Re: [Asterisk-Users] Lobotomized Sipura SPA-3000 configuration needed
Steve Edwards wrote: Return-Path: [EMAIL PROTECTED] Received: from digium-69-16-138-164.phx1.puregig.net[69.16.138.164] (helo=lists.digium.com) by mx.perfora.net with ESMTP (Nemesis), id 0MKv6A-1CUw2b1zZX-0006bH; Thu, 18 Nov 2004 18:50:17 -0500 Received: from [69.16.138.164] (localhost [127.0.0.1]) by lists.digium.com (Postfix) with ESMTP id 331722FFE8F; Thu, 18 Nov 2004 17:48:48 -0600 (CST) X-Original-To: [EMAIL PROTECTED] Delivered-To: [EMAIL PROTECTED] Received: from psmtp.com (exprod5mx126.postini.com [64.18.0.40]) by lists.digium.com (Postfix) with SMTP id ED65E2FFE88 for [EMAIL PROTECTED]; Thu, 18 Nov 2004 17:48:44 -0600 (CST) Received: from source ([63.193.37.160]) by exprod5mx126.postini.com ([64.18.4.10]) with SMTP; Thu, 18 Nov 2004 15:48:53 PST Received: from fs.sedwards.com (localhost.localdomain [127.0.0.1]) by fs.sedwards.com (8.12.8/8.12.8) with ESMTP id iAINmrus027823 for [EMAIL PROTECTED]; Thu, 18 Nov 2004 15:48:53 -0800 Received: from localhost ([EMAIL PROTECTED]) by fs.sedwards.com (8.12.8/8.12.8/Submit) with ESMTP id iAINmrYh027819 for [EMAIL PROTECTED]; Thu, 18 Nov 2004 15:48:53 -0800 X-Authentication-Warning: fs.sedwards.com: sedwards owned process doing -bs Date: Thu, 18 Nov 2004 15:48:53 -0800 (PST) From: Steve Edwards [EMAIL PROTECTED] X-X-Sender: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] MIME-Version: 1.0 Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed X-pstn-levels: (S:84.10653/99.9 ) X-pstn-settings: 1 (0.1500:0.1500) gt3 gt2 gt1 X-pstn-addresses: from [EMAIL PROTECTED] [65/3] Subject: [Asterisk-Users] Lobotomized Sipura SPA-3000 configuration needed X-BeenThere: [EMAIL PROTECTED] X-Mailman-Version: 2.1.5 Precedence: list Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] List-Id: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users, mailto:[EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:[EMAIL PROTECTED] List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users, mailto:[EMAIL PROTECTED] Sender: [EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] Envelope-To: [EMAIL PROTECTED] X-SpamScore: 0.000 I must be getting thick in my old age. The sheer number of options on the SPA-3000 is causing my eyes to glaze over. If anybody is willing to share their configuration I'll post a dummy's guide on the wiki -- http://www.voip-info.org/wiki-Sipura+3000; didn't quite do it for me. All I want to do is have the SPA-3000 configured so that it offers up its FXO and FXS ports to Asterisk -- nothing more, I want Asterisk to be the brains. 1) The SPA should hand incoming calls on the FXO to Asterisk. 2) Asterisk should be able to place outgoing calls on the SPA's FXO. 3) Whatever I dial on the SPA's FXS should be handed to Asterisk. 4) Asterisk should be able to dial the SPA's FXS. 5) If Asterisk is not available, I want to tie the FXO and the FXO together. You might want to see if the SPA-3000 Asterisk Wizard on Voxilla suits your needs, which I believe it will: http://voxilla.com/spa3kasterisk.php -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lobotomized Sipura SPA-3000 configuration needed
Gregory Junker wrote: One problem is that the SPA3K only uses two-stage dialing on the FXO -- VoIP2 path, which means any time someone calls the phone system and gets forwarded to a select SPA3K extension, they hear a dial tone. As far as I can tell, there is no way to disable that. You can have it execute a particular dialplan in the SPA3K but the caller gets to hear the digits as they are dialed into Asterisk. I have my SPA-3000 taking a PSTN line inbound and forwarding it to my Asterisk server after a few rings. I don't hear any dial tone when I do that kind of forwarding. I do it via the dial plan and I also tried it via CFwd SelX Caller/Dest. How are you attempting to do it? -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lobotomized Sipura SPA-3000 configurationneeded
Gregory Junker wrote: Good deal, I looked at the config site but wasn't sure. The SPA3K I have for testing is 2.09, not sure if that makes a difference. You want to upgrade. They mute the dialtone on outbound FXO calls in the later releases. -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SPA-3000 Wizard for Asterisk
For your testing pleasure. Feedback welcome: http://voxilla.com/spa3kasterisk.php -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Sipura 3000 FXO
