[asterisk-users] eagi-sphinx-test how and why
Hi I see eagi-sphinx-test in agi-bin, anyone know how is it supposed to be used and what version of sphinx. Any help will be appraciated mawali ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Project Management Collaboration Software
I have been using netoffice, it is pretty cool http://netoffice.sourceforge.net/ Anyone else? On Thu, 20 Jul 2006, calvis wrote: We are looking at various software packages that do Project Management Collaboration. Since I value the opinions of this list I would be interested in how others are dealing with Project Management Collaboration. By collaboration I mean the sharing of emails, contacts, tasks, and files among team members. If you wish you can send me a post off-list about the solution you are using. Thanks, Charles Alvis Internet Technology Group, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outgoing call bridging
Hi Is there an easy way (without writing a C app) to make asterisk call 2 numbers and then bridge them into one conversatoin (preferably without using meetme). Regards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outgoing call bridging
Thanks for the info. It would be an external program. I have been looking at the originate manager command, but it looks like it would not bridge 2 external numbers. One of the number has to be a local extension. Suppose I want it to dial 2 number on zap and then bridge them, I probably would still have to do a dialplan gimmick, is that right? Regards On Mon, 5 Jun 2006, Tim Panton wrote: On 5 Jun 2006, at 18:40, [EMAIL PROTECTED] wrote: Hi Is there an easy way (without writing a C app) to make asterisk call 2 numbers and then bridge them into one conversatoin (preferably without using meetme). When would you want this to happen? - What is the trigger for the calls? If it is external, you can use call files or a program that sends an 'Originate' to the manager (you can write these in java, perl, ruby python etc). If the trigger is _internal_, look at using a Local channel and some dialplan voodoo. Tim. Tim Panton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialogic channel pricing
Hi As the manual states that Dialogic channel is provided as an add on per price. What does it cost and how can one buy it. Hasn't anyone been able to make a 3rd party dialogic channel using GlobalCall. I do have a couple of dialogic boards that I would like to use, I dont want my old investment to be useless. Regards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LineJack + Asterisk HELP!
Hi Looks like you did not do a make install after compiling the drivers, and it is still loading the stock kernel ixj. Please try doing a make install in the ixj-x.x.x source directory. Hope that helps On Tue, 16 Sep 2003, Bartosz Jozwiak wrote: Yes I fixed it thanks. But I have another problem. I am not so good with linux... so sorry If I am irritating... this is what i got: bmtst:/usr/src/ixj-1.2.1# modprobe ixj /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol try_inc_mod_count_Rsmp_e6105b23 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol register_chrdev_Rsmp_63ef0035 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol request_module_Rsmp_27e4dc04 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol unregister_chrdev_Rsmp_c192d491 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol sprintf_Rsmp_1d26aa98 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol printk_Rsmp_1b7d4074 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol panic_Rsmp_01075bf0 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: Hint: You are trying to load a module without a GPL compatible license and it has unresolved symbols. Contact the module supplier for assistance, only they can help you. /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: insmod /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o failed /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: insmod ixj failed What can I do about it ? - Original Message - From: Daryl G. Jurbala [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, September 16, 2003 1:26 PM Subject: RE: [Asterisk-Users] LineJack + Asterisk HELP! -Original Message- From: Bartosz Jozwiak [mailto:[EMAIL PROTECTED] Sent: Tuesday, September 16, 2003 11:53 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] LineJack + Asterisk HELP! To install driver for LineJack I need kernel source. I have debian, and I installed from apt-get install kernel-source.2.4.20 but while it make ./configure it still asks me for the kernel source. What can be wrong ? [...] Debian kernel sources are tared and compressed. You should see a kernel-source.2.4.20.tar.bz2 or similar in /usr/src. Un bzip2 it, untar it, and make a /usr/src/linux symbolic link to the directory it unpacks to. Daryl ___ 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] call center design question
Hi I would be interested in finding out about your solution, i can send you and email offline if you want to, but if you dont have much to hide, it may be better to post it here. On Tue, 16 Sep 2003, Paulo Mannheimer wrote: Hi Rich, We have done this before. We basically developed a small client that sits on every machine and communicates with * to get information about an incoming call. Contact me off-list and I will be glad to tell you more about the entire solution. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: September 16, 2003 1:39 PM To: Asterisk-users-list Subject: [Asterisk-Users] call center design question Would like to deploy * in a small help desk environment (five to ten people) using call queues and some sort of CTI interface to pop Remedy screen data in front of the help desk person receiving the call. Data to be popped would be based on CallerID. Anyone doing something similar? Anyone interfacing to an external Remedy system? Any reference sites that I could read/learn more of the requirements and/or 10,000 foot implementation? Rich ___ 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] Dialogic channel pricing
