Re: [asterisk-users] Where an extension really is (DUNDi woes)

On 6/15/07, Anthony Francis <[EMAIL PROTECTED]> wrote: Kyle Sexton wrote: > I have two servers setup to do DUNDi lookups against each other. The > scenario is that on server A, I have a wildcard match for extensions > 64XX that rings to a local extension on the server. On server B I > have a 6

[asterisk-users] Asterisk to Panasonic TDA200 with Unicall

I am trying to set up a connection from an Asterisk server with a 2 port E1 card to a Panasonic TDA200 pbx. We are using Asterisk 1.4.4, Zaptel 1.4.3 with Unicall. So far we can get the E1 link up and we can send calls from the Panasonic to an IP phone on Asterisk and even through the other

Re: [asterisk-users] combining AGI with dialplans

In my first message I included the example script: > ***test.php*** > #!/usr/bin/php -q > require_once('phpagi.php'); > $agi = new AGI(); > > $dialstr="IAX2/wayne/[EMAIL PROTECTED]"; > $agi->SetVar("JAMES",$dialstr); > exit(0); > ?> I have found why it didn't work. I needed

Re: [asterisk-users] Unicall + MFC/R2 line dropped immediately afterconnect

Peter Gubis wrote: > Hi, > > connection is already established (i can also hear called person for a > while). Problem is, that the line is every time dropped after 1 second. > I assume, that first billing pulse arrives immediately after link is > established and it drops the line. > > Next week I'l

Re: [asterisk-users] Que on A2Billing

Strange... Got it working now... I can receive incoming call... Changed following parameters in additional_a2billing_sip.conf of the DID to: - qualify=yes canreinvite=no Cheers, Nitesh Guillermo Salas M. wrote: > On Fri, 2007-06-15 at 17:42 -0400, Nitesh Divecha wrote: > >> When I call fr

Re: [asterisk-users] Que on A2Billing

Here is my "sip show peers" hyperion*CLI> sip show peers Name/username HostDyn Nat ACL Port Status 2486543210/2486543210 86.14.22.128 D N 61547LAGGED (66 ms) Now here is the catch, before it used to show the status OK but now its showing LAGGE

[asterisk-users] Asterisk 1.4.anything on FreeBSD?

I was very pleased to learn that 1.4.5 has been released. Unfortunately, I have been beating my head against a wall trying to install 1.4.4 on FreeBSD (6.2). If you have been successful in building 1.4.anything (including addons and zaptel-bsd-trunk), could you please respond, on- or off-list wit

[asterisk-users] Asterisk 1.2.19 and 1.4.5 released!

The Asterisk development team is proud to release Asterisk versions 1.2.19 and 1.4.5. There has been a very large number of bugs fixed since the last release, including crashes and other critical issues. There were 244 commits to the 1.4 source tree and 74 commits to the 1.2 source tree since

[asterisk-users] Accessing Voicemail(Asterisk) -- Remotely either from Cell or Landline

Hi, I was wondering if we can check the voicemails remotely from a cell or a landline number. We have SIP 3 Digit Extensions connected to Asterisk server. If users are away from Desk & need to access voicemails can they dial in to Asterisk PBX & check their messages. I know one can check throu

Re: [asterisk-users] calling

Dean, I would like to use it for this application only. Our phone service is currently provided by our parent organization with is a county government unit. Thanks for your reply. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins

Re: [asterisk-users] combining AGI with dialplans

[EMAIL PROTECTED] wrote: > On 2007-05-15 Tony Mountifield wrote (wrt using AGI scripts to dial out): > >> Can't comment on this one, as I never use AGI to dial. >> My AGIs just set the context, extension and priority, >> and exit to the dialplan to do any dialling. > > (http://article.gmane.org/g

Re: [asterisk-users] Where an extension really is (DUNDi woes)

Kyle Sexton wrote: > I have two servers setup to do DUNDi lookups against each other. The > scenario is that on server A, I have a wildcard match for extensions > 64XX that rings to a local extension on the server. On server B I > have a 6442 real extension that I would like to have ring if ca

Re: [asterisk-users] calling

Hi Carol, Yes Asterisk can be set up to provide this service. >From the tone of your question I'm guessing you may not be currently using Asterisk. Is this all you want Asterisk to do? (eg as an application service rather than provide telephony for the rest of the library as well), or are

