similar to: Parking & SIP Phones

These all work with sip native transfers and sip attended transfers in the example of "query" option. I tested this on my 7960 and it works great. Please others test it. cd /usr/src/asterisk cp contrib/scripts/astxs /usr/bin/ cd apps wget http://www.bkw.org/app_valetparking.c cd .. astxs -install apps/app_valetparking.c ; ; Lets Auto Park the person in the first available slot in

Since www.bkw.org seems not to exist anymore (getting response from some hosting provider), does anyone happend to have a copy of app_valetparking.c from www.bkw.org - the one that should work with * stable 1.0.X ? If so please contact me. One that can be downloaded from www.loligo.com dosn't compile with 1.0.X, and SuperValletParking (www.asterlink.com/svp/) seems to be for * HEAD

Nope ! This is the one that tries to include PRE 1.0.X header file <parking.h>. It cannot compile on * 1.0.X (I have tried also to include <features.h> instead of <parking.h> (as far as I know features.h is successor to parking.h), but still without results). Thanks anyway. Nenad > > Try this > >> Since www.bkw.org seems not to exist anymore (getting

Hi, I've been fussing with this for a while now - and cannot seem to get it to work correctly - or rather as I desire it.. I'm trying to implement a 'find-me' feature in my voicemail. Basically - pressing 1 at the voicemail puts the caller into a queue. The members of the queue are my cell phone and my work DID.. I need the queue to call both numbers at the same time and wait

There is finally an introductory book about Asterisk! It looks like Paul Mahler at www.signate.com wrote it with a lot of help from Digium. I looked at the sample pages, it looks great. __________________________________ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail

Hello, I'm wondering if anyone else has been observing problems lately with 1.4.18 and higher releases of asterisk not properly recording calls. When using MixMonitor, the resulting file is only a few bytes long. I think this is because asterisk is doing Native bridging even though MixMonitor should block that. Did something change around the release of 1.4.18 that would have changed

It seems to never fail - after 3 to 5 minutes SIP -> IAX calls drop audio on one side. I place a call out through voipjet, and call quality is flawless. However a few minutes later the person who I'm talking to can no longer hear me. I can still hear them. What should I look for to resolve this? Has anyone else had this problem? Using last night's CVS this problem still exists.

test only. good luck! james.zhu -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070807/0fd2b827/attachment.htm

Hello, I just got off the phone with my PRI provider, and was told that I am not sending an expected message when I reject a call with a Cause Code for Unassigned(1) and Congestion (42). Busy works fine though... Anyhow, they are seeing the RELEASE COMPLETE message with cause code 1, however the tech told me they expect a PROGRESS indicator with a value between 1 and 10. Any ideas on how

Hello, I've been playing around with two generic X100P analog cards to create a proof-of-concept system before we go ahead and hook up a PRI. I'm running into a reproducible problem with sound quality of bridged calls, and am hoping someone will be able to point me in the right direction. I have in my dial plan a _9. extension so outgoing calls can be made... the first thing is

Does anyone have Music-On-Hold and valet parking, or regular parking working together? No matter how I configure it, I cannot get moh to continue to play after I park a call using either valet parking or regular parking. The only thing I can think of is that I might need to use # transfer instead of sip native transfer? Shouldn't this just work? If needed I can post the config for one

Hello, I've got my dialplan configured to do a double ring when a customer service call comes in, and a normal ring when an extension is dialed directly. When a customer service call is transferred, I want to ring to revert back to normal. In the local extension macro, I have the following ; make sure ring is set to default exten => s,n,NoOp(${ALERT_INFO}) exten =>

It seems Valet Parking and Super Valet Parking won't compile with CVS after 3/4/2005. Apparently some major reworks to Asterisk has broken the out-of-tree modules. Ref: http://bugs.digium.com/view.php?id=3573 The standard Parking lacks functionality and doesn't support contexts. Yet the more current CVS supports several much needed features like Chan Spy and native MOH as well as

Hi Folks, Apologies to anyone who'd prefer not to see this query on this list; but I'm asking because it is probably the forum where I'm most likely to get a good answer! I'm interested in the provenance of the name "normal distribution" (for what I'd really prefer to call the "Gaussian" distribution). According to Wikipedia, "The name "normal

Has anyone tried using a debit/credit card terminal as such: Terminal <-> SPA-2000 <-> Public Internet <-> * <-> PRI I'm hoping someone will tell me they have done this successfully and rarely experience dropped calls. Though I'd like to hear from anyone who has tried and failed as well. Thanks, Trevor Peirce

Is there any way to define call parking parameters for different contexts? For example, if I have a client in context 100 and another client in context 200, can they both define parking positions, say, from 701-710, where 701 in context 100 is different from 701 in context 200? Or even better, can context 100 define parking positions 701-710 and context 200 define parking positions

Hello, In my quest to create several proof of concepts for what can be done with Asterisk, I've run into a bit of a problem. I have a pair of SPA-2000's acting as off premise extensions for an analog line. When a call waiting call comes in, the caller id information makes it though the ULAW codec and displays on the caller id box, however asterisk doesn't seem to want to pick

I don't suppose anyone might know why I hear ringing transposed over itself when I place a call out via PRI? SIP to SIP is fine SIP to IAX is fine SIP to PRI is always transposed I mean sometimes you don't notice it much because it's lined up right, but other times you'll hear a really long ring (starts sounding normal, then sounds "weird" -- like two rings played at

Hi, the topic says it all really. Does the Sipura 3000 detect and report UK clid correctly? thanks Mike

I've seen lots about presence and Polycom phones recently. I've got one here for evaluation but noticed other phones only seem to appear busy when they initiate a call. If they receive a call, they still show as available. Is this a config problem on my part, or is that as far as presence is working right now? Thanks! Trev