similar to: Unavailable/Withheld identification

Hi, I'm trying to set up an Asterisk pbx based on a Fedora Core 1 Linux box (customized kernel version 2.4.24). I want calls from my SIP soft-phones to simply be dumped onto the PSTN line via a BRI (EuroISDN). I have a cheap HFC-S based PCI ISDN card connected to the NT1+ interface, so I need zaphfc. I've read everything I've found at www.voip-info.org, then I've downloaded the

Has anyone managed to get Asterisk to work with any of the Alchemy or Cybergear ISDN switches? Cheers Matthew

On a BT line, how do I determine whether the number on an incoming call has been deliberately withheld (by dialling 141) or is merely unavailable (e.g. because it originated from overseas or passed through some ancient switching equipment) ? In the first case, I want the caller to be played a message to the effect that we are not at home to anonymous cowards but if their business is

HI I want to know Why large enterprises (F500) are not shifting to asterisk as it is going to save them a lot of investment. Are there some problems with asterisk ??? Varun Gupta India __________________________________ Do you Yahoo!? Yahoo! Mail - 50x more storage than other providers! http://promotions.yahoo.com/new_mail

Heres a weird one... Call comes in on mISDN channel. Little bit of dialcode (in a macro) looks up the number in the astdb and puts an name to it. No real magic there, and it works well. Same macro also has parameter passed in to put a prefix on the name - this is set in the DDI handling and is dependent on the number called and allows phone users to see which number was called (company

Hi, now I've got caller id working on my BT line in the UK, I'd like to play a different message to those pesky sort who with hold their outgoing number. How can I do this in my extensions.conf for my [incoming-analog] context? I realise some people may call who are unable to change the way that their system withholds the outbound number, so I'll give them chance to leave a voice

thanks a lot for the reply. i thought of that and i did try to send *exten => _9X.,1,Dial(Dongle/dongle800/#31#${EXTEN:1},120,KT)exten => _9X.,n,Hangup(${HANGUPCAUSE})* but the provider replies back that it is a wrong number. Then i inserted the sim to an ordinary mobile phone and dialed #31# and the number, then the call progressed fine and it restricted the number. What am i doing wrong

Hi All, I'd appreciate some help on how to implement "call stealing". That is, where you dial a code to redirect any call on the system to your handset. I'm getting rid of my BRI service and I'm trying to replace the functionality of my existing ISDN2e PBX (Cybergear Gold) with VOIP and Asterisk. On my ISDN PBX, the short-code *46 does this. For example, if I take a call on

Hi. I am running asterisk 11 and i have usb 3g dongles to make my gsm calls with the following config in extensions.conf exten => _9X.,1,Dial(Dongle/dongle800/${EXTEN:1},120,KT) exten => _9X.,n,Hangup(${HANGUPCAUSE}) By dialing 9 it opens the dongle to make a call. I would like to restrict my caller id. so when i place a call from this dongle, it will send on the other end *blocked number*

I am using LLVM via the Adobe Alchemy project. I'm trying to upgrade LLVM within Alchemy (its v2.1 of opt is crashing), but am running into an issue with llvm-ld. Alchemy is passing -O5 to llvm-ld, but it seems that -O params are no longer accepted by llvm-ld v2.4+ ("llvm-ld: Unknown command line argument '-O5'. Try: 'llvm-ld --help'"). Are -O optimizations

so any ideas, please? On Tue, Apr 10, 2018 at 1:46 PM, Atux Atux <atuxnull at gmail.com> wrote: > after adding the ww: > root at Pbx: /etc/asterisk $ asterisk -rvvv > Asterisk 11.25.3, Copyright (C) 1999 - 2013 D == Using SIP RTP TOS bits > 184 > == Using SIP RTP CoS mark 5 -- Executing > [9211123456 at AllCalls:1] Goto("SIP/500-00000003",

>>> > exten => _9X.,1,Dial(Dongle/dongle800/#31#${EXTEN:1},120,KT) My suggestion would be to add a pause or two before dialing the phone number exten => _9X.,1,Dial(Dongle/dongle800/#31#ww${EXTEN:1},120,KT) D(digits): After the called party answers, send digits as a DTMF stream, then connect the call to the originating channel (you can also use 'w' to produce .5 second

I just bought two new Linksys WRT54G routers. Sveasoft has loaded Linux on this router and included a bunch of Linux tools, one of which is Bandwidth Management. The QoS aspect of this is supposed to be much more granular than the previous solution (Wonder Shaper). I have not been able to find any suggestions for how to impliment QoS (Bandwidth Management) using the web interface of Alchemy.

I'm also running into an issue with llc. Alchemy is trying to use -march avm2 and other avm2 options, but: llc: for the -march option: : Cannot find option named 'avm2'! llc: Unknown command line argument '-avm2-use-memuser'. Try: 'llc --help' llc: Unknown command line argument '-avm2-package-name=cmodule'. Try: 'llc --help' Was avm2 support dropped?

Eeeeck! Asterisk is inserting random digits in dialed numbers. So far I've seen it insert a 2 after the STD (area) code and insert an extra 6 or 7 in the STD code. It's pretty repeatable although the inserted number changes. My Config is: Asterisk 1.4.5, Zaptel 1.4.3, Digium TE205P (rev 02). There's an ISDN PBX on the second span and a BRI euroisdn on the first. Calls from the

I am going to try and use a passive AVM fritz BRI card for my * setup.. Here is the thing.. I need to order my BRI from BT.. The service that looks to be the one to use is what they call ISDN 2e becasue this has the option to setup hunt groups across multiple ISDN2e lines so I could add another line later to get 4 channels.. According to the BT website in order to use the hunt grouping across

Oct 21 10:04:05 alchemy dovecot: imap-login: Login: user=<paulproteus>, method=PLAIN, rip=127.0.0.1, lip=127.0.0.1, TLS Oct 21 10:05:44 alchemy dovecot: IMAP(paulproteus): file message-decoder.c: line 289 (message_decode_body): assertion failed: (pos < input->size) Oct 21 10:05:44 alchemy dovecot: IMAP(paulproteus): Raw backtrace: imap [0x80c8e10] -> imap [0x80c8d1c] ->

Jerry, Jeremy, et. al. Last night's upgrade from cifs-mount-3.2.0-24.1.121 to cifs-mount-3.2.0-24.1.122 on a openSuSE 10.3 x86_64 machine broke mount -t cifs. Now attempting to mount shares results in the following: 11:10 alchemy/home/david/bin] # mount.cifs -v //nirvana/david /mnt/nirvana-david -o username=david,uid=1000,password=notmypass parsing options:

Hi all, I tried to set the calleridname of an incoming call to get different incoming labels displayed for different incoming numbers. This does work for hidden number-calls so I can set the displayed CIDName on my cisco7960 from "CID withheld" to "abc CID withheld" If the incoming CID isn't hidden it works to use SetCallerID but not to change only the CIDName with

All, If the post "Samba 3.6 -> 4.1 change that prevent sharing / - mount error(13)" makes it to the list, I apologize, but it has been over a day and I have no indication that the post ever made it to the list. The issue is that in versions prior to 4.x, I was able to create a root share in samba, but now, something prevents it from being mounted. The relevant smb.conf is: