search for: brancaleoni

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

...e . 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. I noticed that if new messages are recorded with voicemail2 , they're not detected by the message waiting indicator, so the mailbox=XXXX param has no effect, and no message waiting is sent to the phone (sip & adsi, or stutter dialtone) Any hint? -- Brancaleoni Matteo <mbrancaleoni@espia.it> Espia - Emmgi Srl

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------

...er 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"

I wish 'A(x)' was available with AgentCallBackLogin!! :( -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Matteo Brancaleoni Sent: Tuesday, February 03, 2004 6:48 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Playing announcement to called user prior toConfirmation show application dial from asterisk cli: <snip> 't' -- allow the called user transfer the calling user ...

I was wondering if the accountcode flag works with sip channels. I was looking into the debug and ,even if I have the line accountcode=XXX into the users sections of my sip.conf, I don't see it logged into the cdr. Matteo Brancaleoni mbrancaleoni@espia.it Emmegi System Administrator EspiA - EMMEGI Srl - e*solution provider Uffici: Via Pascoli, 37 20129 Milano - Italy Sede Legale: Corso Sempione 67 20149 Milano - Italy Tel. +39 0270633354 Fax. +39 0245487890 http://www.espia.it

Did anyone have a way to make codec negotiation work with Asterisk? This is something I would love to have working as well. I won't need PSTN -> G729 mixing. Just SIP -> SIP using G729 for calling remote offices via VPN, but everything else use G711. -----Original Message----- From: Brancaleoni Matteo [mailto:mbrancaleoni@espia.it] Sent: Wednesday, July 16, 2003 11:32 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Sip codec preferences Hi. I'm experiencing a issue (not big, but important) I have an asterisk installation with a buch of sip phones & analog ones....

...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

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

Hi, I'm having random hangup problems with zap channels. If I turn busydetect off in Zapata.conf, * fails completely to detect a user hangup in the middle of a script. On the other hand, if I turn it on, everything works much better, but long calls tend to be hung up without a motive. Any other parameter that I can try? Any #define that I can tweak and recompile? Will callprogress

I've looked in the obvious places but haven't found a definitive answer to the following: can an IAX extension (an Iaxy phone, for instance) do call pickup via *8? Adolfo

...an active call . The right thing to do is to send a CANCEL to A, since we want to abort the pending INVITE. I'm right ? That's a bug in asterisk ? I've found that using the budgetones phone. They'll go crazy if a INVITE is aborted by a BYE instead of a CANCEL. 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------

...ing project). I've started a horde/chora for me , but was wondering if digium staff could start one from the official cvs rep . Or let someone start it, if they enable rsync mirroring of the cvs itself (also useful for backups ;-> ). Any other idea, question, comments? Matteo -- Matteo Brancaleoni <mbrancaleoni@espia.it> Espia - Emmegi Srl

Hi everybody, I would like to create SIP call flow Diagram under Windows. Is anybody know a program to perform it? I have already Ethereal and I would like an explicit diagram just to show where something have problems... Thanks Ignace

...) option, using A(filename). for example: exten => blah,1,Dial(Zap/blah,30,rA(/my/own/announce)Tt) that doesn't break anything ... feel free to blame me for anything bad this patch could do ;) if for the list is ok, I'll submit to the bug tracker, under a feature-request. Matteo -- Brancaleoni Matteo <mbrancaleoni@espia.it> Espia - Emmegi Srl -------------- next part -------------- --- asterisk/apps/app_dial.c 2003-10-08 00:05:43.000000000 +0200 +++ dial-asterisk/apps/app_dial.c 2003-10-08 00:04:20.000000000 +0200 @@ -337,6 +337,7 @@ struct localuser *u; char info[256], *peers,...

Please, don't hate me because I use Redhat. I am aware that I am asking for problems in running Asterisk on Redhat. I recently aquired a nifty server, moved my digium cards, and installed asterisk. I noticed that one of the four processors was being used at 100% and nothing was working. I tracked CPU utilization back to the Asterisk process. Please, help. James

...1152 (sip_new): Unable to allocate channel structure Also wasn't possible to connect via a unix console... And so on... until I restarted the asterisk proc. What can cause that? I'm running CVS-06/22/03-16:32:23 , on a p4 2.4ghz and 512 mb ram, kern 2.4.21 Thanks a lot, 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------