similar to: Presence Management - use of hint

Playing with hints/presence/BLF on asterisk I've made the following "discoveries". 1. The wiki at http://www.voip-info.org/wiki/view/Asterisk+presence says: "If you add incominglimit=1 to your peer in sip.conf, the SIP channel will notify you when that extension is busy." As "incominglimit" is obsolete you can use "call-limit".

Hello List, Asterisk CLI command "core show hints" gives the list of hint extension configured and its presence status. In command output there is a field called "watchers" and it contains a numeric value of number of subscriptions' registered for that particular extension. So, is there any CLI command to check who the watchers for an extension are? Regards, Rajib Rajib

Hi all- Equipment: Xlite softphone Polycom 301 with SIP 2.1.1 and BootROM 3.2.3 Polycom 501 with SIP 2.1.1 and BootROM 3.2.3 Asterisk 1.4.2 SIP Trunk to FWD I wanted to post this problem as I haven't found it described in any of the past presence threads on here. I use an identical configs for a Polycom 501 and 301. (I actually unplug one when the other is in use). The

I've somehow managed to battle may way through hinting issues with type=peer type=friend and various other oddities and now have presence working correctly on my Polycom 600 and Grandstream GXP-2000 phones. However, on the Polycom I have to press the Buddies softkey in order to see if an extension with a hints priority is in use or not. I've spent all day going through google and my

Hello all, I have a number of Polycom phones 601's and 430's and I'm seeing: Got SUBSCRIBE for extensions without hint. Please add hint to s to context local-hints in the CLI over and over. I have: [local-hints] exten => 110,hint,SIP/110 exten => 111,hint,SIP/111 exten => 112,hint,SIP/112 exten => 113,hint,SIP/113 exten => 114,hint,SIP/114 The hints seem to be

Hi all, I am working on asterisk 1.2.18 zaptel 1.2.17 Polycom 650 polycom 430 SIP version 2.0.3.0131 for IP 650 SIP version for IP430 2.0.3.0127 freepbx 2.2.1 I am trying to configure BLF using asterisk but failed. I would be thankfull if somebody help me. Regards FArooq ********************************************** 1 ********************************************** in my

Hi all, I am working on asterisk 1.2.18 zaptel 1.2.17 Polycom 650 polycom 430 SIP version 2.0.3.0131 for IP 650 SIP version for IP430 2.0.3.0127 freepbx 2.2.1 I am trying to configure BLF using asterisk but failed. I would be thankfull if somebody help me. Regards FArooq ********************************************** 1 ********************************************** in my

Hi all, I am trying to enable SIP SIMPLE communication in my test environment. I have the following env : - one server (192.168.50.126) with Asterisk 11 - 2 clients using pidgin : demo-bob and demo-alice on my 192.168.50.143 I successfully had a phone call between clients. I used the following link to enable SIMPLE messaging between my clients :

Dear all, I have a probability of presence of distribution of a species of interest (varying from 0 to 1 in continuous form) and I have a set of points where I know that species really occurs. But I don´t have points of absence. So, for each true presence I know the estimated presence. I would like to know how can I compute AUC, taking account these Available data. Best wishes,

Asterisk Project Security Advisory - AST-2018-008 Product Asterisk Summary PJSIP endpoint presence disclosure when using ACL Nature of Advisory Unauthorized data disclosure Susceptibility Remote Unauthenticated Sessions Severity Minor

Hi, I've run into a minor snag trying to pass on CALLERID presence from one Asterisk to another via SIP (both running 13.16.0) I have a PRI coming in PBX_A and PBX_A is connected to PBX_B via SIP. PBX_A gets PRI calls on a 4 port Digium card, and each call naturally has its own callerid values and presence. I pass on those calls to PBX_B via SI, and I'm trying to pass on this

I am working on Polycom IP601 console with expansion module. I want to put on the BLF (busy lamp field) feature on all the contact/speed dial names I put on the console but I could not get it to work. *CLI> core show version Asterisk 1.4.13 built by root @ hostname on a i686 running Linux on 2007-11-20 05:26:15 UTC *CLI> sip show subscriptions Peer User Call ID

Hi, On Fri, Dec 30, 2016 at 9:23 PM, Daniel Berlin <dberlin at dberlin.org> wrote: > > > On Fri, Dec 30, 2016, 9:04 PM Sanjoy Das via llvm-dev > <llvm-dev at lists.llvm.org> wrote: >> >> Hi David, >> >> Looking at the original bug, it seems like a straightforward >> undef-propagation bug to me -- SCCP was folding "or undef, constant"

> On 23/03/2020 18:51, Joshua C. Colp wrote: >> On Mon, Mar 23, 2020 at 2:45 PM John Hughes <john at calva.com >> <mailto:john at calva.com>> wrote: >> >> >> Why is asterisk giving an error 500? I can find no reason, there >> is nothing in any log. >> >> >> The sequence number is from the past. The first SUBSCRIBE is

On Fri, Dec 30, 2016 at 9:27 AM, Davide Italiano via llvm-dev < llvm-dev at lists.llvm.org> wrote: > Hi. > I'm sending this email to -dev as this may be of interest of > many/people may have opinions/want to try the change before it goes in > to report problems. > I've been recently working on a patch to integrate `undef` in the SCCP > solver, in the hope of fixing

On Mon, Mar 23, 2020 at 2:45 PM John Hughes <john at calva.com> wrote: > So I've got a bit further with my project to get BLF working between > asterisk and linphone. > > Initially asterisk was rejecting linphone's SUBSCRIBE messages because > they didn't have an Accept: header. I've fixed that and now the initial > SUBSCRIBE messages work and I see all my

Hey Damien, > Generally we prefer to use ssh-askpass for agent notifications. Are you able to use that? Hmm, okay, but it's not clear to me how to make that work. Is what you have in mind documented somewhere? I don't see this specific situation covered in the manpages and a web search doesn't turn up much. I thought ssh-askpass was only invoked when the key is first added to the

Dean Collins just sent out a message a second ago (responding to an earlier posting regarding the new Skype PDA client). He said: "Presence based information is the biggest 'seller' in the IP PBX market at the moment, being able to tell what/where a person is certainly driving a lot of sales through my door." I would like to take a moment to second his message. Every

Does anyone have a working implementation of SIP Presence? I have a new Grandstream GX-2000 phone with the supported hardware and I am not sure how to setup presence with asterisk. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060531/4fc3344d/attachment.htm

My users dial *120 get to an IVR menu that plays their balance and then ask them for a voucher. Ater the balance is played and the request for the voucher is played the user don't hear any other audio from the asterisk box. I can see the asterisk server playing the files to ask for the voucher again but the user cannot hear any thing. Has any one seens this issue with IVRs. I notice a