Sorry for disappointing you, (or actually breathing, probably that might bother you too). Im lazy, actually im so lazy I will ignore most of your email. Please do not lecture me, since I am stupid I will not understand your comments anyway!! Basically, there are a lot of disconnect between what you tell me and what Asterisk documentation is telling me. I want to know the price, and where to find it. And while you talk about me doing research in previous posts Never mind (you are superior anyway) FT On Tue, 16 Sep 2003, Steven Critchfield wrote: You disappoint me. You appear to be a somewhat knowledgeable person since you use pine, but then you don't figure out how to start a new thread. On Tue, 2003-09-16 at 16:54, [EMAIL PROTECTED] wrote: Hi As the manual states that Dialogic channel is provided as an add on per price. What does it cost and how can one buy it. Hasn't anyone been able to make a 3rd party dialogic channel using GlobalCall. had you did a small amount of research at the archives or even looked at recent posts you would have read that the problem is that the dialogic drivers are not GPL compatible. This means Digium must make a version of asterisk available in a non GPL license for the driver to be incorporated. Much the same way that when Sun released StarOffice to the OOo group they had to strip out the parts that had to be licensed in a proprietary way. Or the way that when Netscape released the netscape 5.x code to the mozilla group they had to strip it of anything proprietary. You luck out that Mark doesn't allow any code into the main tree that isn't possible for Digium to release in a license other than GPL. But for this price, and the effort Digium made to produce the driver, you must pay a fee. If you use google to search the archive you will find it to be something like $15 per DS0. I do have a couple of dialogic boards that I would like to use, I dont want my old investment to be useless. This is why ebay is around. Someone will buy them. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialogic Hardware (Take 2)
Please rest assure that I have been following the * development for a while and understand the value the Digium hardware gives me vs any other vendor. Most of the people on this list probably know whats good for everyone else, but I like to find out for myself (I am not a CNN junky). Now the * site mentions Dialogic as supported hardware at: http://www.asterisk.org/index.php?menu=hardware It also mentions in the manual that supprt for Dialogic hardware is avalable from digium. All I want to know is how, where. And is there any other third party channel for Dialogic is available. Now I dont see anything wrong with my question!!. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: your mail
Please try to tell me exactly what steps you did, and I will try to help you. It seems to be a non-asterisk issue so you can just email me directly. Please use a subject line or the spambouncer may not like it. Regards F On Tue, 16 Sep 2003, Bartosz Jozwiak wrote: Hello, I made install. Why I am getting this. My linux is Debian. -- Hi Looks like you did not do a make install after compiling the drivers, and it is still loading the stock kernel ixj. Please try doing a make install in the ixj-x.x.x source directory. Hope that helps On Tue, 16 Sep 2003, Bartosz Jozwiak wrote: Yes I fixed it thanks. But I have another problem. I am not so good with linux... so sorry If I am irritating... this is what i got: bmtst:/usr/src/ixj-1.2.1# modprobe ixj /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol try_inc_mod_count_Rsmp_e6105b23 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol register_chrdev_Rsmp_63ef0035 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol request_module_Rsmp_27e4dc04 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol unregister_chrdev_Rsmp_c192d491 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol sprintf_Rsmp_1d26aa98 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol printk_Rsmp_1b7d4074 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol panic_Rsmp_01075bf0 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: Hint: You are trying to load a module without a GPL compatible license and it has unresolved symbols. Contact the module supplier for assistance, only they can help you. /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: insmod /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o failed /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: insmod ixj failed What can I do about it ? - Original Message - From: Daryl G. Jurbala [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, September 16, 2003 1:26 PM Subject: RE: [Asterisk-Users] LineJack + Asterisk HELP! -Original Message- From: Bartosz Jozwiak [mailto:[EMAIL PROTECTED] Sent: Tuesday, September 16, 2003 11:53 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] LineJack + Asterisk HELP! To install driver for LineJack I need kernel source. I have debian, and I installed from apt-get install kernel-source.2.4.20 but while it make ./configure it still asks me for the kernel source. What can be wrong ? [...] Debian kernel sources are tared and compressed. You should see a kernel-source.2.4.20.tar.bz2 or similar in /usr/src. Un bzip2 it, untar it, and make a /usr/src/linux symbolic link to the directory it unpacks to. Daryl ___ 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 - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialogic Hardware (Take 2)
Thanks for the kind reply, and sorry if Ive been meeing up the threaded mail readers. But this is just half of the story, bacause besides $15 charge, that channel (just like quicknet) only supports incoming calls, but a man must know his limitations!! Regards On Wed, 17 Sep 2003, Alastair Maw wrote: Congratulations on learning how to start a new thread properly. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] phpconfig is out in CVS
I am sorry if I am missing something but!!, How is this any different than using a text editor. What does it give you that (for example) using vi on SSH/Telnet Java Applet that comes with WebMin doesnt give you. In some ways it is actually very limiting. Sorry, but had to ask On Sat, 13 Sep 2003, Darren Poulson wrote: Nice one! Took all of about 30 seconds to install, including downloading from the net. Just got the latest CVS and copied it into the web folder and opened up konqueror. Everything seems to be working fine. Off to do some testing of it now. Cheers, Darren. On Friday 12 Sep 2003 11:34 am, Peter Pauly wrote: On Thu, Sep 11, 2003 at 10:12:50PM -0600, Dave Packham wrote: nope when I click on something on the left I get a FQDN not just the pne you had Hmmm. Further info: it works with Microsoft Internet Explorer. It does not work with Mozilla 1.4 under Linux. It also does work with Mozilla Firebird under Windows. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] phpconfig is out in CVS
And also it require IE for search, which a linux admin will probably not gonna have (I dont use M$ products). BTW there are ways to do that on Mozilla too. On Thu, 11 Sep 2003, Peter Pauly wrote: On Thu, Sep 11, 2003 at 07:57:58PM -0600, Dave Packham wrote: I have put my phpconfig stuff out into the Digium CVS tree. Project name is phpconfig. see it at http://rads.netcom.utah.edu/phpconfig/phpconfig.php Looks cool, but the links don't work on the left. It wants http://phpconfig/phpconfig.php? ___ 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] Is there a web interface to the asterisk system?
