similar to: Denying call transfer to certain extensions

Hi, I have an asterisk server sitting behind a pfsense firewall, I have successfully configured pfsense for NAT traversal, and clients from the internet can call clients inside the network of asterisk, as well as other clients registered with this asterisk server on the internet. The problem now is when a client from the internet do a call, the call disconnects in 10~20 seconds, but during

Greetings All, I was running libvirt-0.9.10 on CentOS 6.3 and it was working perfectly until yesterday when I decided to update to 6.4, which upgraded libvirt-0.9.10 to libvirt-0.10.2. I have a storage pool of type volume group, upon upgrading to libvirt-0.10.2, the disk image gets created as a snapshot on the volume group not as a regular volume. Now every time I create a vm using

Hi, I'm Ossama Khayat, an Arab .Net Framework developer. I'm interested in building an API or COM component, of the speex codec, that is suitable for use in a Windows Forms application. This will be used in an Open Source project that will hopefully be hosted in Source Forge one done. I tried downloading the source code and compiling it but really couldn't do much with it, especially

Hello all, I have a pretty much standard installation of an Asterisk 1.6.1.6 with no PRI cards of any type (full VoIP). Occasionally (it happens every 2 weeks or so), it just stops running. I was using safe_asterisk but it seems that safe_asterisk did not restart it. I do have the core dump file at /tmp/core.myservername-2009-10-20T18:36:20+0200 but it seems it's from an earlier crash. When

Hello all, I have successfully paired my mobile with asterisk, and chan_mobile already run very well, but sometimes when i restart asterisk chan_mobile fails to initialize with the error: chan_mobile.c: Incorrect voice setting for adapter toshiba, it must be 0x0060 - see 'man hciconfig' for details. I have tried several bluetooth adapters, as well as setting Class in

1) It's crashing because LD1 is produced due to LegalOperations=false in pre-legalize pass. Then Legalization does not know how to handle it so it asserts on a default case. I don't know if it's a reasonable expectation or not but we do not have support for it. I have not tried overriding shouldReduceLoadWidth. 2) I see, that makes sense to some degree, I'm curious if you can

I'm curious about this code in ReduceLoadWidth (and in DAGCombiner in general): if (LegalOperations && !TLI.isLoadExtLegal(ExtType, ExtVT)) return SDValue <http://llvm.org/docs/doxygen/html/classllvm_1_1SDValue.html>(); LegalOperations is false for the first pre-legalize pass and true for the post-legalize pass. The first pass is target-independent yes? So that makes sense.

Hi Someone may give me an example of followme() application using in a dialplan (including what to configure in followme.conf) ? I use asterisk 1.6.1 so if your example can match to that release it's will be wonderfull. Thank in advance. Harry. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm trying to use the FollowMe app with Asterisk 1.4.17. I've followed the WIKI page on setting it up but I can't seem to get it to work. Here is my Asterisk version: pbx1*CLI> core show version Asterisk 1.4.17 built by root @ pbx1 on a i686 running Linux on 2008-01-10 12:08:48 UTC Here is a log of when the FollowMe is being called: NOTE: I've tried to use the AstDB as

In followme , is it be possible to have a third option.... Whereas, takecall=>1 declinecall=>2 proposed option transfercall=>3 or, transferring the call directly from followme isn't really neccessary, if the callee could answer the call, and transfer it someplace, that would work as well.... -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, As we know we can configure in astersik like before 5:00pm calls go to reception and after 5:00 pm calls go to some mobile no. One of my client requested that he wants to manually shift the dial plan like above as he has flexiable timing sometime he finishes at 3:00pm some time 8pm. I can not give him freepbx access. Any idea or solution. Regards Farooq --

I've been reading up on followme app for asterisk 1.4 and I have it working but I was wondering if the following was possible: Based on followme.conf present the caller with the option to locate the person: Call comes in (external or internal) and rings extension with followme configured. Before the followme app is initiated the caller is prompted to locate the person (by pressing 1 which

Is there a way to strip the normal features out of FollowMe (call acceptance, etc), and just set followme up to to blind transfer any call to an extension's associated cell number if it is not answered on the extension after 4 rings? Users want followme calls to wind up in their cellphone voicemail and I'm having some issues with ring/answer timing and Asterisk wants to pull the call

Hello list. We just upgraded to 1.6.1.11. We are using real time information stored on mysql databases. That is all running fine. Now, since we upgraded, some member don't get calls from queues. In CLI: "queue show" shows something like: 611 (Local/611 at agents) with penalty 20 (realtime) (*In use*) has taken no calls yet We use the extension 611 in different computers, in the

With our current pbx system, a call comes in from the PSTN to the receptionist. She then hits flash, which puts the caller on hold, calls my extension, says "so and so is on the phone for you", I say "ok put him through", she hangs up and I am connected to the caller. With asterisk@home I can it # then the extension to transfer to and it will ring there. But is there a

Followme is perfect to handle FMFM and it is now also realtime, but it seems impossible to assign some value to a variable, from within the followme to store info for example about the tenant the followme is running under, like instead happen for example in the queue with the setinterfacevar field. I just need to pass a variable from the channel placing the call to the followme to the channel

Guys. This idea has been banging my headfor days now and I feel the need to share with you. Imagine this scenario: all calls come in thru a receptionist, asterisk records all incoming calls, the receptionist's work is to transfer the calls to internal people but some of them are bosses and you know how bosses are, they don't want their calls to be recorded, so, I have been trying to

Hi everybody, What I need to do is to run a context where I'll pass some phones (for example: 3 numbers). I need to make something like a followme, if the first phone is not answered, I'll call the second one, and so on. That dial plan is not the problem, my problem is when I execute the AMI, I'm using the Originate. It needs a channel as an argument, so the context can be executed;

Ahmed, Yes, this is the case, I'm sure many other 'spots' in DAGCombiner use this same check or use a similar check with LegalOperations. It just seems like bad form to have core code that generates an illegal node that legalization cannot seem to handle, unless I'm missing something, which is entirely possible. Potentially we are using the wrong LegalAction, though each I've

Hi Running bristuffed 0.3.0-PRE-1f which is 1.2.1. Using *2 in features.conf for attended transfer. Works well if someone answers. But the following sequence causes issue: 1. Receptionist takes call. 2. *2 then 123 to transfer to extension 123. 3. 123 is busy or does not answer so receptionist hears 123 voicemail 4. How can receptionist reconnect to calling user - could wait for voicemail to