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2009/3/27 Marco Sambo <derwidtel at gmail.com> > I have to try Skip2PBX, integrated into my Asterisk machine, but it seem > more invasive than Gizmo5 opensky. Doesn't it? Gizmo5.com/opensky is a hosted solution SIP to Skype solution meaning there's no software to install on your system. In minutes the system can...

Hi, I have a problem with TDM2400P card. The card is detected ok, I can make a call but only with pulse dialing (not tone dialing) without hear sounds from the headset. When I receive a call, I can to establish a communication, but without hear sounds from the headset. When I dial any phone key, I can hear dtmf tone. I'm using Elastix 1.5.2. These are my configuration files:

Hi, I'm currently using Asterisk 1.4.23.1, and I have a problem (also on previous version). Sometimes, when I try to do an attended transfer to another internal with default feature *2, Asterisk doesn't make it (it doesn't play 'pbx-transfer'). Sometimes on second time, Asterisk make transfer correctly. I have this problem on variuos type of SIP phones (GrandStream, Aastra,

Hi, I have a question. How can I configure my sip.conf to make a SIP phone busy on incoming and outcoming calls? I explain my problem. When SIP phone receive a call and then I try to call that phone, I find it busy. When SIP phone make a call and I try to call that phone, I find it avaible and it rings but I want to find it busy. I configure sip.conf like following: [10] type=friend qualify=yes

Hi, I've a problem: I can't configure DAHDI with ech canceller OSLEC. I have Asterisk 1.4.24 and DAHDI 2.1.0.2. I compiled also OSLEC. But when in /etc/dahdi/systems.conf I insert value echocanceller=oslec,1-4, command dahdi_cfg -vvvvvvvvvvvv give me an error about oslec. Someone can help me? Thanks Marco -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, someone has used the voice recognition software named Sphinx??? Can he tell me how to use and its performance??? Thanks Marco -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090408/4acd09c5/attachment.htm

Hi all, I try to install FOP. It's very nice. In documentation I red that from my dial plan I can launch a popup window with UserEvent() application. I try to follow FOP documentation but I can't popup anything. My structure is: - server 1: Asterisk system - server 2: FOP system - client On client I connect to FOP panel, but I don't see any popup. Someone can help me to configure FOP

Hi all, I have a question. I have a VPN and I want to use a SIP softphone on my notebook using with asterisk. But I have some problem with firewall and port. Someone knows which ports I should open on my firewall??? I can't connect ... Thanks all. Marco -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I've a problem: I update my asterisk to version 1.4.25, and the attended transfer doesn't work. A call B, B press *2 and voice announce to digit internal and select internal of C. ---- CORRECT ---- A hear music on hold and B talks with C. ---- CORRECT ---- If B press *0, the call return to A. ---- CORRECT ---- if B hangup, ...... also the call hangup Someone can help

Hi at all ! I've a problem and I don't know how to solve it. My configuration is the following: ISDN LINE ---> PATTON (SIP) ---> ASTERISK in asterisk my sip.conf for sip patton account is the following: [general] port=5060 bindaddr=0.0.0.0 context=default language=it limitonpeers=yes notifyringing=yes [acc1] context=fromPSTN_Ext1 type=friend qualifiy=yes host=dynamic

Hi all, I have 2 question. I have a call center queue with 5 agent; the following are the configuration files: *queue.conf* [name_of_queue] musicclass = default announce = queue-name_of_queue strategy = ringall servicelevel = 60 context = callcenter timeout = 60 retry = 5 wrapuptime=15 autopause=no maxlen = 0 announce-frequency = 60 periodic-announce-frequency=30 announce-holdtime = yes

Hi all, I want to try to use a USB phone with Ekiga under Linux (Debian Lenny). It works: I can receive and make calls. But some buttons of USB phone don't work properly. In particular, button *, #, and hangup have wrong key mapping. Someone have tried a USB phone ???? Thamks all Marco -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I have a question: how can I see hints of a remote Asterisk in IAX2 trunk?? I want to set BLF on my phones to look state of other phones also in other Asterisk server. Someone have any idea or solution? I use Asterisk 1.4.24. Thanks all Marco -------------- next part -------------- An HTML attachment was scrubbed... URL: