similar to: Any asterisk echo demo servers ?

Hi, I'd like to have local extensions accessible through SIP uri (like Joe@company.com), but at the same time for convenince to be also extension with number (like 100) for more convenient dialing thought softphones that support only numeric keys. Can this be done ? Since I'm newbie, I'd really appreciate small example... Thanks in advance, regards, Robert.

Hi, I'm trying to get Asterisk working on P4 2.8 server behind NAT and Firewall (all ports we're set according to instructions) on DSL line. When pbx connects to Digium demo server( I'm located in Slovenia, Europe), it gets first few words, then silence and then comes back when enumerating dial possibilities ("4 for accounting ...). Same happens from SIP or IAX local extension.

Hi, we're testing Asterisk 1 RC 2 behind ordinary router and NAT. Since we're sharing network with web server it seems like voip packets are not coming through fast enough (Digium demo dies after few seconds...). It's the same if I make direct calls (passing Asterisk) so we conclude it's network problem - it also work normally outside our router... I wonder what solutions can we

Hi, I'm trying to implement dynamic routing of incoming calls to local extension if previous outgoing call was unanswered. But after I do Goto to s-NOANSWER, variable ${EXTEN} changes to 's-NOANSWER'. I guess this is normal, but I don't understand why ? How to workaround on this one ? Thanks in advance, regards, Rob. [outbound-capi-ISDN] exten => _0.,1,NoOp(Calling ISDN

Hi, I'd like to use DISA properly for my case - I'd like to handle it right, if user when in DISA doesn't dial any number - how does Asterisk return from DISA cmd ? I'd like to dial some default number if user doesn't dial anything or give him some message - but I don't know what gets executed after DISA if nothing is dialed .... I'm reading this on wiki, but

Hi, I have following problem. Asterisk is connected to ISDN router on BRI interface. ISDN PBX is connected to another channel of BRI interface. Now I'd like to route all incoming calls first to Asterisk and then if caller wants to talk to extension on ISDN PBX then I'd like to route call to another capi channel but free the current one. Is this possible at all or do I need to take 2 capi

Hi, thank you for last info. we've tried to use te110p but failed. We're quite surprised that cable wasn't included with the card as any documentation, at least on HW setup and installation, yet cable pinout for connection to PRI interfaces.... 1. We have followed instructions on your site and from Beronet guide, but card just keeps blinking and nothing happens (also no useful

Hi, I'd like to use Asterisk with ISDN interface and normal analog interface to door phone (or any other low cost connection type to door phone). What would be your recomendations for needed HW in Europe? Is it possible to have this in one PCI card? Are there any lower cost voip door phones? Thanks in advance, Robert.

Hi, I'm testind Diax. I have flashing note about 1 new voice message. Can I hear it somehow from Diax gui, or must I call pbx to get message ? Thanks, Robert.

Hi, I had music on hold working but now don't know what happened. I get : WARNING[...] res_musiconhold.c:334 moh0_exec: Unable to start music on hold (class '') on channel SIP... Any ideas what is wrong ? Regards, Robert.

Hi, I'd kindly ask for any guidance how to setup Netmeeting to work with Asterisk. I've setup Asterisk as Gateway, selected GSM codec, and I'm able to call local extensions (no calls into PBX functions) but get no sound. Any hint, advice ? Anyone using Netmeeting (maybe also windows messenger) with Asterisk sucessfully ? Thanks in advance, regards, Robert.

Hi, I'd like to implement scenario to send user to operator's queue by default (if doesn't dial any extension) but only if there is operator agent logged, so user could get response. If not I'd like to send it to voicemail... Any quick advice ? Thanks in advance, Robert.

Hi, I have P4 2.8 HT machine with Asterisk. I spot 3 processes of mpg321 taking 20% of CPU each even when no call is made to Asterisk or any other PBX activity is on ? Is this behaviour normal ? Regards, Robert.

Hi, I've found approximate same pricing for both. Sipura 2100 seems to have more features... What are differences between those two ? What about their reliability (specially regarding fact, that they deal with analog phones) ? Thanks in advance, regards, Rob.

Hi, I'm curious if anyone is using firefly as SIP client and if qualify=yes works for it. In my case Asterisk just keeps retransmitting of OPTION SIP message and Firefly doesn't seem to respond - but all this could be just wrong settings ? Anyone has working SIP configuration with qualify ? Thanks , Rob.

Hi, what would be general choice between those two options? They are related to two different things: security, performance. But what would I loose if I run as non-root without realtime priority ? Thanks in advance, regards, Rob.

Hi, at the university department we have quite old Panasonic 2+6 PBX (1BRI + 6 POTS for outputs) and 25 local analog extensions. We would like to add Asterisk with 1 fresh BRI line and possibly integrate with existing equipment (we would like to crossover between both pbxses). What would be most efficient way to do this ? Thanks in advance, regards, Rob.

Hi, I spot weird behaviour of latest Firefly 3rd party on my laptop. Sometimes it comes to state that it won't start (hangs on "Initializing" ) and it again works after system restart... Didn't yet figured out how to recreate it..... Any similar experience ? Also - how can I force Firefly to make outgoing calls (also sip or iax calls) through Asterisk ? I'd like to

Check your FW-1 tracker and see if any sip packets are dropped during call initiation. I had this problem and it went away when I upgraded the BT's firmware to the latest (16). Beware, though, that people on the list claim that this firmware breaks functionality of the message button and autoanswer. I haven't checked this yet, cause I can't afford to go back a version. I prefer a

Hi, I'd like to setup oneway connection - so asteriskB can place calls on asteriskA and be safely authenticated with rsa keys. I just don't get any response on asteriskA. I've generated pair of keys: name.key, name.pub and put them on both servers - is it right to only have name.key on asteriskA and name.pub on asteriskB ? I get everybody is busy ... on asteriskB, and none