[Asterisk-Users] test
sorry test. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] ASTCC: Asterisk Calling Card Solution
Hi Mark. Thanks for release this application! It's works well !! great !!! mack On Wed, 14 Jul 2004 00:24:08 -0500 (CDT) Mark Spencer [EMAIL PROTECTED] wrote: I am hereby announcing the immediate availablity of ASTCC for *alpha* testing. ASTCC is an AGI script and CGI script which greatly simplifies the task of creating a calling card application on Asterisk. Just check it out of Asterisk CVS as module astcc: export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot cvs login(password is anoncvs) cvs co astcc ASTCC is, of course, distributed under GNU GPL. I hope you enjoy! Mark ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[3]: [Asterisk-Users] Video/H323/SIP
no, that mean use with gk.because that isn't bridge by *. Does anyone knows another way can do that? mack_jpn On Thu, 24 Jun 2004 23:10:43 +0900 (JST) Isamar Maia [EMAIL PROTECTED] wrote: Nakano San, Have you tried to make * only to route the connection and they just talk point-to-point without * bridging? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PTHREAD_MUTEX_RECURSIVE in appradius-1.0
Hi all How can I fix this problem? Regards, mack_jpn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Video/H323/SIP
I tryed it. but callee cannot answering with video in SIP. # surely videosupport=yes in sip.conf H.323 is works well but I think stilln't support over * yet. mack_jpn. On Thu, 24 Jun 2004 14:03:10 +0200 Michael Devenijn [EMAIL PROTECTED] wrote: I found this tool, but didn't have the time to test it... http://www.dylogic.com/sito/ArticlesDMD/mirial.html -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of shabanip Sent: donderdag 24 juni 2004 13:59 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Video/H323/SIP Is there any software based solution to establish a video connection with * and sip protocol? - Original Message - Hi, -Original Message- It's already possible to use VideoPhone with Asterisk. I'm planning to buy 2 of them. Anybody using any Video SIP phone with asterisk? Yes, we're using the WVP-2000. Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip videosupport
Hi all I was tryed to connect to mysip.ch scs_client by siemens that isn't works well. Does anyones knows to work H/W or S/W applictations in asterisk SIP videosupport? Regards mack_jpn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Can't compile asterisk.
Hi could you please tell me your enviroment about this problem? and how do you fix it ? please disclose, when are you okay. Regards mack_jpn On Fri, 2 Apr 2004 12:40:52 +0900 岩田 伸介 [EMAIL PROTECTED] wrote: Hi. At last, I can compile asterisk. I had compiled low version of ncurses, glibc readline termcap and so on.. Finary, I coud compiled asterisk. Thanks a lot!! Thanks for reply. Of course, I had already read ML like follows. This case errors are almost same, perhaps. But, tell the truth, I can't understand What I have to do? I had recompile some version automake source, and tried to recompile asterisk. The result does'nt go well. What's wrong? automake? term.c?termcap?curses?readline?etc... I am confusing Give me help! Is this same result? http://lists.digium.com/pipermail/asterisk-users/2003-November /027321.html mack_jpn On Thu, 1 Apr 2004 01:08:44 +0900 岩田 伸介 [EMAIL PROTECTED] wrote: hi. I got these compile errors while install asterisk. readline and openssl are compiled using gnu source, and kernel version is 2.4.17. Compile errors message is follows. Someone cleared this problem? Please, help! Regards. -- gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o config. o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o call erid.o fskmodem.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o ast _expr.o dsp.o chanvars.o indications.o autoservice.o db.o privacy.o astmm.o enum .o srv.o dns.o aescrypt.o aestab.o aeskey.o -ldl -lpthread -lncurses -lm -lresol v editline/libedit.a db1-ast/libdb1.a stdtime/libtime.a make: pwd: Command not found ？？editline/libedit.a(editline.o_a): In function `term_move_to_line': /usr/src/asterisk/editline/term.c:554: undefined reference to `tgoto' snip --- 代理店どっとこむ ポータル＆自動音声応答装置、IP電話、IPPBX複合システム開発、 UMS・業務支援・応対窓口・在宅コールセンター開発請負 http://www.dairiten.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't compile asterisk.