Re: [asterisk-users] Que on A2Billing

On Fri, 2007-06-15 at 17:42 -0400, Nitesh Divecha wrote: > When I call from my cell to the above DID, it hits on the Asterisk and > I > see A2Billing trying to call SIP/2486543210, but it fails because > Asterisk says "Unable to create channel of type 'SIP' (cause 3 - No > route to destination)

Re: [asterisk-users] Que on A2Billing

2486543210 is my SIP-Friend which I created manually and associated with one of the card number. My ATA is registered to Asterisk using the about DID Number. So I want when I call the above number, it should ring on the ATA. When I call from my cell to the above DID, it hits on the Asterisk and I

[asterisk-users] calling

I would like to find out if Asterisk will allow you to call multiple phone numbers with a pre-recorded message. I currently use a calling system software to contact people to let them know they have a book waiting for them at the public library. I'd also like to know if the software will allow yo

[asterisk-users] scaling with SMP

Is there a way to cause asterisk to benefit from running on a machine with more than two cores? I only see two processes running, with one at a very low priority and the other at a very high priority. I'm guessing one is managing the other. Thanks, Mark _

[asterisk-users] Where an extension really is (DUNDi woes)

I have two servers setup to do DUNDi lookups against each other. The scenario is that on server A, I have a wildcard match for extensions 64XX that rings to a local extension on the server. On server B I have a 6442 real extension that I would like to have ring if called. It seems that DUNDi is

Re: [asterisk-users] Que on A2Billing

Thanks man... That really helped me to move couple of steps. Now I see the incoming calls are going in proper direction... I know I am still missing a small piece here... I did ADD the Destination as a SIP/2486543210, assigned the card number, enabled VOIP_CALL, and enabled Active. When I dial th

Re: [asterisk-users] Que on A2Billing

On Fri, 2007-06-15 at 17:01 -0400, Nitesh Divecha wrote: > Thanks man... That really helped me to move couple of steps. Now I see > the incoming calls are going in proper direction... I know I am still > missing a small piece here... I did ADD the Destination as a > SIP/2486543210, assigned the car

Re: [asterisk-users] Community PBX?

On 6/15/07, Kyle Sexton <[EMAIL PROTECTED]> wrote: I'm wondering if anyone out there is running a community PBX for their local Asterisk User Groups or area Linux groups. I've been thinking of setting one up but am stuck as to what services to provide that people would actually find useful. I

Re: [asterisk-users] Run as root?

Malcom Kemp wrote: > In looking at the safe_asterisk script, it would appear that it is > encouraging the running of the Asterisk application as root user. My > natural inclination is to run it as a non-privileged user. What is > recommendation? > the recommendation is not to run as root, and h

Re: [asterisk-users] app_rxfax vs (iaxmodem+hylafax)

On 6/15/07, Steve Underwood <[EMAIL PROTECTED]> wrote: > Paradise Dove wrote: > > can anybody help me to choose the most reliable fax solution for * . > > after googling the net i found that there are at least two solutions > > for this, app_rxfax+spandsp and iaxmodem+hylafax. > > > > - what's th

[asterisk-users] Community PBX?

I'm wondering if anyone out there is running a community PBX for their local Asterisk User Groups or area Linux groups. I've been thinking of setting one up but am stuck as to what services to provide that people would actually find useful. I know that I could setup simple SIP->SIP to allow ever

Re: [asterisk-users] Que on A2Billing

On Fri, 2007-06-15 at 14:20 -0400, Nitesh Divecha wrote: > You said change the context for SIP Customers to > "context=a2billing-did", do I have to create this context or > A2Billing > will generate by itself? > The a2billing package comes with some examples, you must have to create the a2bill

[asterisk-users] combining AGI with dialplans

On 2007-05-15 Tony Mountifield wrote (wrt using AGI scripts to dial out): > Can't comment on this one, as I never use AGI to dial. > My AGIs just set the context, extension and priority, > and exit to the dialplan to do any dialling. (http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.use

[asterisk-users] FXS card with 3-way call, transfer and call waiting.

