similar to: SIP softphone volume control?

I would appreciate some help with this. I read the GPL license and basically it says you can do whatever you want with the software (sell, modify) as long as you include the source code, the License and make any changes you make available in the same manner to all others. My questions is this: If I develop an external application (say a Call Center application or a GUI management application)

I want to be able to demo * on a notebook at a client's site. This means no FXO gateways; just 2 sip phones (like SNOM) and maybe a softphone (GnoPhone?). I already have RH9 running on my notebook. I would like to have one SIP phone dial and go through IVR before making a choice and ringing the other phone extensions. Of course the notebook would have to be running Asterisk. How can i setup

Hi U can visit the http://iaxclient.sf.net for some opensource underdevelopment softphones. Take Care Obaid Amin Syed >From: Chris Albertson <chrisalbertson90278@yahoo.com> >Reply-To: asterisk-users@lists.digium.com >To: asterisk-users@lists.digium.com >Subject: [Asterisk-Users] Good W2K softphone >Date: Fri, 3 Oct 2003 23:00:13 -0700 (PDT) > > >I haven't

Chris Albertson wrote: |What firmware version do you have? program version 1.0.4.39 -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 1878 bytes Desc: S/MIME Cryptographic Signature Url :

Hi, My installation at home use two NAT translations before it reaches the linux box where Asterisk is running on. I use DSL with a Wireless router which fwd all packets to an Windows 2003 box an this windows box it NATing the UDP and RTC packets to my linux box. If I try to connect to it from outside I get this error : Nov 30 22:19:02 WARNING[1106250672]: chan_sip.c:673 retrans_pkt: Maximum

On the cheap side, the ITX or even MicroATX machines work great. These are commodity items, so they tend to be far less expensive than custom solutions. Various manufacturers, but we've had very good success with any of the AOpen MicroATX boards and their slimline MicroATX case: Aluminum: http://usa.aopen.com/products/housing/A340-series.htm Steel:

I have just checked the Openbsd box on the if interface. fxp0: flags=8843<UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST> mtu 1500 address: 00:02:55:30:54:28 media: Ethernet autoselect (100baseTX full-duplex) status: active inet 192.168.1.1 netmask 0xffffff00 broadcast 192.168.1.255 inet6 fe80::202:55ff:fe30:5428%fxp0 prefixlen 64 scopeid 0x1 xl0:

Many of you are familiar with how lousy Festival sounds. AT&T has a product, NaturalVoices, that sounds much better. There are male & female voice fonts for US/UK/Indian English, French, Spanish, and German. I am considering offering a linux-based text-to-speech engine based on the NaturalVoices runtime. An asterisk module would also be provided, making it easy to add natural sounding

Is anyone using Asterisk as the gatekeeper/proxy for videophone calls? Thanks, --Ernest

I can't send mail to any addresses in nufone.net; they all get rejected by a spam blocker. And their website is gone, too!! The URL leads to a "parking site." My accounts still seem to work, but I wonder WTH is going on? Thx. B.

I have 8 lines coming into an existing PBX system and am looking for a cost effective way to replace the existing system with Asterisk. We need some of the features in Asterisk, including its ability to support remote offices (long distance savings). At first glance this appears to require a T100P card and a channel bank, but that seems rather expensive. My estimated price on that would be

I need to get your thoughts on something.. :) I am trying to create a system to process the CDR call logs for department accounting.. I think there are two ways of doing it.. Either I can create an AGI that will run on the "h" extension and will lookup the last entry that matches the account code of the call that just ended in the MySQL CDR and calculate the call cost immediately..

Hi All Has anything been done to fix the issue where the * box is sat behind a nat firewall? Regards Mark

Hello, today I found this security report regarding Asterisk SIP Security. http://www.securiteam.com/securitynews/5LP0720B5G.html Maybe It could help somebody who isn't using a newer than 15th of August cvs version. Best regards Lubo

ChanIsAvail returns the channel ID plus "-<session>". How can I edit ${AVAILCHAN} to remove this session ID, so I can use its contents in a subsequent Dial statement? Dialing on Zap just gives a warning, but dialing a SIP channel completely errors out. ------ extensions.conf snippet------------- ; ; Main Home number (8901) ; ; Bedroom1 exten =>

Hey.. Has anyone played around with Asterisk on the Itanium2, Opteron or Athlon64?? Can Asterisk (or Linux for that matter) actually make good use of a 64bit system?? Later..

I'm first to buy 5 pack. Even for > $30.

I have a machine that crashes every so often. I believe the following macro is responsible (gotoif,$[${ARG3}] in particular). The macro works as expected: if ARG3 is defined - hop over assignment. But my hunch is that it gradually chews up memory. ; This macro is puts voicemail in an alternate mailbox (if ARG3 defined - otherwise Mailbox matches extension). [macro-stdexten] exten =>

Yes folks it's me a Newbie. Remember I am also a non-Linux person trying to learn. I have a production Server running Asterisk .5 12/02/03 CVS, and would like to upgrade it to the new .71. Has anyone come up with instructions (Documentation for us newbie) on how to do this? My server is running Mandrake 9.0 which I know nothing about! Sorry if this sounds stupid but all the instructions I

Hi . I already set-up sendmail on another network server thus it would be nice to use that one or is sendmail on * server required!? I had a look in the archive but couldn't find any information where to set the mail server from localhost to my network server . Cheers Ralf -------------- next part -------------- An HTML attachment was scrubbed... URL: