Re: [asterisk-users] Installing Asterisk on to CentOS 4

Anthony Francis wrote: Ove Aursand wrote: Abdul wrote: Hi expets, I have installed Asterisk 1.4.11 on CentOS4 successfully without any error. But when i am trying to start asterisk with following cmd i am getting unknown command. [EMAIL PROTECTED] ~]$ asterisk -

Re: [asterisk-users] Flash IDE

Disable DMA on that drive. Thee HD/DOM/CF-card does not support DMA and linux tries to "DMA" it. On 9/11/07, Mojo with Horan & Company, LLC <[EMAIL PROTECTED]> wrote: > > For real! I see the BIOS, then I see GRUB "Loading stage 1.5" and then > a good 60 seconds go by before the kernel and initrd

Re: [asterisk-users] Partitioning DSL input

Although you can find a router with QOS or dedicated bandwidth feature, I would suggest a QOS enabled Switch. Any IEEE802.1p enables switch,(these days less than $100 for 16 port) can do the job. you cant do alot when your traffic reaches internet, thats why most you can do is up to your modem. cos

Re: [asterisk-users] Cisco UC 500

I'm trying to get some more information on this myself as its a new product from Cisco. What i know, Cisco attendant console works with skinny,Cisco page and SLA also works wiht skinny and not SIP. So its either having these or SIP. On 9/10/07, Drew Gibson <[EMAIL PROTECTED]> wrote: > > Jeremy M

Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)

I liked the queue game concept! although it could be cruel! On 9/11/07, Steve Totaro <[EMAIL PROTECTED]> wrote: > > > http://vonmag.com/editorial/web-exclusives/mark-spencer-digium-is-growing-up > > Seems the Adtran relationship goes way back... > > Thanks, > Steve Totaro > >

Re: [asterisk-users] Linux-HA and Asterisk

On Tue, Sep 11, 2007 at 04:30:03PM -0400, Mike Clark wrote: > Yes, the Asterisk boxes were on private addresses. The Polycoms are also > behind a NAT. Yes, I tried using externip in sip.conf and this allowed > registration, and calls to be placed, but no audio. Unfortunately, > Polycom does not

Re: [asterisk-users] Another State Of The Punctuation Mark question - Vonage

On Tue, Sep 11, 2007 at 03:53:56PM -0400, Alex Balashov wrote: > On Tue, 11 Sep 2007, Jeff Bachtel wrote: > > Broadvoice can't handle multiple lines being billed to the same account > > and using the same SIP credentials, which is probably not too large a > > deal for a 4 line install, but would

Re: [asterisk-users] New Project: AskoziaPBX

On Mon, Sep 10, 2007 at 10:04:31AM +0200, Michael Iedema wrote: > This is not a live-cd but rather an image that must initially be > written to a disk, so a dedicated machine is needed. After that, the > entire system is upgradeable through the webGUI. You might want to note: http://www.webgui.

[asterisk-users] bug in 1.2.24

It is not a bug. attended Transfer is using Local channel, if you have a look the debug log from CLI, you will see why it fails. To solve this problem, enable recording before the calls go into the queue. Exten => ,1,MixMonitor(...) Exten => ,2,Goto(ext-queue, , 1) This will ensu

Re: [asterisk-users] TDM400P not answering or making calls

Hi, Thanks for spotting that, however that was a copy-and-paste mistake, 'channel = 3' was is in my zapata.conf. I wish it were that simple! Thanks again, Tom On 11/09/2007, Carlos Rojas <[EMAIL PROTECTED]> wrote: > Heloo, > > I think that your error is: > > zaptel.conf: > > ---

Re: [asterisk-users] Flash IDE

Ed W wrote: > I worried a lot about the same, in the end I went for a small laptop > drive for "safety" (it's inaudible) > > However, this came up on slashdot recently and if you search around the > logic seems to be that: > > - Flash rewrites quite a few times > - The good stuff has wear level

[asterisk-users] IAX2 NAT issues

Hello, I am playing around with IAX2 and I have encountered a problem trying to setup an asterisk box through NAT using IAX2. This is the problem: Asterisk box => Advanced Firewall => Internet => User's router => User The user can register, the server can answer, calls can be made. Asterisk b

