similar to: dial pattern

Hello, I look at voip-info for a simple billing application . I wish to calculate price to pay according to the datas stored in cdr table (unixodbc/mysql). what do you advise me ? Harry __________________________________________________ Do You Yahoo!? En finir avec le spam? Yahoo! Mail vous offre la meilleure protection possible contre les messages non sollicit?s http://mail.yahoo.fr Yahoo!

--- hgaillac-sip@yahoo.fr a ?crit : > hello, > > How asterisk could support res_snmp even this module > don't help to monitor all asterisk features? > > monitoring asterisk with snmp would be a good > thing. > Which solution ? > > Harry > --- Kristian Kielhofner <kris@krisk.org> a ?crit : > > > hgaillac-sip@yahoo.fr wrote: > > > I

I upgrade 1.2.7-1 to 1.2.9 but asterisk is not stable I 've lost call SIP<->ZAP. channels. i can't hear sound because of res_snmp.so . Is it a b?ta release ?? I downgrade to 1.2.8 or 1.2.7 I do hope 1.4 will be a real stable realease Harry __________________________________________________ Do You Yahoo!? En finir avec le spam? Yahoo! Mail vous offre la meilleure protection

some body can help me with astrisk server . i have problemes with the message is notice [5483] : chan_iax2.c: 5325 register_verify : no registration for peer 'xxxxxxxxx' from (xx.xx.xx.xx.) can you explan me wath's the master thank's __________________________________________________ Do You Yahoo!? En finir avec le spam? Yahoo! Mail vous offre la meilleure protection

hello to all, I advice you to not use CDRtool from ag-projects : Fisrt ag-projects talk about is product like a gpl software however they don't provide at least some documentation for non commercial users . try to call them !! i'll offer you some money . You can not Call them for some advices ... It's really a bad product don't waste your time to setup it. this enterprise must

Are you making progress?<DIV> <P class=MsoNormal style="MARGIN: 0cm 0cm 0pt" align=center><SPAN lang=EN-US><FONT face="Comic Sans MS" color=#002060 size=3>Dear Friend,</FONT></SPAN></P> <P class=MsoNormal style="MARGIN: 0cm 0cm 0pt" align=center><SPAN lang=EN-US><?xml:namespace prefix = o ns =

hello, how can I do to drop decimal after the comma please for example for tthis line > print(P) [1] 62.000000 1.000000 7.661290 5.200000 17.100000 2.318801 how canI do to keep only 62 1 7.66 5.2 17.1 2.32 thanks __________________________________________________ ble contre les messages non sollicités [[alternative HTML version deleted]]

Hello, Polycom ip soundpoint support ACD login/logout . Can we configure asterisk with polycom ACD support? Regards Harry ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international. T?l?chargez sur http://fr.messenger.yahoo.com

Anyone know how to let asterisk deal with the progress codes coming from the carrier? The problem I am having is when a customer calls an invalid number the carrier tells me the call is invalid via a progress code but doesn't route me to a recording (this number is invalid). Instead they hang up on me causing a fast busy or sometimes hold up the call with dead air for 15 to 30 seconds then a

Hello, I patched asterisk patched with the latest t38 support . I would need some people for tests. Regards harry ___________________________________________________________________________ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services pr?f?r?s : v?rifiez vos nouveaux mails, lancez vos recherches et suivez l'actualit? en temps r?el.

Hello, >From some days ago, when i made changes in web interface to asterisk that comes with trixbox (freepbx), this dont reflect the changes in asterisk configuration. I has reviewed the file permissions in /etc/asterisk and all files are writable to asterisk user. In freepbx all appears to be ok (i dont see any errors...). Anyone can help me with this problem? Thanks in advance, PS.

Hey all, I've been trying to get MoH to work from the line-in on my soundcard, but as of yet have had no success. I found this script that should allow for it to happen: http://www.sineapps.com/news.php?rssid=722 The script, when run as the asterisk user, works properly and streams sound to stdin. But when Asterisk starts MoH it stops it immediately afterwards with no explanation. Has anyone

Hello, Digium does not provide snmp support to monitor their cards ! Anybody has tried Sangoma product A104 Quad T1/E1 or others ? Regards harry ___________________________________________________________________________ Yahoo! Mail r?invente le mail ! D?couvrez le nouveau Yahoo! Mail et son interface r?volutionnaire. http://fr.mail.yahoo.com

Hello everybody, I would like to implement "callback" function. When I call a person and his extension is busy I can press, for example, 5 and get a callback when his phone is not busy anymore. When I create a call file and copy it to spool call folder asterisk makes a call. One problem is that when extension is still busy my phone rings and I get busy tone of the person who I am

Hello, How many digium cards is supported per asterisk server ? Regards Harry ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international. T?l?chargez sur http://fr.messenger.yahoo.com

Un texte encapsul? et encod? dans un jeu de caract?res inconnu a ?t? nettoy?... Nom : non disponible URL : <https://stat.ethz.ch/pipermail/r-help/attachments/20080523/7061f532/attachment.pl>

Can a temporary message in Asterisk voicemail be de-activated so that the "regular" unavailable and busy messages are played. I have several users who are stuck with the temporary message. Thanks Mark

Hello, I'm trying to figure out how to change the redial, thus far if I hit redial it will redial the last called I made that was answered, not the last call I made that was not answer. I'm using Asterisk 1.8 Thanks, Motty

Hi List I need a quality US 800 DID over IAX for my Asterisk server, preferably one that doesn't cost the earth. Any suggestions please? Thanks -- Chris Blunt Entropy IT Ltd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061220/4919f3cb/attachment.htm

Every few seconds I get the following message: == Parsing '/etc/asterisk/manager.conf': Found == Connect attempt from '127.0.0.1' unable to authenticate I'm trying to track down where it's coming from. I've used TCPDUMP & NGREP to monitor 127.0.0.1, no data's flowing. I've tried loading Asterisk with no modules, tried loading with a naked