similar to: G729 licencing with asterisk, how does it work ??

Hi, I succesfully install spandsp chan_misdn and digium card. the rxfax works fine and I get the fax result by email. I would like to do the same using a Patton gw + zaptel but I can't receive fax anymore, the call comes in from ISDN in the Patton gw, patton sends it to asterisk, asterisk run a macro to make a tif file using rxfax, the tif file is correctly created but with a 0 size the call

Hi everybody I hope that somebody can help me with the following .... I have 2 quadbri cards 2 - 1t0 cards 1 pabx alcatel 4200 I would like to connect my asterisk to the alcatel , I installed bristuff 0.3.0-1p , loaded the zaphfc driver in NT mode configured zaptel and zapata , it works great. then I removed the 1 t0 card, added the quadbri loaded qozap : insmod qozap.ko ports=15 ( 4 ports

Hi everybody, I'm testing asterisk@home 0.4, looks great so far I was working when I have been alerted by a bip comming from the * pc... I connected a screen to it and saw that there was a message which looked like : Message from syslogd@asterisk1 at Thu Feb 10 09:01:00 2005 ... asterisk1 so I stopped asterisk, type mail and got a strange mail saying that user xxxx@yahoo.com could

Hi, Does the G.729 module support adding more licences??

Hi all: I've just purchase some licences of G.729 codecs, and I like to bring up * using /etc/rc.d/init.d script. Does anyone knows how to start in the "old" way? Thanks in advance, Gus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031016/6dd07c4b/attachment.htm

hi, My question is may be a bit out of scope but I don't know where to turn, I have a 1760 with a ccme 24 user licences 1 bri card. I want to configure a bri card in a cisco 1760 on port 0/0, the card is new, seen by the router, show isdn status gives layer 1 desactived , layer not activated, what ever I do, no shutdown command / shutdown command, etc , the green OK light never turn on,

Hi- I'm using IAX between two boxes, where one box is located in US and the second in Europe. I'm trying to achieve the best voice quality and mainly reliability between these boxes and looking for hints and experience of others. Facts: - Asterisk 1.0.7 - RTT varies from 130-170 ms, depends on time and actual Internet throughput Questions: - What is the sugested codec for such setup?

Hi, One of the key features of Asterisk is that we can install it on many hardware platforms. We've done our best to script this installation process, so that, in case of hardware failure, we can re-install Asterisk on another platform. The question I have is how can we adapt our process so that Digium's G729 licences (or other licenced software) could be installed without asking too

Hi, I successfully connected 2 servers via IAX but I'm pulling my hair to connect 2 extra servers , Anyone connected 3 or 4 servers together ? is it possible ? I d like to share the dialplan so _2XXXX goes to server A _3xxxx goes to serverB _4xxxxx goes to server C etc from the 4 servers any example of which one is peer, which one is user or friend would help me :-) thanks jl

I haven't been able to find the answer online, and am not currently able to conduct an experiment to find the answer... I understand that in a SIP call where G729 has been negotiated as the preferred codec, a G.729 licence is not consumed until there is a need to perform transcoding, e.g. play a non-g729 sound, or do voicemail, or enter a Meetme, etc. What happens when a SIP call in progress

Hi I am trying to make this setup work phone1---g729---asterisk1---sip---asterisk2---g729---phone2 I have tried several configurations but none worked I keep getting transcoding errors I have installed one g729 licence on each asterisk, but I can't verifiy because the show g729 command is not available, I use 1.2.17 Do I need 2 g729 licences per asterisk ? Do I need to register

I am having some problems with g729 with SIP and ZAP channels. 1) I have two g729 licences. Very frequetnly (I don?t know what triggers the error) I get the following warnings and error when I try to place a call via SIP to my X100P. The only way to get out of this is through a restart of *. When the error ocurrs there are no other calls in place. Any ideas? Error Opening channel:2 not

Greetings all, I'm trying to improve the codec selection on a few of the asterisk boxes we have to keep the g729 licences free for calls from ATAs that don't support anything apart from g711 and g729. GSM seems to offer noticably inferior call quality (at least when using a softphone + decent headphones), but it's about where I want the bitrate to be. I know there are lots of Speex

Hi, I found E400P quad PRI card quite cheap (749USD): http://www.govarion.com/product_info.php?cPath=1&products_id=2&osCsid=68cdd6e3d08754 in comparison to te410p (approx 1500 USD ) http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TE410P Now newer generation with HW echo canceling emerged (te411p). I'm not sure in what things those two cards

Hi, I am still toying with the idea of going ahead with using the G.729.. Can those using it tell me about some of your experiences using G.729.. Things like and problems you had running it, the voice quality and anything else you can think of... I have read in the archives that asterisk has to be run with -c.. Is this still the case? and if so does this mean that * can't be run using the

Hello, If asterisk.org can't provide you documentations have a look here : http://www.digium.com/index.php?menu=product_detail&category=software&product=ABE I do hope some people understand my posts. Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger

2009/1/27 Olivier <oza-4h07 at myamail.com> > > 2009/1/27 Olivier <oza-4h07 at myamail.com> > > Hi, >> >> I carefully followed instructions in README file lasting with : >> /root/register >> ... blabla >> asterisk -r >> CLI> restart now >> >> Then asterisk -r fails with : >> # asterisk -r >> Asterisk

Hi, I sometimes play around with extreme parameters for distributions and found that qbeta is not always monotone as the following example shows. I don't know whether this is serious enough to submit a bug report (as this example is near to the limitations of floating point arithmetic). Josef > x <- qbeta((0:100)/100,0.01,5) > x [1] 0.000000e+00 1.253990e-201 1.589622e-171

Hello, I'm quite new to R but tried to learn as much as possible in the last few months. My problem is that I would like to estimate the model of Leon et al. (2005). I have shortly summarised the most important equations in the following pdf file: http://hannes.fedorapeople.org/leon2005.pdf My main question is now how could I introduce these two additional terms into the Likelihood

Hi.., we have two asterisk server interconnected to each other through IAX2 trunk in two separate office. with this bellow configuration do we need to have Licensing for using G729 codec???? Office A --------T1 ----- Astrisk TE05P----------------IAX2----------------Astrisk Box -2 | |