search for: misdial

...(our fax number isn't in the 100 number range). If you just hang up the sending fax will often try a few times before finally giving up. Our outgoing fax is connected to the PBX (not asterisk), and we can do a blind transfer to that which will print it out, but right now the fax is printing a misdialled fax and it's up to about 3 meters long and still going. I have an asterisk server plumbed into the PBX via an ISDN trunk, so I'm thinking that if I could map an extension to that which would just 'eat' any fax we transfer to it, it would save some paper. Any fax coming in on the...

....conf> exten => #,1,Background(goodbye) ; Notify caller exten => #,2,Hangup() ; Hang up exten => t,1,Hangup() ; Hang up if timeout exten => i,1,Playback(invalid) ; Play "invalid ; extension" if caller ; misdials an extension Basically, I expect asterisk to load the two as separate contexts, and I could swear that it used to. In fact, when I set the verbosity higher, asterisk is definitely still loading them as separate contexts. As of yesterday, though, when I have this format, asterisk won't accep...

...ARG1},360) should work then? >>> >>>> Why in the world would you ever want to do that anyway? >>> >>> >>> >>> Because there is no dial tone detection on the FXO card or module >>> As you have seen on the list, many complaints about misdialing >> >> >> >> Yes, a common problem. Put a w in just like above. This has been >> discussed on the mailing lists many times. >> > " Show application dial" does make reference to using w AFTER the > initial number is called. Others indicate th...

...because extension not found. I thought that it might be worth adding a line to my fail2ban filter, but am looking for a hand with the regex. I have come up with: NOTICE.* .*: Call from '' to extension '.*' rejected because extension not found but I realize that anyone misdialling a valid extension a few times gets cut off. Can someone suggest an improvement? (How could I limit this to 4 or more digits dialled for example?) Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attach...

...pecial extensions. I put this at the very end of the last context in my dialplan and it does show up at the end as expected when you do a show dialplan I've tried matching h t and i to no avail... when voicemail terminates it still always plays my fatfingers catchall that is intended only for misdialed numbers. It's like voicemail is trying to go somewhere that is invalid as it terminates I just do not know what that somewhere is! I must be missing some really simple point here :-) Thanks! Steve ;normal extension & voicemail exten => 4102,1,Dial(SIP/4102,44,tT) exten =>...

I recently purchased a Asterisk Developer's Kit (TDM) and now have it outfitted with 2 FXO modules and 2 FXS modules. I'm not using the X100P modem card that came with the kit. I'm having problems with dialing out on my POTS line. Successful dial out is intermittent. About 50% of the time the call goes through. The other 50% it is dialing the wrong number. ( I can hear the error

...1 call Center. > > Regards > > Shahnawaz > > On 2010-01-29, at 2:41 PM, Kevin P. Fleming wrote: > >> Leif Neland wrote: >> >>> 2: Often callers are answered with an automated message "This is 911, >>> please hold", just to give pranksters/misdiallers a chance to hang up >>> before "disturbing" the operator. Unless 911 records the incoming >>> call >>> right from the start, they will never hear the "im-at" message. And >>> even >>> if they do, they have to know the message is t...

Hi there, I am running a PBX under asterisk 1.6. I have few FXO analogue lines connecting to PSTN. These lines are in a hunt group. I trying to make my extensions to dial 91, but this is a bit scary, I mean if somebody make an emergency call after hours and without completing call is not able to tell his/her location. How can I make 911 call center to know the exact location of my extension. I

Hi All. I've been experimenting with SLA on Asterisk 1.4.13 (patched up to 1.4.14). I am using a SIP channel for my "trunk" line. On the whole things are good, but I have noticed that if I misdial an outgoing call, i.e. I get 404 "Not Found" in the SIP trace, then the trunk line just drops, rather than presenting an error tone or message to the user. ======================== Here is my sip.conf (cleaned to protect the innocent): register => 5000:password:username at vsp5000/5...

Hello everyone. I'm new to Asterisk but got some basic functionality going last night and I'm just giddy to have my own PBX ;-) Sorry if these are silly questions: My Asterisk server has the TDM22B (2 FXO, 2 FXS) interface. I have a very basic PSTN line coming in from the phone company, I tried to get the most no-frills line possible (didn't pay for caller ID, voice mail, etc.). I

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:

...39; in macro 'common' == Spawn extension (a100pub, 314, 2) exited non-zero on 'SIP/gss-cc05ceb8<ZOMBIE>' == Using SIP RTP TOS bits 176 == Using SIP RTP CoS mark 5 Then we see the destination callee attempting to pick up the call and is the output of our routine to catch misdialed/unknown extensions: -- Executing [701 at a100:1] GotoIf("SIP/jasiii-cc05ceb8", "0?:_.,1") in new stack -- Goto (a100,_.,1) -- Executing [_. at a100:1] Answer("SIP/jasiii-cc05ceb8", "0.5") in new stack -- Executing [_. at a100:2] Playback(&q...

Here's a possible bug, or more likely, I'm just missing something. We have a pots card in one of our asterisk boxes. Its a simple asterisk setup with one FXO/FXS card and basic static extensions file, etc. When we dial out over the pots line, 4 out of 5 times, it will work. However, every 4 or 5 times, we get an error back from the provider that says "The number you have dialed.....

Hello, I'm trunking with an ITSP that, when treating an outbound to an unknown destination, either: - send a SIP error code (I can't be more explicit, at the moment), - or cast a pre-recorded audio message using Early Media. At the same time, I'm also trunking with Contact Center solution which doesn't support Early Media. Beside asking my ITSP to treat calls consistently or

i need to know something in the zaptel configuration as it seems i can configure detecting the busy tone and hangup after number of busy tone counts, that was great but the problem is sometimes the pstn line has no dialtone and when i try to make call it continue dialing while not having a dialtone! while it should say "all lines are busy/congested" how can i configure that?? i already

I am trying to tweak my dial plan and I am running into a problem. Sometimes my VoIP out bound calls do not complete on overseas calls(busy or just a hang-up). Is there a way in the dial plan to automatically dial out of my PRI when something like this happens. Either by time limit by a failure event? Any point in the right direction would be great Thanks, CLI output (cleansed to protect the

...may be down at one end or the other. 2. The span or WAN is not connected correctly. 4 04 Send special information tone. Indicates that the called party cannot be reached for reasons that are of a long term nature and that the special information tone should be returned to the calling party. 5 05 Misdialed trunk prefix (national use). Indicates the erroneous inclusion of a trunk prefix in the called party number. 6 06 Channel Unacceptable. Indicates that the channel most recently identified is not acceptable to the sending entity for use in this call. 7 07 Call awarded and being delivered in an E...