Is this same result? http://lists.digium.com/pipermail/asterisk-users/2003-November/027321.html mack_jpn On Thu, 1 Apr 2004 01:08:44 +0900 岩田 伸介 [EMAIL PROTECTED] wrote: hi. I got these compile errors while install asterisk. readline and openssl are compiled using gnu source, and kernel version is 2.4.17. Compile errors message is follows. Someone cleared this problem? Please, help! Regards. -- gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o config. o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o call erid.o fskmodem.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o ast _expr.o dsp.o chanvars.o indications.o autoservice.o db.o privacy.o astmm.o enum .o srv.o dns.o aescrypt.o aestab.o aeskey.o -ldl -lpthread -lncurses -lm -lresol v editline/libedit.a db1-ast/libdb1.a stdtime/libtime.a make: pwd: Command not found ？？editline/libedit.a(editline.o_a): In function `term_move_to_line': /usr/src/asterisk/editline/term.c:554: undefined reference to `tgoto' snip --- 代理店どっとこむ ポータル＆自動音声応答装置、IP電話、IPPBX複合システム開発、 UMS・業務支援・応対窓口・在宅コールセンター開発請負 http://www.dairiten.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Softfax/spandsp
Hi Steve. On Thu, 18 Mar 2004 22:06:46 +0800 Steve Underwood [EMAIL PROTECTED] wrote: snip There is now a new tarball at ftp://ftp.opencall.org/pub/spandsp/spandsp-0.0.1b.tar.gz Please try this, and report any problems you find. This version has the following changes: A floating point exception has been fixed A problem with the software not properly Some fax machines send a little less than the specified 1.5 seconds of training test data, so the training test failed every time. I now only look for 1.25seconds of training test data. Some fax machines do not correctly initialise the scrambler in their V.29 transmit modem. I have changed the software to tolerate this. Some fax machines send a burst of ones before the burst of zeros that forms the training test data. I have changed the software to tolerate this. Many thanks for your great release rxfax function is works well with my Canon MFC Multipass B-30!! mack_jpn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] IPC5000 (WIP-5000 from hitachi cable)
That is fahdtel's product? http://www.dairiten.com/modules/mylinks/visit.php?lid=217 It looks very cool. mack_jpn On Mon, 15 Mar 2004 06:23:22 + Miguel Cavazos [EMAIL PROTECTED] wrote: thanx for the review michael, could you send some pictures of the phone? can you tell how long does the battery lives? signaling what do the menus have how do you configure it etc? maybe after you do a full testing we can do a Wisip vs. IPC5000 working futures. Miguel Cavazos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Grandstream TFTP Config
On 16 Mar 2004 01:08:27 + Hermann Wecke [EMAIL PROTECTED] wrote: On Mon, 15 Mar 2004, Matthew Marlowe wrote: I can confirm 1.0.4.53 is bad as well. :) 1.0.4.50 has been working fine for me. I received the 1.0.4.54 firmware. So far, so good. No new problems. How can I get this firm? that isn't in here ;-) http://www.grandstream.com/BETATEST/ mack_jpn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] segfault with alsa
Hi all I was tryed to use asterisk with alsa following versions. alsa-driver-1.0.2c alsa-lib-1.0.3rc1 but an error occured about that. Program received signal SIGSEGV, Segmentation fault. [Switching to Thread 1074440928 (LWP 2141)] 0x41114e03 in snd_pcm_hw_param_set_near (pcm=0x80e9578, params=0xbfffe04c, var=SNDRV_PCM_HW_PARAM_RATE, val=0x1f40, dir=0x0) at pcm_params.c:786 786 unsigned int best = *val, saved_min; Does anyone knows fix this problem? mack_jpn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Web based UA
yes I know two Web based iax client.
