similar to: Sipura, Asterisk, *0, and Call Waiting

Is it possible to set up the following? public IP Asterisk Server to ata 186 behind a XP server firewall. I think my biggest problem is that I don't know how to make XP forward the RTP port to the private ata address. I would put up some configs, but was hoping that someone one who has this working can share the working configurations incase mine are all messed up (which is likely.)

Dear R People: Is there a way to forecast with GARCH modeling as found in tseries, please? When I use the predict command, I get an output of length 100, regardless of what I put in the n.ahead steps. R Version 1.9.0 Thanks in advance. Sincerely, Laura mailto: lauraholt_983 at hotmail.com download! http://toolbar.msn.click-url.com/go/onm00200413ave/direct/01/

I would like the ability for a sipura sip device to instantly connect to an asterisk server as soon as the sipura sip device goes offhook and before any digits are pressed. This way asterisk can provide the dialtone and the dialplan. This also allows me to play a greeting to the phone before giving them a dialtone. Is there any way to do this, like possibly having the sipura device dial a

I am trying to configure the FXO port on a Sipura-3000 for use with Asterisk. 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

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. /carmi

Hello, One of the Sipura 2k's I'm using has a dialtone that occasionally fades to static when the user picks up the line. Are there any settings that I can check that would affect this? Regards, Christopher

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Well, a bit of a newbie using Asterisk and trying to get some Sipura SPA-841 phones to talk to the Asterisk server. Not doing something right apparently.... If I turn on debug for the extension in the CLI (sip debug ip xxx.xxx.xxx.xxx) I see frequent 5 packet attempts by the server to contact the phone, but seems to always be failing. The status

Hello, In my quest to create several proof of concepts for what can be done with Asterisk, I've run into a bit of a problem. I have a pair of SPA-2000's acting as off premise extensions for an analog line. When a call waiting call comes in, the caller id information makes it though the ULAW codec and displays on the caller id box, however asterisk doesn't seem to want to pick

All- I'm attempting to set up a Sipura SPA-3000 in Sao Paolo, Brazil. Not being a portuguese speaker, I'm having a rough time of finding the relevant information on how to make the thing pick up the PSTN line and make an outbound call. The sipura in question works fine on a bench connected to a POTS line in the US, but is now plugged in in Sao Paolo. The immediate thing I notice is that

Having an issue with the CIFS VFS module. One of our fileserver has a directory structure with hard drives mounted into that directory structure, so --> C:\volumes\case\1 is a hard drive C:\volumes\case\2 is a hard drive If I share out and mount up the volumes folder, I can see the case folder, and if I navigate into that folder it shows '1' and '2' as links to nothing

>So in my sip.conf I put many variations of what I thought should go in >there, >finally including (to no avail): I have had to play this guessing game as well with other codecs/config file settings in general. And I think this email touches on something that has been troubling me for sometime. >You should allow "g726" only. I think one problem area of documentation

Well, after some more poking around, I discovered that this problem was not neccessarily a simple matter of getting Samba properly configured. For the benefit of the list, I've posted the following URL which takes you to a Debian bug report which starts out with a post describing the exact problem I'm seeing. As it turns out, it may have more to do with a problem with file locking

Hi guys, I'm interconnecting an Asterisk box with a Lucent Definity PBX by means of FXO/FXS ports on a TDM2400 card. Everything works well, except for one little thing. Every now and then somebody (from an Asterisk extension) will call another extension on the Lucent Definity PBX and they hit their voicemail. They caller leaves their message (or not) and hangup, BUT the Lucent sometimes

Dear all i have one confusion about how to dial outgoing call through asterisk like when i press 0 i got dial ton of exchange for outgoing call my setup is [sip_phone]-----[*]----[mediant2k]-----[Avaya_PBX]------e1-----[Exchange_PSTN] now i want to setup whn i press 0 in my sip phone i got dialton of PSTN so i can call outside people is there any special configuration to give

I know this isn't an asterisk thing, but since the recommendation to get one came from here I figure lots of people out there have one. I read the docs, and it says that in order to do a blind transfer I should hit "flash", then dial "*__" then the number. Now, how on a normal phone do I dial "asterisk underscore underscore"? Can someone tell me how doing a

Hi all, Thought I should say something - it''s been a little while. :) Anyway, I''m still futzing with how I want the Ruby API to look. The more I look at the Perl version, the more I think it was too faithful to the C API. Right now I''m looking at the trigger() method. For example, I don''t see any reason to make the "type" key take a hash as an

Hi there everyone! I've bought a Sipura SPA 3000, and succesfully connected it to my Mac, where I installed Asterisk 1.4.0. Both ports (FXO and FXS) are well configured). However, living in Brazil, I'd like to know if there are optimal settings to my PSTN that I should enter into the config of the device. I experience a little bit of echo on the FXO probably because I raised the gain of

Hi, I'm trying to install a VB program of genealogy : Visuged; (http://www.visuged.org/accueil.php3 )with the last version of Wine. The installation seems to work fine and I can launch the app. (the appareance is better with the last version). When I want to import a gedcom file and after save it as a mdb base, I get the following message in a box : "<Erreur n?-2147221166>

I suggest getting rid of the hash style arguments fior new() for this package. In this particular case, you already have a class and accessors, so you can just use the yield trick, e.g. rb_yield(self) if rb_block_given_p(). Thus you can do: mm = Mmap.new do |m| m.namespace = file m.swapfile = sfile m.size = 1024 end Looks just as nice, and is a lot less code on the C side of

Dear R People: Here is an interesting question: >library(chron) >xt <- seq.dates(from="01/01/2004",by="days",length=5) >xt [1] 01/01/04 01/02/04 01/03/04 01/04/04 01/05/04 > #Fine so far >as.POSIXct(xt) [1] "2003-12-31 18:00:00 Central Standard Time" [2] "2004-01-01 18:00:00 Central Standard Time" [3] "2004-01-02 18:00:00 Central