Re: [asterisk-users] TDM400P periodic sound clicks on FXS

On Wed, Sep 12, 2007 at 12:33:02AM +0300, Costa Tsaousis wrote: > Matthew Fredrickson wrote: > >Costa Tsaousis wrote: > > > >>Hi, > >> > >>I am having periodic sound clicks (2-3 per second) on all FXS of a > >>TDM400P when the remote end is my VoIP provider. However: > >> > >>- recording the con

Re: [asterisk-users] TDM400P periodic sound clicks on FXS

Costa Tsaousis wrote: > Matthew Fredrickson wrote: >> Costa Tsaousis wrote: >> >>> Hi, >>> >>> I am having periodic sound clicks (2-3 per second) on all FXS of a >>> TDM400P when the remote end is my VoIP provider. However: >>> >>> - recording the conversation on the asterisk, does not have the

Re: [asterisk-users] TDM400P periodic sound clicks on FXS

Matthew Fredrickson wrote: Costa Tsaousis wrote: Hi, I am having periodic sound clicks (2-3 per second) on all FXS of a TDM400P when the remote end is my VoIP provider. However: - recording the conversation on the asterisk, does not have the glitches, although I can hear them on a real p

[asterisk-users] Chan_sip Entry

Hello, I am trying to get to Jain Sip softphones to call one another via an Asterisk server. When I call from phone 1 to phone 2 there is audio transmission both ways, but when I call from phone 2 to phone 1 I don't get audio transmission and reception both ways. When I look at the asterisk l

Re: [asterisk-users] TDM400P periodic sound clicks on FXS

Costa Tsaousis wrote: > Hi, > > I am having periodic sound clicks (2-3 per second) on all FXS of a > TDM400P when the remote end is my VoIP provider. However: > > - recording the conversation on the asterisk, does not have the > glitches, although I can hear them on a real phone. > - My VoIP pr

Re: [asterisk-users] 56k modem configuration

Andrea Spadaccini wrote: > Ciao Matthew, > >> I would be very surprised if chan_modem actually works... I don't think >> I've *ever* seen it setup before. > > Well.. So there's no hope to make that modem work with Asterisk, right? Unless someone speaks otherwise, I would say that the most accur

Re: [asterisk-users] New Installed X100p

Steve Totaro wrote: > Matthew Fredrickson wrote: >> Steve Totaro wrote: >>> I have had Digium tech support tell me to do the same thing >> I'm hoping that wasn't the final conclusion in the tech support >> debugging process. If it was, than I am very sorry to hear that, and >> will make note

Re: [asterisk-users] Linux-HA and Asterisk

Jeff Bachtel wrote: > On Tue, Sep 11, 2007 at 10:32:14AM -0400, Mike Clark wrote: > >> We have gotten stuck trying to get a highly available Asterisk cluster >> fully functional. We used Linux-HA with Asterisk 1.4.11 on privtae IP's >> behind the virtual public IP. I got as far as getting phon

[asterisk-users] TDM400P periodic sound clicks on FXS

Hi, I am having periodic sound clicks (2-3 per second) on all FXS of a TDM400P when the remote end is my VoIP provider. However: - recording the conversation on the asterisk, does not have the glitches, although I can hear them on a real phone. - My VoIP provider to my VoIP phones through the s

Re: [asterisk-users] TDM400P not answering or making calls

Heloo, I think that your error is: zaptel.conf: --- fxsks=1 loadzone= uk defaultzone = uk zapata.conf: [channels] language=en context=incoming signalling=fxs_ks busydetect=yes busycount=4 callprogress=no relaxdtmf=yes callwaiting=no callwaitingca

Re: [asterisk-users] Installing Asterisk on to CentOS 4

Ove Aursand wrote: > Abdul wrote: >> Hi expets, >> >> I have installed Asterisk 1.4.11 on CentOS4 successfully without any >> error. >> But when i am trying to start asterisk with following cmd i am >> getting unknown command. >> >> [EMAIL PROTECTED] ~]$ asterisk -vvc >> -bash: asterisk: comm

Re: [asterisk-users] Another State Of The Punctuation Mark question - Vonage

On Tue, 11 Sep 2007, Jeff Bachtel wrote: > Broadvoice can't handle multiple lines being billed to the same account > and using the same SIP credentials, which is probably not too large a > deal for a 4 line install, but would quickly become unmanageable for > anything larger. So it is not e