the one is
Babar Shafiq's iaxclient.ocx
http://www.geocities.com/babarnazmi/index2.htm
and another one is
Dan's activediax
http://www.laser.com/dante/diax/diax.html
so usefull :-)
We can find topics at here.
http://iaxclient.sourceforge.net/
mack_jpn
On Wed, 25 Feb 2004 13:08:11 -0600
Jonathan Moore [EMAIL PROTECTED] wrote:
Found this link from a google search
http://www.dairiten.com/webiax/
--
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508
Quoting [EMAIL PROTECTED]:
You may be right here. I was thinking of an ActiveX plug-in. I don't expect
them to use public internet kiosks so they should be able to use the ActiveX
approach. I was hoping that something IAX based could be found as it would
make the connectivity easier and open port risk reduced.
Michael
Original Message
Subject: Re: [Asterisk-Users] Web based UA
From: Jonathan Moore [EMAIL PROTECTED]
Date: Wed, February 25, 2004 11:16 am
To: [EMAIL PROTECTED]
I think xten is supposed to have an active X control version of their
softphone that would probably do what you are talking about.
On Wed, 25 Feb
2004, Michael Graves wrote:
Hello All,
Does anyone here have any experience with web based soft clients for
*?
I'm thinking about putting a page up on our corp web server that
would
let staff in the field connect to our in-house phone system via the
internet. This could help staff making overseas calls while on
trips,
without demanding that they use a particular laptop/soft phone. They
could use an PC on a broadband connection.
Thanks,
Michael
--
Michael Graves [EMAIL PROTECTED]
Sr. Product Specialist www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]
It is dangerous to be correct about matters when the established
authories are wrong. - Voltaire
** Tag(s) inserted by Bandit Tagger98 -
http://www.gbar.dtu.dk/~c918704
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Jonathan Moore
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Winfield Public Schools
Office 316-221-5100
Fax 316-221-0508
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Re: [Asterisk-Users] Best Linux Distribution
On Tue, 13 Jan 2004 16:55:03 -0500 (EST) Greg Boehnlein [EMAIL PROTECTED] wrote: I was toying with the idea of making an Knapterisk distribution based on Knoppix. I've got my hands full right now, however, and can't really get involved in another project. Does anyone try it? http://featherlinux.berlios.de/ mack_jpn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Your config files
On Thu, 8 Jan 2004 21:25:55 -0500 Steven E. Frazier [EMAIL PROTECTED] wrote: Is there an easy way to get all of your sound files, do you have ftp or just http? Thanks. Steve I think WebDAV is easy way. but I stilln't try it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Handytone 286 RTP Problems
Hi Mike I know exacty same situation about BT100 that sometimes lost any packets. like a DoS attack for BT100? ;-( mack_jpn [EMAIL PROTECTED] asterisk]# ping 192.168.XX.XX PING 192.168.XX.XX (192.168.XX.XX) from 192.168.XX.X : 56(84) bytes of data. 64 bytes from 192.168.XX.XX: icmp_seq=0 ttl=250 time=2 usec Warning: time of day goes back, taking countermeasures. 