|[TK]D-Fender||ariel_, If you aren't running around screaming like the rest of us .... you clearly have failed to comprehend the depth of the problem ;)|
|ariel_||ok I will step out of it.....|
|action||ariel_ goes back to working on his 22 call center asterisk boxes that have full cdr reporting and transfers work fine.....|
|ariel_||goes back to working on his 22 call center asterisk boxes that have full cdr reporting and transfers work fine.....|
|redax||[TK]D-Fender: if I'd call some kind of ForkCDR at the time chan_sip interconnects the cannels, even that would be sufficient|
sorry, but dont understand ariel, there's no ivr at all..
|plasmid||[TK]D-Fender, 'morning or|
|Fladimir_bg||When i set H323 phone with ooh323.conf with Trixbox 2.0, H323->SIP - there is audio and it is wotking but when i call SIP-> H323 the phone is ringing but there is no audio. Any ideas ?|
i created custm trunk but i heard that for Incoming call i have to put context = from-pstn but i don't know where to put this context
|jbot||Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums at http://www.trixbox.org/modules/newbb/|
|docelm0||ACK! TRIXBOX SUCKS!|
|bsd_tech||the new trixbox is ugly|
|Bobocop||is there any variable containing MWI text sent in sip notify? Or do I have to change it in chan_sip.c and recompile?|
|fd__||h'loh, any idea how to extend zaptel or sip error capability from congestion,busy,chanunavail to include numbernotinuse and phoneunreachable ?|
now it's just chanunavail whether a cellphone is just turned off, or the number is not in use
i'd like to record my own text saying "the number you called can't be reached at this time" and "the number you called is not in use" ..
i want to do this with a TDM card AND a sip-trunk to a gsm gateway
tried googling but no .conf will give this to me straight, and i really don't want to edit c code and recompile (trixbox in use)
|elusive||I have setup one asterisk here, with 2 sip accounts, and I am using 2 linksys PAP2, with g729 codecs...|
when I call, the otherside, can hear my voice perfectly, but I the voice of the otherside, arrives like a mess here
any suggestion on what should I do?
|ctp_||hi folks. anyone here knows how many msn's are possible using sipgate? i need 6 msn's for our purposes.|
|fd__||elusive try to see it's the same codec|
i got a similar problem when all was not alaw, there was an ulaw component somewhere in between
trashy, raspy voice, barely made out the words
may I msg you?
I just want to paste 3 lines for you, to have a look
|kiwoneka||good afternoon to all|
i recompiled zaptel 1.4.1, now i dont have dial tone
|THX2000||who needs a dial tone? :P|
|Modcuts||Am i right in thinking that sll and tls manager interface is not folded and part of asterisk distro at the minute?|
|SomethingISODD||Hello all question is there anyway i could setup dids to use them for dialup internet connection??|