search for: mbrancaleoni

...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

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

...#39;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

<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

...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

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 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

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

...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. I have 2 1 sip phone that...

...wing community seems to be very active these days. Hopefully when the time comes for our county's transition to VoIP we may be able to go for an Asterisk-based solution. -- Tony Kava Network Administrator Pottawattamie County, Iowa -----Original Message----- From: Brancaleoni Matteo [mailto:mbrancaleoni@espia.it] Sent: Monday, 24 November, 2003 12:12 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Picking an open channel (FXO port) for outbound calls hi. use groups :) zapata.conf group=1 signalling=blah channel=1-2 etc etc then in extension.conf, just use exten => _9.,1,D...

Hello, For those interested in overcoming the problem with some NATs and Firewalls in regards to tftp. I found a nice little tftp server here: http://freshmeat.net/projects/jtftp/?topic_id=87 I tried it and it works great. Regards, Andres. -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

...ns 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

...e from another phone) btw, If someone is interested, I can submit a patch to the bugtracker. I won't do it until that's usefult for someone... since is a very special features that probably no one will ever use lemme know. Matteo. -- 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 Iaxtel: 1-700-56-62458 - ext 911

...ease would not have been possible without your help! * Seg fault in chan_local - local_pvt_destroy (closes issue #15314. Reported by sroberts. Tested by davidw, lottc. Patch by davidw.) * T38 reinvite started from Asterisk (closes issue #15373. Reported by dcolombo. Tested by dcolombo, mbrancaleoni. Patch by mbrancaleoni.) * manager keeps creating /tmp/ast-ami-XXXXXX files (without deleting) when a single manager client remains logged in (closes issue #15730. Reported by zmehmood. Tested by zmehmood. Patch by junky.) * BASE64_DECODE() adds garbage end end of decoded string...

...ease would not have been possible without your help! * Seg fault in chan_local - local_pvt_destroy (closes issue #15314. Reported by sroberts. Tested by davidw, lottc. Patch by davidw.) * T38 reinvite started from Asterisk (closes issue #15373. Reported by dcolombo. Tested by dcolombo, mbrancaleoni. Patch by mbrancaleoni.) * manager keeps creating /tmp/ast-ami-XXXXXX files (without deleting) when a single manager client remains logged in (closes issue #15730. Reported by zmehmood. Tested by zmehmood. Patch by junky.) * BASE64_DECODE() adds garbage end end of decoded string...

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

...he .inf for the Digium vendor and device ID. I have not tried this, but since the MD3200 modem works that way in Linux, the X100P may work that way in Windows. Then you would have a $100 winmodem! Let us know what you find out. Jeremy -----Original Message----- From: Matteo Brancaleoni [mailto:mbrancaleoni@espia.it] Sent: Friday, April 16, 2004 9:28 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Windows Drivers for Wildcard FXO Card /me set ban on *winzoz*drivers* they doesn't exist. wildcard are only for linux and only for asterisk. but you can port the driver to window...