64 bytes from 192.168.XX.XX: icmp_seq=1 ttl=250 time=969 usec 64 bytes from 192.168.XX.XX: icmp_seq=2 ttl=250 time=766 usec 64 bytes from 192.168.XX.XX: icmp_seq=3 ttl=250 time=746 usec 64 bytes from 192.168.XX.XX: icmp_seq=4 ttl=250 time=829 usec 64 bytes from 192.168.XX.XX: icmp_seq=5 ttl=250 time=725 usec 64 bytes from 192.168.XX.XX: icmp_seq=6 ttl=250 time=735 usec 64 bytes from 192.168.XX.XX: icmp_seq=7 ttl=250 time=703 usec 64 bytes from 192.168.XX.XX: icmp_seq=9 ttl=250 time=670 usec 64 bytes from 192.168.XX.XX: icmp_seq=10 ttl=250 time=728 usec 64 bytes from 192.168.XX.XX: icmp_seq=11 ttl=250 time=711 usec 64 bytes from 192.168.XX.XX: icmp_seq=12 ttl=250 time=701 usec 64 bytes from 192.168.XX.XX: icmp_seq=13 ttl=250 time=707 usec 64 bytes from 192.168.XX.XX: icmp_seq=14 ttl=250 time=693 usec 64 bytes from 192.168.XX.XX: icmp_seq=15 ttl=250 time=692 usec 64 bytes from 192.168.XX.XX: icmp_seq=16 ttl=250 time=678 usec 64 bytes from 192.168.XX.XX: icmp_seq=17 ttl=250 time=673 usec 64 bytes from 192.168.XX.XX: icmp_seq=18 ttl=250 time=699 usec 64 bytes from 192.168.XX.XX: icmp_seq=19 ttl=250 time=683 usec 64 bytes from 192.168.XX.XX: icmp_seq=20 ttl=250 time=696 usec 64 bytes from 192.168.XX.XX: icmp_seq=21 ttl=250 time=714 usec 64 bytes from 192.168.XX.XX: icmp_seq=22 ttl=250 time=704 usec 64 bytes from 192.168.XX.XX: icmp_seq=23 ttl=250 time=701 usec 64 bytes from 192.168.XX.XX: icmp_seq=24 ttl=250 time=691 usec 64 bytes from 192.168.XX.XX: icmp_seq=25 ttl=250 time=670 usec 64 bytes from 192.168.XX.XX: icmp_seq=26 ttl=250 time=690 usec 64 bytes from 192.168.XX.XX: icmp_seq=27 ttl=250 time=698 usec 64 bytes from 192.168.XX.XX: icmp_seq=28 ttl=250 time=713 usec 64 bytes from 192.168.XX.XX: icmp_seq=29 ttl=250 time=723 usec 64 bytes from 192.168.XX.XX: icmp_seq=30 ttl=250 time=703 usec 64 bytes from 192.168.XX.XX: icmp_seq=31 ttl=250 time=694 usec 64 bytes from 192.168.XX.XX: icmp_seq=32 ttl=250 time=685 usec 64 bytes from 192.168.XX.XX: icmp_seq=33 ttl=250 time=727 usec 64 bytes from 192.168.XX.XX: icmp_seq=34 ttl=250 time=720 usec 64 bytes from 192.168.XX.XX: icmp_seq=37 ttl=250 time=687 usec 64 bytes from 192.168.XX.XX: icmp_seq=38 ttl=250 time=704 usec 64 bytes from 192.168.XX.XX: icmp_seq=39 ttl=250 time=686 usec --- 192.168.XX.XX ping statistics --- 40 packets transmitted, 37 packets received, 7% packet loss round-trip min/avg/max/mdev = 0.002/0.695/0.969/0.126 ms On Sun, 04 Jan 2004 20:16:31 -0800 Mike Machado [EMAIL PROTECTED] wrote: I am trying to get the handytone 286 to make a very simple call to * and having problems. It registers with * just fine, but when I place a call (to echo test, for example), the RTP stream seems to have problems opening. Here is there error I get in *: snip ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Java?
On Thu, 01 Jan 2004 17:50:32 +0100 Philipp von Klitzing [EMAIL PROTECTED] wrote: Hi! We needed the client browser to be open all the time for dynamic data to load without the page refreshing. After looking at all of our options we decided on programming it ourselves using flash rather than java. Apache + Mysql + PHP ( Ming + Actionscript ) + Asterisk is good. Dynamic effective,Easy coding and Fast response :-) That's an excellent suggestion, I agree with Ray. Masakazu, do you think you could provide a working sample either here on the list or in the Wiki? Cheers, Philipp yeah. surely ok. but please just a moment to disclose my code. because that is very evalution code at now. bit buggy ;-) I think AMP + ming + actionscript + * + ecasound + xoops makes us good CRM ( one of anti-claimer ) enviroment. http://www.wakkanet.fi/~kaiv/ecasound the interface of voice by asterisk. and translate to mp3 by ecasound. and play that realtime stream by ming. mack ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Java?