Randy Bush wrote: i come from an automated ip backbone world where we generated configs automatically from sql data tied to the back office and sales systems. i want to have a shipping person take a new spa3k out of the box, plug it into an ether, hit the 'Confirm' button on the customer order fulfillment screen, wait 30 seconds, and then stick the puppy in the outbound shipping box. There's no reason a SPA configuration couldn't be auto-generated from a SQL database. There's the issue of interfacing that with the SPA Compiler to compile a configuration file and put it somewhere (e.g. a web server), but that should be easy to do. It's fairly trivial to set up a box as a DHCP server and tftp server with a spa.cfg (where is 2000, 3000, etc). The DHCP server sets the tftp-server option. This config simply sets a provisioning rule that says go get your next config from https://some/website/$MA.cfg; (where the device substitutes $MA for its MAC address). This URL would be the location where your auto-generated device configuration would reside. Optionally, you can also have it load new firmware. The spa.cfg file sits in the root of the tftp server. Basically, all you have to do is unbox a new SPA, plug the box in, wait several seconds, wait for the status lights to stop blinking, and the box is ready to ship. I did something similar with about 40 SPA-2000s and the entire process was painless and quick. It doesn't even need to have the real configuration from the SQL database yet, it can pick that up when it gets plugged in next time. -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura-3000 - silent dial out on FXO port
Benjamin on Asterisk Mailing Lists wrote: When I connect to the Sipura to dial out on the PSTN line connected to the Sipura's FXO port, it gives me the dialtone of the PSTN line and then I can hear the DTMF for the number I dialled beforehand. It does work but the customer perceives this delayed second DTMF feedback as unprofessional and the sipura as a toy. I wonder if there is anything that can be done to keep the channel to the caller silent until after the Sipura has sent the DTMF out on the PSTN line. Upgrade your firmware to the latest release. They solved that problem in the more recent releases (2.0.10 and above, IIRC). -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Opensource Sipura Profile Compiler for SPA2K, 3K
Kristian Kielhofner wrote: 1) There is a lot of code in the dump from /admin/advanced. Note that if you're interested in only changing a few parameters, you need not post everything. 2) The password is all *'s (not good to PUT it back like that). See previous point. -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
Steve Maroney wrote: Hey guys, Im about to sign up for VoicePulse Connect. Of course, I plan on using my asterisk server to register = with the service. I would rather sign up with VoicePulse via SIP instead of VoicePulse Connect. My asterisk box is behind another Linux box serving as my nat/firewall. Does anybody think I will have a problem ? Should I stick to IAX and VoicePulse Connect or can I use VoicePulse with SIP ? You can use VoicePulse Connect with either SIP or IAX2. The non-Connect plans require an ATA for a primary line -- a SIpura SPA-2000 or SPA-3000, though they previously supported the Cisco ATA-186. You can get Open Access lines (adjuncts to primary lines) that use generic SIP credentials. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Interruptable SayUnixTime
I'd like to announce the time when people call and hit my voice-menu context, but the complaint is that the time announcement isn't interruptable. Is there any way to make SayUnixTime interruptable? -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Blind Call Transfer using Sipura 3000 + asterisk
On Wed, 2004-08-11 at 12:00, Karun Chemudugunta wrote: I am using Sipura 3000 to receive PSTN calls and forward those calls to asterisk for voice processing and after that, I am transferring call to extension through FXS port on SPA 3000. Currently, media of call is trombone through asterisk. i.e achieving blind transfers on asterisk with SPA 3000. Is it possible to stop trombone and send back media path on Sipura FXO to FXS. Basically you want to eliminate the back and forth traffic between the Asterisk server, right? That'd be nice. :) -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA-3000 as a regular Asterisk FXO device?
