similar to: What changed in Directed PickUp between 1.6.1 and 1.6.2 ?

Hello We have recently upgraded our internal PBX from 1.4 to 1.8. This made the BLF lamps on our Polycom phones stop working. After a lot of googling and a lot of testing, I have been unable to find a solution. I did try to change the call-limit value from 4 to 1, and this actually made BLF work (noone suggested this, and what documantation I can find states that this option is deprecated). This

Dear all i'm planning an upgrade of some asterisk installation from 1.4.32 to 1.6.0.28 (as i think it should be the most stable now). Reading the UPGRADE-1.6.txt file i've noticed that: * SIP: The "call-limit" option is marked as deprecated. It still works in this version of Asterisk, but will be removed in the following version. Please use the groupcount functions in the

Hi, I would like to test the following COLP use case : Alice and Bob are both using a SIP phone registered on a Asterisk 10 server. Alice dials Bob's extension. While Bob's phone is ringing, Asterisk updates Alice phone screen with Bob's name, so that at a glance, Alice can check she dialed the correct number. Before diving into Asterisk documentation, I would be happy to be

Does anyone have use for a broadcast/annouce app? I wrote SystemAnnounce which will play a specified file to all active channels (in an UP or bridged state). This was originally to tell users to get off the system, but there are several other uses... I also wrote a new CallPickup and CallPark app, both of which work more as expected (supply actual extension numbers, etc). Let me know if there

I don't understand this behavior. Why does the every data point get trashed by data.matrix when there is one non-numeric element in the array? Thanks. > temp GDP CPIYOY 19540 2098.1 garbage 19632 2085.4 0.9 19724 2052.5 0.8 19814 2042.4 1.1 > data.matrix(temp) GDP CPIYOY 19540 4 4 19632 3 2 19724 2 1 19814 1

Dear all, I have some questions, I'm sure it's pretty stupid for most of you, but I need you guys to help me. Here are my questions: 1. Music On Hold, it doesn't play any sound on the parked call or hold call. But if I do ps-ax, it shows mpg123 .....( I forgot the exact line). I'm using slackware 9.1 2. I have fxs 3 port, and in my zapata.conf I have included callpickup=1-4,

Hi all, I am having a problem with ChanIsAvail. It always returns the same result, regardless of whether an extension is available or not. It always returns 0 Unknown Status. This is my dialplan. exten => _2XX,1,ChanIsAvail(SIP/winsor_${EXTEN}|s) exten => _2XX,2,Verbose(0, ${AVAILSTATUS}) exten => _2XX,3,GoToIf($[${AVAILSTATUS} = "1"]?4:5) exten =>

Greetings all, First time poster, Sorry if this has been answered here before. We recently replaced a failed 1.4x asterisk PBX at a customer location. Voicemail access was setup when the customer dialed *8, This worked in 1.4. Now, Running 1.6 (I know it's old I had to load it quickly, And that's what I got working first. It'll get upgraded to 1.8 soon). The strange part is *8 no

Finally found a nice date chooser script (http://projects.exactlyoneturtle.com/date_picker), but am a little confused how to implement it. So I put this where I need the date picker: <a id="_name_link" href="#" onclick="DatePicker.toggleDatePicker(''name'')" class="demo_link">[ choose date ]</a> <div

Hello, I'm using Thomson/Technicolor ST2030S hardphones with Asterisk 1.6. Changing from 1.6.1.18 to 1.6.2.6, I can see a change in Pickup's behaviour and I'm a bit confused about it. With 1.6.2.6, when extension 7791 is calling extension 7792, I can see INVITE messages coming in and out Asterisk. I can also see a NOTIFY message advertising this call to subscriber 7793, for instance.

So I'm having trouble getting call-pickup working. Got a few different SIP phones (cisco 7940's and SPA-841s) all with pickupgroup=0 in sip.conf. I can't seem to get it working. This *is* possible from SIP phones, right? Do I need to add anything to my dial-plan? -- Paul A. Dugas Dugas Enterprises, LLC paul@dugasenterprises.com 1711 Indian Ridge Drive

Hello. Once again, I came up with a problem for which I can't seem to find a solution. I'm not able to make BLF work with Thomson ST2030 phones and Asterisk (1.2.13). I've set up hints in dialplan, as well as Subscibe keys on the phone. The LED status gets updated according to the associated line status. However, when a phone is ringing, If I try to pickup the call by pressing the

hi folks, Im pretty sure this has been covered before but I just wasnt able to find any answer. Im having troubles with the call pickup feature, is just not working for me. whenever I press *8 or 200 or anyother. nothing happens and sometimes I also get" nothing to pickup". I have read this might be a bug although I havent found any patch for it. does anyone have any ideas? Im using

Hello ! I am having troubles setting up Busy Light Fields (BLF) in asterisk 1.4.18 The things work up to 80%, I can transfer the call by BLF button and I can see the green (free) status and red (busy) status. What I cannot do is to accept the call when someone rings a remote extension. The BLF button starts to blink in red telling me that the call is ringing on remote extenson, but if I press it,

<snip> Replaces: pickup-9582-c0a80101-d-4 at 192.168.101.102 <snip> This Replaces header refers to RFC3891 which is not yet supported in Asterisk (see http://www.voip-info.org/wiki/view/Asterisk+SLA) This INVITE fails with : <snip> chan_sip.c: Trying to pick up 7792 at subs <snip> app_directed_pickup.c: No target channel found for 7792. If I'm dialing *87792 instead

Does anyone have any idea what can cause nmbd to begin gobbling CPU? We are running Samba 2.2.4 on HP-UX 10.20, having upgraded from Samba 2.0.10 about a month ago. Our clients are a mixture of NT 4.0 and Win2K. Today, we have begun to see widespread client drive disconnections, and nmbd is averaging 60%+ CPU usage with peaks up to 100%. We have heard reports of network problems at our site

Hi all, Consider this ARM assembly code of a C function: 00008124 <foo>: 8124: push {r4, r6, r7, lr} 8126: add r7, sp, #8 8128: mov r4, r0 812a: ldrsb.w r0, [r2] 812e: cmp r0, #1 8130: itt lt 8132: movlt r0, #85 ;

Hi, I'd like to implement the following: Extension 101 calls 102 but 102 is busy and has no voicemail so 101 is sent to a custom IVR that says something like "extension $EXTEN is $DIALSTATUS. Please try again later or dial $CODE now to notify you as soon as $EXTN is available.". So the "notification" part is what I'm trying to figure out. The extensions are SIP (but

Hello, Does anyone know when we will see asterisk 1.6.1 (and/or 1.6.2) binary packages at packages.asterisk.org? Greets. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100403/6acc4c2c/attachment.htm

Hi A question, i have upgraded a beta serveur from Asterisk 1.6.1 to 1.6.2 and now all SIP Relatime user are rejeted : [Oct 5 05:39:22] DEBUG[15081]: chan_sip.c:21639 handle_incoming: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 5 05:39:22] DEBUG[15081]: chan_sip.c:21658 handle_incoming: Ignoring SIP message because of retransmit (REGISTER Seqno 44199, ours 44199) [Oct 5