Re: [asterisk-users] Linux-HA and Asterisk

On Tue, Sep 11, 2007 at 10:32:14AM -0400, Mike Clark wrote: > We have gotten stuck trying to get a highly available Asterisk cluster > fully functional. We used Linux-HA with Asterisk 1.4.11 on privtae IP's > behind the virtual public IP. I got as far as getting phones registered > and being abl

Re: [asterisk-users] Another State Of The Punctuation Mark question - Vonage

On Tue, Sep 11, 2007 at 09:32:06AM -0700, Eric Chamberlain wrote: > For several years now, we've used VoicePulse Connect > for our Asterisk IAX and SIP trunks. > Ravi and KP are both technical guys and know Asterisk extremely well. They'd better be good; their busi

Re: [asterisk-users] Another State Of The Punctuation Mark question - Vonage

On Tue, Sep 11, 2007 at 08:56:53AM -0400, Jay R. Ashworth wrote: > There was a flurry of "Vonage is going to unlock SIP" activity last > year; did anything productive ever come of it? > > Are *you* using your Vonage lines directly into Asterisk? > > In lieu of that, for a 4 line small business th

[asterisk-users] bug in 1.2.24

GUys.. I dont know if this is a known bug or not but I just tested and replicated this one over and over again. It involves call transfer from calls that entered the pbx via a queue.. say a call comes in and its thrown in a queue, somebody answers the call but then wants to transfer the call to so

[asterisk-users] TDM400P not answering or making calls

Hello, I have recently purchased a TDM400P card with one FXO expansion card, and I'm having problems. The card does not pick up incoming calls. Asterisk detects the ringing line and rings various SIP phones as required. When a sip phone answers, the sip user hears nothing and the PSTN user contin

Re: [asterisk-users] Flash IDE

Gordon Henderson wrote: > One thing to note and this might well shaft you is that they use POI mode > rather than DMA (or at least the ones I'm using do) so they will really > crowbar the bus & cpu when doing transfers to/from them, however with only > 4-6 people and not doing much like writing

Re: [asterisk-users] stop log/debug messages into /var/log/messages

Hi Benjamin; I am also interested in the same issue, but I would like to know how you can know where these logs are stored (in which file and path)? I readed that syslog, can you please help me about that? Regards Bilal Ghayad Mobile: 00965 9849460 --- >When you access the A*k console,

Re: [asterisk-users] 56k modem configuration

Ciao Matthew, > I would be very surprised if chan_modem actually works... I don't think > I've *ever* seen it setup before. Well.. So there's no hope to make that modem work with Asterisk, right? Thanks, -- Dr. Andrea Spadaccini Multimedia Technologies Institute - MTI S.r.l. Web: www.x-voice.

Re: [asterisk-users] exit ChanSpy with DTMF

On 9/11/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > > Part of a supervisor menu I'm writing requires that I allow the > supervisor to choose to ChanSpy a channel from the main menu then return > back to the menu (dialplan) to choose other options when she's done. Is > there a way to 'exit'

Re: [asterisk-users] Flash IDE

CF flash deviced work fine provided that a) The CF has a wear leveling controller inside (not all do, especially the cheap ones) so even a ext2 filesystem wan't create problems b) You use a distro with read only (or partial write) filesystem .i.e logs to ram or remote server etc Other than that w

Re: [asterisk-users] New Installed X100p

Matthew Fredrickson wrote: > Steve Totaro wrote: >> I have had Digium tech support tell me to do the same thing > > I'm hoping that wasn't the final conclusion in the tech support > debugging process. If it was, than I am very sorry to hear that, and > will make note of it. > Thanks, yes,

Re: [asterisk-users] 56k modem configuration

Andrea Spadaccini wrote: > Hello everybody, > I've got a 56k usb modem, lsusb says: > > Bus 002 Device 002: ID 0572:130 Conexant Systems (Rockwell), Inc. > > I'd like to let it work with Asterisk. I think that I should use chan_modem > and/or chan_modem_bestdata, but I found little or no documen

Re: [asterisk-users] New Installed X100p

Steve Totaro wrote: > I have had Digium tech support tell me to do the same thing I'm hoping that wasn't the final conclusion in the tech support debugging process. If it was, than I am very sorry to hear that, and will make note of it. -- Matthew Fredrickson Software/Firmware Engineer Di