On Tue, 2004-08-03 at 14:17, Mike Benoit wrote: My SPA-3000 finally arrived and I'm trying to get the FXO port on it to work as if it was a X100P card as far as Asterisk is concerned. I have Asterisk dialing out over the SPA-3000 FXO port no problem. The issue I'm having problems with is having the SPA-3000 automatically forward all incoming PSTN calls to the Asterisk mainmenu context (or ext I guess). Currently the SPA-3000 answers the call, then I hear a modified dial tone, which if I dial any extension + #, it will ring a SIP phone no problem. So now I just need to get it to that automatically. For the dial plan associated witn inbound PSTN calls, use a dialplan like so: (S0:666). 666 is the extension to my voice menu. This is the hotline syntax in the dial plan on the Sipura. -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 3000 PSTN disconnect in the UK
On Fri, 2004-07-30 at 23:59, Chris Stenton wrote: Anyone else got a Sipura 3000 in the UK? Apart from CID not working it also seems to not notice any of the line state changes on the PSTN when the remote party terminates the call. It only recognises the offhook signal which gets sent much later. Inbound Caller ID on the FXO port is only Bellcore FSK (North America) at this time. You might have to play with the PSTN Disconnect parameters to obtain the optimal result for detecting hangups. What might give you a hint as to what parameters to turn off and/or adjust is the Info tab. Look at the PSTN Disconnect Reason when the SPA-3000 finally does give up the PSTN line. -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Has anyone tried using a Sipura-3000 as an FXO device for *?
On Sat, 2004-07-31 at 17:43, Kevin Walsh wrote: As I said, I don't have one of these yet. Do you happen to know what the box would do if the dialplan said to route the call to :@gw0 and that port was already in use? You'd probably get a Fast Busy if dialing from the FXS port. If coming in from Asterisk, I think you'd get the appropriate SIP message saying the line was busy (503?) -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Has anyone tried using a Sipura-3000 as an FXO device for *?
On Tue, 2004-07-27 at 15:52, Carmi Weinzweig wrote: I am considering using Sipura-3000s as FXO devices for my * system. Has anyone tried them in that configuration? They interest me because they need no PCI slots and therefore no drivers. I would much prefer not to have any special kernel requirements for my system. A number of us are using SPA-3000s for this exact purpose, including myself. Works pretty well. -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VSP? Looking for advice.
On Thu, 2004-07-22 at 15:58, Chris Shaw wrote: As for multiple lines, they do offer multiple virtual numbers but unless you want it to look like you have multiple lines, you don't need to do that... VoIP by nature supports multiple calls to the same phone number without the need for Trunk Lines I don't know if they *support* having multiple simultaneous calls (I do know it works though I've tried...), but I wouldn't think they would have too much of a problem with it because they support 3-way calling which is basically the same thing as having 3 simultaneous conversations... They do let you make multiple simulatenous calls, but be warned that each additional outgoing stream will be charged at their overage rate (except for three-way calling). -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT + iConnectHere Broken in 1.0RC1
On Thu, 2004-07-22 at 21:35, Lion Templin wrote: I've been using * CVS code from May of this year and was able to connect to iConnectHere and receive calls with * being behind NAT. Now that I've upgraded to 1.0 RC1, this no longer works. I've tried setting nat=yes in places, externip, et al with no success .. even though the code I was running from back then worked without that. It's also broken in the current CVS stream as well, or at least that's what I've found. -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VSP? Looking for advice.
On Fri, 2004-07-23 at 12:50, Chris Shaw wrote: There shouldn't be an overage rate though if you're on the unlimited plan like I am... Not according to the CEO of BroadVoice: http://www.voxilla.com/voxstory71-nested-order0-threshold0.html -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone heard of BroadVox direct?
On Wed, 2004-07-21 at 21:42, Jay Milk wrote: Just received: Cognigen is very proud to announce the official launch of Broadvox Direct, a new VOIP service. Broadvox (http://www.broadvox.net) sells the service directly, and has since March of 2004 I believe. Cognigen is simply a reseller of said service. A number of other VoIP providers private-label Broadvox service in one form or another. I have used their service since December of 2003 (kind of a beta tester). Initially, they shipped the SPA-2000, but are about to start shipping their new DTA which is based on the Mediatrix 2012. I've been beta testing their new DTA and, at least in the ways I use, it works really well. Supports faxes and analog modems too, which is a bonus. They are not Asterisk friendly in the sense they require ATAs and won't provide credentials. At the moment, I have the SPA-2000 plugged into the FXO port on my SPA-3000 so it goes into my Asterisk server. I'll swap it for the new DTA once they move my phone number over to it. :) -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spa-3000 review?
