similar to: *1.4 Received SIP subscribe for unknown event package: call-info

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

Mostly to debug/test BLF, is there a softphone or another app. which can subscribe to hints on Asterisk? Heck, it doesn't even need to be able to do calls :-) Leif

Appearently SPA941 is less than a SPA942 without two ports, poe and backlight. There is less features too, it doesn't support BLF. Is it possible to hack 942-software into 941, or is there another workaround? Leif -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091112/0a6cbf82/attachment.htm

I have 3 phones, SIP/3, SIP/6 and SIP/9 SIP/3 subscribes on hint on SIP/9 Phone 6 calls phone 9, blf on phone 3 flashes until 9 picks up, then it is steady red. That's correct. But when 9 hangs up the hint goes to "InUse&Ringing", the light on 3 is still flashing. It keeps flashing until somebody calls 9 and hangs up again. -- Executing [9 at local:1]

I'm trying to implement call parking with asterisk and Cisco SPA504G phones: features.conf parkext => 700 parkpos => 701-702 context => parkedcalls I defined one of the unused keys to park the calls: Key2: fnc=sd;ext=700 at 10.0.1.103;vid=1;nme=Park I also defined two other keys to pickup/unpark the calls: Key3: fnc=blf+sd+cp;sub=701 at 10.0.1.103 Key4: fnc=blf+sd+cp;sub=702 at

I want some cheap ip-phones with auto-answer, to work as paging system at dinnertime. Options, please. Leif

The vast majority of what I've done with Asterisk has been with the Grandstream GXP-2000's. These phones work great for us for everything *except* speaker quality is quite poor and appears to be half-duplex. So now that we've bought and are using 40 GXP-2000's we're doing some testing on other phones. I've bought a Polycom 301 & 501 as well as a Linksys SPA942. While

We are having an issue with the SPA962/932 combo where the phone and the sidecar will reboot unexpectedly ? could be onhook, could be on a call, doesn?t seem to matter. I read that certain early firmware revisions could cause this so I?m running what was a week ago the newest available (5.1.18). A call to Linksys support suggested that I ensure that the phones are using a recent firmware version

I'm running asterisk on a Freebsd with 2 Nic's. Inside NIC is 192.168.5.x where the phones are. Outside NIC used to be a public IP with the ISP's device set to bridging, but the new WiMAX router only offers me the public ip 94.18.x.x on the outside, and forwarding everything to 192.168.1.50 on the "Outside NIC" Some of the phones are being disconnected with Asterisk

Dear Asterisk Users, Just a quick reminder that the Early Bird Discount for AstriCon Europe ends on Saturday, April 15 2006 (US Central Daylight Time). Register today for AstriCon Berlin, AstriCon Paris or AstriCon London and save 20% ($70.00 USD) off the standard price. Register Now: http://www.astricon.net/register/ Each of the three AstriCon Europe events promises to be a great opportunity

I'm looking for a way for linux to query a pc if user X is on, and has used the pc recently or the screensaver is not active. If so, I'll route a call for user X to the phone near that PC. Ideas, anyone? Leif

I made a patch to app_dial.c to make Dial(ext1&ext2&ext3,tumeout,B) return busy when just one extension is busy. http://www.neland.dk/app_dial.c.diff It works, but... I can't figure out setting/reading an option. It looks fairly easy, but the flag is always set. *** app_dial.c.org 2009-11-04 22:15:50.000000000 +0100 --- app_dial.c 2009-12-01 09:29:19.000000000 +0100

When phone A registers, I want phone B to ring, when picked up, it should call phone A and connect the phones. Translated: When GF in Mexico powers up laptop where soft iax-phone registers automatically, I want to talk to her asap :-) How to? Leif

Asterisk 1.4.29 BLF-SUBSCRIBE go to internal IP (ngrep output): U 2010/03/02 11:34:06.013515 212.78.xxx.xxx:2048 -> 62.134.xxx.xxx:5060 SUBSCRIBE sip:12 at 62.134.xxx.xxx SIP/2.0..Via: SIP/2.0/UDP 212.78.xxx.xxx:2048;branch=z9hG4bK-d28tfohos0vh;rport..From: <sip:K922002626 at 62.134.xxx.xxx>;tag=vyx8c0trgx..To: <sip:12 at 62.134.xxx.xxx>;tag=as13e7cb7c..Call-ID:

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

Dear R list, I have a recurring problem with the languageR package, specifically the aovlmer.fnc function. When I try to run the following code (from R. H. Baayen's textbook): # Example 1: library(languageR) latinsquare.lmer <- lmer(RT ~ SOA + (1 | Word) + (1 | Subject), data = latinsquare) x <- pvals.fnc(latinsquare.lmer,

Hola buenas noches, tengo un problema bastante gordo. ¿A alguno le ha dejado de funcionar las funciones pamer.fnc y mcp.fnc con la nueva versión de R? La semana pasada formatee el ordenador y ahora scripts antiguos no funcionan. La cuestión es que me precupa que no funcione el ejemplo de tutorial del autor. Os dejo un script que debería de funcionar y no lo hace

Hi, I'm evaluating what could keep me from upgrading production systems to 1.6.2. As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I met an issue with BLF-pickup which kept me from going further. Have you met other issues I should include include in my checklist ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

In relation to our CRM-system I'd like to send a query to asterisk who is extension xxx talking to. When the operator enters the page with customer data, the crm should send a query to asterisk, to get the cli of the call the operator is having. If the number is matching the customers number in crm, a record will be made, if it is not, a popup "Are you talking with this customer

Running postfix+dovecot+mysql on a mac-mini, and trying to solve this puzzle. Is this a Mac issue?, a MySQL issue? or something I haven't configured in? You can see from the output of the samples shows in the first case, that SHA512-CRYPT seems to be Unknown. The code I used is from a Linux based tutorial on setting up Postfix+Dovecot+MySQL on a site. Everything is generally going ok, apart