similar to: Asterisk as a SIP client, Need to auto-answer

We're looking at using 4 or 8 port T1 cards with echo cancellation and are evaluating brands to go with. We know that Sangoma has excellent solutions especially when it comes to echo. But we still have to hear about actual performance of a Digium card using the same Octasic DSP echo canceller. Would appreciate hearing something on this. --------------------------------- Sponsored Link

Hi Friends, Recently, I bought NOKIA E70 mobile. I have configured my mobile to connect with my Asterisk server depends on the information available in Internet. But, its telling that "Registration failed". If anybody configured this mobile for Asterisk server, please tell me the step by step configuration or please tell me a good website link to do this. Looking forward to your

Hi Friends, Thank you for your response. Yesterday only, I configured my Nokia E70 mobile and its working fine. For group members convenience, here I am giving the configuration: Configuring the Nokia E70: Go to Menu - Tools - Settings - Connection - Sip Settings - Profile name: Olivetalk Service Profile: IETF Default Access Point: Olive Public user name: sip:102@202.xxx.xxx.xxx Use

From: "Time Bandit" <timebandit001@gmail.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Date: Tue, 7 Nov 2006 08:53:51 -0500 Subject: Re: [asterisk-users] Follow Me problems > Today we appear to have discovered our first bug. We have an extension > setup to "followme" by ringing that extension

Take a look at freepbx 2.2 beta. We have made both ringgroups and follow-me have a call confirmation option. When used, the ringgroup/follow-me extensions that are outside lines (like your cell phone) must confirm they want the call (press 1 to accept, 2 to decline). All the while the caller hears ringing (or MoH if chosen). If no answer, they are sent on to where ever else you want them to go

Hi all, How can i originate a call from someone outside my sip-network (for example my PSTN home number) to one of my SIP number? I can originate a call from my SIP-network using this parameters in Originate call command : Channel = SIP/0041435215301 Context = default Exten = 00982166501553 Priority = 1 CallerID = 0041435215301 this works with out any problems I initiate a call from one of my

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Ok here is what I''m trying to accomplish. Right now we''ve only 3/4 T for internet access. This new program they want requires a full T just for it. My proposed solution is to bring in a DSL or wireless link for some cheap access for this new application. What I would like to do is setup a linux box with 3 nics. Traffic would go like so client-->linux box-->firewall

Dear all, I have a data.frame that includes a series of demographic variables for a set of respondents plus a dependent variable (Theta). For example: Age Education Marital Familysize Income Housing Theta 1: 50 Associate degree Divorced 4 70K+ Owned with mortgage 9.147777 2: 65

I've got one that I haven't been able to solve. Hopefully someone else has had this issue. I'm using the paging script in free pbx, which appears to: Send a sipheader autoanswer, Create a conferece Add the phone to the conference But if the user hits the page extension, all the phones auto answer, and if they hang-up before the phones join the conference I end up with dozens of

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sorry if I did not reply to this, answers below: --- Josh Green <josh@resonance.org> wrote: > On Tue, 2006-07-25 at 06:37 +0200, Josh Green wrote: > > A user of CRAM (http://swami.sourceforge.net/cram.php) sent in a > bug > > report related to decoding of CRAM files. This issue occurs with > > flac-1.1.2 but not previous versions (such as flac-1.1.1). Note >

Hello, I need to activate the permissive parameter for a asterisk PCI card. But this parameter doesn''t exist in /sys/module/pciback/parameters/ . I tried to activate this option in the boot option (pciback.permissive) ... When I want to use my PCI card in a domU : Nov 20 10:41:55 myhost kernel: pciback 0000:07:01.0: Driver tried to write to a read-only configuration space field at

Hi all, I am using asterisk1.2(vicidial). I am using like pbx . In this how can I confugure the internal conference calls. suppose I have A,B,C,D,E users these all peoples should be internal conferece . for them i was give 101,102,103,104,105 extensions. For this scenario what can I do exact configuration in dialplan and any to edit confugration files please help me . and how can they cut the

hallo, i tried to setup my extentions,conf like this but it never jumps to the busy part (102) asterisk always plays the unavail msg, also when i am connected to another iax channel (conferece room) and no more channel on my client is available. could sombody give me a hint what could be wrong? thanks , alex snd*CLI> -- Accepting AUTHENTICATED call from 81.135.10.114, requested

I am trying to read some data off the zillow site. Newbie to xml, html, parsing and the xml package. I've been able to load the web page I'm interested with the following code but I'm not sure of the next step to get the information I'm interested in into R : library(XML) url <- "http://www.zillow.com/homes/511 W Lafayette St, Norristown, PA_rb" doc <-doc <-

I'm running samba 3.0.11 on SuSE Linux Pro 9.2. RPM's: samba-doc-3.0.11-0.1 samba-client-3.0.11-0.1 samba-3.0.11-0.1 samba-winbind-3.0.11-0.1 libsmbclient-3.0.11-0.1 Kernel : Linux printserver 2.6.8-24-smp #1 SMP Wed Oct 6 09:16:23 UTC 2004 i686 i686 i386 GNU/Linux The server is currently serving up only printing to a network of 170ish users. Printing was originally being handled by

I am using Samba 2.2.3a-12.3 on a Debian server. All the clients but one are XP home edition. I have a share on the server which all workstations are able to see, view, write, edit, etc. After 15 minutes of inactivity the share obtains a red X over the share icon. I am able to reactivate the share by clicking on the red x with no problems. I understand that Microsoft, in their wisdom, has a