On Wed, 31 Dec 2003 21:19:10 +0200 Stephen Karrington [EMAIL PROTECTED] wrote: We needed the client browser to be open all the time for dynamic data to load without the page refreshing. After looking at all of our options we decided on programming it ourselves using flash rather than java. snip Apache + Mysql + PHP ( Ming + Actionscript ) + Asterisk is good. Dynamic effective,Easy coding and Fast response :-) --- Masakazu Nakano. Dairiten.com - an open source VoIP and Ubiquitous Portal site in Japan. http://www.dairiten.com/modules/news/ powered by xoops at http://www.xoops.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Quality Survey.... :P
but that's web admin page sometimes no respond and some hang-ups ;-P I think that still dangerious for bussiness use. mack_jpn On Wed, 24 Dec 2003 08:08:13 +0200 Lubomir Christov [EMAIL PROTECTED] wrote: I have to say that Grandstream phones are REALLY GREAT products for the price of ~ $65 !!! Lubo Brian West wrote: Today class we are going to be talking about the wonderful line of GrandStream products. Or should I say BarbieTone phones? Who else is having MAJOR issues with the grandstream products? How many times have you been told upgrade upgrade upgrade? How many of you have paperweights, granted the phone is light as a feather and couldn't weight papers down in the first place? How about that ring tone, really dandy eh? Who else is irked about the the GAPS crap? It should slurp down cfgMACADDRESS.txt and we shouldn't have to pay more for that option. Have you had Message Waiting Indicator issues? Have you had issues with the Hold button and flash button? Have you had issues with sip transfers? Has the grandstream product line made you want to hurt someone? Care for some matches and lighter fluid? Was the response from grandstream support able to take care of your problems? I own a grandstream phone and I guess I just don't use it enought to see alot of these problems but the consensus on #asterisk is they are CRAP and everyone should stop buying them till they get their act together. A few people in the asterisk community have offered to write IAX firmware for the phones but grandstream has give them the run around. If they can't create stable and usable firmware they should atleast let the info out to let someone write IAX firmware for the damn thing. BOYCOTT GRANDSTREAM Thanks, bkw_ PS: then again you get what you pay for, 10 dollar phone with a 65 dollar pricetag. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- 代理店どっとこむ 代表 中野 ポータル＆自動音声応答装置、IP電話、IPPBX複合システム開発、 UMS・業務支援・応対窓口・在宅コールセンター開発請負 http://www.dairiten.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] First version of the ActiveX version of DIAX (0.1.0) available for download
I need more instruction too. like a http://www.dairiten.com/webiax/ On Mon, 22 Dec 2003 00:02:08 -0200 Carlos Arnt [EMAIL PROTECTED] wrote: I put like the readme.txt say the code in my web page , put the OCX in the same Directory, but not work. Did has any problem ?? Thanks The code: html head meta http-equiv=Content-Type content=text/html; charset=windows-1252 titleNew Page 1/title /head body OBJECT ID=diax CLASSID=CLSID:E873C973-33F5-49DD-95AB-56259BD89A77 CODEBASE=diax.ocx /OBJECT /body /html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RxFAX application
Hi sergio On Fri, 19 Dec 2003 14:49:15 +0100 Sergio Serrano Revuelto [EMAIL PROTECTED] wrote: Hi all, I have tested RxFAX application through X100P card. When Fax arrive i obtain the next trace: snip 5 (0.01679,-0.16590) - 0.02781 6 ( -0.04451, 0.75304) - 0.56904 7 ( -0.01415,-0.29305) - 0.08608 Fast carrier down Segmentation fault And i obtain 8 byte tif file. Any Idea? I have installed tiff-3.5.7 and spandsp-20031021. I get same result. but the end part looks like that. Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down -- Hungup 'Zap/1-1' with no segfault I'm tryed with tiff-v3.6.0 ( use with tar balled headers ) and spandsp-20031021 Does anyone have good result? Regards. mack_jpn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RxFax
