similar to: Fax and SIP Device

I just can't get kphone to register with asterisk, i can make calls to the demos and even get into the mailbox but kphone cannot register. Here's my story. Can you help me?? Please I have installed asterisk on debian using apt-get install asterisk. I have configured an extension in extensions.conf as follows exten => 8003,1,Dial(Sip/8003,${RINGTIME},rt) exten =>

Hello ! I 'm trying T.38 faxig with spandsp using rxfax/txfax as fax terminal. As another endpoint I 'm using Grandstream HT 486 ATA and a fax machine. Has anybody success with the HT486 as T.38 terminal ? ATA as originator: I managed only onetimes a successfull T.38 fax session. The other times the HT486 did not initiate a re-invite with T.38 parameters. Or shall the Terminator

Before I relate the actual problem, some context. Callcentre environment, a few users testing a new digital dialer... 1. Agents are using Grandstream ATA HT486 and a small analogue dialpad with a headset. 2. SIP connection to Asterisk-1.2b1 3. IAX2 connection to ITSP provider. The call is initially set up in the following way. 1. Agent calls into a meetme conference room and subseqently stays

Does anyone know if the HandyTone 486 has the option to turn the two ethernet ports into either a switch/hub, or does it have to do NAT ? Thank you, Steve Maroney

Grandstream owners, I just noticed that there is a new firmware release, for those that are interested: http://www.grandstream.com/BETATEST/ Doug

Hi, I needed to create config files for downloading to Grandstream devices and made a little script for it. It seems to work fine for the HT486. The script needs a config file (cfg.in) which is in this format: P2 = blah P10 = hrm ...etc... The configfile may contain comments (starting with '#') and empty lines. Mind that the 'gnkey=0b82' shouldn't be in the configfile, as

I read the thread about what IP phone is best for business deployment with great interest. Our need is slightly different however. We are deploying VoiP as a value-add with our high speed internet service and are having trouble finding the right SIP analog terminal adapter. In order to support people's existing phones and wiring we need to use an ATA. 1) The first priority is we want

Hello all, I've started implementing iLBC on some of the ATAs we have floating around clients' homes, but I'm coming against this error message with most of them: codec_ilbc.c: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (38)? The ATAs in question are various Grandstream models - the HT486 being the predominant one. Looking at the list archives, it's

Greetings all; I have an odd problem - Hangup isn't hanging up, instead Asterisk carries the flow going in the extensions.conf, and the next matching extension gets run. Not good. My extensions.conf (highly simplified) looks like this: [pri] include => dids include => SIPlookup [dids] exten => 13015555555,1,Wait,3 exten => 13015555555,2,Answer exten =>

We have a Dlink DVG-1120M and were surprised that it was able to handle 2 simultaneous conversations to 2 seperate phones using only 1 MAC address and 1 IP address. So we asked ourselves..why can't other 1 MAC/1IP devices do this as well? I have a Grandstream 486 that has 1IP and 1MAC. But I don't see anywhere in sip.conf to add a second line to a device. Is this possible? Can this only

Hi All Is there any special configuration needed to send and receive faxes on an ATA device? I am using G711.a with a Grandstream Handytone 486. I can send faxes from a fax machine on the ATA, but receiving doesn't work. I get the fax signal, but it just doesn't continue. The LAN is used purely for VoIP traffic. Garth

Hi. Does anyone know of any small SIP phones (and preferably have some experience of using them and happy to recommend them)? By 'small' I mean a single-piece phone, with dial buttons in the handset, so that it can be carried around easily in a laptop bag. Something like http://maplin.co.uk/images/Full/35493i0.jpg (which is unfortunately just a standard analogue telephone).

Well, my faxes passes through asterisk successfully, however I still have some problems about fax reception by rxfax. The softfax answers, and negotiates transmission, however then as some stage of communiation something is wrong. But I have nothing more but this log: Jun 2 00:10:21 DEBUG[16900]: chan_zap.c:4242 zt_read: DTMF digit: * on Zap/10-1 Jun 2 00:10:22 DEBUG[16900]: chan_zap.c:4242

Hello all, After compiling successfully Asterisk and AMPortal, I cannot make the fax module work. Asterisk does not start (unless I remove the modules or mark them as Noload in modules.conf) with the following error: Jun 3 20:55:25 VERBOSE[3328]: [app_rxfax.so]Jun 3 20:55:25 WARNING[3328]: /usr/local/lib/libspandsp.so.0: undefined symbol: dds_modf Jun 3 20:55:25 WARNING[3328]: Loading

Hi, Is it possible to set a variable for a context for all extensions? I haven't been able to find it. I want something like this in extensions.conf: [from-iaxfwd] exten => .,1,RING=r3 exten => 123456,1,Goto(from-pstn,s,1) [from-internal] exten => .,1,RING=r2 include => ext-local [ext-local] exten => 1,1,Dial(Zap/1,${LONGTIMEOUT}) exten =>

I have been trying to setup faxing with a recent CVS-HEAD. I have downloaded and compiled spandsp-0.0.2pre18 and gotten apps_makefile.patch, app_txfax.c and app_rxfax.c I'm not suprised that the patch failed. Does anyone know what changes need to be made for this to work? I have very little Fedora experience and no experience in changing make files so this is all new to me. Thanks,

I got two phones on an ATA-186 (601, 602) and two phones on the TDM22B (603, 604). I have two lines on the TDM22B. I cannot figure out some of the problems: 1. 601 dials via ZAP/3-1 to local phone number at PSTN: ringing pickup on PSTN (empty) still ringing in the phone set 601 2. call from PSTN back: 601 picks up ... everything works !!! No caller id shows up 3. For testing I have only one

I have been trying to get faxing working with stable but I have had no luck since cvs 1.0.4. I've tried 3 versions of SpanDSP and the system answers the fax but looks like it isn't training properly. Lee -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Bob Goddard Sent: 06 August 2005 23:44 To:

I can dial from 601 to a public number. The public number rings. I pickup and hear nothing, while on 601 it keeps ringing. (BTW, is it right to say "ringing" on the active phone?) The *CLI> doesn't show me anything useful: Executing Macro("SIP/601-8238", "dial-pstn|88097880|") in new stack Executing SetGlobalVar("SIP/601-8238",

Good day! Have a weird problem with HT-286 and Conference room. I use Asterisk CVS-HEAD-06/04/04. Here it is: When HT-286 get into the conference room first and nobody in that room everything seems ok (with any codec HT286 allowed), but when HT-286 get into conference room when somebody already there, have got different HT behavior: 1. When HT use GSM codec => it connects to conference room,