search for: efendiyev

...J 1200 printer shared by samba (with LPRng). The problemm is: when printing from W2K clients users cannot change print options (like portrait/landscape page orientation, number of copies etc). When printing from Win98 clients all is ok. Could someone help vt with this problemm? -- Sincerely, Elman Efendiyev elman@megacom.com.ua

...; [332] context = 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

...mode. I mean setup 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

...s option do in SIP<->SIP calls. I don't need a transcoding, only a signaling conversion, and this is possible with some softswitches, so i wondering what about asterisk. Same question about H323<->H323 calls I'm using NuFone Network's H323 cahhel Thanks -- Sincerely, Elman Efendiyev PROTECH INC.

...rrect me if I'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

...ke this: Try to 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

...tmfmode = rfc2833 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

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

...rnel and zaptel versions, maybe zaptel.conf and zapata.conf) configuration here. I think this info could be very useful for many people and we can make detail compatibility list in wiki. There is some hw examples list in wiki already but I think not enough detailed. Thank you! -- Sincerely, Elman Efendiyev PROTECH INC.

...verheating and voltage problems with a hardware (controlling over IPMI), this hardware (with another HDD and software versions) worked fine about year with asterisk restarts manually only for a version upgrade. Could somebody point me the way to debug this problem? Thank you! -- Sincerely, Elman Efendiyev

Hello, I'm following the thread "Asterisk@Now vs Trixbox", and I have a similar question: if someone is going to install Asterisk, FreePBX and A2Billing, should you advice him/her to use Trixbox ... or a custom "step by step" installation on a distribution of his/her choice? Thanks Stefano

...EASE COMPLETE 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

Hello everyone! I tried to send a fax over SIP with an Asterisk Server in the middle (no Digium Cards, etc. installed, everything is SIP only, the PSTN-Gateway is external). Whenever I start sending a Fax to a PSTN destination, the Call gets answered and asterisk tries to build a native bridging: -- Attempting native bridge of SIP/sip.westend.com-082fd1b8 and SIP/xxx-3ef8 Then the following

...ace Subsystem: Unknown device 795e:0001 Flags: bus master, medium devsel, latency 64, IRQ 9 I/O ports at e800 Memory at fe6fd000 (32-bit, non-prefetchable) Capabilities: [40] Power Management version 2 I appreciate any ideas. Thank you. -- Sincerely, Elman Efendiyev

...e people giving their opinion on the subject I proposed: "Trixbox vs. custom install". You've all been very helpful. I try to summarize what has emerged from the various messages. Forgive me if I miss or forget something or if I simplify too much some of your messages... - Elman Efendiyev says that you should install from sources if you need customized setup. He suggests using Slackware for Asterisk installations. - Lee Jenkins suggests using a GUI (like Asterisk@Home), while admitting there are some problems with dialplan customization. He has also written his own Asterisk GUI...