similar to: DISA and notransfer/reinvite?

Hi All, I'm using DISA on my * server to avoid overseas toll charges when making calls to Western Europe from my cell phone. I have DISA working with a DID from a VoicePulse Connect account. The outgoing call to Europe is also made via Voicepulse Connect. I see that the IAX media path is bridging the inbound call to the outbound call so that the media stream entirely bypasses my server once

'http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20DISA' This should help. -----Original Message----- From: Michael Graves [mailto:mgraves@mstvp.com] Sent: Wednesday, January 26, 2005 9:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] dialplan logic for conditional DISA on incomming 800 number Hi All, I have an 800 number from

Hi All, I have an 800 number from Clearpath. Good folks, highly recommended. I'd like to be able to use the 800 humber for DISA access as well as a published number that I give to my customers. Does anyone here have any example of dialplan logic that would handle normal 800 incomming calls, then allow me to press a key, enter pin etc to get into my DISA sequence? Thanks, Michael --

My setup is this: Analogue phone attached to a Linksys PAP2 | Asterisk | VoIP provider I have put the PAP2 in 'batphone' mode where when you pick it up it immediately dials the 's' extension in the pap2_incoming context in Asterisk, where asterisk answers the call and does a DISA(no-password, internal). I do this because it means I can centralise all of my dialplan logic in

On Wed, 1 Dec 2004 12:37:13 -0800 (PST), Ben Kirkpatrick wrote: > Do you find it difficult to manage four LD providers? > Can you show me part of your LD Macro and how it's used? > > I'm toying with two LD providers now, but don't have failover setup. >Just using each one for what they are best at (least cost). > >Thanks, >--Ben Kirkpatrick > > Not

Hi All, Has anyone on-list tried using a USB style adapter like the VTA-1000 to provide some form of gateway from Skype into *? If so, how well did it work? Would I need to combine the VTA-1000, which appears to be USB-to-RJ11 and a standard FXO port? Would it be possible to provide a more direct approach? Some of my associates use Skype, which I'd prefer to avoid. However a single

Manny, You really need to try Astlinux. See www.astlinux.org. It does pretty much what you desire. Also see my recent article about Aslinux embedded on a Soekris Net4801 (http://www.tomsnetworking.com/Sections-article153.php) Michael Graves Sr Product Specialist Pixel Power Inc mgraves@pixelpower.com o(713) 861-4005 o(800) 905-6412 f(713) 864-8668 c(713) 201-1262 > -------- Original

Hello All, I have a small * server in my home office with several IP phones. The system is not fully in service yet as I'm still hunting for a cost effective FXO adapter that I can rely upon for my two primary PSTNs. That said, I'd like to move it into service for another application...which brings up a question. I'd really like to stop making international calls from my cell phone

OK. Now I show my ignorance. What's the difference between a T-1 and a channelised T-1? I see that Covad's voip service (formerly GoBeam) requires a channelised T-1. Then I read recently on the list that many T-1s being installed are actually HDSL....which would be not a T-1 at all...right? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist

Hello, all. We are having a problem where audio for sip channels is dropping upon reinvite. Perhaps it reflects a misunderstanding of what reinvite does. We are running Asterisk 1.6.1.1 on CentOS 5.3. SIP is set to canreinvite=nonat. We have tried RTP with strictrtp set to both yes and no. We have also tried extending the Asterisk rtp port range to accommodate the differing default ranges of

Hello all. I am currently using Asterisk 11.7.0 (also tried Asterisk 12, but same behavior) and is having an issue when it comes to reINVITE on BYEs. Apparently one of the SIP providers that I am using does not always process reINVITEs correctly, and would return a 500 Internal Server Error message on some (but not all) of these transactions. To get around this issue, I have been using

Hi, I put reinvite=yes in my sip.conf. For testing, I restricted the codecs to alaw. I have no modifiers in my dial command. Thus, there should be no reason not to reinvite. Call (sip, authenticated) comes in and is forward via SIP (not authenticated) to another asterisk box. Unfortunately, media path still passes through the asterisk box in the middle. Using sip debug I even can't find

Hi all, So the scenario is: A -> Asterisk -> B after B send back 200 OK Asterisk is answering the call to A. Directly after the Answer Asterisk generates a ReInvite to A and the only difference between the 200 OK sdp and the reInvite sdp are the offered codecs which are forwarded from B to A. Here i do not understand why this could not be done in the 200OK to A? As far as i understood

How about it. Anyone use these guys? Their rates look ok. I'm looking for alternatives to VoicePulse Connect that provide DIDs in Houston TX. Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves@mstvp.com o713-861-4005 o800-905-6412 c713-201-1262

Is there some concensus on where to set callerid when making outgoing calls via an ITSP over IAX? Is this best accomplished in IAX.CONF or EXTENSIONS.CONF? Also, tech support at one ITSP told me that the SetCIDName command doesn't do anything. Is this something that might be unique to their server? Or a general statement of fact? Thanks, Michael -- Michael Graves

I see that ABPTech are now offering this SIP Wifi phone. Does anyone have an experience with it? I've considered the Pulver/Zyxel wifi phone but would really rather have something that can handle WEP or WPA. I run an 802.11g wlan with 256 bit WPA encryption. This Hitachi phone looks very much like a cell phone, which could be ideal. Michael -- Michael Graves

Is anyone here able to make calls to FWD via IAX? I haven't beenable to for some while. I'd like to get to the bottom of the problem. There's been little response in the FWD support forum thus far. I can call my own number and it rings my server, but I cannot call any other number. It generates and error reporting "everyone is busy at this time." Michael -- Michael Graves

Does anyone on-list have any experience with this device? It's a 6 port FXO<>SIP gateway. Michael Graves -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves@mstvp.com o713-861-4005 o800-905-6412 c713-201-1262 fwd 54245

Does anyone have any experience using this DLink quad FXO <> SIP gateway with Asterisk? I'm still looking for an analog interface that I can live with, having tried X101p, SPA-3000 and TDM400...all with less than desirable results. Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power

Hello, I bought a copy of this software in the hope of bridging Skype into my * box. It sort of works but the audio is all distorted and there's huge latency. Does someone have this working well with their * server? According to the wiki someone does, but I don't know who. I'm needing advise on what to fiddle with to get it working correctly. Thanks Michael -- Michael Graves