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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:

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 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

Hello, I try to install Sambo on a Dreambox (Linux 2.6 based on a small PowerPC) I place in attachement the log file Please help me ! Thank you very much for your help Thierry Vorms

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:

I know this issue has been covered with at least 2 different patches, and probably a dozen different discussions, however I'm a bit unclear as to what my options are. I have a DSL line coming in with 8 IP addresses going to an OpenBSD firewall doing 1:1 NAT for machines behind the firewall. My asterisk box is one of these machines, and I'd like to allow foreign SIP clients

Please google my father's name Sambo Dasuki for possible investment of capital in your country Please reply by email on interpi148 at aol.com

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...

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, 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 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:

...ares from the server itself with smbclient, the but windows box never sees it. What I didn't mention is that system is also the internet gateway for the network. It has two network cards, one for the local network, and a second one hooked to a DSL modem. What seems to be happening is that Sambo is sending out broadcasts on the internet link, and never talking on the local LAN network. Which is incredibly unsurprising because TCP/IP routing is configured with the internet link as the default gateway. Now I'm stuck. How can I tell samba to only broadcast on the local domain Thank...

Hi all, in witch way can I put in a log file the asterisk console? I have tried with some settings in file logger.conf but the log not contain the same debug that I can see with "asterisk -rvvv". I need it in debugging purpose for tracking some bug. Thanks Enrico. -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type:

Hi I am in the middle of move a small business over from legacy PABX + PSTN lines to VOIP infrastructure. I borrowed a spa9000 to place between the PABX and the PSTN lines. I have had this going for a while (>5 months) and it has been working fine (some issues with echo and other minor things), which is why I am moving to asterisk. I bought a tdm410 with 3 fxo + fxs. The fxs is connected to

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 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