similar to: PickUp a call with feature pickup (*8) from a IAX2 channel

Hi, I have a working asterisk 1.4.0 with Mysql Realtime configuration, and today I encountered this error. Now, I have no acces to any information in mysql realtime, so nothing work now !!!!! [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Everything is fine. [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions

Ia had a server overload today because someone did a call forward to their own extension. To do a call forward I write a key called CFWD with the extensi?n number and number to dial . The main script tests if the key/value exists and dials the number stored in the database. What is an easy way to prevent dumb people from creating a loop? -- Telecomunicaciones Abiertas de M?xico S.A. de

I have searched for some time but I have not found an asnwer on how to fix the CDR when a call is transferred. The problem is that if someone dials a cell phone and then transfers the call to another extensi?n the CDR for the cell call stops and there is no way to track that the call was transferred so we can bill correctly. Many people have asked this question but there is no answer, only a

This question is a clone of my stackoverflow question which never got answered (o_O). Therefore I am posting it here. I would really like some inputs if possible. I am currently building some applications which make use of HDF5 files. I have already taken a look at the hdfgroup website with regards to dataspace <https://support.hdfgroup.org/HDF5/doc/H5.user/Dataspaces.html> and I think I

im trying to use monitor() aplication with "b" option, to start the recordigin just once the conversation has actuallly begun. It works fine with a sip extensi?n, but when i use a zap channel, it records all the channel bridging, including the ringing sounds... could you please help me with this issue? ill keep reporting thanks. -------------- next part -------------- An HTML

Greetings to everybody. My question is that it?s impossible to pick up a call from ZAP, IAX or mISDN with my Ext Key of my GrandStream. It always give me a Spawn Message on CLI and a ?603? error on my LCD GrandStream. Exactly from my CLI screen i get this message -- Executing NoOp("SIP/11-096c2ac0", "Probando 1 ") in new stack -- Executing

1. What does this have to do with R? 2. If it concerns computational biology, the Bioconductor Help list may be a better place to post. Cheers, Bert Bert Gunter "The trouble with having an open mind is that people keep coming along and sticking things into it." -- Opus (aka Berkeley Breathed in his "Bloom County" comic strip ) On Tue, Aug 1, 2017 at 2:28 AM, Koustav Pal

Hello, I'm creating a loop to work with vegan, to get a species abundance curve. Here I send the script I've created and also an excel file to prove what it can do. Well, I have a database with 20 years, and each year we have sampled 19 stratum, and in each estratum we have carry out some sumpling. Then, with the script that I've sent I've got to calculate the species abundance

Hi guys! I?m using Asterisk 1.2 with mISDN support. I have problems with Pickup calls with my Grandstream Buttons . I set up on Dial Plan this: Exten => _**XXX,1,Pickup(SIP/{EXTEN:2}) but it doesn?t work if the call comes from mISDN. So, I wanna do something to this: Exten => _**XXX,1,SendDtmf(*8#) because if I introduce *8# into my telephone i can pickup a call from everywhere. BUT the

Mandi! Rowland Penny via samba In chel di` si favelave... > I would undo that, it appears to be wrong. OK, i've undo also i. > I have tested this on a Ubuntu 22.04 computer and it works, so I have > updated the wiki page: > https://wiki.samba.org/index.php/PAM_Offline_Authentication Apparently works as expected: root at dane:~# wbinfo -K gaio Enter gaio's password:

1. This relates to the package *rhdf5* and its implementation of the HDF5 dataspace interface. I am asking for an example of how other people who use this package make use of the HDF5 data space interface exposed by the library. Longer answer: As per my understanding, the dataspace interface exposes data locations within a dataspace, but even while retrieving data from an hdf5 file using methods

Im trying to write a controller spec, but when I run it I get an error like the one below. I found Bug6707 (http://rubyforge.org/pipermail/rspec-devel/2007-March/002147.html ), but I get the error no matter how I run the spec ( whether I use rake or not) Any ideas? I am using 0.8.2 The Error: C:/rails/InstantRails/ruby/lib/ruby/gems/1.8/gems/rspec-0.8.2/lib/spec/expectati ons/sugar.rb:14:in

From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Kevin Larsen Sent: Tuesday, June 2, 2015 4:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Forward loop protection... > Ia had a server overload today because someone did a call forward > to their own extension. To do a

Hello list, this is my first post on this list. I have a server with Asterisk and a Linksys SPA3102 with 3 SIP phones. I have configured the SPA PSTN line as trunk to receive and send calls. I can call outside from SIP phone throw the PSTN line and all is OK, the problem is when I receive a call from the PSTN, on the out caller phone there is a demo playback. I want to redirect the call to a

On 22/05/2023 10:14, Marco Gaiarin via samba wrote: > Mandi! Rowland Penny via samba > In chel di` si favelave... > >> I would undo that, it appears to be wrong. > > OK, i've undo also i. > > >> I have tested this on a Ubuntu 22.04 computer and it works, so I have >> updated the wiki page: >>

Im glad to let you know that finally I invested some time to make work Unicall in Asterisk 1.4, I must say not much testing could be done since I have no hardware available ( cards, servers ), however a friend was able to test it with a couple of calls with success, I need you to test this and report some feedback. The sources are available in: http://moy.ivsol.net/unicall/soft-switch/r1b1/

I have a customer that is complaining that any call coming in from Nextel gives a fast busy. We are running Asterisk 1.4.7.1 with Zaptel 1.4.3 and all the MFC/R2 patches and libraries. All other calls go out and come in, just Nextel seems to have this problem. The phone company technician connected a PBX emulator on the line and that one could receive the calls from Nextel. The E1 is provided

Hello, I am using macros to give the ability to a call-receiver to 'accept' a call. However, any keypress connects the caller. Anyone have any suggestions about how to re-engineer this so that the receiver can deny the call, or press other keys to do other actions, without connecting to the user? Thanks, Jason Wolfe

Hi list: I'm having the next problem, it appear that the application ChanIsAvail is not working on Asterisk 1.4.5 always return me 0 in AVAILSTATUS. I add my dialplan and the output to the cli. THanks. In the example i'm dialing from extension SIP/112 My DialPlan Secction: [macro-callonlyiffree] exten => s,1,ChanIsAvail(${ARG1}|s) exten => s,n,NoOp(${AVAILCHAN}) exten

On 19/05/2023 12:02, Marco Gaiarin via samba wrote: > > I'm trying to enable offline auth in a Ubuntu 22.04 box, following: > > https://wiki.samba.org/index.php/PAM_Offline_Authentication > > using standard ubuntu samba package (4.15.13+dfsg-0ubuntu1.1). > I've enabled workaround 'lock directory = /var/cache/samba'. I would undo that, it appears to be