similar to: Persistent variables disappear when dialing Local extension

Kevin P. Fleming wrote: > Falck Kenneth wrote: > > My persistent variables (_XXX or __XXX) don't persist when I dial a > > Local extension. I'm doing a forked dial where the other channel is > > SIP and the other Local. Is this a known problem? Using Asterisk 1.0.9. > > Variable inheritance is a CVS HEAD feature, it is not > supported in 1.0.x. Thanks, I

Kevin P. Fleming wrote: > Falck Kenneth wrote: > > Thanks, I was misguided by > > http://www.voip-info.org/wiki-Asterisk+Variables which > didn't mention > > this. > > You are more than welcome to edit the page to make it obvious > to the next reader :-) You're quite right - I added a little note there to warn others now. It's a great Wiki anyway,

The hypervisor side changes for XSA-46 require the tool stack to now always map the guest pIRQ before granting access permission to the underlying host IRQ (GSI). This in particular requires that pciif.py no longer can skip this step (assuming qemu would do it) for HVM guests. This in turn exposes, however, an inconsistency between xend and qemu: The former wants to always establish 1:1 mappings

If I do T.getArch() == xxx TLI.setUnavailable(LibFunc::copysign) then this works at generating a call instead of not being able to select the ISD::FCOPYSIGN, but I don't know why I don't need to do this for other LibFunc functions (such as floor, etc... these generate call just fine)? Thanks, Ryan On Tue, Jun 14, 2016 at 11:58 AM, Ryan Taylor <ryta1203 at gmail.com> wrote:

Thanks a lot for all the suggestions! Unfortunately, it still gives problems. Most common error message is "ast_realaudio_callback Failed to write frame" after "paying the beep". Then it says "User disconnected". Also, it doesn't react to any extension entered and doesn't do any forwarding (as it should in "exten =>

Hello, I'm using asterisk 1.0.7. with a somewhat advanced setup with IAX and CAPI as channels. A call comes in via IAX2 and should be redirected to CAPI. So I wrote the following dialplan: [fromiax] exten => _8XXX,1,Answer exten => _8XXX,2,Dial(CAPI/265:B${EXTEN:1},,r) [fromcapi] exten => 265,1,Answer exten => 265,2,Dial(IAX2/PoC/11@from-lw) exten => 265-BUSY,1,Busy exten

Aloha! Concisely, I''ve been trying for several days to get VGA/PCI passthrough working. Regarding VGA vs PCI passthrough, I''m not entirely certain of the difference? Does Xen perform a different set of operations when passing through a VGA card versus any PCIe card? I''ve read the wiki and it seems the answer is "yes;" however, when passing through a

Asterisk Users, I have 3 FXO modules with the TDM400P Digium Card. I can dial into the Asterisk rings my Sip phone, but dialing out with my SPA941 phone through the zap channel is a problem. I keep getting this message on the Asterisk CLI. What am I doing wrong? Thanks in advance. -- Executing [103 at default:1] Dial("SIP/200-006fa300", "{Zap/g0/{EXTEN:1}") in new

Managed to implement this on asterisk v1.4.24.1, Also, Hangupcause updating to user field. However, this only works on the edge of my voice network (demarcation point) It does not work on my internal routing boxes as I use IAX to route between remote sites. I was thinking of using some sort of SIP variables to transport these results over the IAX trunk.. Any bright ideas folks???

Hi! I have just installed asterisk and got it to work in a way, but I can't dial out via asterisk. Capi seems to be configured correctly as the system can answer incoming calls or route them to a sip phone. But as soon as I try to call out, asterisk just dies. The output in that case is: *CLI> -- Executing Dial("SIP/florian-386b", "CAPI/1234567:2437826|60") in

Hi all, I have a VOIP phone with the PA1688 chip; my firmware is V1.42.028. This IP phone is registered in an Asterisk PBX and I've a problem when I dialing internal number. If I dial an internal number, like for example 102, the IP phone takes 35 seconds to send the number to Asterisk; here below the debug output 192.168.0.75: first 192.168.0.75: 1 192.168.0.75: 10 192.168.0.75: 102

Hello Everyone, I am trying to diagnose issue with my IAX2 extension not working. When I have iax2 set debug on all I see is this: *Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ * * Timestamp: 00003ms SCall: 00130 DCall: 00000 [64.229.229.111:64823]* * USERNAME : 100* * REFRESH : 60* * * *Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type:

Hi everyone, For some reason MixMonitor doesn't record when it should; It actually shows the MixMonitor line just fine on the CLI. How can MixMonitor be debugged for things like privilege issues or filename issues? **I had this working at one point and then stopped working. Not sure what I changed. System Info: Asterisk 1.4.21.2 Queuemetrics 1.6.3.0 [queuedial] ; this piece of dialplan is

Hi, i am using the n-way-call dialplan solution found on voip-info. i have added its entry in applicationmap of features.conf file. the problem is......its not working. to activate the n-way call i dial *0 but nothing happens. i have played around with dtmf and codec settings but no success. the extensions and sip configuration is below if you want to have a look. I dont have any clue why its not

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.

I am trying to make a system where a conference user can invite others to join. I am running the 1.2 version of asterisk, so can't use the example on voip-info.org. With use of the X flag on a meetme conference to exit with a single digit, I can get people to join me in a conference with exten => _XXX,1,Dial(${THEIR_EXTEN},,dG(conference-context^${CALLERID}^1)) where the

Apologies if this is a resend, but I've sent this 12 hours ago and still can't see it on the list. Hi, I've just started to setup my phones with Buddy watch. Basically, it all works fine when using the simple example on the wiki: exten => 123,hint, SIP/some_sip_reg exten => 123,1,SIP/some_sip_reg BUT, what I need to do is dynamically decide where the hint checks for buddy

On Tue, Jun 7, 2016 at 1:57 PM, Ryan Taylor <ryta1203 at gmail.com> wrote: > Tim, > > Currently, I have to do multiple things: > > 1) create some setLibcallNames in XXXISelLowering.cpp to generate correct > naming for RTLIBS. > 2) lower ISD down to an RTLIB for some calls (and then do solution 1 on > those to get correct names) These solve a related but different -

Hello, i'm using the parking feature. When the call is parked by A (number *15) , B is correctly parked, by A did not hangup automatically. Here are the dialplan [local] exten => _XXX.,1,Wait(0) exten => _XXX.,n,Dial(SIP/${EXTEN:0}@trunk_sip_2,0,TK) exten => _XXX.,n,Dial(DAHDI/4/${EXTEN:0},0,TK) exten => _XXX.,n,Playback(callbox-thinkro-trunkDefautIndispo) exten =>

Hi everybody, I have a problem using parking for outgoing call. A is an local sip phone. A is using the local extension : [local] exten => _XXX.,1,Wait(0) exten => _XXX.,n,Dial(SIP/${EXTEN:0}@trunk_sip_2,0,TK) exten => _XXX.,n,Dial(DAHDI/4/${EXTEN:0},0,TK) exten => _XXX.,n,Playback(callbox-thinkro-trunkDefautIndispo) exten => _XXX.,n,Hangup() exten => h,1,Hangup() When A is