Re: [asterisk-users] Flash IDE

Juan Sandro wrote: > Hi > > We have a number offices accommodating 4-6 people each hence it is very > important for PBX to be fanless and silent. We have been looking at using > IDE flash disks also called DOM. The performance tests we have done so far > satisfy our requirements, however we are con

[asterisk-users] SIPAddHeader cmd from Realtime MySQL, not getting all the 'appdata' field

Hi All, I'm doing some simple paging functions and using the SIPAddHeader cmd. * 1.2 branch. Using it in the extensions.conf file, it works fine: exten => _*2XX,1,SIPAddHeader(Call-Info: sip:\;answer-after=0) in * console: lab2*CLI> -- Executing SIPAddHeader("SIP/204-0818dcd0", "Call-Info:

[asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)

http://vonmag.com/editorial/web-exclusives/mark-spencer-digium-is-growing-up Seems the Adtran relationship goes way back... Thanks, Steve Totaro ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocat

Re: [asterisk-users] Another State Of The Punctuation Mark question - Vonage

Vonage has a "business" offering, but they aren't really structured to provide business quality support. I wouldn't use them for a business. For several years now, we've used VoicePulse Connect for our Asterisk IAX and SIP trunks. Ravi and KP are both technica

Re: [asterisk-users] Asterisk on NGINX Server?

>Was the AGI Server to write dialplans in any programming language in >Asterisk assumed to be configured for the apache web server? No, it's not assumed to be for any web server at all... AGI scripts can be written in any language that reads from STDIN and writes to STDOUT, or can listen on a ne

[asterisk-users] dtmfmode rfc2833 and info

I have two asterisk machines setup. M1 is asterisk 1.4.11 connected with a PRI M2 is asterisk 1.2.23 connected to M1 over sip. When M2 calls out through M1 and tries to use SendDTMF() in an agi I get varied results. 1) In sip.conf if dtmfmode=rfc2833 I do not hear the sendDTMF() 2) In sip.conf if

Re: [asterisk-users] Prevent multiple sip registrations

On 9/11/07, Rizwan Hisham <[EMAIL PROTECTED]> wrote: > My requirement is to prevent registrations for aan account if that account > is already registered with a user. That is a perfectly valid requirement. This is not a SIP protocol issue. This is a SIP Registrar implementation/policy issue. If a

[asterisk-users] exit ChanSpy with DTMF

Part of a supervisor menu I'm writing requires that I allow the supervisor to choose to ChanSpy a channel from the main menu then return back to the menu (dialplan) to choose other options when she's done. Is there a way to 'exit' ChanSpy and continue down the dialplan? Or is a caller stuck in Ch

Re: [asterisk-users] canreinvite

The others answered correctly personal I like using rtp debug. As for making sure in the DialPlan that the RTP goes end to end without asterisk. 1. Make sure they both use the same codec and protocol. 2. Don't put any options in app_dial, like tTwW or anything else that will force asterisk to stay

Re: [asterisk-users] canreinvite

> How can I know that the traffic went directly between > the endpoints and did not go via the asterisk? I'm sure there are many ways to do this one way would be to do rtp debug on the cli and watch for media packets another would be to do tcpdump on the command line and watch for packets ther

Re: [asterisk-users] canreinvite

Don't know about IAX. As for SIP, You will know what ip address and port the audios should be transmitted to by looking at the sdp session. Just goto the * console and enable sip debug. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bilal ghayyad Sent: Tu

Re: [asterisk-users] Prevent multiple sip registrations

On 9/11/07, Rizwan Hisham <[EMAIL PROTECTED]> wrote: > The whole point of doing this is because if the user gives away his > username/password to his friends or relative and allows them to use his > account, that way we r gona have a lot more traffic in our asterisk server. > Also we charge our use

[asterisk-users] Linux-HA and Asterisk

We have gotten stuck trying to get a highly available Asterisk cluster fully functional. We used Linux-HA with Asterisk 1.4.11 on privtae IP's behind the virtual public IP. I got as far as getting phones registered and being able to place calls that rang and you could answer, but there was no a

Re: [asterisk-users] Prevent multiple sip registrations

Hmmm. Then SIP is not your solution. SIP servers have no ability to tell one user from another if they share secrets. Strongly suggest that you change the ethos behind how you're manage you're users. Unfortunately, with the business plan as-is, you're using end-user "trust" not to abuse the system