Hi all I was tryed to use rxfax compiled with no problem. and I get the FAX over pstn.but it only first 3 bytes. like that. - II* - but not enough more. How can I fix this problem? and I found two files of tiffiop.h and tif_dir.h at /tiff-v3.6.0/libtiff Is that right? or required FTP version of those files? mack_jpn # sorry for my strange english ;-) On Wed, 17 Dec 2003 14:02:53 -0600 Tilghman Lesher [EMAIL PROTECTED] wrote: On Wednesday 17 December 2003 13:55, Alvaro Parres wrote: i have check at internet, that some one use RxFax application for recive faxes... Where i can get this application, becouse i have the cvs of today and it does not have application??? It won't be included in CVS, either, until the author declares it stable. You can retrieve the current package from opencall.org. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Web Interface for CDRs
like that but poor :-)
// CDR access
$connect = @mysql_connect (localhost,USERNAME,PASSWORD);
mysql_select_db(asteriskcdrdb);
$result = mysql_query (select dstchannel from cdr where dstchannel like
'%IAX%',$connect);
$iax = mysql_num_rows($result);
$result = mysql_query (select dstchannel from cdr where dstchannel like
'%SIP%',$connect);
$sip = mysql_num_rows($result);
$result = mysql_query (select dstchannel from cdr where dstchannel like
'%Zap%',$connect);
$pstn = mysql_num_rows($result);
$result = mysql_query (select dstchannel from cdr where dstchannel like
'%H323%',$connect);
$h323 = mysql_num_rows($result);
mysql_close($connect);
printEOH
IAX {$iax}
SIP {$sip}
PSTN {$pstn}
H323 {$h323}
EOH;
mack_jpn
On Tue, 9 Dec 2003 09:55:33 -0500
Bruce Hedreen [EMAIL PROTECTED] wrote:
Does anyone know where that nice .php is that was written to access the
CDRs from mysql DB?
Bruce W. Hedreen
Computer Technologies of Eastern Carolina, LLC
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Re: [Asterisk-Users] NTT FSK - Japanese Caller ID
Hi Isamar maybe I think disclose your code to CVS is best and fast :-) mack Hi folks, I'm trying now to play with fsk_modem.c and callerid.c to get the Japanese callerid working and I already got to make some steps.. I don't know if anybody accomplished that already... but Since two or more minds think better than one, send private messages anyone who is interested on that and can help... Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Change the all announcement
Hi Hachison. you need voice actor and sampler software and codec conversions. mack_jpn On Fri, 21 Nov 2003 19:00:07 +0900 Hachison [EMAIL PROTECTED] wrote: Hello I can cange busy and unavaile message by voicemailmain. I would like to change the Voicemail and Voicemailmain prompt itself. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX version 0.9.2 available for download
On Tue, 11 Nov 2003 09:30:41 +0200 Dan [EMAIL PROTECTED] wrote: Hi, - Original Message - From: Masakazu Nakano [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, November 11, 2003 2:18 AM Subject: Re: [Asterisk-Users] DIAX version 0.9.2 available for download On Mon, 10 Nov 2003 17:40:02 +0200 Dan [EMAIL PROTECTED] wrote: and need 'callto:' support :-) Why you need this? Give me an example. 1) netmeeting and cuseeme are already support. 2) make a call easier with groupware and/or portal. ( 'tel:' is good too ) I really do not understand what do you mean... Can someone else explain what is about? What is 'callto:' support? What is 'tel:'? What about groupware? oh... I'm sorry. Now I wish to making SOHO platform with twiggi and asterisk. twiggi is very cool LAMP + IMAP based groupware. http://www.twiggi.org/screenshots.php 'callto' is like that for M$ Windoze. http://msdn.microsoft.com/library/default.asp?url=/library/EN-US/netmeet/nm3_1l4o.asp and tel: is implementing some mobile cellurer carriers (such as i-mode). using 'make a call'. in Germany. http://sten-schmidt.net/imode/taglist.html mack ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX version 0.9.2 available for download
Hi Dan Could you please compile with MBCS ( multibyte charactor supports ) okay? http://msdn.microsoft.com/library/default.asp?url=/library/en-us/vccore/html/_core_international_enabling.asp mack ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX version 0.9.2 available for download
