similar to: Configuring BLF or Asterisk presence/Hints feature

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, I ma using Asterisk 1.2.18 & FreePBX 2.2.1. I have assigned every users in office with Polycom with 2 extensions as below 555 8555 I have configured Follow-me to ring when the users doesn't picks the phone on line 1(555) after 10 seconds & then ring the line 2(8555). But this is not a standard telephony which I have been advised to change like below. If someone calls

Hello , I want to enable recording for a few extensions. In sip.conf it is defined as record_out=Always record_in=Always under the section of extension.but it doesn't work. Extensions are defined in the extension_additional.conf file like exten => 10,1,Macro(exten-vm,10,10) exten => 10,hint,SIP/10 exten => ${VM_PREFIX}10,1,Macro(vm,10,DIRECTDIAL) I can't be sure

Version - Asterisk SVN-trunk-r12793M (1.2.4) I have 4 Grandstream GXP 2000 phones configured. However at the moment, I have had to disable BLF, Hints, and Call Limiting due to an extremely annoying bug which seems to make the phones channels "lock" in busy after a call has been hungup. If I do a show hints after say extension 200 has hung up I get the following -= Registered

In testing Asterisk 11, I've found that Asterisk doesn't seem to be sending BLF updates to SIP peers that have subscribed to a hint looking at a Custom device if that Custom device state is RINGING or RING_INUSE. All other states seem to be working correctly. The hint section of the dialplan is: [hints] exten => _3XX,hint,Custom:${EXTEN} Console shows the following for core show

Hi All, I have registered "abhijit" for SIP in asterisk Server. I am able to register my softphone (SJPhone) to the server using the name "abhijit". But whenever I try to make any calls I am gettinh the following error message:- *CLI> -- Registered SIP 'abhijit' at 172.20.28.85 port 5060 expires 120 May 26 07:34:52 NOTICE[2761]: pbx.c:1738 pbx_extension_helper:

Hi, I'm still in the process of debugging this, but I have a gotoif statement that looks like this: exten => 26,1,GotoIfTime(7:00-18:00|mon-fri|*|*?ext-queues,210,1) exten => 26,n,Goto(ext-local,${VM_PREFIX}127,1) I have others setup the same way that also seem to have the same 'issue'. The issue is that they work, but they seem to require (and I don't understand why) a

Today is a strange day. My asterisk server is suddenly saying that all extensions are on hold. All my hints are like this: -= Registered Asterisk Dial Plan Hints =- 4101 at hints : SIP/4101 State:Hold Watchers 0 4100 at hints : SIP/4100 State:Hold Watchers 0 4002 at hints

Are all of the sip phones in the same context? > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Robert Jenkins > Sent: Tuesday, January 16, 2007 1:44 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: [asterisk-users] Polycom IP601 - some hints working,

All temporary files are kept in a single directory. At the end of the autotests that one directory is deleted. Signed-off-by: Darryl L. Pierce <dpierce at redhat.com> --- autotest.sh | 20 +++++++++++--------- 1 files changed, 11 insertions(+), 9 deletions(-) diff --git a/autotest.sh b/autotest.sh index c9f8a2d..d658cf3 100755 --- a/autotest.sh +++ b/autotest.sh @@ -40,6 +40,7 @@ # an

Hi, I am using Asterisk 1.2.24, I have written my dialplan to land with an IVR with the same time if the customer knows the parties extensions they can dial directly, but what happens is sometimes its working and sometime its not working. My extensions.conf as follows, [incoming] exten => 052477302,1,Wait(2) exten => 052477302,2,NoOp(${CALLERIDNUM}) exten =>

Hi, I configured Buddy Watch function on my Polycom IP 601. It works well, until I make a reload of Asterisk. After reload, if I give the "show hints" command in Asterisk's CLI, it says that there are no watcher for the extensions that I configured. Before the reload in the CLI appears: -= Registered Asterisk Dial Plan Hints =- 3002 : SIP/3002 State:

Hello. I am running Asterisk v1.6 on Ubuntu 10.10 with FreePBX v2.8. When I call the voicemail for any of my extensions, the call just dies. On a softphone, I get no sound whatsoever; it just hangs up after a couple of seconds. On my handset attached to my SPA-3102, it get a sound like when you leave an analogue phone off the hook. I have three extensions setup and they all do the same thing.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! Are hints not yet implemented in res_jabber? I have this here: exten => 3000,hint,gtalk/gtalk_account/mari.xxxxxxx at gmail.com But the hint doesn't show any difference. It always shows online on the phone and core show hints always shows that: 6003 at internal : SCCP/6003 State:Unavailable Watchers 0 6002 at internal :

.... does anyone onlist have any 1st-hand experience (good or bad) w/ DirectDial.com for procuring computer components ? Thanks in advance :-). I am interested in them as a potential source for parts to build an Opteron box running Linux, probably CentOS, to bring this slightly on topic .... -- William A. Mahaffey III ---------------------------------------------------------------------

Hi all, I've just updated to current CVS, and have 2 polycom IP phones, one is a IP600 and the other is a IP300. The IP600 shows the status of the IP300 and a ZAP line quite nicely, but the IP300 won't show the status of the IP600.... Is there any additional debug apart from "show hints" to see why this might not be working ?? -= Registered Asterisk Dial Plan Hints =-

I have meetme working with BLF on polycom phones however when meetme is not actually being used by anyone the 'status' of meetme becomes "idle". Which the Polycom phone sees and produces a clock symbol and FLASHING red LED. Are there any 'tricks' or work-arounds to change this status to something that does not blink the phone's LED making it look busy when meetme is

Struggling with an IP650 and 7 IP335s this morning. I have the following hints defined (courtesy of FreePBX 2.9): extensions_additional.conf:exten => 300,hint,SIP/300 extensions_additional.conf:exten => 301,hint,SIP/301 extensions_additional.conf:exten => 302,hint,SIP/302 extensions_additional.conf:exten => 303,hint,SIP/303 extensions_additional.conf:exten => 304,hint,SIP/304

Hello, I'm trying to supervise an existing Voicemail box with a BLF button on Debian's asterisk 13.14.1 system. I mostly found this [1] document. I added in a context a line like: exten = *7000,hint,MWI:31 at default With "core show hints", I can read this: *7000 at subs : MWI:31 at default State:Unavailable Presence:not_set Watchers 1 My questions

Group I'm having trouble getting hints to work correctly using SVN-branch-1.4-r59289 I have hints working on several other systems but I must be missing something this time around. VoIPGW*CLI> show hints -= Registered Asterisk Dial Plan Hints =- 30@default : State:Unavailable Watchers 3 29@default :