Re: [asterisk-users] Asterisk 1.4.11, res_features.so, SegFault

Open a bug in http://bugs.digium.com/ including all the information you provided here. Also remember to read the bugs guidelines before openning the bug, this might be already reported. Regards On 9/11/07, Bruce McAlister <[EMAIL PROTECTED]> wrote: > Hi All, > > I have a really strange issue occ

Re: [asterisk-users] canreinvite

Dear C F; So in that case, if I placed canrenvite=yes for both endpoint, it is not condition that traffic will be directly via the endpoint while signaling via Asterisk as still Asterisk should detect whethor it is necessary to stay in the path or not? Please advise. How can I know that the traffi

Re: [asterisk-users] Prevent multiple sip registrations

My requirement is to prevent registrations for aan account if that account is already registered with a user. On 9/11/07, Adrian Marsh <[EMAIL PROTECTED]> wrote: > > But then how do you know which is the "correct" user? > This is where the whole point of secrets/passwords should come into > play.

Re: [asterisk-users] Prevent multiple sip registrations

The whole point of doing this is because if the user gives away his username/password to his friends or relative and allows them to use his account, that way we r gona have a lot more traffic in our asterisk server. Also we charge our users a fix amount of money every month for their account so if

Re: [asterisk-users] nat=yes

Dear Benjamin; So in that case, when we set nat = yes? For what we do this? C F, I have nat=yes set by default for all my extensions(with canreinvite=no). And things work fine. Bilal, about Asterisk sending packets to public/private : Asterisk will send packets to the public IP advertised by th

Re: [asterisk-users] Connecting Legacy Pbx With Asterisk With FXS.

On 9/11/07, Olivier <[EMAIL PROTECTED]> wrote: > Hi, > > So, if you dedicate PBX ports to serve as a trunk, you're likely to loose > the abilty to forward DID calls : when a call for an Asterisk user comes > into Panasonic PBX, it will be forwarded to Panasonic FXS trunk ports. > Then, Asterisk sho

Re: [asterisk-users] Prevent multiple sip registrations

But then how do you know which is the "correct" user? This is where the whole point of secrets/passwords should come into play. If no-one else knows his details, then no-one else can register. In the land of IP, you can't even guarantee that a remote ends IP will be the same from minute to minute..

Re: [asterisk-users] Connecting Legacy Pbx With Asterisk With FXS.

You can do that. Just make sure you have a proper dialplan in asterisk, among others make sure you teach your users and configure properly how to transfer back to the Panasonic, you will need to use app_flash with features.conf On 9/11/07, Sanspareils Greenlans <[EMAIL PROTECTED]> wrote: > Sir, >

Re: [asterisk-users] Prevent multiple sip registrations

On Sep 11, 2007, at 7:29 AM, Eric "ManxPower" Wieling wrote: > Rizwan Hisham wrote: >> well he does not have access to hi sip settings, so he cant edit the >> host= every time he moves or registers from anyother >> place. >> Actually he should be able to register from anywhere in the world >>

Re: [asterisk-users] Installing Asterisk on to CentOS 4

Abdul wrote: Hi expets, I have installed Asterisk 1.4.11 on CentOS4 successfully without any error. But when i am trying to start asterisk with following cmd i am getting unknown command. [EMAIL PROTECTED] ~]$ asterisk -vvc -bash: asterisk: command not found [EMAIL PROTECTED] ~]$ I

[asterisk-users] Another State Of The Punctuation Mark question - Vonage

There was a flurry of "Vonage is going to unlock SIP" activity last year; did anything productive ever come of it? Are *you* using your Vonage lines directly into Asterisk? In lieu of that, for a 4 line small business that doesn't need to pay Vonage $150 a month, who? Broadvoice? Someone else?

Re: [asterisk-users] Flash IDE

On Tue, 11 Sep 2007 04:04:27 -0500, Juan Sandro wrote: > >Hi > >We have a number offices accommodating 4-6 people each hence it is very >important for PBX to be fanless and silent. We have been looking at using >IDE flash disks also called DOM. The performance tests we have done so far >satisfy ou

Re: [asterisk-users] Prevent multiple sip registrations

Rizwan Hisham wrote: > well he does not have access to hi sip settings, so he cant edit the > host= every time he moves or registers from anyother place. > Actually he should be able to register from anywhere in the world but once > he has registered with us, i dont want anyone else to register wit

