similar to: make h323 native transfer on stablished call

Hello, there has been a post to this list somewhere arount april which said that it is possible to use a Sangoma BRI A500 card with zaptel and asterisk bristuff. That is, without sangoma_brid and sangoma_mgd daemons and without woomera channels. Could anybody give me a short hint how to configure this? I tried wanpipe-driver + zaptel + asterisk-bristuffed, but I couldn't get zaptel to

Hello Does anybody who have experience in installing the h323 modules in asterisk. I try'd it many times an spend xxx hours to install it but didn't get lucky so far. I have asterisk 1.0.1 running with bristuffed 2.0 I'm willing to pay for this. Sjaak

Folks: does anybody have an idea? how to capture the DTMF digits to a file, after an extn asnwer? then POST it to a url? Regards, JR

Hi, I have the following dialplan. Problem is, if the user authenticates, * starts counting as billable seconds even if i hangup the phone before the called party answers..And also as disposition.. it accepts all calls authenticated as 'answered' If i commentout the authentication line everything works as it should be. How should i use authentication that, it should accept it as aswered by

Guys. Can you dial 800 and 888 toll free numbers using FWD? how do you dial them cause I tried using 1800xxxxx and 1888xxxxx and I simply get a "nobody can asnwer the call" signal on asterisk. Can you dial 800 toll free from FWD?

I have searched for some time but I have not found an asnwer on how to fix the CDR when a call is transferred. The problem is that if someone dials a cell phone and then transfers the call to another extensi?n the CDR for the cell call stops and there is no way to track that the call was transferred so we can bill correctly. Many people have asked this question but there is no answer, only a

Hi to everybody, i tried to find an asnwer before posting this, but most astcc billing issues i searched refer to the case when no billing occurs at all. In my case i get only initial charges and any subsequent minute does not count for billing. In my iax.conf i entered the "notransfer = yes" but nothing changed. My test card and test calls are the following TEST-CARD en N/A

Hi all, I have encountered this problem: if the caller is connected to the callee using Dial() command called from extensions in extensions.conf, there is no problem. But if the same caller and callee are connected using an AGI->exec('Dial'...), the line is disconnected when asnwer. There's a problem bridging. If the codecs are the same on both ends then there is no problem.

In glm, is there a way to specify the family to be exponential, or Gamma but with alpha=1? This is required by Dobson Exercise 4.3d, page 48. But without specifying alpha=1, the asnwer is the same as in the book. If reply to the list, please cc me a copy. Thanks. -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.- r-help mailing list -- Read

Hi All, I'm running the now almost-to-be upgraded R 2.11.1 on a Intel Mac, and on a Ubuntu machine, but the problem I see is the same. I noticed the following behaviour: 407585.91 * 0.8 [1] 326068.7 -- the right asnwer is 326068.728 round(407585.91 * 0.8, 2) [1] 326068.7 -- same issue 407585.91/100 [1] 4075.859 -- the right answer is 4075.8591 I have no saved .Rwhatever in my environment,

This question appears to have been asked previously, but not answered. the last response I can find to this previous thread is here: http://tolstoy.newcastle.edu.au/R/help/04/07/0126.html. The asnwer was to provide debugging info, not an answer. So the problem is that I'm trying to use lda on my dataset. You can download my data here: http://northstar-www.dartmouth.edu/~jgilbert/nolda, I used

Using Digium's "Asterisk Developer's Kit (TDM)", I've been trying to make an outside call by copying sample.call to /var/spool/asterisk/outgoing. I want the VoiceMailMain to run when the call is answered. The call is made correctly but, as you probably know, the application starts as soon as the call is made. I see there are two solutions: Using callprogress=yes in

Hello, I'm having some problems recording calls. Here is the situation. Call come in Zap/1-1, rings through to SIP/101. Call is answered, talk talk talk, hangup Zap/1-1-in.wav is full of loud static Zap/1-1-out.wav sounds fine Call comes from SIP/101, rings to SIP/102 Call is aswered, talk talk talk, hangup SIP..in.wav sounds fine SIP..out.wav sounds fine. So, it's something about

Hello friends of R list, Im a physician and Im not that good in statistics. I have posted similar email in the epi-sig list but no one aswered so far. Im cunducting a systematic review on two diagnostic test for a particular tropical disease. I found a free software that make most of the analysis called MetaDiSc. However there is a paticular analysis that I wuould like to try that it is not

Paul Madley wrote > >Unfortunately (as far as I'm aware) this is a bug in the 1.4.17 >release, and therefore I don't think any config changes will fix it. >We've been told to roll back to our previous 1.4.13 installation. It >also seems to manifest itself in "ghost ringing" as I've called it; >place a call to a SIP extension, then put down the

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Good evening, I have something strange, when I add an ALERT_INFO variable to a ring group, the invite generated contains 2 lines with Alert-Info and my phones return a 400 Bad Request... I've checked in my config files, there is only one line with Set(__ALERT_INFO..... Any idea?? PS: I'm using Asterisk Asterisk 1.4.13-BRIstuffed-0.4.0-test4

I've got a problem to make run Starcraft in local area network . I can see in the ipx screen the name of the network game (sometimes) and when i click it returns me an error I try to play with another computer which runs on win98 Please asnwer me . I read about ipx_route but i don't know how ti use it

I installed samba 1.9.16p11 on a Unix(BSDI 2.1); on my network also exist two NT servers and some Win95 and WfWg machines. Here the fun begin: 1.- my smb.config is: ; Configuration file for smbd. [global] auto services = chris atoledo printcap name = /etc/printcap load printers = yes guest account = pcguest [homes] comment = Home Directories browseable = no guest ok = no

Hi All, In the past I have used Samba in a good way, as now in some different setup with existing FreeIPA for my DNS (which will be seperated later) and user management I'm searching for information what I can do with two together as there is a Ldap difference as know. As I use FreeIPA and I want to avoid Linux and Windows not mixed on the same AD server I want to stay with FreeIPA, also

Andr? Peters writes: > > I have a problem with a user who uses a wireless carrier that keeps > > changing his IP as he travels throughout the city. From the perspective > > of our dovecot IMAP server, the user keeps logging in from another IP, > > and after a short while, hits up against the mail_max_userip_connections > > limit. It takes 30 minutes before those