similar to: "help"

you can not ping that address because ICMP is turned off. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Deon Rodden Sent: Monday, July 26, 2004 2:22 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Broadvoice problems again Attn: James Greetings, C:\>ping 147.135.8.129 Pinging

How does one do translation for calls that come in from other pbx's where the incoming caller ID is an internal extension number on their pbx? Eg. when I get a call from Free-World-Dial the CID shows up as "429102" which is essentially their internal extension number sans any routing prefix. To dial the number back I need to dial the extension with FWD's routing prefix

I'm toying with adding a feature request to provide some sort of gain setting for voicemail when accessed from "certain" interfaces. Maybe something like voicemail=6.0 (db) within a specific channel section of zapata.conf corresponding to a pstn line. Situation: 1. Someone calls into asterisk and leaves a voicemail. The sound is recorded at some volume well below 0 db, and is

Adam Jackson (1): twm 1.0.4 Alan Coopersmith (2): Explicitly require libSM & libICE since twm calls them directly Change twm_CFLAGS to AM_CFLAGS to make automake-1.10 happier James Cloos (3): Rename .cvsignore to .gitignore Add *~ to .gitignore to skip patch/emacs droppings Replace static ChangeLog with dist-hook to generate from git log Wolfgang

Hello, I have a problem with connecting a Digium X100P card to a Brazilian analog line. Can somebody help me out with this problem? My /etc/zaptel.conf is loadzone=br defaultzone=br fxsks=1 My /etc/asterisk/indications.conf [general] country=br [br] description = Brazil ringcadance = 1000,4000 dial = 425 busy = 425/250,0/250 ring = 425/1000,0/4000 congestion =

On Tue, 15 Jan 2019, Hans Wennborg wrote: > On Sat, Jan 12, 2019 at 12:49 AM Jordan Rupprecht <rupprecht at google.com> wrote: >> >> >> >> On Fri, Jan 11, 2019 at 6:26 AM Martin Storsjö <martin at martin.st> wrote: >>> >>> Hi, >>> >>> The COFF support in llvm-objcopy is in a pretty half-finished state at the >>>

Jordan, I have sent the patch via Phabricator: https://reviews.llvm.org/D69188 Let me know if I got it right. -- Tobias On Thu, Oct 17, 2019 at 7:12 PM Jordan Rupprecht <rupprecht at google.com> wrote: > > Tobias, > I don't have much experience with ARM, but from your report and Peter's explanation of why LLD does it, I agree we should be consistent with LLD and keep the

How do you setup the timing in Meetme conference? I have a x100p and tdm4x card. When I dialing to my conference I get a request to schedule in the past error message. thanks -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Saturday, April 10, 2004 10:48 AM To:

Hi everybody, Is the only way to use asterisk _not_ as root to change the permission of all the directories where asterisk need to create a file? ("/var/run/", "/var/log/asterisk/messages") any help will be appreciated, Cyprien

Hi, I compiled asterisk and chan_h323 on an Opteron in 64 bit mode. In the h323's Makefile I replaced in line 24 CFLAGS += -march=$(shell uname -m) by CFLAGS += -march=k8 and also tried CFLAGS += -m64 -march=k8 Both solutions do compile, but when starting asterisk, a load error occurs: undefined symbol: _ZN14H323Connection24OnUserInputInlineRFC2833ER15OpalRFC2833Infoi When I grep

Hi All, Last week I started hearing a huge amount of local end echo on incomming calls. I am using a Sipura SPA-3000 as my FXO connected to an SBC POTS line. Echo cancellation is enabled in the SPA firmware. As a test I switched to a Digium X100 card the still lives in my server but the echo was about the same. I have both Polycom IP600 and SNOM 200 phone, which both hear the echo. I'm

I believe a good amount of them (if not most of them) have already been ported! Off the top of my head, I remember that asan, tsan, msan, hwasan, the kernel santizers, and sancov have been ported. I don't think ubsan has been ported yet though. You can also check if other passes you need run under the new PM by checking PassRegistry.def. On Fri, Aug 2, 2019, 14:20 Jordan Rupprecht

http://yro.slashdot.org/article.pl?sid=04/08/04/2212251&tid=158&tid=95&tid=103 Probably some of you already saw this. Now, beyond discussions regarding the legitimacy of such a ruling (whether they have the legal, moral or whatever right to enforce it), there's the technical aspect. Suppose i provide VoIP services using Asterisk, and i fall under the incidence of the FCC ruling

With the 1U Apple G5 server at a good reliability/storage/pricepoint level... got to thinking about compiling Asterisk on OS X.. at least for SIP phone call switching, voicemail, etc. Has anybody attempted this? Email me off list if this is too dev-heavy for the user list. Thanks, Ted W -----Original Message----- From: asterisk-users-request@lists.digium.com

Extreme N00b, I am getting the error message "a target does not exist" when running the make install inside the zap directory, probably pretty common, possibly a package I didn't install, just need some insight on it. The same occurs with the libpri and asterisk. -----Original Message----- From: asterisk-users-bounces@lists.digium.com

-----Original Message----- <snip> Is this possible with asterisk? Anyone have a sample dialplan? -other than the problem outlined below I would try something like S,1,wait(20) S,2,voicemail(uwhatever) S,3,hangup That should ignore the call for 20 seconds and then leave a message in the unavailable greeting for 'whatever' then hangup That leaves another problem -

I had my asterisk configuration working very well with broadvoice, but it stopped working this afternoon. I plugged the Cisco 7960 phone I used for my origional signup (just a few days before they offered generic BYOD) and it works fine. I did notice it seems to do all of its comunication through proxy.broadvoice.com (I used tcpdump). I have never contacted broadvoice about using asterisk

It's actually a bit weirder than you might think. The CommandLine parser will happily eat as many dashes as you care to write, e.g., `----sections` is the same as `-sections`. On Tue, Apr 16, 2019 at 2:11 AM James Henderson via llvm-dev < llvm-dev at lists.llvm.org> wrote: > As I think I said elsewhere, I find it weird that LLVM tools accept long > arguments with a single dash,

----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Saturday, September 13, 2003 7:55 PM Subject: Asterisk-Users digest, Vol 1 #1279 - 16 msgs > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

> Well that didn't work....I now get this error > > > Jan 12 16:56:21 NOTICE[4989]: app_dial.c:746 dial_exec: Unable to > create > channel of type 'SIP' > == Everyone is busy/congested at this time > -- Executing VoiceMail("IAX2/iaxfwd@65.39.205.121:4569/5", "b") in > new > stackJan 12 16:56:21 WARNING[4989]: