search for: matteo

...PROVIDER g711 ----------------------> g711 | g711 ---------------------------> g711 rtp rtp So the problem is that UAC does not hear audio at all. Any idea? (Asterisk version: 1.4.33.1). Thanks in advance, Matteo -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110613/977b15f9/attachment.htm>

...the folder inside the guest is to set the access mode to 'squash' but all file operations are performed by the user running QEMU. If I ran QEMU as root, the passthrough mode works as expected. Is there the possibility to configure the passthrough access also for non-root users? Thanks, Matteo Matteo Lanati Distributed Resources Group Leibniz-Rechenzentrum (LRZ) Boltzmannstrasse 1 85748 Garching b. München (Germany) Phone: +49 89 35831 8724

...it get only the first digit of 10 or 11 ... resulting into bad * or # detection, since they're picked up as 1 . So I think asterisk expects a signal=* or signal=# . but, who's right? asterisk or the budgetones? I haven't found any doc on the net that reports what values to use... -- Matteo Brancaleoni Espia System Administrator Email : mbrancaleoni@espia.it Web : http://www.espia.it Phone : +39.02.70633354 - ext 911 IAX(2): guest@213.140.14.155 - ext 911 or tel:17005662458 - ext 911

Hi. Has anyone experienced with the new firmware .77 ? There's Day Light Saving time now, but haven't time to play with it, till now. Matteo. -- Matteo Brancaleoni Espia System Administrator - IT services Website : http://www.espia.it Email : mbrancaleoni@espia.it

...df1[df1$id==i & df1$t==t,"x"] = x } } df = rbind(df,df1) This is also too slow, and my PC gets stucked. I don't want to go for a matrix, because I'd loose the column names and everything will become too much error-prone. Any suggestions on how to do it? Thanks in advance, Matteo -- Matteo Richiardi University of Turin Faculty of Law Department of Economics "Cognetti De Martiis" via Po 53, 10124 Torino Email: matteo.richiardi at unito.it Tel. +39 011 670 3870 Web page: http://www.personalweb.unito.it/matteo.richiardi/

Matteo, I fully agree with David: please read the posting guide. Anyway, the error message says it all: "the estimated variance of the individual effect is negative". See e.g. the "basic panel" chapter (10 or 11) in Wooldridge's "Econometric Analysis of XS and Panel Data&quo...

...ntil answered. With the m(...) option in the Dial command (like the example below) asterisk provides only music on hold while the phone rings. exten => s,n,Dial(SIP/XXXX,30,m(default)) I can not use queues because the requirements is to have 1 call and not a lot of calls. Thanks in advance, Matteo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100610/926fead3/attachment.htm

Dear R team I have a problem with the installation of the RCOR package; any time I try to install it I get the error message: Error: package 'ROCR' was built before R 2.10.0: please re-install it. What can I do to solve this problem? Thank you Kind regards Dr. Matteo Pettinari Senior Assistant Cardiale Heelkunde matteo.pettinari@uzleuven.be<mailto:matteo.pettinari@uzleuven.be> tel +32 16 34 09 65 fax +32 16 34 46 16 UZ Leuven | campus Gasthuisberg | Herestraat 49 | B - 3000 Leuven | www.uzleuven.be<http://www.uzleuven.be/> [[alternative HTML v...

Hi. I was wondering if the zplex in the dev kit could be configured to have all fxs ports, instead of the standard 8 fxo + 16 fxs. If so, anyone managed to do that? Matteo. -- Matteo Brancaleoni Powered by RedHat Linux 8.0 Linux User #153521 -----BEGIN GEEK CODE BLOCK----- Version: 3.12 GS d? s:- a- C+++ UL++++ P+ L+++ E- W+++ N++ o K- w-- O- M-- V-- PS PE- Y PGP++ t 5 X+ R tv- b++ DI D+ G e h! r++ y ------END GEEK CODE BLOCK------

...a MeetMe() conference room * the callees are Page()d, then the dynamic conference room is connected to the previous one I'm wondering if Asterisk is relaying DTMF (SIP info or RTP) from the caller to the callees. I found option 'F' for MeetMe, but I have no control on Page(). TIA, Matteo

...re up app_queue . So what I expect, when the operator hangs up, his phone will automagically rings playing the announce "from-queue" and bridge it with the call that's waiting. So, I'm correct? Anyone experienced that or could give me a better way to handle that? Thanks a lot, matteo. -- Brancaleoni Matteo <mbrancaleoni@espia.it> Espia - Emmegi Srl

Hi, All Is there a provision for "AbsoluteTimeout" application to notify called and calling party of the reason why the call suddenly ended? This way, the parties will be much better informed, hence they will/should not think that their VOIP/telco provider(s) are providing bad service. Ta SJ

...> >and this happens also for other versions of gdal I have downloaded. > >The biggest problem is that also other packages like spgrass6, spGDAL and >spmaptools are dependent from gdal....so i cannot go on.... > >Can I ask you to help me , please?? > >Thank you > >Matteo > > > > > > > > >

Hia .... i try to use the fonction Getvar of asterisk to get a variable myDNIS that i have define. i use it as follow Action: Getvar Channel: SIP... Variable: myDNIS but asterisk don't know it .i have the response as follow Response: Error Message: Invalid/unknown command does everybody meet this problem . i try all possible combination and nothing help please ..!! :-( thanks in advance

...hether I choose alaw, ulaw, gsm as codec (besides changing cpu load of course). How can I debug it? I'm using A* 1.6.2 and both linphone 2.x and 3.x. I just need a console scriptable softphone, so maybe there's an alternative to linphone (which seemed good enough anyway!)... Thank you, Matteo

...imit the number of concurrent sessions in IAX, globally or on a per-user basis. That could be needed for security purposes (to prevent dos attacks), to limit bandwidth / cpu usage, or to not allow more than N guest connections, for example. Any other VoIP channel support that? (like SIP, MGCP) Matteo. -- Brancaleoni Matteo <mbrancaleoni@espia.it> Espia - Emmgi Srl

<P>hi,</P> <P>Can anybody pls tell me how to activate loggin CDR on mysql db. I tried&nbsp;editing the /etc/asterisk/cdr_mysql.conf file and recompiling asterisk, it didn't work. Normal loggin on Master.csv file works fine.</P> <P>Thank you inadvance,</P> <P>Surajee</P> <P>&nbsp;</P><br> --------------This mail sent

The subject pretty much says it all. I have a customer who would like to have an option where a caller can leave a voicemail in such a fashion that it would be simultaneously delivered to a set of mailboxes all at once--the idea is "trouble ticket" type operation where multiple technicians will *each* get the vm. He prefers that, if we can do it, to a "shared mailbox"

Hi, I have this error, I have a digium card TE110P Tiger3xx When I'm load the dirvers by this command modprobe wcte11xp I got this error "Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected FATAL: Error running install command for wct1xxp " when I'm Lunch the command ztcfg -vv I got this error.

Anyone having problems getting transfer to work here is the answer... It appears the manual is incorrect.. The manual says: 1)Press "Transfer" button. 2)Dial the target extension. 3)Hangup the phone. This will disconnect the call.. Here is how it can be done.. Matteo gave this solution.. (thanks) NOTE:'Use # as Dial Key:' must be set to YES To trasnfer: 1)Press "Transfer" button. 2)Dial the target extension. 3)Press "#" key. The call is transfered and you can hangup. If you are like me and have extensions that use the "#"...