search for: callforward

Hello, i am trying to enable call forwarding on asterisk 1.6 with asterisk-gui If i set my stdexten as follows (with the lines i marked) everything seems like working. But if i make any change on asterisk-gui and apply it.. it recreates the macro-stdexten and deletes my configuration regarding to it. So where should i add my call-forward configuration??? Where am i making a mistake??

Hi All, I'm using asterisk 1.2.15 and call forwarding doesnt work for me. from my extensions.conf: ; Unconditional Call Forward exten => _*21*X.,1,NoCDR exten => _*21*X.,2,Set(DB(CFIM/${CALLERID(NUM)})=${EXTEN:4}) exten => _*21*X.,3,Playback(vm-saved) exten => _*21*X.,4,Hangup exten => #21#,1,NoCDR exten => #21#,2,DBdel(CFIM/${CALLERID(NUM)}) exten =>

Hi everyone, I am facing lots for problem with CDRs in 1.6 and above versions,its shows wrong records when I do transfer(caller side and calee side),callforward,call parking.Is the present CDRs in 1.6 is enough for Complete billing.?What I need to do to make it proper.Please help me on this. Thanks Nikhil

...se asterisk originates a new call to his cell and then bridges the two. so he told me, a partner of his, at his office does the same thing, and when he does it, the callerID shows up as coming from the initial caller, not from his office. so here's the schematic: customer -> our office ---callforward--> cellphone so should I call AT&T and ask them to unlock our callerID so I can set the outgoing callerID to the customer's number in my dialplan? or is there some other way to handle this? I appreciate any input, Thanks! -- Gerard Saraber Network Admin. Rarcoa, Inc (630) 654-2580 x...

...that they have to use FXO ports, but it is realy cheaper, so customers don't really care. In any case, there are 2 issues that I can't get solved, and they are not interested in helping. 1. When they tell you that they are putting all your lines in a hunt, it realy is not a hunt but just CallForwarding No Answer/Busy, what that means is that if I have asterisk setup to first ring a phone for 5 times and then go to an IVR and answer the phohe, it will go to the next line and stop ringing the first line, and therefore never end up in Voicemail or my IVR. 2. No CPC, hung channles, blank voicemail...

I have put together this call forwarding and dnd config: I'm sure it can be dome with macro's but I couldn't figure that out... anyone care to input. 74 Turns DND on my phone will not ring, drops caller to voicemail... 73 Turns DND off 72+ext forward your extension to another extension and voicemail is left at the forwarded extension. 71 turns off call forwarding. ; dnd Could

...I don't want to dial on the first channel as it's my main number and all calls are received by that channel if the line isn't busy. I don't have to manage it for incoming calls (it can't be done anyway) as it's done not on my side but on the side of my provider (with the callforward on busy thing) but for outgoing calls the asterisk server must manage it. -put all of the zap channels in a group and specify that you want the outgoing calls to choose from the back of the group. Look on the wiki for the g and G options to the Dial()application. <snip> So if you think the...

I did not really spend too much time on looking at call forwarding and wonder if someone could help me. It seems that for setting call forwarding on the Polycom phone itself, only "forward all calls" will work. The other call forward function like "forward if no-answer for n rings" or "forward if busy" does not work at all on the phone. If that is the case, it

...To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Is existing CDR in Asterisk is enough for complete billing Hi everyone, I am facing lots for problem with CDRs in 1.6 and above versions,its shows wrong records when I do transfer(caller side and calee side),callforward,call parking.Is the present CDRs in 1.6 is enough for Complete billing.?What I need to do to make it proper.Please help me on this. Thanks Nikhil -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to A...

Hi I am using asterisk 1.6.1.1 ,and trying to do cdr billing.But the problem is when I do transfers,callforward,callparking cdr records are not proper.Below is example 'A' called pstn number and 'A' transfer the call to 'B',In this case cdr is genrating as 1 . caller - A ,Callee - PSTN ,uniqueid=923479129.234 2. Caller -A , Callee -C uniqueid = 939389037.343 Ther...

Has anyone implemented "hot seating" in any neat way? This where people can log in to any phone in the company and have their calls/voicemail come to that particular handset.....

