search for: callednumb

.... I want to get called number in features.conf. I am defining a feature in features.conf and that feature got executed on pressing a particular DTMF key sequence. As I want to execute my own application on pressing that key which will use called number. testfeature => 3,peer,AGI,StoreNumber|<CalledNumber> Here I want to use called number in place of <CalledNumber> tag. When I use any variable *${DIALEDPEERNUMBER} *then it does not resolve the variable in features.conf. if i use following then it does not work. testfeature => 3,peer,AGI,StoreNumber|*${DIALEDPEERNUMBER} *StoreNumber...

...der [mynumber] type=peer username=mynumber fromuser=mynumber secret=mypass fromdomain=mysipprovider host=mysipprovider insecure=very --- Any help would be deeply appreciated! Best regards, Atle Soma - soma@netpower.no NetPower Int. AS ---- # U myhost:5060 -> mysipprovider:5060 INVITE sip:callednumber@mysipprovider SIP/2.0..Via: SIP/2.0/UDP 212.33.133.1 95:5060;branch=z9hG4bK40e0db6c..From: "4721677891" <sip:4721677891@mysipprovider >;tag=as11f66db2..To: <sip:callednumber@mysipprovider>..Contact: <si p:4721677891@myhost>..Call-ID: 5ae4b68378e5edf8081a79253ddf8...

I have been discussing an asterisk solution with a company that has a custom written dialogic based solution. The issue is that their dialogic solution can read callerid from incoming calls, even if the callerid is blocked. I have read before that Asterisk can do this, and they want me to make sure that their new system will be able to do this. A quick poke around inside the zaptel source code

...1); } else { debug(" Call Loop Anaylsis for ".$v{callednum}." = NO LOOP"); $AGI->exec('Set',"__".$v{cfnum}."=".$v{callednum}) } return; } $AGI->exec('Set',"__".$callednumber."=1") if ($calltype !~/^Local/); if (callfwd_loop_check($callednumber,$callfwdtonum)) { return; } $AGI->exec('Dial',"Local/+".$callfwdtonum."\@localcontext\&SIP/+".$callfwdtonum."\@sipproxy|180"); I hope this all makes sense! :) Tha...

Hi, I would like * to treat a SIP extension as a normal extension, when it comes to the busy functionality. In other words, if someone tries to call the SIP phone and there is already an ongoing conversation, the new caller should get a busy message/tone Is there any parameter that I can set? Is this something that should be configured at my softphone? Best, PHM

...bin strategy. If all the gsm lines are busy, the call must be sent to the pri lines with a linear strategy. here is the dialplan: exten => gsm,ChanIsAvail(EXTRA/r2&DAHDI/g1) same => n,GotoIf($["${AVAILORIGCHAN}" = ""]?unavail,1) same => n,Dial(${AVAILORIGCHAN}/${CALLEDNUMBER}) The problem is that the ChanIsAvail function is breaking the round robin strategy. When this function is executed i can see in the console this message: Hungup 'EXTRA/3-1' and when the Dial function is executed only the EXTRA/1 channel is selected, never the EXTRA/3 one. that happen...

Hi everyone, Using Asterisk 1.6x here with a TDM PRI. I have to run a campaign for about 5000 numbers and then put the call to agents right away and pull up the CRM based on the number dialed. So, I am going to be doing some PHP+Ajax work. I am familiar with spool files but I don't like the fact that I can't read the status of the call in real-time. However, I know that it's the

...t - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Blocked Callerid > > > Do they have an 800 number? If so perhaps their 800 > number provider is doing it via DTMF. Search around on the > internet, I believe the standard format for the DTMF is > *CALLERID*CALLEDNUMBER* (or perhaps reversed). > > > On 2/1/06, pdhales@optusnet.com.au <pdhales@optusnet.com.au> > wrote: > > I have been discussing an asterisk solution > with a company that has a custom written dialogic based solution. > > The issue is that their dia...