On Mon, 10 Nov 2003 17:40:02 +0200 Dan [EMAIL PROTECTED] wrote: and need 'callto:' support :-) Why you need this? Give me an example. 1) netmeeting and cuseeme are already support. 2) make a call easier with groupware and/or portal. ( 'tel:' is good too ) mack ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX version 0.9.2 available for download
thanks Dan! and I found an error. please fix. strange function in call,when it stilln't registered ( like a no home server... ). On Sun, 9 Nov 2003 19:53:36 +0200 Dan [EMAIL PROTECTED] wrote: Hi all, As promise, the new prerelease (0.9.2) is now available for download from the followiing locations: http://www.laser.com/dante or http://www.geocities.com/tdanro snip ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Hi Dan. thanks for good application! and I wish 'no with installer' package about that. because I think use with USB-memory device in any places (ie.net-cafe.) is that need registry setting or not? On Sun, 2 Nov 2003 22:21:09 +0200 Dan [EMAIL PROTECTED] wrote: Hi all, I have developed a full featured Windows IAX phone based on LIBIAX library . It is now in a prerelease version (0.9.0) and you can download it for free from my web page: http://www.laser.com/dante or http://www.geocities.com/tdanro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 802.11 phone review: WiSIP
I found it. but that webite is chinese BIG-5. take care. http://www.mpn.com.tw/index-big5-PRODUCT.html and that already released by Fujitsu. http://www.net-2com.com/jp/product/hw/wireless_ipphone/ mack_jpn On Fri, 03 Oct 2003 19:03:10 -0500 Steven Critchfield [EMAIL PROTECTED] wrote: On Fri, 2003-10-03 at 18:09, John Todd wrote: Hello - Here's my first impression review of the first SIP 802.11 phone. I got my hands on the first one sold, so that perhaps makes me the first person to have a real 802.11 SIP phone commercially in the US interworking with Asterisk. Whee! Can someone point me to other commercially shipping phones to prove me wrong? http://www.loligo.com/asterisk/misc/WiSIP/WiSIP-review.html There was another 802.11 phone from Symb ??? I don't remember the name. It was supposed to run SIP on 802.11b. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] Recommended OS
and I think use those cvs with rh7.3 and apt for RH is works well :-) mack_jpn Tilghman Lesher wrote: On Monday 22 September 2003 22:37, Steve wrote: On Monday 22 September 2003 11:25 pm, Andrew Joakimsen wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael A. Miller Sent: Monday, September 22, 2003 10:40 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Recommended OS Is there a recommended OS that Asterisk should be used with? I have been trying to get Asterisk running on Red Hat 9.0 with little success. I've been running it on RH9.0 w no problems. Ditto with getting new updates and recompiling. Are you aware of the software requirements? bison cvs gcc kernel-sources libtermcap-devel newt-devel ncurses-devel openssl096b openssl-devel readline41 readline-devel Actually, readline should not be necessary anymore. There's now an implementation of readline included in the source (BSD-derived). -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PHP Web interface for Asterisk
I bote 'with PHP' too :-) it's so easy to access text strings,web,db,sounds(ecasound) and agi :-) --- Masakazu Nakano. Dairiten.com - an open source VoIP and Ubiquitous Portal site in Japan. http://www.dairiten.com/modules/news/ powered by xoops at http://www.xoops.org On Thu, 26 Jun 2003 21:50:53 -0600 Dave Packham [EMAIL PROTECTED] wrote: ok guys I have a PHP GUI that will be great for both of you. direct editor to the whole file intact OR click to go to an extension. I will post a link to it tomorrow morning... as soon as I can get it off my production server HEHE it can do CRC checks on the *.cnf files and it will allow you to edit and parse out for you all your config entries with complex cnf files and default sample confs. it does login verification on the manager.conf as well as read/write features based on the manager.conf... I am getting it ready to give to Markster to include (if he wishes) into the cvs tree. I would accept any constructive/positive as well as well thought out