similar to: SIP<->H323 calls without proxying RTP

Displaying 20 results from an estimated 300 matches similar to: "SIP<->H323 calls without proxying RTP"

2002 Jul 11
3
Printing from W2K clients
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
2004 Sep 13
4
Unknown RTP codec 72 received
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
2004 Nov 23
1
Fax over SIP Problems (sorry for this topic ...)
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
2007 Feb 12
3
Trixbox vs. Custom install
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
2004 Sep 06
1
T.38 "pass-thru"
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
2004 Jul 25
1
Busydetect problems
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
2004 Sep 17
1
How would you handle a fax without T.38orG.711uLaw?
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
2004 Jul 24
1
Please help I fear I have missed something very important! but what?
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
2007 Feb 11
0
TE110P working hardware configurations
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
2007 Feb 07
9
Digium TE110P
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
2004 Jul 12
0
IP Soft Phone with FAX
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
2004 Jul 24
0
PBX functions and different channels grouping
Hi All, I need to replace old analog PBX with Asteriskl and X-Lise SIP SoftPhones as client phones. First: I have problems with implementation of PBX functions. I need and unsuccesfully tried theese functions (took info at http://voip-info.org/wiki-Asterisk+PBX+functions) Call Pickup: Supported in the standard installation (*8 - defined in res_parking.c +54) - Just don't understand how to
2004 Jul 28
0
D-Link DG-104SH H323 problemm
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
1997 Apr 01
0
R-beta: Re: R-alpha: windows advice
> > Start Up > ======== > Second, under Unix we have a Rprofile in the library directory but users > can also have their own Rprofile files in different directories. Since > R is not launched from a command line it seems hard to arrange this > under windows. > If an image gets saved I was planning to link it (via a double click) to > the R.exe and in that instance I
2004 Sep 20
2
H.323 call problemm (no sound)
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
2007 May 11
0
Asterisk crashes
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
2007 Oct 19
1
Using register => to let Asterisk register to another softswitch via SIP
Hi All; Alot of softswitches or PBX's does not accept to manipulate any SIP call without being registered firstly. So that means, I need asterisk to register firstly then I can route my calls to that SIP trunk. In IAX2, we use the register => , so what shall we do in Asterisk? And how its format will be (if we will use register)? Or what is the solution? Regards Bilal
2010 Dec 03
0
Wine release 1.2.2
The Wine maintenance release 1.2.2 is now available. What's new in this release (see below for details): - Support for animated cursors. - Translation updates. - Various bug fixes. The source is available from the following locations: http://ibiblio.org/pub/linux/system/emulators/wine/wine-1.2.2.tar.bz2 http://prdownloads.sourceforge.net/wine/wine-1.2.2.tar.bz2 Binary packages
2006 Mar 07
3
Re: [asterisk-dev] Is there a way to define an outbound proxy in sip.conf ?
Hello, I use both ser/asterisk . In fact i wish asterisk to forward all the sip requests which are not handled by domain=domain.tld in sip.conf Here is a diagram: The sip agents use the Sip proxy as an outbound sip proxy and domain=domain.tld . When the sip agents dial sip:user@otherdomain.tld so the request is sent to sip proxy and so to Asterisk. I wish Asterisk to Look up the
2003 Nov 12
3
DIAX 0.93 with some sound improvements and not only...
Hi all, DIAX 0.9.3 is available for download from the same place: http://www.laser.com/dante or http://www.geocities.com/tdanro The new DLL contain the latest updates made by Steve in the iaxclient library. Still just IAX1 is supported (for the moment). What's new in 0.9.3? - accept blank passwords; - accept for registration/calls host names, not only IP Address; - password no