Re: [asterisk-users] Prevent multiple sip registrations

On 9/11/07, Rizwan Hisham <[EMAIL PROTECTED]> wrote: > well he does not have access to hi sip settings, so he cant edit the > host= every time he moves or registers from anyother place. > Actually he should be able to register from anywhere in the world but once > he has registered with us, i dont

Re: [asterisk-users] Prevent multiple sip registrations

well he does not have access to hi sip settings, so he cant edit the host= every time he moves or registers from anyother place. Actually he should be able to register from anywhere in the world but once he has registered with us, i dont want anyone else to register with my asterisk using his crede

Re: [asterisk-users] Flash IDE

On Tue, 11 Sep 2007, Juan Sandro wrote: > > Hi > > We have a number offices accommodating 4-6 people each hence it is very > important for PBX to be fanless and silent. We have been looking at using > IDE flash disks also called DOM. The performance tests we have done so far > satisfy our requirem

Re: [asterisk-users] Prevent multiple sip registrations

On 9/11/07, Rizwan Hisham <[EMAIL PROTECTED]> wrote: > We cant do that. Thats becoz the original user may change his/her location > which will result in change of ip address. We have to set host=dynamic for > allownig the original user to register from anywhere. > So any other ideas? So, if he can

Re: [asterisk-users] Prevent multiple sip registrations

We cant do that. Thats becoz the original user may change his/her location which will result in change of ip address. We have to set host=dynamic for allownig the original user to register from anywhere. So any other ideas? On 9/11/07, Adrian Marsh <[EMAIL PROTECTED]> wrote: > > I believe you can

Re: [asterisk-users] Prevent multiple sip registrations

I believe you can use the host= to configure the allowed IP in sip.conf From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rizwan Hisham Sent: 11 September 2007 11:30 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk

[asterisk-users] Prevent multiple sip registrations

Hi all, Is there anyway i can prevent multiple sip registrations from different IPs using single username in asterisk. Does asterisk provide any aid in this respect? As far as my knowledge is concerned i dont think there is any support for this in asterisk, so i think i'll have to makeup a script w

Re: [asterisk-users] Connecting Legacy Pbx With Asterisk With FXS.

Sir, I want to dedicate two or three Panasonic port to communicate with Asterisk and vise-versa. I am having Panasonic pbx 1232. Rajeev. > Hello, > > 2007/9/10, C F <[EMAIL PROTECTED]>: > > Which Panasonic PBX? > > > > On 9/10/07, Sanspareils Greenlans <[EMAIL PROTECTED]> wrote: > > > Sir, > >

Re: [asterisk-users] Installing Asterisk on to CentOS 4

I installed it using yum from the atrpms repo and it all seems to work. Did you compile from source? On 9/11/07, Abdul <[EMAIL PROTECTED]> wrote: > Hi expets, > > I have installed Asterisk 1.4.11 on CentOS4 successfully without any error. > But when i am trying to start asterisk with following cm

Re: [asterisk-users] Installing Asterisk on to CentOS 4

Add /usr/sbin to your PATH, or run /usr/sbin/asterisk. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or

[asterisk-users] Installing Asterisk on to CentOS 4

Hi expets, I have installed Asterisk 1.4.11 on CentOS4 successfully without any error. But when i am trying to start asterisk with following cmd i am getting unknown command. [EMAIL PROTECTED] ~]$ asterisk -vvc -bash: asterisk: command not found [EMAIL PROTECTED] ~]$ I checked modules and o

[asterisk-users] Flash IDE

Hi We have a number offices accommodating 4-6 people each hence it is very important for PBX to be fanless and silent. We have been looking at using IDE flash disks also called DOM. The performance tests we have done so far satisfy our requirements, however we are concerned with DOM durability.

Re: [asterisk-users] Siemans SIP/PSTN phone S450

Adrian Marsh wrote: > Hi All, > > Just added a Siemens DECT SIP/PSTN S450 phone to login to my A*k server, > and I see "Got SIP response 405 "Method Not Allowed" back from > 192.168.3.64" but the phone seems to work ok. > > Any ideas where it falls over in the SIP protocol? I've included this >

[asterisk-users] Asterisk 1.4.11, res_features.so, SegFault

Hi All, I have a really strange issue occuring where if I run "show dialplan" or "dialplan show" or "dialplan show parkedcalls", then asterisk dumps core. It only appears to happen with contexts that are created within res_features. I am able to display all my other dialplans, but, every time I t