slightly negative comments and diffs... :) Dave Packham U of Utah [EMAIL PROTECTED] 6/26/2003 7:04:59 PM Well, for *, I fall into the newbie category (not for telephony, VOIP, Internet, *NIX, C, etc - those I've been doing since the Internet had 3 nodes :-) and each technology mentioned was a newborn) I believe making it easy for folks to enter the * world will do nothing but sell Digium products, expand/improve *, etc. Keeping it in a 'you have to be an expert hacker' world will not. I personally would assist in a PHP (I assume) web GUI effort, and will definitely contribute 'simple' but complete mini-examples of conf files for * - that seems to be something lacking at present for a newbie like myself. And - yes - I've read the manual from end-to-end several times already :-) My testbed is a dual Xeon RedHat box (shows as a 4 CPU setup to top), tomorrow should provide a 4 FXS card and an FXO card and I have a fully deployed H.323 VOIP environment (Altigen) to play with. I'll snag a PRI card after I get things squared away - * will be my PBX backup to the Altigen until such a time as it proves itself superior... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Gary Sent: Thursday, June 26, 2003 5:34 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Web interface for Asterisk I tend to agree with Steven on this... If the web form makes it easier for the newbies why not, its just another option It could even be expanded to be a dialplan for dunnies (woops, i meant dummies:-) interface Considering all it is, is an interface to write out a .conf file On Thu, 26 Jun 2003 17:04:28 -0700, Steven P. Donegan wrote: I disagree - for many tasks a GUI would be just fine, for others direct coding would do the trick. They do not have to be mutually exclusive. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jeremy McNamara Sent: Thursday, June 26, 2003 4:42 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Web interface for Asterisk That GUI is going to dramaticly limit the flexibility of your config. The only way you can make a GUI config work with Asterisk is if you have a very very specific task you want to accomplish, but even then you still will have issues as your requirements change with time. Stick with what the AstGod has bestowed upon us It will save you many headaches. Jeremy McNamara Dylan VanHerpen wrote: Hi everybody, I've been tinkering with a web based interface for Asterisk. I tried to stick as closely to the current configuration format as possible. The web interface should help to do things a little easier (sort by extension, context, do bulk changes). www.packetbell.com/asterisk Feedback appreciated! Dylan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP Provider
Hi We can found a couple of ITSP at Jasomi networks's PR. http://www.jasomi.com/pr_deployment.html Does anyone try it? mack On Thu, 12 Jun 2003 19:48:43 +0200 (MEST) Martin Dommermuth [EMAIL PROTECTED] wrote: Hi, * Erik Lagerway wrote/schrieb: There is a provider in the US - www.AddaLine.com, who just launched a SIP service with some great rates for North America I have been using their service for months and I am extremely happy with the service. looks like Germany is again laggin behind all others in the communication field. Or I asked at the wrong place. There might not be to many people from Germany in this list. Anyway, thanks for the answer. CU MartinD: -- +++ GMX - Mail, Messaging more http://www.gmx.net +++ Bitte l臘heln! Fotogalerie online mit GMX ohne eigene Homepage! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM Encoders..
check here http://www.metadecks.org/software/sweep/ --- Masakazu Nakano as [EMAIL PROTECTED] On Tue, 18 Mar 2003 17:35:39 + WipeOut . [EMAIL PROTECTED] wrote: Hi, What utilities do you guys use to record your IVR messages?? I know there is the record application in * but I was wondering if there were any freeware Windows or Gnome utilities that would make it a lot quicker to record a large number of files.. Also what parameters do the files need to be encoded with, what I mean by that is are they required to be 8000 hz, mono, or similar?? Thanks. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