Wolfgang S. Rupprecht wrote: Interesting. I'm at -current +/- a day and do see a NAK/retry-with-md5 exchange when I do a sip debug. The md5 authentication is also NAK-ed. Well you got farther than I got when I was having problems. :) My fear was that it was expecting the calling user to use their own username in the validation instead of asterisk using the shared secret with a shared user-id. Asterisk should use whatever credentials you define as HTTP Username/Password in the SPA-3000 configuration. -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spa-3000 review?
Wolfgang S. Rupprecht wrote: Have you gotten asterisk to work for dial-out to the PSTN when using a md5 authentication? What I discovered via tcpdump was that the Asterisk box wasn't responding to the authentication request for whatever reason. I couldn't get it to work until I upgraded to the latest CVS release. Once I did that, I could do it with authentication. -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spa-3000 review?
On Wed, 2004-07-14 at 14:33, Dameon D. Welch-Abernathy wrote: I experience a some echo. It can be minimized by adjusting the SPA to PSTN Gain and PSTN to SPA Gain values to quieter values. I believe it can range from -12 to 12. 12 is about twice as loud as 6, -12 is twice as quiet as -6. I was incorrect here, it can be anywhere between -15 and 12. -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] spa-3000 review?
On Wed, 2004-07-14 at 16:56, Kevin Walsh wrote: I'll rush out and buy one for use at home as soon as they support the UK (BT) phone system for Caller*ID and distinctive ringing etc. (as the SPA-2000 does for UK phone handsets). The FXS port on the SPA-3000 supports that stuff, but the FXO port does not. Sipura is looking at supporting that stuff on the FXO port in about 1-2 months, if what they said during beta was correct. -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spa-3000 review?
On Wed, 2004-07-14 at 09:43, Rich Adamson wrote: Since the 3000 has been out for a little while, has anyone done a review of the product? (couldn't find anything on google for wiki). There's a review of the SPA-3000 on Voxilla. I know of a number of people using the SPA-3000 with Asterisk, as do I. -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spa-3000 review?
On Wed, 2004-07-14 at 14:18, Mike Benoit wrote: Dameon and Wolfgang, Have either of you experienced echo when making a call from the FXS port to the FXO port on the SPA-3000? I experience a some echo. It can be minimized by adjusting the SPA to PSTN Gain and PSTN to SPA Gain values to quieter values. I believe it can range from -12 to 12. 12 is about twice as loud as 6, -12 is twice as quiet as -6. -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where can i get an UK SIP account with UK number?
On Wed, 2004-07-14 at 11:41, Johannes van Hulst wrote: Can somebody help me with some names of good UK SIP providers? I am looking for a UK number to connect to my asterisk server. www.gossiptel.com provides UK numbers www.iconnecthere.com provides UK numbers There are probably others. -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 calls through IAXTEL.com
On Tue, 2004-07-13 at 05:38, Steve Woolley wrote: 1) if I dial 800 numbers, like (800)555-1212, I get a bunch of silence and the following in my log: -- Starting simple switch on 'Zap/97-1' -- Executing NoOp(Zap/97-1, ) in new stack -- Executing Goto(Zap/97-1, intern-post|818005551212|1) in new stack -- Goto (intern-post,817005556226,1) -- Executing Dial(Zap/97-1, IAX2/myusername:[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack -- Called myusername:[EMAIL PROTECTED]/[EMAIL PROTECTED] -- Call accepted by 69.73.19.178 (format GSM) -- Format for call is GSM -- IAX2[iaxtel-outbound]/3 stopped sounds The call never seems to go through. I am also having that problem. I started having that problem within the past few days. Maybe there's some problem at iaxtel? 2) Not knowing any other way to test, I have simply picked up my asterisk SIP and analog phones and dialed my own 700 number (700)555-6226 to which I get a bunch of silence and the following in my log: You can try calling me at 700-650-4330 and see if you have a problem and help me see if I got my IAX2 setup correct. :) -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP client to IAXTel 800/888/877/866 dialing thru Asterisk
