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Good afternoon, I'm trying to write a simple callback context, but i need to hangup an incoming call and then call the origin number back, the problem is that asterisk stops processing the call after Hangup() application then it is not able to dial the origin number back. Sorry for the grammatical erros. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Good Morning, I'm thinking about buying the asterisk six-months online course, Have somebody here that bought that course? What is your opinion? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110606/3008703d/attachment.htm>

Anyone successfully using a HP 8000 DN or HP 5 Lsi (with JetDirect module and assigned IP) with samba? Subject says it all. If you're successfully doing this, what configs made it work? FAQs tell how to set up with a print server, but what if the printer itself is the server? Trust me, the HP doesn't know jack about port 139. Using samba 2.0.4 on Solaris 7 and NT 4.0 SP5 with

Hi everyone, Sometimes i am having problems with Zap channels on asterisk 1.2 (Disc-OS 1.1), after some calls, the channel continues in use, even after hanging the call up, then i need to run the "soft hangup Zap/<zapchannel>" in the asterisk CLI to release the channel. Here is my zapata.conf: [trunkgroups] [channels] language=pt_BR context=default usecallerid=yes

Hi, Does anyone know if CID is already working with Digium TDM800P card using DTMF signalling? (I'm brazillian) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101124/bdcb693c/attachment.htm

Good morning, I have a digium wctdm24xxp in my asterisk box, i am not able to see the callerid when the call is incoming from the fxo line, i live in Brazil, how can i change the signaling from fsk to dtmf? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I would like to receive some suggestions about dialplans written in lua, actually my dialplan is written in ael, but i'm having some problems with it. I noticed that asterisk just translates ael to the old extension language, does it do the same with lua? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm having a weird problem with asterisk (asterisk18-1.8.12.2_1). Every call on the system, whatever it comes from the PSTN or from local extensions, when we hit the '#' button to transfer the call, asterisk just disconnects it, without any error or log, here is my current features configuration: Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8