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...9;m wrong) I think the best solution would be a software phone with fax support by T38, but would be happy to find software phone able to faxing in realtime by any metod. Is such softwate exists at all? Can anybody suggest me a solution for my problemm? Thanks. -- Sincerely, Elman Efendiyev elman@earlinvest.com

...call ZAP channel, if it's busy (or another problemm with it) then try to call H323 channel, if busy again try to call IAX2 channel I found info only for ZAP channel grouping and dialing channels simultaneously Please help me with theese problemms Thanks! -- Sincerely, Elman Efendiyev elman@earlinvest.com

Hi guys. I have a XP100P Clone , and the busydetect dont work for me.. PSTN---Asterisk---Sip---Asterisk----PBX Any call from pstn side dont disconnect ... I have no disconnect supervision and busydetect dont work... Please Help me. Zapata.conf [channels] echocancel=yes usecallerid=no hidecallerid=no rxgain=0.0 txgain=0.0 signalling=fxs_ks callprogress=no context=entrada channel=>1

...3 callerid = <233> ------------------------------------------------------------------------ ------------------------ When I using DG-104SH and netmeeting (without asterisk) its ok in both directions Could anybody explain what I'm doing wrong? Thanks -- Sincerely, Elman Efendiyev elman@earlinvest.com

...up like this: Hardware gate with T.38 ------ Asterisk ------ Hardware gate with T.38 But I had troubles with this setup (no faxing) while two gates conneted directly with same network path without Asterisk able to faxing without problemms Where I'm wrong? -- Sincerely, Elman Efendiyev elman@earlinvest.com

asterisk-users-bounces@lists.digium.com wrote: > Isn't it possible to use T.38 for interconnecting hardware gates > supporting T.38 with asterisk using SIP REINVITE? > I'm not shure but but think its's might be possible because after > reinvite traffic goes directly from one gate to anotger, not over > Asterisk We've seen a problem here with asterisk. Wehn

...channelsOpen = 0 -- Unknown [111.222.111.222] has cleared the call == Spawn extension (test, XXXXXXXXXXXX, 1) exited non-zero on 'SIP/234-d01b' == H.323 Connection deleted. Could anybody point me what I'm doing wrong Thanks. -- Sincerely, Elman Efendiyev elman@earlinvest.com

Sorry about this, I have been struggling with the basics of my asterisk config. I set up two sip peers and two phones. And I set up lots of dial masks for outgoing calls, all my outgoing calls were working great, however incoming calls were a different matter altogether, I cannot get incoming calls to work. So I have gone back to a very basic FWD config, with one phone which as far as I am aware

...= full-access type = friend disallow = ulaw insecure = no username = 332 secret = xxx host = dynamic nat = yes dtmfmode = rfc2833 callerid = <332> Could somebody tell me whay this "Unknown RTP codec 72 received" means and how to fix it? Thanks. -- Sincerely, Elman Efendiyev elman@earlinvest.com