similar to: Where to find local FXS settings ? [SOLVED]

2009/3/17 Olivier <oza-4h07 at myamail.com> > > > 2009/3/16 Olivier <oza-4h07 at myamail.com> > > Hi, >> >> I'm rather new to this domain so I may be doing stupid things without >> being concious of that. >> >> I've got a Patton MATA I'm trying to setup as T.38 fax adapter. >> Whenever I connect a fax machine (Dell

Hi, Some ATAs (SPA3102, M-ATA, ...) have a long local FXS settings list such as : FXS port gain, Ring Waveform Frequency ... 1. My understanding of these is that those settings define how calls coming from SIP side, trigger a signal which will in turn, ring analog device. is this correct ? 2. Are those values local for each country ? 3. Shall I change them according the analog device plugged

Hi, I'm rather new to this domain so I may be doing stupid things without being concious of that. I've got a Patton MATA I'm trying to setup as T.38 fax adapter. Whenever I connect a fax machine (Dell MFP1815dn) or a phone to it, I can successfully send a fax or talk to the other end. Whenever I connect a fax modem (Dell Inspiron 6400 laptop), I keep getting "No signal. Line is

2009/12/15 Olivier <oza-4h07 at myamail.com> > > > 2009/12/15 Steve Howes <steve-lists at geekinter.net> > > >> On 15 Dec 2009, at 10:42, Olivier wrote: >> > Unfortunately, it seems macro expansion doesn't occur in Line1 tab : >> > when I type $A or $(A) or ${A} or $GPP_A or $UID1 in User ID field >> > (in Line1 tab), asterisk

2009/1/27 Olivier <oza-4h07 at myamail.com> > > 2009/1/27 Olivier <oza-4h07 at myamail.com> > > Hi, >> >> I carefully followed instructions in README file lasting with : >> /root/register >> ... blabla >> asterisk -r >> CLI> restart now >> >> Then asterisk -r fails with : >> # asterisk -r >> Asterisk

2009/2/26 Olivier <oza-4h07 at myamail.com> > I must add I tried spandsp0.0.6xxx as a warning message advised me to do so > (using 0.0.4 would be ok for me but current trunk doesn't allow this > anymore, it seems). > > > 2009/2/26 Olivier <oza-4h07 at myamail.com> > > Hi, >> >> With 0.0.6pre3: >> # ./build.sh >> CMake Warning (dev)

Leg/Transaction Does Not Exist" and obviously not taken into account as endpoint GUI remains unchanged. Looking deeper into this here are : NOTIFY message accepted by S450IP NOTIFY sip:7531 at 192.168.100.197:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK06adc48b;rport From: "asterisk" <sip:asterisk at asterisk>;tag=as4ea953db To: <sip:sip:7531 at

2009/2/5 Olivier <oza-4h07 at myamail.com> > Hi, > > Here http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 is a > table listing ATA/Gateways combinations. > Could anyone successfully set a Patton M-ATA to work with another one, > using Asterisk 1.4 ? > > Is reinvite (canreinvite=yes) necessary or not ? > > Regards > > Replying to myself, I

Replying to myself, I've just read in 1.6.1 announcement that a new Incomplete dialplan application is the one that provides what I'm looking for ... 2008/10/3 Olivier <oza-4h07 at myamail.com> > Hi, > > If my memory serves me right, there was thread (in dev mailing list ?) > explaining how we could implement 2 stages dialing with SIP endpoints: > user dials 1234

2009/3/9 James Sneeringer <jsneerin at gmail.com> > On Mon, Mar 9, 2009 at 11:13 AM, Olivier <oza-4h07 at myamail.com> wrote: > > Anyway, whenever I'm typing make menuselect, app-fax is greyed out as in > my > > opinion, spandsp libriaries have not been found. > > > > Maybe, I should have typed something like (as suggested > >

2008/12/9 Olivier <oza-4h07 at myamail.com> > > > 2008/12/9 Tilghman Lesher <tilghman at mail.jeffandtilghman.com> > > On Tuesday 09 December 2008 09:14:11 Olivier wrote: >> > Hi, >> > >> > In voicemail.conf: >> > ; Supported values: >> > ; 'filename' filename of a soundfile (single ticks around the >>

2009/5/8 Olivier <oza-4h07 at myamail.com> > Hello, > > Here (http://downloads.digium.com/pub/telephony/codec_g729/README) are > instructions to install G729 software. > (I think I followed instructions step by step but g729 license doesn't seem > to show up). > > My question is : > Is the command bellow still up to date ? > > >g729 show I suddenly

2009/12/3 Olivier <oza-4h07 at myamail.com> > Hello, > > How can you parse a comma separated list using function CUT and AEL ? > > I've tried but it displays error message (though is seems to find the > correct value) : > > STRING=101,102 > VAL=${CUT(STRING,\,,1)}; > NoOp(VAL is ${VAL}); > > Cheers > Sorry for the noise but I mixed up with another

(Im' answering cc the list, so the knowledge keeps there, and maybe some more qualified answers become). Am Mittwoch, den 15.10.2008, 18:00 -0700 schrieb Francisco del rosario: > Hey Rodolfo... Need some help from you ... > I need to know what hardware do I need to make SIP calls if I set-up > asterisk > So the situation is that I have a PC and configure the software of my PC to

Hi everyone, I'm hoping you can give me some pointers. I have a requirement to draw multiple (103) xy line plots onto one output device. Ideally the plots should be displayed in a hexagonal grid (example at www.maladmin.com/example.jpg). I can calculate the locations for each waveform but am wondering how to create multiple plotting areas. I have come accross references to a package grid

Hi, Changing the line bellow helped to get incoming calls but I add to remove secret= option in sip.conf (otherwise, Patton wouldn't respond to 407 Auth required challenges). If someone could enable secret and still get incoming calls (in any SmartWare 5.X), please, do not hesitate to share here ... interface sip IF-ASTERISK bind context sip-gateway ASTERISK route call dest-table

2009/12/4 Olivier <oza-4h07 at myamail.com> > Hello, > > Has someone successfully used this QUEUE_VARIABLES() function (in > 1.6.2-rc7) ? > I tried to use it as I'm using SIPPEER() but without success. > > A previous question about it remainded unanswered ( > http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/224466). > > Regards > How can

Hello, I intend to buy a FXO/FXS device from Linksys. I'm thinking about SPA3102. What you guys think about it. Is ok, is working with asterisk, can i use it like voip peer. Thank you for your advice. Jonson. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070117/93bc7fdb/attachment.htm

Hi, Digging on this case : 2009/5/26 Olivier <oza-4h07 at myamail.com> > Hi, > > In my sip.conf, I've got : > [general](+) > ; register=>tcp://trunk4ipbx:password at 192.168.100.129<trunk4ipbx%3Apassword at 192.168.100.129> > register=>trunk4ipbx:password at 192.168.100.129<trunk4ipbx%3Apassword at 192.168.100.129> > > When

"In computer software standards and documentation, the term deprecation = is=20 applied to software features that are superseded and should be avoided.=20 Although deprecated features remain in the current version, their use = may=20 raise warning messages recommending alternate practices, and deprecation = may indicate that the feature will be removed in the future. Features = are=20