search for: misdialled

Displaying 17 results from an estimated 17 matches for "misdialled".

Did you mean: misdialed
2006 Nov 02
2
fax eater
...(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 10...
2005 Feb 17
1
Having trouble with extensions in an include file and retrieve_extensions_from_mysql.pl
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
2005 Aug 23
1
Wait before dialing ( was Pause during dialing to enter another number)
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 =>
2011 Dec 29
2
Interesting attack tonight & fail2ban them
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
2005 Aug 18
1
Newbie Trying to make 'catch all extension' but is catching voicemail exit!
Greetings, Running CVS HEAD about 3 weeks old, I have been beating my head trying to get this to work properly.. Or at least figure out what's going on. Maybe I have done things wrong... I have created a 'catch all' extension at the end of our last context where all phones & voicemail extension exist. This catch all is included in all and works quite nicely except when voicemail
2004 Jul 27
1
Dial out problems with Digium TDM400P card.
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
2010 Jan 31
0
asterisk-users Digest, Vol 66, Issue 75
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
2010 Jan 28
2
911, location
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
2007 Nov 29
1
SLA: Handling of errors in outgoing call
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.
2005 Sep 21
3
Caller ID and Call Parking on an analog PSTN line?
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
2009 Aug 25
1
followme app
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:
2009 Jul 01
2
Multi-tenant parking broken in 1.6.1.1?
Hello, all. With the assistance of very helpful folks, our brand new multi-tenant setup seems to be working smoothly from start to finish with just a bump or two. The biggest is parking. Now that we got most kinks worked out, I'm a little more comfortable in trying to resolve this. There seem to be two problems: 1. Parking assigns parking spaces from the default group no matter
2007 Jun 04
2
FX Dialing Odd
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.....
2016 Feb 03
4
How to deal with error messages passed as Early Media
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
2005 Aug 28
5
Detect Dialtone
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
2006 Jan 26
6
Fail over to Pri on VoIP connection failure
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
2004 Apr 10
1
Archive Post ISDN Q.931 disconnect cause codes
Keywords T1 Q.931 isdn disconnect cause codes itu standard libpri Dont know if anyone wondered what q.931 cause codes are but i wishwe could get these back into the dial plan as a var Standard Q931 Codes Decimal Value Hexadecimal Value Definition 1 01 Unallocated (unassigned) number. This number is not in the routing table or it has no path across the ISDN cloud (network). 1. Check routing