Hi, I would like to understand how those features (subject) work on fxs ports. Unfortunately I don't have a digium card with this kind of port, then any help will be appreciated. I tried to gather some information from google and this list history, but I still need some help. 3-way-call - As I c

Re: [asterisk-users] Que on A2Billing

Thanks Man... Do I need to change my context in sip.conf to "context=a2billing" or should I leave it to "context=default"? You said change the context for SIP Customers to "context=a2billing-did", do I have to create this context or A2Billing will generate by itself? Cheers, Nitesh Guiller

Re: [asterisk-users] Transfer caller direct to voicemail

Hi Wes, thanks for the suggestion but I have gone a simpler route suggested by Leonardo Kamache with exten => _*[1-3]XX,1,Wait(1) exten => _*[1-3]XX,n,Voicemail(${EXTEN:[EMAIL PROTECTED]|u) exten => _*[1-3]XX,n,Hangup() I had assumed the "*" would have been eaten by features in features.conf b

Re: [asterisk-users] Recomender Server specs for 250 con-current calls

Transcoding plays a major role you can find some info @ voip-info ... http://www.voip-info.org/wiki-Asterisk+dimensioning ~Vamsi On 6/6/07, Vidura Senadeera <[EMAIL PROTECTED]> wrote: Dear All, I looking to implement asterisk solution for 2000 sip registrations and expecting con-current c

Re: [asterisk-users] Que on A2Billing

On Fri, 2007-06-15 at 12:19 -0400, Nitesh Divecha wrote: > Thanks everyone, > > OK, I got everything working... I manage to create a SIP Customer with a > real DID number and configured an ATA with the DID number. ATA can login > and can make calls out without any issues. > > But incoming calls

Re: [asterisk-users] app_rxfax vs (iaxmodem+hylafax)

Olivier wrote: > > 2007/6/15, Steve Underwood <[EMAIL PROTECTED] > >: > > ... > > The t38modem > program from openh323 does this, and it has to do some nasty > things to > work. :-\ > > Steve > > Is this openh323 project alive ? > Latest news date

Re: [asterisk-users] app_rxfax vs (iaxmodem+hylafax)

On Fri, Jun 15, 2007 at 05:41:36PM +0200, Olivier wrote: > 2007/6/15, Steve Underwood <[EMAIL PROTECTED]>: > > > >... > > > The t38modem > >program from openh323 does this, and it has to do some nasty things to > >work. :-\ > > > >Steve > > > >Is this openh323 project alive ? > Latest news date fro

Re: [asterisk-users] Que on A2Billing

Thanks everyone, OK, I got everything working... I manage to create a SIP Customer with a real DID number and configured an ATA with the DID number. ATA can login and can make calls out without any issues. But incoming calls are failing... As soon as the call hits Asterisk, A2Billing script ru

Re: [asterisk-users] WaitExten not responding on key presses

Jack wrote: > Hi, > > I have the problem that WaitExten is not responding to key presses. You haven't answered the call when announce=0 - you need to answer the call before you can get input information! Change your code to: [hotline] exten => _X.,1,Set(CALLERID(name)=Hotline) exten => _X.,n,

Re: [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes

Matthew J. Roth wrote: > In the meantime, I'm looking for insights as to what would cause > Asterisk (or any other process) to idle at the same value, despite > having similar workloads and twice as many CPUs available to it. I'll > be working on benchmarking Asterisk from very low to very high

Re: [asterisk-users] app_rxfax vs (iaxmodem+hylafax)

2007/6/15, Steve Underwood <[EMAIL PROTECTED]>: ... The t38modem program from openh323 does this, and it has to do some nasty things to work. :-\ Steve Is this openh323 project alive ? Latest news date from 2003 (http://www.openh323.org/) ! ___ -

Re: [asterisk-users] mISDN problem

On 6/13/07, Josu Lazkano <[EMAIL PROTECTED]> wrote: > > How can I saw the status of the ISDN??? > ...try "misdn show stacks" or "misdn show config". You can also increase debug level in /etc/misdn-init.conf... Output will end up in /var/log/asterisk/misdn.log Cheers, -- Ex Vito _

Re: [asterisk-users] app_rxfax vs (iaxmodem+hylafax)

Paradise Dove wrote: > can anybody help me to choose the most reliable fax solution for * . > after googling the net i found that there are at least two solutions > for this, app_rxfax+spandsp and iaxmodem+hylafax. > > - what's the differences between these two? > - which one's better? why? >

Re: [asterisk-users] g729 codec

On Fri, Jun 15, 2007 at 09:12:54AM -0400, Kevin Smith wrote: > Hi everyone, > > Simple question that I haven't been able to find a direct answer to. We > currently have call recording with our asterisk system. The files, I am > assuming since that is the codec we are using, are being recorded in

[asterisk-users] can ENUMLOOKUP query multiple DNS servers without having to replicate the same code for each server?

