similar to: Fwd: Re: SIP behind NAT, workaround to make W Snel's very welcome fix work both for inside *and* outside clients

--- "Olle E. Johansson" <oej@edvina.net> wrote: > From: "Olle E. Johansson" <oej@edvina.net> > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Asterisk on FreeBSD > Date: Mon, 27 Oct 2003 08:24:22 +0100 > > Rich Adamson wrote: > > >>My Asterisk (fresh CVS) takes 98% of the system load on my FreeBSD > server.

Your log file almost looks like a bug in Asterisk doesn't it? Why call poll() with a zero timeout while passing only one FD? and then why do the read when there is no data? Read the man pages for all the system calls Take a look at the source chan_sip.c /* Wait for sched or io */ res = ast_sched_wait(sched); if ((res < 0) || (res > 1000))

Hello everyone and welcome to my first post to the list! After studying for a couple of weeks, I finally built * for the first time last night, and of course had the same SIP-behind-NAT woes that plague all of us who use NATted connections. It was therefore with no small joy that I read the fix for that that Walter Snel proposed (q.v.:

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

Hi Chris, I know this is quite an old email, but I was browsing through the archive :) I am currently working on "embedding" asterisk in one of Allwell's STB's. The idea is more or less exactly like yours. The STB will be solid-state and contain OS, Asterisk, Basic configuration and voice files on a flash disk. It will boot up and get a network share via DHCP. This network

The disadvantage (depending on what you're looking at doing) is that they have to be in an operating computer Hmmm ... maybe my wish list for a self-contained FXS should expand to a self-contained one or two FXS and one FXO with software config and at least two ethernet ports (like one to internet connection and one to hub/switch for extra connections). This would be a nice self-contained

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:

Another issue to consider is T1 framing. If your application is putting bits onto the T1 at the rate of 1.544 Mbit/s then the T1 would need to be unframed. I don't believe this is an option in zaptel! If however, it is putting bits on at a rate of 1.536 Mbit/s and adding 8000 bit/s for framing then you may be able use the suggestion below. Don Pobanz On Thursday, April 03, 2003 3:28 PM,

No it's not broken, I just did a CVS update recently. Try downloading from CVS and then and if you have problems quote the error messages in your e-mail. --- "Lic. Edwin Mamani Z" <emz@mail.iseye-bo.net> wrote: > > Hi > I am interest with asterisk, but I like download It and the link is > break > > Please, Any people give me this software > > tanks

I'm working on contributing two things for Asterisk 1) STUN suport, this will allow asterisk to detect any NAT firewalls and enable eventual self-configuration with respet to NAT 2) A GNU "autotools" based build system. This will enable developers to make their code more portable and for features to be enabled/disabled as compile time. As a first step

Is there a way to set to audio gain for each SIP extension? I see in the docs this can be done for zaptel but I don't see it documented for SIP. It would be nice to be able to make the various kinds of extensions have equal volume. ===== Chris Albertson Home: 310-376-1029 chrisalbertson90278@yahoo.com Cell: 310-990-7550 Office: 310-336-5189 Christopher.J.Albertson@aero.org

Hello, I've subscribbed to "IConnect". I use it eclusively for outbound calling. I like the rates they charge but people I call complain about the audio quality. They say it sounds like I'm using a "cheap mic." or they complain about echo. The sound is very clean at my end. I'm using a Bundgtone phone with meadi routed through Asterisk to IConnect. It's

How about a PCMCIA Zapata interface?? Asterisk and its strength kick ass as a test unit. Can't do some of the things a T-byrd can do but the Telco techs freak when you tell them its your PBX!!! ) 10. Re: The Smallest Asterisk Server Ever? (Panny Malialis) Message: 10 From: "Panny Malialis" <panny@hotlinks.co.uk> To: <asterisk-users@lists.digium.com> Subject: Re:

FWIW: I order a lot of Dells. My boss is cheap. That being said, I *like* Dell, it's a very well designed box. It's been said many times that Dell does not innovate, instead they copy and improve and I firmly agree with the "improve" part - they are a dream to work on. Some things to watch out for with Dell: 1. They typically tack on a shipping charge of $139 Cdn (yes they

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

Hi everybody, I have 3 TDM400P installed in a machine,and though the 4 ports of the first card work fine, some ports on the other two have low or no signal and a noise instead. Can someone help? Thanx __________________________________ Do you Yahoo!? Exclusive Video Premiere - Britney Spears http://launch.yahoo.com/promos/britneyspears/

It is a cable 4-5 meters long that has handssets connected I don't think its a matter of a distance __________________________________ Do you Yahoo!? Exclusive Video Premiere - Britney Spears http://launch.yahoo.com/promos/britneyspears/

I wasn't thinking of using the conference system as the basis. I was thinking more along the lines of: 1) Setup a second extension on the Cisco phone named "INTERCOM" enabled for auto-answer 2) Create a call group on asterisk to dial that "INTERCOM" extension on every phone that will participate 3) Add a feature code that would dial the intercom extension and connect

My God why does he pentagon have an moronic idiot like you working for them !?!?!?!?!? Friggin amatuer hour or what! __________________________________ Do you Yahoo!? Exclusive Video Premiere - Britney Spears http://launch.yahoo.com/promos/britneyspears/

Hi list, I am facing the following problem, i need to make the following scenerio work exten => _900.,1,Dial(Zap/g1/xxxxxxxxxx,25,r) exten => _900.,2,Wait(3) exten => _900.,3,SendDTMF(${EXTEN:1}) I am using PRI-ISDN with T400P card. Searching through archives, i found that we can add some 'w' s to the dial string. I tried that using on both PRI and Analogue channels but it