similar to: Asterisk as H323 gateway

Hallo, we have started playing with asterisk about one month ago, and we do like very much what we are experiencing. Now we would like to take some step further towards "standardizing" installed modules, functionalities, tools etc. The "wall" we are facing now is: choosing the right tool for * management. We tried AMP, very powerful but incomplete (CAPI is very important to

Hi all, I have a question about apply function. Is that possible to pass some non-default arguments in the function we want to apply ? For example: if "mat" is a matrix and I want to use the "tabulate" function on its row. The command apply(mat,1,tabulate) works but I have problem with this one apply(mat, 1, tabulate(nbins=4)). Any clue ? Thanks, Eric -- Eric

I'd like to add a custom #pragma directive in clang for research purposes. I'm familiar with the LLVM pass-writing interface but new to the rest of LLVM and clang. I've looked into Simone Pellegrini's work on a #pragma interface (https://github.com/motonacciu/clomp and http://llvm.org/devmtg/2013-04/pellegrini-slides.pdf), but I'm in no position to determine if this approach

I have just bought several Sipura SPA-841 SIP phones, and after some testing I have found out that the volume received by other parties when calling using the handset is very low. I've been able to reproduce this problem in the 3 phones I've tested so far. I've tried tweaking several configuration options but nothing I has helped so far. Has anybody else experienced this problem?

Hi, I am trying to define DID Routes via AMP (last version 1.10.008) I succeded in defining single DID route, one per extension, let's say i.e. DID number 0101234567 set destination to extension 567 DID number 0101234555 set destination to extension 555 and so on Now I was trying to define only one route to a custom application DID number 0101234XXX routes to Custom-App

I am using Asterisk (Debian unstable packages) with an OH323 connection to my provider. Everything is working except for the generation of ringback tones when I receive inbound calls from the PSTN. My provider tells me that we're sending call progress indications and that because of this they're expecting us to generate the ringback tone. Does anybody know how to configure this in

are you giving answer()? ..o-------------------------------------------------------o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Juan Jose Comellas Sent: Friday, September 30, 2005 10:32 AM To: Asterisk Users

Hello everybody, for all of you that have searched for a real fax solution, look no further. We now have T38 faxing. Please contact me for more information. Thanks Michael D. Schelin ShellTel 626-814-2354

We intend to use Asterisk with a very large dialplan (with a lot of functionality for 3000+ users). Each user will be able to change several of his parameters in the dialplan, so we will be forced to reload the diaplan constantly. Has anybody else any previous experience with a similar installation? There are some things that we'd like to know, if anybody can help us. These are: - Is

I'm having some trouble dialing phone numbers in Argentina with Digium Zaptel cards. Does anyone have some sample configuration that works with Digium TDM04B cards in Argentina? I'm mainly referring to the /etc/zaptel.conf and /etc/asterisk/zapata.conf. I have two Zaptel cards: the first one has 1 FXS port and 3 FXO ports; the second one has 4 FXO ports. My current configuration is

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Hi, all. I am building a SER+asterisk PBX airming at around 10k persons' usage. For authentication purpose I am in favor of ldap storage, while I am not sure the current ldap module for asterisk(0.9.9.2) is stable enough? sorry I do not master the proper testing mechanisms to find out myself. Thanks in advance.

Does anyone have any input into integrating asterisk with a alcatel 4400 PBX. Acording to what i've found is that Alcatel uses R2 for E1 -- regards Vikram (http://www.vicramresearch.com)

I am trying to connect two Xten eyeBeam Video Phone No problems in voice connecting. I tryed to modify my sip.conf [general] language=it videosupport=yes ; enable Asterisk video support port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=h263 allow=gsm allow=ulaw allow=alaw ; H.263 is our video codec ;

I am trying to use language=it in asterisk I downloaded the sound package and installed it I added country=it in indications.conf language=it in sip.conf language=it in iax2.conf everything ok in call from sip and from iax The problem arises in outside call, coming trom CAPI Trunk I try language=it in capi.conf: no result: always language=en I found a german forum, and it seems to be a

I have an ISDN card, Billion ISDN PCI Card I tried to use the ZAPHFC, I patched the kernel, I did anything (also followed reccomandation on use on Suse Linux Professional 9.2 --my box is) using bristuff last version. In the end I succesfully compile zaphfc, but I am not able to use the card (a lot of problem running zapcfg, a loto of problem starting asterisk saying about wrong anything (from

I have the following problem when calling from outside my asterisk box : ********************* Extension 'my ISDN phone number' in context 'from-pstn' from '' does not exist. Rejecting call on channel 0/1, span 2 ********************* The card is zaphfc configured (group=2), the calls form internal to outside are perfect. If I had a chan_capi card, I shoud add

Hi all, I was reading http://bugs.digium.com/view.php?id=2639 and it seems that anthm's great native MoH patch only works on HEAD. Does anyone have a version of the native MoH patch that works on 1.0.8? If so please point me to its location or email it off-list. Thanks and regards, Patrick

On 07/04/2012 10:35 AM, Raghavendra, Prakash wrote: > Hi > > I wanted to know the status of OpenMP (or in general auto > parallelization) support in > > LLVM. > > I read some threads discussing about possible effort in that direction > in 3.1 onwards, > > but did not see any further discussion on that. > > If someone in the list knows the updated info or can

Is there any way to have more specific disposition codes in the CDR? Currently there are only 3 values: ANSWER, NO ANSWER, BUSY. In this way, when i call a cell phone that is switched off i get "NO ANSWER", while i would like to be able to log that the call is not answered because "The customer you have dialed is unavailable at the moment". The same for "non