...I don't want to dial on the first channel as it's my main number and all calls are received by that channel if the line isn't busy. I don't have to manage it for incoming calls (it can't be done anyway) as it's done not on my side but on the side of my provider (with the callforward on busy thing) but for outgoing calls the asterisk server must manage it. The only example I saw for outgoing calls were using only one FXO card (X100P I think) so I'm sure it's not that hard but I have no clue how to type it in asterisk. That brings me another question. I'm curren...

...; take a look to the compile howto. Park stuff is not compiled by default speeddial = ; you can add an empty speedial if you want an empty button (7960, 7970, 7920) speeddial = *97,voicemail, cfwdall = off ; activate the callforward stuff and softkeys cfwdbusy = off dtmfmode = inband ; inband or outofband. outofband is the native cisco dtmf tone play. ; Some phone model does not play dtmf tones while connected (bug?), so the default is inband imagever...

...e with 4 agents that login/logout dinamically. If there are no agents, queue timesout and gets derived to another queue that somebody answers as last resort or waits there. If there is at least one agent logged in, but it is busy, dialparties.agi detects that that extension has no callwaiting, no callforward, no voicemail, and hangs up the call inmediately with a "nobody is available to take your call right now" message, making the queue useless. My PSTN connection is an AS5300 in SIP, my extensions are analog phones connected to an Audiocodes MP108-FXS with SIP. This is the output from C...

...erNameForProxy - PasswordForProxy - UserNameForRegistrar - PasswordForRegistrar - # Programmable Keys. Key functionality must go before key values. ProgrammableKey1 OneTouchDial ProgrammableKey2 OneTouchDial ProgrammableKey3 OneTouchDial ProgrammableKey4 CallForward ProgrammableKey5 TwoTouchDial ProgrammableKey6 DoNotDisturb ProgrammableKey7 VMA ProgrammableKey8 Mute # One and Two-touch keys. Must go after Programmable keys functionality definitions. # Refer to Programmable and Fixed Function Keys for usage guide # OneTouchKeyX value i...

...waiting) exten => *70,5,Playback(activated) exten => *70,6,Macro(hangupcall) exten => *71,1,Answer exten => *71,2,Wait(1) exten => *71,3,DBdel(CW/${CALLERIDNUM}) exten => *71,4,Playback(callwaiting) exten => *71,5,Playback(de-activated) exten => *71,6,Macro(hangupcall) [app-callforward] ; dialed call forward app - forwards calling extension exten => _*72.,1,DBput(CF/${CALLERIDNUM}=${EXTEN:3}) exten => _*72.,2,Answer exten => _*72.,3,Wait(1) exten => _*72.,4,Playback(loligo/call-fwd-unconditional) exten => _*72.,5,Playback(loligo/for) exten => _*72.,6,Playback(lo...

...waiting) exten => *70,5,Playback(activated) exten => *70,6,Macro(hangupcall) exten => *71,1,Answer exten => *71,2,Wait(1) exten => *71,3,DBdel(CW/${CALLERIDNUM}) exten => *71,4,Playback(callwaiting) exten => *71,5,Playback(de-activated) exten => *71,6,Macro(hangupcall) [app-callforward] ; dialed call forward app - forwards calling extension exten => _*72.,1,DBput(CF/${CALLERIDNUM}=${EXTEN:3}) exten => _*72.,2,Answer exten => _*72.,3,Wait(1) exten => _*72.,4,Playback(loligo/call-fwd-unconditional) exten => _*72.,5,Playback(loligo/for) exten => _*72.,6,Playback(lo...

I used to get bad echo and also voice breaking up during a conversation, so I upgraded the machine (from a PII 300 to a Athlon 1.8GHz) and have resolved the voice breaking up issues but still seem to have a fairly bad echo on *both* sides of the conversation. Currently I am only using the system for outbound calls, so I am always calling from the digium USB port phone and out to the telco using a

How many times do you ask for help here before getting a respone? Every single thing I do No matter what I get busy extensions. I am willing to pay someone to help here. Anybody got a clue? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/d0d1326c/attachment.htm

Have asterisk@home 1.2.1 The server is on an internal network eg 10.10.10.10 It is NAT'd 1:1 via Checkpoint firewall to external public IP eg 50.50.50.50 The remote IAX2 phone (ATCOM320) is configured to call 50.50.50.50 on extension 1055. Outbound calls to 1055 work perfectly. Inbound calls from 1055 get picked up as if it were an external call (see below) and goes straight to the ring