Through my Asterisk server, I am trying to use IAXTel to dial 800-type numbers (yes, I know I can do the same thing with FWD and others via SIP, but I wanted to play with IAX a little). It appears I'm running into some sort of a codec mismatch or something because it's not working right. The SIP client is a SPA-3000. In iax.conf, I have something like the following: [General] disallow=all allow=ilbc allow=adpcm allow=g726 allow=gsm allow=ulaw allow=alaw register = user:[EMAIL PROTECTED] [iaxtel] type=friend host=iaxtel.com secret=password auth=rsa context=from-iaxtel inkeys=iaxtel The sip.conf has similar settings in the General section related to codecs. The codec on the SPA-3000 is forced to G.711 (the only thing it talks to directly is my Asterisk server). When I make a call to a number on IAXTel (e.g. 1-700-VOXILLA), it works fine. If I try and call an 800 number, I don't get any audio. However, it appears the IAX side of the equation isn't negotiating a codec right. Furthermore, the SIP side is choosing the wrong codec (though I've tried putting allow=ulaw first and it still didn't help) grover*CLI iax2 show channels Peer UsernameID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format 69.73.19.178 (None) 3/0 1/0 0ms ms ms UNKN 69.73.19.178 phoneboy5/00102 00019/00017 00099ms ms 0010ms UNKN 2 active IAX channel(s) grover*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format 10.0.0.250 53 2b890d18-49 00101/00103 ILBC 1 active SIP channel(s) Is there a problem with iaxtel? Any ideas? Asterisk CVS-HEAD-07/06/04-01:33:49 built by [EMAIL PROTECTED] on a i686 running Linux -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FINALLY! a good book about Asterisk.
On Thu, 2004-07-08 at 22:02, Kevin Walsh wrote: In my experience, the royalty is 10 to 15% of the *wholesale* price, which means it's more like two to three bucks a book. Trust me, unless you're Stephen King, John Grisham, or someone like that, you don't make all that much writing books (at least directly). All the more reason for the author to consider OpenDoc publishing. At the time I started the first book (which technically, someone else started, I ended up taking over), I don't think that was an option. By the second book (which was just an update of the first), I couldn't do OpenDoc because I was basically under contract with the publisher. Authors of technical books tend to make more money out of consulting anyway, as they're seen as an expert in their field. That assumes, of course, the author still wants to work in that field by the time the book is published. :) -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FINALLY! a good book about Asterisk.
On Thu, 2004-07-08 at 16:44, Sathya wrote: $50 is a bit steep for a book Usually author of a book makes 10 to 15 percent of the cover price. So whoever who wrote this book will get 5 bucks a book. In my experience, the royalty is 10 to 15% of the *wholesale* price, which means it's more like two to three bucks a book. Trust me, unless you're Stephen King, John Grisham, or someone like that, you don't make all that much writing books (at least directly). -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP hackers gut Caller ID
On Wed, 2004-07-07 at 05:29, Chris Foster wrote: I hope NuFone doesn't drop asterisk-set-able callerid's after this article; i've been wanting that feature from voicepluse for a long time. My VoicePulse Connect line allows you to set Caller ID. -- PhoneBoy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How do I disable '#' to transfer a call?
I don't see anything on the Wiki or in the documentation about disabling this feature. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do I disable '#' to transfer a call?
On Tue, 2004-07-06 at 01:51, Shaun Ewing wrote: Easy, just don't include t or T in the dial string options. I guess I was searching for the wrong question in the documentation: disabling the transfer feature instead of enabling it. :) I'm only interested in disabling the # when I *make* a call as that's where I'm likely to hit an IVR, so I guess that means removing the 'T' option. Thanks for the help. -- Dameon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How do I disable '#' to transfer a call?
On Tue, 2004-07-06 at 08:35, Andrew Thompson wrote: I'm only interested in disabling the # when I *make* a call as that's where I'm likely to hit an IVR, so I guess that means removing the 'T' option. That means you just want to remove the t/T from your outbound dialplan, not inbound to your extension. I figured that out once I read the documentation a little. :) -- Dameon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
