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...;iax" returns exactly nothing. But searching on iax2 does start to dig up some good stuff. Sorry for the hassle. Tough day. -brian -------- Original Message -------- Subject: Re: [Asterisk-Users] IAX config documentation Date: Mon, 19 Apr 2004 21:22:44 -0400 From: Brian Cuthie <brian@systemix.com> To: asterisk-users@lists.digium.com References: <40843FD1.6030109@systemix.com> <1082409966.1421.14.camel@Steven.basesys.com> I know that this stuff is. What I'm looking for is an overview of how these features work in the context of IAX. For instance, trunking is a...

I have two 1-700 numbers from IAXTel. Both get registered from the same Asterisk server. I can make and receive calls on each without any difficulty. What I can't figure out how to do is route the incoming calls differently based on which 1-700 number is dialed. I must be missing something obvious. Thanks -brian -------------- next part -------------- An HTML attachment was scrubbed...

Anybody figured out how to close a VoicePulse Connect! account? As bad as their web site is at most other things, the notion of actually closing an account doesn't appear to have even been contemplated. -brian

After downloading the latest CVS head and testing it with the Cisco 7960 (SIP v7.2), it appears that it's still necessary to patch rtp.c to avoid audio dropouts. I'm quite sure my gateway provider is running an older version of Asterisk, and I suppose that this may be the root cause. But I mention the issue here because it seems like it would be a mistake to ship Asterisk 1.0 if it

...ion of the function * The combined value of the bounty * Licensing for the code - I would suggest that all bounties should be disclaimed so they may be candidates for the CVS Happy easter from one of your friendly bug-marshals! /O --__--__-- Message: 6 From: "Brian Cuthie" <brian@systemix.com> To: <asterisk-users@lists.digium.com> Subject: RE: [Asterisk-Users] No ringing tone with IAXY (and other bits and bobs) Date: Sat, 10 Apr 2004 08:50:52 -0400 Organization: Systemix Software Reply-To: asterisk-users@lists.digium.com What version of the Asterisk code are you running?...

In the absence of "The Definitive Guide to Asterisk Dial Plans" book, I'd like to do something possibly unique with the formatting of extensions in my dial plan, and am having trouble. We use VoicePulse connect, which gives us local DID for inbound and outbound calls (even though DTMF tones are not working in Voice Pulse Connect at the moment). To dial local numbers, you have to

...rset="iso-8859-1" Content-Transfer-Encoding: 7bit X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 5.50.4522.1200 X-MimeOLE: Produced By Microsoft MimeOLE V5.50.4522.1200 --__--__-- Message: 7 Date: Mon, 12 Apr 2004 15:42:27 -0400 From: Brian Cuthie <brian@systemix.com> To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] OT appologies to list Reply-To: asterisk-users@lists.digium.com There's more than a little irony here, given that one of their products is called "Email Blaster." -brian Linus Surguy wrote: >[I'm so...

...n. I upgrade the phone to 7.1 and it installed the Universal Application Loader. Now I'm getting Protocol Application Invalid after it reads tftp SIP(MAC).cnf Any ideas? Again sorry this is off topic --__--__-- Message: 9 Date: Thu, 08 Jul 2004 08:28:47 -0400 From: Brian Cuthie <brian@systemix.com> To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID Reply-To: asterisk-users@lists.digium.com The real problem here is that people shouldn't be using callerid as an authentication scheme. Lots of people have had the ability to set arbitrary cli...

It seems that each time I get a new checkout of * from CVS my Cisco 7960 works worse than before. I know this stuff's in flux, so I mention this in case it's news. Anyone else having trouble? What I'm seeing (er, hearing) is really choppy audio. The previous version I had installed had fairly frequent audio dropouts (not present when I make the same calls through the same * box

Hello, I'm looking for an ISDN BRI-U interface for use in the US. I'm primarily interested in using the BRI as a trunking interface into the PSTN with Asterisk. Naturally, cheaper is better. I currently use a Nortel Norstar system with BRI-U trunks, and really like the digital PSTN interface. Would really like to replace the whole mess with Asterisk but want to keep the ISDN trunks.

Let's say that I have a call coming in to Asterisk through a TDM400P and going out through SIP to someone on the Internet. Is there any configuration option that would allow me to do silence suppression on the RTP stream generated by Asterisk on behalf of the TDM400P connected user? SIP phones allow me to do this easily, but I'd like to be able to conserve upstream bandwidth on calls that

So what are people using these days for SIP or IAX to PSTN gateways. 1. Do any of the standard companies (Packet8, Broadvox, Vonage, etc.) allow you to use your own SIP device (phone or something like *) instead of the interface hardware they usually provide? 2. What about latency and reliability? 3. Finally, do any of the providers deliver more than one call via SIP? In otherwords, if

I've setup the following in extensions.con: exten => 2200,1,Ringing exten => 2200,2,Wait(2) exten => 2200,3,Answer exten => 2200,4,Zapateller exten => 2200,5,Macro(stdexten,2205,SIP/2205) This works as expected if I dial from a SIP phone on my desk. However, if I dial in from the PSTN (through a SIP provider) it fails while trying to play ths SIT with: Apr 8 18:53:12

Should FAX transmission generally work through Asterisk and a TDM400P connected through a PSTN gateway? At first blush I'd think that if they're all g.711uLaw encoded that it would work. But experience shows otherwise. Is there a better way to do FAX? -brian

Is there any documentation on configuring IAX between * machines? I've noticed references to many topics in the config files, including: - dialplans - trunking - authentication - transfers But before I go and try to grok 8000 lines of source (in one file, no less) I was hoping that somewhere there exists even something like a man page that describes the configuration options.

I've been setting up a couple of * boxes with IAX trunking between them. But I've been seeing some strange IAX behavior. Asterisk version is latest CVS-04/21/04-18:10:19. Here's what I'm doing: the boxes are peers, and I have setup my iax.conf file to look something like this: << machine1 >> [iaxuser] type=friend username=iaxuser secret=foo auth=md5

Hi everyone, I have a TDM400 configured with 4 FXS ports, each connected to a caller-id analog trunk port on a Nortel system. Outgoing calls work great. But on incoming calls it appears that loop current is getting dropped momentarily about 10 seconds after the call is answered. Since the Nortel system is programmed to recognize this as remote party hangup it is causing all incoming calls to

Anybody know an easy way to adjust audio level of recordings made in Asterisk (using the 'record' application)? I've noticed that recordings using the "wav" format are about twice the level of those made using "WAV" or "wav49". Unfortunately, the "wav" recordings are uncompressed and about 10 times the size of the other formats. -brian