search for: misdialed

We have a 100 number indial range and every so often get fax calls on our voice numbers (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

Folks, I've been running asterisk successfully using the extensions.conf and voicemail.conf. Now that I've got asterisk happily looking up MySQL tables for the VM configuration, I decided to try out the contributed script /usr/src/asterisk/contrib/scripts/retrieve_extensions_from_mysql.pl I edited the script so that its output goes to a separate extensions_from_mysql.conf file. The

Started a new thread as my problem is somewhat different than the OP. Seems his somewhat different problem doesn't work as advertised either. Eric Wieling wrote: > I don't know what the problem is, but this is what I use and it works on my analog FXO port. > exten => _9NXXNXXXXXX,1,Dial(${PSTN}/w${EXTEN:1}) So, I modified slightly to fit my dialplan: exten =>

I happened to be in the cli tonight as some (208.122.57.58) initiated a simple attack - just trying to make long distance calls from outside context. Although harmless, this went on for several minutes as the idiot just used up my bandwidth with SIP messages. Here's and example: [2011-12-28 22:53:42] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension

...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 => 4...

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

Hi Shahnawaz Have you considered how you are going to address location issue for Mobile users calling 911. You should think of SS7 MAP/TCAP to atleast know their Cell ID Regards Sam > Thanks very much everybody who contributed their thoughts. I would try > to get some DID's so that each physical location can be identified by > 911 call Center. > > Regards > > Shahnawaz

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.

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(&quo...

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 Est...