similar to: MOH - White noise, static

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I have a medium sized business that is interested in implementing * as their PBX system. They currently have a Panasonic system with Panasonic handsets that they are going to replace Asterisk with, as the current system is maxed out, and they don't even have voicemail capabilities. I have been considering using an Adtran Atlas 550 with FXO and

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 OK, so I have a local extension that a phone can call to take it to voicemail. I don't want it to exit out to a fast busy tone, as I would rather it allow the user to simply continue on and call a new number (without having to physically release the line first). The [intern] context is where everything goes by default (sip.conf for example has

Hello all, I would like to document some things I am doing with asterisk, but would prefer to do this from a web interface. I am unfamiliar with any software that allows you to create online documentation from a web interface. Ideally I will be able to create documentation online from a browser, which then when saved, is immediately ready to be read online. Perhaps I can setup different authors

> I'm still having my (apparantly) timing problem, but I'll do some more > testing and make a separate thread for that. I'm generating an > outbound call through Asterisk. The inbound audio is good, but the > outbound audio is sometimes staticy. This seems to happen only at the > start of the call. These are short test calls, just playing a "weasels > ate our

Hi, Is it possible to use asterisk with Free World Dialup (FWD) ? Did someone manage to make it work? how? Best, -P -- __________________________________________________________ Sign-up for your own personalized E-mail at Mail.com http://www.mail.com/?sr=signup CareerBuilder.com has over 400,000 jobs. Be smarter about your job search http://corp.mail.com/careers

Hello, I recently purchased a Wildcard X100P FXO Card primarily as a learning tool and to provide myself with voicemail and a few other perks that are better than an answering machine. I followed the setup in the FAQ, and it works, just not well. The sound is very choppy and staticy whenever a call is placed. Sometimes it clears up, and sometimes it degrades very badly. I also recently

Hi All, I have just compiled the newest version of mpg123 on a RedHat 9.0 system (mpg321 has not been installed) and I am using the newest CVS version of asterisk. Whenever I place any mp3 files in the /var/lib/asterisk/mohmp3/ directory, Asterisk crashes a horrible fiery death. If mp3s exist in that directory, then I can't even start Asterisk. If I start it without files then copy

> output of zttest: > Opened pseudo zap interface, measuring accuracy... > 99.938965% 99.951172% 99.938965% 99.963379% 99.963379% 99.951172% > 99.951172% Ah. Don't be scared. This is actually -correct- when using RTC. Let me demonstrate: [root@dhcp5 zaptel]# ./zttest -v Opened pseudo zap interface, measuring accuracy... 8192 samples in 8195 sample intervals 99.963379% 8192

I had posted earlier asking about a Snom200 communicating with a C7960 and lots of noise in one direction. Turned out the problem was created by me removing the allow=all statement in sip.conf. Someone had suggested that statement is no longer needed, and using allow=ulaw, etc, had an issue where one or more deny's had to be used as well. By adding allow=ulaw in the sip.conf file, the Snom

hi all, I don't know how to setup asterix to work as PBX. If I want just basic configuration with 2 SIP phones (snom, ata), what all I have to write in the configuration files, or respectively in the configuration of ata and snom ? If there is any good documention available, send me URL too. All (ata, snom) are behind firewall (nat) and astrix is on the public IP, but I can move for

Hello! Attached to the bottom of this e-mail is an edited version of an e-mail I originally wrote to Digium tech support regarding Ouch and Power alarm errors I have been receiving on my TDM400. It contains a great deal of detail regarding my setup. In the end, I have found that one of the 5 phones I'm trying to make work with Asterisk is contributing to the generation of these errors.

Ok, I'm a dumb-ass today. I failed to see the "available = no" in the service entry. Sorry for wasting your time. Rob Naccarato "Civilized men are more discourteous than savages Sys Admin because they know they can be impolite without Sheridan College having their skulls split, as a general thing." Oakville, Ont. Canada

Hello Samba, I have a samba/cups problem I need help with. I'd appreciate an advice you can offer. Please see my thread with cups below: Ryan Suarez wrote: > Ryan Suarez wrote: > >> Michael R Sweet wrote: >> >> >>> Ryan Suarez wrote: >>> >>> >>>> Greetings, >>>> >>>> We're

---------- Wiadomo?? przekazana dalej ---------- Od: Stefano Stabellini <stefano.stabellini at eu.citrix.com> Data: 4 czerwca 2010 14:13 Temat: Re: [Xen-devel] Xen 4.0 in debian packages? Do: Thomas Goirand <thomas at goirand.fr> DW: "xen-devel at lists.xensource.com" <xen-devel at lists.xensource.com> On Thu, 3 Jun 2010, Thomas Goirand wrote: > ?ukasz Ole? wrote:

I have a TDM20B and asterisk compiled fine. The drivers have been loaded (wcfxs and run ztcfg -vv, I see the drivers loaded with lsmod). Asterisk starts up fine. I am using the default configuration files that are made when you do a "make samples". I was wondering if someone had a link or website that stepped someone through this kind of setup. What I want to do right now, is use a

>Date: Tue, 8 Dec 1998 11:00:42 -0500 (EST) >From: Rob Naccarato <rob@sheridanc.on.ca> >To: samba@samba.org >Subject: Can't connect to shares w/2.0beta3 on OSF1 > > >I was running 1.9.18p10 on a test DEC Unix 4.0d box here and I decided to >give 2.0beta3 a try. For some reason, I cannot connect to any of the >server's >shares or even get a list of

Hello, Maybe I'm missing something here. What is the proper way to use RTC with ztdummy now? I'm using -HEAD from a day or two ago on Linux 2.6.11.11. In zaptel/Makefile, I changed CFLAGS to: CFLAGS+=-I. -O4 -g -Wall -DBUILDING_TONEZONE -DUSE_RTC #-DTONEZONE_DRIVER and I get.. make -C /lib/modules/2.6.11.11/build SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory

Hi there, It is my understanding that blktap2 has been removed from the built of xen-hypervisor-{i386|amd64} because of the OpenSSL licensing and its incompatibility with the GPL, but apparently it is possible to not use OpenSSL at all. What is your thoughts about this? Could blktap2 support be added? Should it be packaged separately and be sent to non-free? Thomas -------- Original Message

Hi all. I have some questions: 1) Is there a way to get a full log of the calls (incoming and outgoing) 2) How is the intregation of Mysql and Asterisk. At witch Aplicattions. 3) And of the Extension a) I have a Support Call Center. Almost all the time all the extensions are busy, and some calls at hold. Is there a way that when some

I'm sorry this is somewhat offtopic, but I do plan to use this to help me create documentation for the * project.. so I guess it is somewhat on topic :) Anyways, I am looking for some sort of document control system. It should act somewhat like a CVS where it keeps previous versions, allows people to submit documentation, keeps track of who has what document open etc.. etc.. The