similar to: Asterisk 1.8 Multiple Parking Lots

Hello, all. I'm having a deeply frustrating time getting multiple parking lots to work and am wondering what I am doing wrong. I am using Asterisk 1.6.1.1. I defined two separate parking lots in features.conf as follows: [parkinglot_a100] ; SSI context => a100-parking parkpos => 900-920 findslot => next [parkinglot_a10] ; EBC context => a10-parking parkpos => 800-820

Hello, all. With the assistance of very helpful folks, our brand new multi-tenant setup seems to be working smoothly from start to finish with just a bump or two. The biggest is parking. Now that we got most kinks worked out, I'm a little more comfortable in trying to resolve this. There seem to be two problems: 1. Parking assigns parking spaces from the default group no matter

We are planning to have about 100+ parking lots defined in features.conf , each with about 4 unique park positions. Asterisk will be handling all the parking and unparking (we don't exclusively use Park/ParkedCall in the dialplan): [parkinglot_a] parkpos => 1-4 context=parked [parkinglot_b] parkpos => 5-8 context=parked As far as I can tell, Asterisk adds/removes extensions to the

Has anyone managed to get multi-parking lot call parking working correctly? I've had several attempts at it, and never seem to be able to get it to go properly - (actually, at all): I've most recently done this with 1.6.1.x, and now 1.6.2.x, with no luck in either case. What I've been "trying" is the following: features.conf [general] parkext => 100 [featuremap]

I have been trying to get the dynamic parking working. For some reason when I park a call using this method the console says it is using the default parking context not the one I am trying to specidfy. It also is playing the parked extension to the caller. I am transfering the call to an extension that is doing a goto to the context below. Any ideas or examples on how to make this work.

Hi all, We have been using asterisk 1.8 for some while now, together with asterisk 1.6. We have the following problem. In asterisk 1.8 when you leave a voicemail, the person at that extension is notified by email that he has a voicemail. The voicemail is attached to that email along with other details like: duration of vm, caller name, caller number, date. The duration is always reported lover

I would like to hire someone to help us tweak our asterisk system for Snom phones. We would like to enable things like: One touch recording One touch park orbits Presence Please contact off-list if you will be able to help. Thermal -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi.... I'm a little confused here... trying to setup a parking lot... lot is setup.... but how do I send calls to the parkinglot? If I allow #700 transfer, it seems I can only transfer on inbound calls... if I use a T in my dialplan I can only transfer on outbound calls... additionally pressing # to use an auto attendant elsewhere causes asterisk to try to transfer... any thoughts? Is there

Does anyone have any examples of using realtime database driven call parking lots. I am on version 1.8.x My goal is to be able to do database driven multi-tenant parking lots with out adding sperate entries into Features.conf for each lot. I also need to be able to use the same parking extension pool for each tenant but sand box them into sperate lots. We have been able to do this for every

Hi, Is there a variable that contains the Sip Peer name? I was using ${CALLERID(num)} for outgoing calls, but when a call is being transferred, that variable contains something else. I need a variable that is always set to the SIP Peer's name. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all. I am trying following scenerio for call park & pickup. voice is flowing established between B & C, after call-pickup ( instead of A & B ). can anyone please clarify why it is happening like this, ( or ) do i need some more configuration for park&pickup ? A B

Hello using asterisk 1.8.32.3 I am not able to make a call with video support. I do not know what I am missing to make this video call. Codec h264 should be supported. sip*CLI> core show codecs Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INT BINARY HEX TYPE NAME

Hello, I am trying to create a dynamic call parking lot using https://wiki.asterisk.org/wiki/display/AST/Application_Park But this manual is not enough to fix my problem : Asterisk keeps trying to park the call in the default parking lot : [Jul 2 11:32:14] -- Executing [3333 at from-770000:5] Set("SIP/testacc77000-00000002", "PARKINGDYNAMIC=parkinglot_test") in new

Hello in sip.conf I have ; videosupport=yes Kind regards. J. On 20-04-17 13:09, Marcelo Terres wrote: > I suppose that you enable the video support on sip.conf, right? > > Regards, > Marcelo H. Terres <mhterres at gmail.com> > IM: mhterres at jabber.mundoopensource.com.br > https://www.mundoopensource.com.br > https://twitter.com/mhterres >

Hi. we are experimenting a strange issue in our PBX. By example: if we dial to the 100, the call is answered in 199. We dont have any redirection for that, but the cli show the same issue when request show peers. Aditionally, the user 100 use the ip address 192.168.11.100, and the cli show connected the user from 192.168.11.160 (that ip is assigned to the user 199) PBX*CLI> sip show peers

How can we set the music on hold class using the Dynamic Parking lots? The variables set the PARKINGLOT, PARKINGDYNAMIC, PARKINGDYNPOS,PARKINGEXT,PARKINGDYNCONTEXT I can't find a PARKINGMOH variable. This is becoming a big issue. We are using the current release 11. version We have to be able to set the MOH dynamically I just can't find the mechanism. Any ideas? Thanks

Hi gurus!!! I have a Freepbx with Asterisk 1.8.25.0 with a sip trunk on the pstn Every minute asterisk sends an OPTION Request, i beleived that it's related to qualify functions. The every minute annoyng answer of the pstn is "403 Forbidden". Some people told that asterisk is not sending the username in the OPTION, required by the pstn. Taking a look of the example of rfc3261.txt

Hi list, I enabled the transfer function in my dialplan, but I may configure it wrong because sometime when I call a SIP extension number from one FXS port, the SIP side will hear word "transfer", I hear nothing, after that, the call conversation is fine.I'v had this problem for a long time, could not get clue where I configure it wrong. here is my related config part: sip.conf:

Greetings all, I'm trying to connect to an AT&T teleconference, but the call is never marked as ANSWERED by asterisk and therefore won't bridge and continue. The only work-around I've come up with so far is to dial like this: Exten => 744,1,Dial(Zap/g1,,p) The "private" mode keeps the line open without trying to do a bridge, but requires the

Hi After recently upgrading to 1.8.3 I have noticed that the nat setting for my peer in my sip table is not making it into the realtime cache. For example * Name : 501 Realtime peer: Yes, cached Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : pack-local Subscr.Cont. : <Not set> Language : AMA flags :