reasonably significant (4 seconds) delay between when the SPA-3000

You can even set it to zero. Mine works well when in zero. The line pick up immediately :> -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chris Stenton Sent: Thursday, February 09, 2006 6:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SPA-3000 VOIP-PSTN

Ok, here's a weird one. I've attached a spa2000 to asterisk, and got the two phones to register as exten 706 and 707. I can call exten 708 (a cisco 7940) from 707 and everything works fine. I can call exten 708 from 706 and everything works fine. When I make a call to either 706 or 707 from any phone, the phone attached to the spa does not ring. However, if I pick up the appropriate

This post is simply documenting a spouse-friendly way of using the spa-3000 as both a fxs and fxo port for basic soho environments in the US, allowing asterisk to participate as needed/wanted. All home phones are connected _only_ to the spa-3000 fxs port. The incoming home pstn line is connected _only_ to the spa-3000 fxo port. Defined Line 1 (fxs) to register with asterisk via sip (extn

I hooked up a SPA-1001 with asterisk yesterday and all works well except the phone doesn't ring. The phone I'm using has a LCD display so I can see the call come in. (with caller id info) I can answer and complete the call but it's just not ringing. The phone rings if pluged into a POTS line so it's not the phone that's the problem. I've used the SPA-1001's web

Hi, As mentioned we have the problem that sometimes (could be up to a view times a day) for the calling party (SIP Device) you here ringing. The called party however answered the phone, but hears nothing. The calling party keeps ringing until the phone is hangup. First I thought maybe the card or the server has a problem, so I changed from a PCI beronet 4bri to a Junghanns 4bri PCIexpress and

Howdy, Excuse the neophyte questions... I was wondering: (1) what's involved in setting up a call with encrypted media (I'm on a cable network and don't want my calls snooped); (2) is there a cheat-sheet for configuring Sipura handsets/hardphones like the SPA-942, and in particular for message-waiting indicator and shared-line appearances? (3) my PSTN service provider that I

After purchasing the Uniden TRU9480 and then the Panansonic 5672, both of which do not have "phone company" compatible FSK/stutter MWI, I finally got smart and found out just which Panasonic phones have this feature. Only the following 5.8G models in their current line have FXO compatible MWI. I purchased the 5771 unit and one remote. I have confimed it does in fact work with Asterisk

Hi all, There are a lot of posts around the web about my question; unfortunately I have not been able to get any of the solutions to work. I'm using Asterisk 1.6.2.8 under CentOS 5.5. I'm trying to get call pickup working for the secretaries that monitor their bosses' phones. The BLF and the speed dial works great on the Linksys phones. Call pickup is the problem. My features.conf

SNOM phones can be configured using files on a TFTP server. On Thu, Apr 9, 2015 at 11:14 AM, jg <webaccounts173 at jgoettgens.de> wrote: > > Does anyone know how to program Snom phones using a Mac addresses like > what is done with the Ciscos. I have about 50 extensions to be programmed > and I am hoping there is a way on Asterisk to program extensions on the > snom

Hi Can anyone please guide us if there's any way of disabling alerting/ringing in asterisk when a call is placed to any subscriber. What we want is the channel establishment as it happens during a call progress but the subscriber should not ring. Is this possible in asterisk? Regards, Amber, Sarosh & Naheed -------------- next part -------------- An HTML attachment was

Hello Everyone, I have a new problem where when placing the call, asterisk will automatically go into music on hold until the call is connected (ie, no ringing). It was kind of confusing because sometimes `SESSION PROGRESS` takes longer than others, during this time we are in MOH. The call does eventually connect and the MOH stops. When debugging I saw the following debug message: [Sep 10

Hi I'm having difficulties getting up my TE110P (running as a E1) when I connect it to the PRI. If I start the server with a loopback connector everything seems fine and the led is green but when I connect it to the PRI the flashing starts .... I can't seem to find anything in the log that suggests what could be wrong .....Aug 8 11:33:00 DEBUG[1369]: Updated conferencing on 31,

Setting up pick/callgroups really tightly is the way to go - one of our techs here does that all the time for clients. PaulH On Wed, 2007-06-20 at 20:51 +0930, Tom Lanyon wrote: > Hi list, > > Does anyone have any advice on the following: > > Incoming calls to our office come in on a SIP trunk. Since all our > offices/desks are in close proximity, we would like just a

Hello, I've got various phones (mostly SPA-922) behind NAT registered to Asterisk. I've set nat=yes and canreinvite=no, and everything seemed to work great with 1.2.17. After upgrading to 1.4.2 using users.conf and macro-stdexten my spa-922 can't call other extensions. -- Executing [23@default:1] Macro("SIP/22-b72006f0", "stdexten|23| SIP/23") in new stack

After searching around, I've been unable to to find any relevant info on this. Perhaps the group can help? I am seeing something strange with a new Sipura SPA-3000 (and I've noticed this also with an IAX softphone): When I dial 777, this dialplan (in extensions.conf) is run: exten => 777,1,Dial(Zap/1/2345678) exten => 777,n,Hangup The number is answered by the called

Hi All. I've been experimenting with SLA on Asterisk 1.4.13 (patched up to 1.4.14). I am using a SIP channel for my "trunk" line. On the whole things are good, but I have noticed that if I misdial an outgoing call, i.e. I get 404 "Not Found" in the SIP trace, then the trunk line just drops, rather than presenting an error tone or message to the user.

anyone using/experimenting with this new feature in asterisk 1.4? is anybody able to post some info how to use and what features are supported? I have general knowledge how SLA should work, ie. monitor status of another line like BLF with additional features like answer ringing call, barge into existing call on shared line and make conference call or resume call, that was put on hold on