similar to: app_queue and ARA

Hello, Is there an app_snmp for asterisk-1.2.3 ? 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

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

Hello, I use both ser/asterisk . In fact i wish asterisk to forward all the sip requests which are not handled by domain=domain.tld in sip.conf Here is a diagram: The sip agents use the Sip proxy as an outbound sip proxy and domain=domain.tld . When the sip agents dial sip:user@otherdomain.tld so the request is sent to sip proxy and so to Asterisk. I wish Asterisk to Look up the

Hello, Asterisk sent back 407 proxy authentication . How can avoid this ? I set insecure=very without success in sip.conf and my sql server . 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

Hello, I wish to set a sip uri sip:info@mydomain. I use ser for authorization and authentication (registrar rtpproxy and outbound proxy) I use asterisk 1.2.5 with realtime . the info is used as a hunt group so i add in extension.conf [info] exten => info,1,Answer() exten => info,n,Dial(Sip/84,10) exten => info,n,Dial(Sip/85,10) exten => info,n,Hangup Ser forward sip:info@mydomain

Hello, my config : aah 2.6 (asterisk 1.2.4) , centos 4.2, 1 TDM11B (1 Fxo / 1 fxs ), 1 phone, 1 softphone I'm in France When someone from PSTN calls and hangs up before the call is answered, internal extension keeps ringing until timeout occurs. PSTN line keeps busy. Hangup detection doesn't work. I've played with different paremeters (callprogress, busydetect, busycount,

Dear User, Anybody could dial these sip uri : sip:info@nxs.yi.org (french) sip:music@nxs.yi.org (music 60s) sip:support@nxs.yi.org (french) Thanks for help ___________________________________________________________________________ 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

Hi all, look at these lines. I created a queue named info when a caller (extension 86) place a call he is put on queue he sould hear MOH . What's the meaning of : Jan 29 14:35:30 WARNING[2591]: file.c:509 ast_openstream_full: File 100 does not exist in any format Jan 29 14:35:30 WARNING[2591]: file.c:821 ast_streamfile: Unable to open 100 (format ulaw): No such file or directory Regards

Hello, I posted a lot of mails may be asterisk is not able to accept sip calls from internet !? My english is not fluent i try my best ! My problem I use ser+asterisk. For local calls there are no problem (PSTN or IP) Now i wish to receive calls from other internet domain but asterisk ask for authentication 407. IS IT possible to Disable authentication for incoming calls to my sip uri ?

The list of your interest is R-help not R-sig-mac stefano Il giorno 26/gen/06, alle ore 01:20, Sylvain Charlat ha scritto: > Hi, > > Is there any simple way to get histogram for different levels of > factor? > > Say you have the following data set: > > Island Sp.diam > Moorea 1.21 > Moorea 1.27 > Moorea 1.28 > Moorea 1.22 > Moorea 1.28 > Rurutu

Hi all, Is there a solution to solve this ? ASTERISK 1.2.4 || Internet===SER/OPENSER=====Nat==[private net] || sip agents rtpproxy/mediaproxy Sip agents use SER/OPENSER as an outbound sip proxy and asterisk as a registar server, pbx functions, ... SER/OPENSER look for domains in URI. if domains are handled by SER/OPENSER

Hi all I'm a newbie in asterisk.I install asterisk server successfully. I configure this server to traverse NAT. Using Xlite clients, i make a call between 2 local networks through Internet.Asterisk server is installed on a host with public IP. client A (in the LAn A) and client B (in the LAN B) are registered. When i make a call from the LAN A to the LAn B, everything goes well.But, when

Message: 6 Date: Fri, 03 Mar 2006 17:32:47 +0000 From: Conrad Wood <asterisk-users@conradwood.netasterisk-users@conradwood.netasterisk-users@conradwood.netasterisk-users@conradwood.net> Subject: Re: [Asterisk-Users] Problem with NAT!!! To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID:

Hi all I'm a newbie in asterisk.I ant to know how i ca configure asterisk to work with the nathelper module of openser to fix the nat problem! Thanks in advance! bets regards Serge --------------------------------- Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international.T?l?chargez

Hi all, May I have to patch asterisk-1.2.x with this patch http://bugs.digium.com/bug_view_page.php?bug_id=0002859 to configure an outbound sip proxy in sip.conf ? Regards Harry ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et

Hi all i want to know where i can find french sounds for asterisk. I don't have any studio to register good sounds. Bests regards Serge ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et

Hello, Here is part of my extensions.conf. I set both absolute and response timeouts according to the "day" context. I wish to asterisk hangup after 60s and 10s to play or replay the annoucement . Asterisk doesn't jump to T extension. How can fiox this problem ? harry ... [day] exten => s,1,Set(TIMEOUT(absolute) = 60) exten => s,2,Set(TIMEOUT(response) = 10) exten =>

Hi all I want to use conference in Asterisk. I configure a conference room in meetme.conf (as conf => 600,1234) and extensions.conf as (exten => 600,1,MeetMe(600,i,1234)) . When i call the extension 600, i have the following message in the asterisk logs: WARNING[7758]: pbx.c:1688 pbx_extension_helper: No application 'MeetMe' for extension (conference, 600, 1) == Spawn extension

Hello, I wish to play a recorded announcement on reply to 4xx 5xx 6xx messages . According to the status a audio file would be played from asterisk server via ser to the caller How can I configure a such feature ? My configuration: Ser act as an outbound sip proxy . Asterisk a sip media server and registrar. sip agents ---- SER -----Asterisk Harry

Hi all I complile asterisk 1.2.4 successfully.I install festival successfully and i configure asterisk to work with festival.But When i call the festival extension configured in extensions, the festival application is executed well (i see it in the log) and must read the text (hello world).But i'm hear no voice. What's the problem? Thanks