similar to: CE certification for Europe

Hello, On an Asterisk 1.4.33.1 in a simple scenario: [test] exten => _X.,1,Dial(SIP/12345 at peer01,,,) exten => i,1,Hangup(${HANGUPCAUSE}) exten => t,1,Hangup(${HANGUPCAUSE}) exten => h,1,Hangup(${HANGUPCAUSE}) I have noticed that no matter what value we set in the Hangup(<cause code>) commands, if the call is not answered by peer01 for any reason, the actual cause code

Hello all, We have been testing Asterisk RC2 with the native H323 channel driver. We followed the instructions with the needed OpenH323 and PWLib versions and everything compiled ok. Operation of the driver seems ok, except from 2 main points: 1) Audio is passed between the two ends of the call only after the call is answered. This was not the case with previous versions of Asterisk (0.9.2

Hello, I just tried to compile OH323 0.6.3b on a RH9 machine with Asterisk 1.0 RC2 installed but failed. I applied the patch to the required OpenH323 library according to the instructions, and set the proper directories in the Makefile. Here is what I receive after I issue make: ******************************* g++ -DP_USE_PRAGMA -fno-rtti -ffunction-sections -fdata-sections -D_REENTRANT -

Hello, Does anyone know any solution to this? Or is Asterisk-Dev a more suitable list to ask this question? Best regards, Vlasis Hatzistavrou. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Vlasis Hatzistavrou Sent: Thursday, February 16, 2006 3:43 PM To: asterisk-users@lists.digium.com Cc: 'Vlasis

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I've done quite a bit of googling and haven't found a solution to my problem. I've got the Digium dev kit (wctdm11b) set up and working. I've compiled spandsp and can receieve faxes from eFax (www.efax.com) but the pages are blank. The page count is correct, in that if I fax a two page document, my tiff file has two pages, but they are white blank pages. I found one similar

I'm using get_data in an AGI script and am having a problem when, after a long time in my IVR, when I ask for a 10-digit phone number, the first few tries are always invalid -- the number it reads back is very strange, almost like the DTMF tones from other answers were being cached and then dumped on the call to get_data. Anyone ever experienced this before? I have to do some major

Hi, A glance at sourceforge.net/projects/openh323 Help Forum made me wonder if this location is the one to use (I got trouble in the past when google pointed to an obsolete site) : some quite old messages remain unanswered. Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I've been playing with T.38. I observed that mostly but not always, it's the "calling endpoint" that reINVITE the other party to drop current SIP/G711 session and start a new T.38. But sometimes, it's also the callee party that reINVITE the calling party. Which is the "standardized" or most common, way to start a T.38 session ? Shall it come from callee or

asterisk-users-request@lists.digium.com wrote: >Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can

I know R has a steep learning curve, but from where I stand the slope looks like a sheer cliff. I'm pawing through the available docs and have come across examples which come close to what I want but are proving difficult for me to modify for my use. Calculating simple group means is fairly straight forward: data(PlantGrowth) attach(PlantGrowth) stack(mean(unstack(PlantGrowth)))

I'm little bit confused which Digium hardware is certificated for use in Europe. It looks like new cards are certificated, like TE4XX series. What about TE110 or TDM400P? Can someone confirm that? Thanks, David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060309/1d1c65a1/attachment.htm

Hello, I have been wrestling with installing the CAPI drivers for AVM Fritz in order to use chan_capi with Asterisk. I use an SMP machine, RH 9. I have found rpm's for CAPI and AVM drivers (namely: capi4k-utils-2003.06.16-08.mungo.RH9.i686.rpm and kernel-2.4.20-8- avmfcpci-03.11.02-08.mungo.RH9.i686.rpm), but I believe that they support only single processor machines. I've already

> On 30/10/2018 14:40, Simon Matter wrote: >>> On 30/10/2018 06:46, Simon Matter wrote: >>>>> On 10/29/18 1:55 AM, Simon Matter wrote: >>>>>> To me it seems like, if they are smart, they will try to push IBM >>>>>> POWER >>>>>> and RedHat Linux together to establish real competition in the >>>>>>

Mark Rousell wrote: > On 30/10/2018 17:14, Simon Matter wrote: <snip> > Yup. When I looked at IBM Power machines before (maybe about a year ago, > not sure), there was actually a pricing tool on the website. You could go > through various options for machines (GPUs, CPUs, storage, memory, etc.) > and get a price. Annoyingly I didn't record detailed pricing info but, as

Is there any Ruby on Rails certification available for RoR developers? I am thinking of getting into RoR development, but would like to get certified if possible so I have something to stick on my resume. Thanks

hi, any certification on centos ? if yes where can i buy study materials ?

Out of interest is there any estimated date for the TDM400 FXO modules receiving A-tick certification? And has anyone compared the FXO modules with the X100P on Australian exchanges/equipment? Do they perform any better than the X100? Cheers, Chris Lee

A client asked me to come with a system that will pass certification with PPP in the UK. Google is not being helpful :( It has something to do with recording calls in case the PPP requests a copy. Supposedly their rules were relaxed on August 1st if that helps. Any clues (links?) will be appreciated. -- Thanks in advance,

Hello, Does anyone know whether chan_sip in Asterisk supports DTMF in clock rates other than 8000? I looked for telephone-event/16000 in the changelog and in Jira but no luck. Any help would be appreciated. -- Best regards, Vlasis Chatzistavrou.