search for: espia

<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

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 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

...pickup calls arriving to the console 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

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

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

...rst 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 to all. I've put on line a cvs viewer for asterisk source code. Is based onto the suite horde+chora. The website is http://asterisk.espia-net.net The cvs modules shown are * asterisk * asterisk-addons * zaptel * zapata * libpri * libr2 * libiax * libiax2 * gnophone * phpconfig * gastman all revisions, branch , comments & whatever cvs is has been preserved. this could be a sort of mirror. I've installed such system, since I p...

...ator 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

...d 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

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"

...ll. 'P[(x)]' -- privacy mode, using 'x' as database if provided. 'g' -- goes on in context if the destination channel hangs up 'A(x)' -- play an announcement to the called party, using x as file see last param ... Matteo. -- Matteo Brancaleoni Espia System Administrator Email : mbrancaleoni@espia.it Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): guest@213.140.14.155 - ext 201 Iaxtel: 1-700-56-62458 - ext 201 _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digiu...

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

...egotiation 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's ou...

I typically start asterisk with the safe_asterisk script: 22865 pts/3 S 0:00 /bin/sh /usr/sbin/safe_asterisk 22867 pts/3 S 0:31 asterisk -vvvg -c 22871 pts/3 S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 For-You.mp3 for.m 22873 pts/3 S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 For-You.mp3 for.m But when I do a "stop now" from the CLI, the mpg123 causes a

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:

...ce I was annoyed this morning, I wrote this simple php script to output channel status from asterisk manager. <disclaimer> that's very bad written, nor commented... I wrote that just for fun </disclaimer> and if someone will use that / improve it , just lemme know. http://asterisk.espia-net.net (wrote with php 4.3.3 and depends on Event: StatusComplete, so a recent * cvs version is needed) -- Brancaleoni Matteo <mbrancaleoni@espia.it> Espia - Emmegi Srl

Need some help with modules.conf, and basic RH9 linux skills. I've installed the new TDM04B 4-port FXO card and its working. After a reboot, when I do lsmod I see the wcfxo module but not the wcfxs even though both are listed modules.conf. If I "modprobe wcfxs", then lsmod has both modules showing. The wcfxs module is the last one in the modules.conf. Is the order of entries

I'm using the stable branch. Is voicemail or voicemail2 deprecated? TKS Paul pmahler@signate.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040502/1b0ab572/attachment.htm