similar to: PBX functions and different channels grouping

Hello, As I understand * don't supports T.38 in Zap channels (please correct me if I'm wrong, BTW is there plans for such support?) I believe it's should support T.38 in "pass-thru" 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

Hi, I need to send and receive faxes over VoIP in realtime. I mean: user ? calls from VoIP network to fax machine on PSTN, but starts voice conversation with user B on that fax machine. Then users agree to send a fax (any direction), pressed "start", completed fax transmission and then continue a voice conversation. This is one of generic ways to use analog fax machine. As I understand

Hi, I'm using D-Link DG-104SH (H323 4 port FXS gateway) with analig phone connected to it and X-Lite softphone as endpoints with * When I calling from X-Lite to analog phone it's ok When I dilaling X-Lite from analog phone, X-Lite si ringind but whei I picked up X-Lite connection drops IP of DG-104SH is 192.168.1.3, H323 ID is GW1 X-Lite number is 233 Here is * output: -- Executing

Hi all, I'm having trouble with H.323 outbound calls, * connects but there is no sound in both ways. I'm using X-Lite as SIP client with GSM codec and dialing to ITSP (which using cisco, I think) over H.323 with G.729 codec. I have 4 digium G.729 licenses installed and this is onli one call. I tested my * with another ITSP over SIP and G.729 codec and there was all ok Here is my configs

Hi, I have Slackware 8 Linux Box with Samba-2.2.5 and HP LJ 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

Hi all, I get "Unknown RTP codec 72 received" message in console when call in progress, but all seems OK. Calls is from X-Lite SIP softphone to PSTN over voicepulse connect (IAX) and to FWD echo test (SIP). But this message only with one SIP client, others (X-Lite too) not giving this message. All X-Lite settings are identical. Asterisk is last cvs version This what I see in console

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

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

Hello, Could somebody tell me is it possible to use asterisk without RTP proxying in SIP<->H323 calls? I mean exactly what canreinvite=yes 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

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

Helo, I have a troubles getting to stable work of Digium TE110P card (mailed some time earlier in the list) - I can't get 100% pseudo zap interface accuracy (zttest), so getting HDLC aborts and call drops. I tried number motherboards, hardware and software configs according to info in wiki, thisl list and number of websites - no luck. So I ask everyboby who successfully use Digium TE110P card

Hello, I have very annoying problem with asterisk 1.4.4: Every evening when I have peak load asterisk crashes, "peak load" is only over 20-30 sip-to-h323 simultaneous calls. Nothing special in logs after crash. Load average never was higher than 0.3, asterisk never uses more than 12% CPU (according to top). Tried SVN versions - same result. Both h323 and sip peers has only one codec

Dear R community, I would like to create my own R package files, but I find some problemm for R versions >1.9. When in previous versions of R I could write a simple text file, to have a functioning file package, now I found that is neccessary to implement also binary copies of the file. I cannot understand, reading from R manuals, how it is the correct procedure to create these binary files.

Dear Community, I've a Work -Shift Scheduling Problem I'd like to solve via constraint linear programming. Maybe something similar to http://support.sas.com/documentation/cdl/en/orcpug/63349/HTML/default/viewer.htm#orcpug_clp_sect037.htm Can anybody suggest me any package/R examples to solve this? If it's needed more details of my little problemm I can provide. Thanks in

Dear all, is asterisk capable to make a call for 24 hour without break ? My dial string in extension.conf is : Dial(SIP/[ext_no]@[pbx_name]) I dont use any dial parameter. The problemm is, my customer complain that the call was cut after 4 hours. Thanks in advance, Ikka Jakarta, Indonesia -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear Dovid, thx for the input. for timer in sip.conf, I used default setting. This is some of the result for "sip show settings" RTP Keepalive: 0 (Disabled) RTP Timeout: 0 (Disabled) RTP Hold Timeout: 0 (Disabled) Session Timers: Accept Session Refresher: uas Session Expires: 1800 secs Session Min-SE: 90 secs Timer

Hello, I try to install on my ubuntu ruby on rails server. I have install ruby,rails,gem and all files but doesn''t work. When i run the server with webrick works perfect but when i use apache i get this message: We''re sorry, but something went wrong. We''ve been notified about this issue and we''ll take a look at it shortly. Here are my installed app versions:

Hello everybody. First of all thanks to all the 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

hi where i should load the module for the trasncoder wctc4XX (lspci shows TC400P) thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Helo, I have problem with Digium TE110P connected to CISCO 3640 (port on NM-HDV-2E1-60) wth PRI E1 link. I use CISCO now for testing but when I tried with real PBX problem was exactly same. I have this messages in Asterisk conole and log sometimes: NOTICE[1115] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Usually 2-5 such messages in series, can be repeated after 10