similar to: MACRO-INCOMING-CALL-TO-EXTENSION

hi ppl :D my configuration is as follows, i run (let's call it machine 2) debian etch 4.0 and asterisk 1.2, i use voiceone (www.voiceone.it) as an interface to manage asterisk, I have a grandstream/handytone 486 as a sip device, no PSTN line or anything like that all SIP only. I have a machine (machine 1), which functions as my router and machine 2 and sip device are behind it, grandstream box

Yo all, As there has been some intrest, here's my updated version: I post it to "-dev" as well as "-users", as it may be of intrest to both. Inspired by the example in the tips & tricks-section of "http://www.junghanns.net/asterisk/", I built a more elaborate set of features. Currently, my implementation supports call- forward unconditional, on no answer

Hi The only way to have CD service is using: DAHDISendCallreroutingFacility(<destination-5551212>, <original-my-number>, cfu|cfb|cfnr|unknown) or it is possible also with Dial() command before answering the call also ? Regards Babak -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have just installed a Sirrux PC140 card for the first time. Managed to build the drivers and get * to load them on FC4, but it does not work. It seems that layer1 in the ISDN is not even activated. The same ISDN lines connected to a Samsung DCS works so it is not the lines. I am including my sirrix.conf and the output of some of the * Srx commands below. Any pointers would be

hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but before i was using asterisk 1.4 and had the same problem, it concerns an italian voip/sip provider called eutelia/skypho, my problem is the following one: when i start my pbx my skypho account is working fine, meaning that e.g. incoming calls are shown in the asterisk CLI and caller and callee can hear each other when

What is CFU and CFNR? > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Michel Koenen > Sent: Tuesday, August 30, 2005 1:46 AM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] How to use * and # as part of > numberindialcommand > > > From: "Damon

Dear All, I can not transfer call in progress. What's wrong with my macro? I think tT flags is enough right? [macro-stdexten] exten => s,1,Set(temp=${DB(CFU/${ARG1})}) ; Get CFU key exten => s,2,Set(DNDStatus=${DB(DND/${ARG1})}) ; Get DND key exten => s,3,GotoIf($["${temp}" = ""]?5) ; If not existing, goto priority 5 exten =>

Hi! I'm here again with echo canceller problem... :-( I think I've done everything to enable echo canceller feature, but it still doesn't work... Can anybody tell me if there is some error or something missing in this configuration please? I'm using Eicon Diva Server 4Bri. http://www.eicon.com/worldwide/products/MediaGateways/disv4bri.htm?techspec=1&regID=4999 Card

Dear All, I have a problem with call transfer. When I dial a number and then I want to transfer current call to an extension, I'm getting disconnected. Transfering incoming call works fine. I'm using macro for dialing. extensions.conf: [from-internal] ignorepat => 9 exten => 200,1,Macro(stdexten,200,SIP/dzalewski) [macro-stdexten] exten => s,1,Set(temp=${DB(CFU/${ARG1})})

Hi I installed Asterisk 1.8.7 with CD ISO(Elastix 2.2) I want to use DAHDISendCallreroutingFacility Application on a PRI link(LIBPRI Already installed). according to https://wiki.asterisk.org/wiki/display/AST/New+in+1.8 Asterisk 1.8 include this application but I cannot see it with "core show applications" Do I need to install mISDN or other modules for using that ? Regards M.Shirazi

Hello List Next question where google did not spit out an unsable answer. When redirecting a call with Transfer, I would like to correctly indicate the reason. I did try this: exten => XX,1,NoOp(Call to ${EXTEN} from ${CALLERID(all)}) exten => XX,n,Dial(SIP/ZZ) exten => XX,n,set(REDIRECTING(reason)=cfb) exten => XX,n,Transfer(SIP/YY) I did try with 'reason'

Hello List, I am having a strange issue with a trixbox system we installed for a client, and I would appreciate any help on this one. The issue is that occasionally when they go to answer an inbound call from the Sangoma A200 - there is no one there, and they are presented with dial tone. The calling party is hung up. A bit of background: The client actually has two systems install (one at

Yo, Inspired by the example in the tips & tricks-section of "http://www.junghanns.net/asterisk/", I built a more elaborate call divert-feature. This one validates if the extension a call-forward is to be set to is actually valid for the current context and additionally saves this context into the DB and always uses it to originate the divert from, as you can't expect the

Yes, you need the latest ToT version of llvm or you run -loop-vectorize -earlycse -instcombine -simplifycfg The bitcast essentially is a noop to satisfy the type system. This is how your example looks like for me: vector.body: ; preds = %vector.body, %vector.ph %index = phi i64 [ 0, %vector.ph ], [ %index.next, %vector.body ] %.lhs = shl i64 %6, 2

Problem: *Call* *waiting* comes in, I press flash to answer it, and the first caller gets disconnected after 3 seconds. This is all ZAP - no VOIP. System: Analog stations and trunks running on a pair of TDM400's. It does NOT have * call* *waiting* from the phone company, and I have enabled it in all my conf files. The trunks are set to FXSKS and the stations are FXOKS. I am not using *call*

The loop vectorizer relies on cleanup passes to be run after it: from Transforms/IPO/PassManagerBuilder.cpp: // Add the various vectorization passes and relevant cleanup passes for // them since we are no longer in the middle of the main scalar pipeline. MPM.add(createLoopVectorizePass(DisableUnrollLoops)); MPM.add(createInstructionCombiningPass());

Add 5-argument handling for paravirt ops patching of PAE functions. Signed-off-by: Zachary Amsden <zach@vmware.com> diff -r dbe11208916f include/asm-i386/paravirt.h --- a/include/asm-i386/paravirt.h Thu Apr 19 11:40:55 2007 -0700 +++ b/include/asm-i386/paravirt.h Thu Apr 19 12:04:16 2007 -0700 @@ -308,10 +308,9 @@ unsigned paravirt_patch_insns(void *site * return value handling from

Add 5-argument handling for paravirt ops patching of PAE functions. Signed-off-by: Zachary Amsden <zach@vmware.com> diff -r dbe11208916f include/asm-i386/paravirt.h --- a/include/asm-i386/paravirt.h Thu Apr 19 11:40:55 2007 -0700 +++ b/include/asm-i386/paravirt.h Thu Apr 19 12:04:16 2007 -0700 @@ -308,10 +308,9 @@ unsigned paravirt_patch_insns(void *site * return value handling from

The instcombine pass cleans up a lot. Any idea why there are still shufflevector, insertelement, *and* bitcast (!!) etc. instructions left? The original loop is so clean, a textbook example I'd say. There is no need to shuffle anything.At least I don't see it. Frank vector.ph: ; preds = %L5 %broadcast.splatinsert1 = insertelement <4 x

I've been using the following macro to ring SIP and IAX devices for a few seconds, and then add on a cell phone if there is no answer on the SIP or IAX device. Periodic problems began a few versions ago and now the problem happens every time with 1.2.9 and 1.2.9.1. The problem is that when a call from the PRI falls through to voicemail, the call is dropped before the voicemail greeting