Hi all, Does ENUMLOOKUP can query multiple DNS servers without having to replicate the same code in which the only thing replaced is the server? If I use ENUMLOOKUP(${exten}), Asterisk will parse enum.conf file to find the list of DNS servers in order of preference to be queried, but, I p

Re: [asterisk-users] Run as root?

Hi Malcom, my advice is to run asterisk as non-privileged user. And do not user safe_asterisk script if u can...it cannot check if it running and you can have many safe_asterisk running in memory...moreover this fills your CLI with a lot of annoying "remote unix connection" messages. Giorgio

Re: [asterisk-users] app_rxfax vs (iaxmodem+hylafax)

Paradise Dove wrote: > can anybody help me to choose the most reliable fax solution for * . > after googling the net i found that there are at least two solutions > for this, app_rxfax+spandsp and iaxmodem+hylafax. > > - what's the differences between these two? > - which one's better? why? >

[asterisk-users] g729 codec

Hi everyone, Simple question that I haven't been able to find a direct answer to. We currently have call recording with our asterisk system. The files, I am assuming since that is the codec we are using, are being recorded in the g729 codec. Is there a way to listen to these calls, say on windo

Re: [asterisk-users] Unicall + MFC/R2 line dropped immediately afterconnect

Hi, connection is already established (i can also hear called person for a while). Problem is, that the line is every time dropped after 1 second. I assume, that first billing pulse arrives immediately after link is established and it drops the line. Next week I'll be able to play around with tim

Re: [asterisk-users] Slightly OT:CSU on Digium cards, and it's requirement

C F wrote: > No, I installed the system, it goes from smart jack to the PRI card in > the Panasonic KX-TDA200 thru the 0290 card > >> When I first started working with T1's, most CSU's were external. I still >> have several of them in storage in fact... and I still use external >> CSU/DSU's on my

[asterisk-users] Run as root?

In looking at the safe_asterisk script, it would appear that it is encouraging the running of the Asterisk application as root user. My natural inclination is to run it as a non-privileged user. What is recommendation? + This e-mail was checked by the TecInfo Content Scanning Servi

Re: [asterisk-users] Slightly OT:CSU on Digium cards, and it's requirement

On 6/15/07, Nick Seraphin <[EMAIL PROTECTED]> wrote: > > On Thu, 14 Jun 2007, C F wrote: > > > > Bottom line is, no matter what the FCC says... and if somehow you managed > > > to get it to work without a CSU... I believe the phone company would have > > > a fit if they knew you connected equipment

Re: [asterisk-users] My Kernel

Did you reboot your box after the kernel upgrade ? - Original Message - From: "bilal ghayyad" <[EMAIL PROTECTED]> To: Sent: Friday, June 15, 2007 1:02 AM Subject: [asterisk-users] My Kernel Hi List; I did yum install kernel and yum install kernel-devel, now when I type 'uname' -a I ha

[asterisk-users] hangup during voicemail announcement drops all calls

Hi, I'm using Asterisk 1.2.18 on a Debian Etch box. I have two phones a Gigaset C450IP and a Snom 360. Suppose someone is calling the Gigaset phone and a second call comes and is redirected to the voicemail: if the new caller hangs up during voicemail announcement, Asterisk drops the first call.

[asterisk-users] Error: Unable to allocate RTCP socket: Too many open files

Hi, I have a Intel Xeon Dual Core server, with 3 GB RAM, running Centos 5.0, Asterisk 1.4.4 and mysql 5.0. It is a kinda high-traffic box, with about 60 concurrent calls. The profile of calls on this box are: Incoming: via a Sangoma A101 via SIP from anothjer SIP server Outgoing all calls tha

Re: [asterisk-users] Sending text to a phone that's no in-use ...

On Fri, 15 Jun 2007, Philipp von Klitzing wrote: > Hi! > Anyone know if it's possible to send a line of text to a phone that's not currently in-use? What I want is: SendText(SIP/101, "Hello World") but that doesn't exist ... >> >> Snom's or Grandstream GXP2000's I'm