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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 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

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

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

...the wiki on "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 insta...

Thank you to everyone who has offered so far! I've had formal offers from Martin List-Peterson, William Suffil, Greg Varga, Brian Cuthie and Ed Guy (hopefully I haven't forgotten someone....!) Now we just have to decide where the best spot to host it is. What do you guys think? For this week, I don't care if this is a one off. At some point I'd like to have a weekly conference, and if we can get it hosted permanently...

...ions.conf to achieve what you are > looking to achieve. > > Some thoughts: > > What do you want to happen when one of the call takers has all 6 lines > in use? > > Have you considered using queues to do what you need? > > -Chris > > On 10:08 AM 5/22/2004, Brian Cuthie wrote: > > > >You might consider using the Cisco SIP phones. They're smart enough > >to accept incoming calls for as many call appearances you have with > >the same SIP registration. > > > >-brian > > > >Tor Roberts wrote: > > > >&g...

Hello. I have another issue. When I call in, everything is processed correctly, including voicemail, but I don't hear any ringing/ringback. exten => s,1,Zapateller(answer|nocallerid) exten => s,2,NoOp exten => s,3,Playback(pls-wait-connect-call) exten => s,4,Dial(${PHONE1}&${PHONE2}&${PHONE3}&${PHONE4},15,Ttm) exten => s,5,Answer exten => s,6,Wait(1) exten

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.

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

...lephone cards for the UK (WipeOut) 3. Re: Re: Analogue telephone cards for the UK (Paul Tyreman) 4. No ringing tone with IAXY (and other bits and bobs) (Chris Orme) 5. Nothing to do? Go bounty-hunting! (Olle E. Johansson) 6. RE: No ringing tone with IAXY (and other bits and bobs) (Brian Cuthie) 7. RE: No ringing tone with IAXY (and other bits and bobs) (Rich Adamson) 8. Extensions and Include (Kevin ) 9. RE: No ringing tone with IAXY (and other bits and bobs) (Brian Cuthie) --__--__-- Message: 1 Date: Sat, 10 Apr 2004 10:53:15 +0100 From: Iain Stevenson <iain@iainstevenson...

....723 (Eric Wieling) 2. RE: G.723 (Steven Critchfield) 3. Re: Voicemail storage in DB (James H. Cloos Jr.) 4. Voicemail config from database (AJ Grinnell) 5. oob to inband dtmf over rtp (James H. Cloos Jr.) 6. OT appologies to list (Linus Surguy) 7. Re: OT appologies to list (Brian Cuthie) 8. Zapateller issues (Mark Phillips) 9. RE: Zapateller issues (Sean Cheesman) 10. RE: Zapateller issues (Darrin Johnson) 11. Re: Re: Voicemail storage in DB (James H. Thompson) 12. Re[2]: [Asterisk-Users] OT appologies to list (Stephen Karrington) --__--__-- Message: 1 Date: Mon, 12...

...voicemail & SIP friends from mysql (=?iso-8859-1?q?Umar=20Sear?=) 6. Re: ISDN, AVM C4, HFC-cards and echo (Junaid Saeed Uppal) 7. RE: ISDN, AVM C4, HFC-cards and echo (Robinson Tim-W10277) 8. Question about Cisco IP Phone 7960 (Hall, Eric M.) 9. Re: VoIP hackers gut Caller ID (Brian Cuthie) 10. Re: VoIP hackers gut Caller ID (Stuart Baggs) --__--__-- Message: 1 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID From: <tpanton@attglobal.net> Date: Thu, 8 Jul 2004 11:59:00 0100 Reply-To: asterisk-users@lists.digium.com Because if p2...

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

Hi, I am setting up a dispatch center where will have 4 call takers, all with Polycom IP 600 Sip phones. Each phone will be setup with 6 extensions each. When a new call comes in, the first extension on all the phones will ring. This works fine, the problem is when one of the dispatchers is already using her first extension and another call comes in. What happens now is that the remaining 3

What you and so may others on this lise seem to forget is that Cisco is a company offering bsuiness products for businesses. Businesses typically pay by check and wire transfer, especially for items such as this. If you want home-user pay-by-credit-card service, buy products from Belkin's home line and similar. Oh...what's that? None of these cheesy Stocked-at-Costco hardware