I have asked this question in the start, but then realized that this qustion does not generate any response at all. Going through the archive I find out that lot of people have claimed to have made web gui, but when I see them, I stick to vi. It should be very easy to do, generate config files (or meta config files) from web gui. I think the webmin modules that you may have seen on the site is a failed (incomplete ??) attempt to do so. Another thing is that a configuration interface is what makes you product seem superior to others. So if people make good gui's for *, the probably keep them to themselves as a competetive advantage. Regards FT On Tue, 9 Sep 2003, Buddy Edwards wrote: Is there a basic web interface to the console to the asterisk system like webmin? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Chan_h323/g729 - X100P connecting to non-DigiumPartner
Hi The endpoint seems to be running Radvision h323 stack, and I know chan_h323 works with Radvision, there could be a couple of reasons!! 1) You dont have G729A in the capabilities of remote endpoint 2) The packetization interval is way off The best way would be to run ethereal or dump323 and see what is being negotiated. Also try to use fastConnect on both sides and force same packetization, (you can use my patch posted a couple of days ago to force packetization interval in G729 in chan_h323) Isamar Said I have on Chan_h323 with G729 and X100P trying to connect to a Planet VOIP400 gateway box(http://www.planet.com.tw) I uncommented g729 in the Makefile and I'm setting g729 in h323.conf I'm receving in my side: 1:20.906 H225 Caller:810f070 h323ep.cxx(1537) H323 Clearing connection ip$localhost/4112 reason=EndedByRemoteUser and the other side(Planet) says: 15- RADH 2 HSMU RAD: cmHookSend(masterSlaveDeterminationAck) 11- HSMU 0 Remote capabilities list: 0- HSMU 0 [1] g729AnnexA: Audio Receive 0- HSMU 0 Try matching local element: 0- HSMU 0 [1] g7231: Audio Receive and Transmit 0- HSMU 0 Try matching local element: 0- HSMU 0 [2] g729: Audio Receive and Transmit 0- HSMU 0 Try matching local element: 0- HSMU 0 [3] g711Ulaw64k: Audio Receive and Transmit 0- HSMU 0 Try matching local element: 0- HSMU 0 [4] t38fax: Data Receive and Transmit 0- HSMU 0 Try matching local element: 0- HSMU 0 [5] g729: Audio Receive and Transmit 0- HSMU 3 HSMU 2: Capabilities: NO MATCH FOUND! 0- HSSM 2 HSMU 2: SM H245 WAIT COMPLETE - error release 1- HSMU 2 HSMU 2: abort timer in state H245 WAIT COMPLETE 0- HSSM 2 HSMU 2: SM H245 WAIT COMPLETE == RELEASING 10- RADH 2 HSMU RAD: cmHookSend(endSessionCommand) 3- RAD 2 HSMU 2: cmEvCallControlStateChanged(cmControlStateTransportDisconnected, - ) Anybody has any idea? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intresting.. hrm
I have seen it in Linuxworld t ostel's booth. It is actually very close to (but bigger) the design I suggested in the list a couple of days ago. Beside long boot time a and huge memory, it seems to be running fine. The huge memory requirement is probably what drives the cost this high. Regards On Fri, 22 Aug 2003, Brian West wrote: And it runs linux. http://www.zip4x4.com/ZIP4x4.htm Anyone seen one? bkw ___ 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] Intresting.. hrm
Linux ensures that custom firmware is a piece of cake. It actually makes pulling one feature out and putting another in as easy as on you computer. It is matter of adding a .so file and an executable to your filesystem. Switching stacks is as easy as killing one process and starting another. I have been working on a linux phone that used to be vxWorks and linux gives my unlimited flexibility. Custom firmware is a nightmare (and using a nowhere compiler like I have seen suggested here) and a huge can of worms. Regards On Sat, 23 Aug 2003, Brian West wrote: OUCH.. its a nice phone but really is it that nice.. it runs linux so I wonder how hard custom firmware would be? bkw On Sat, 23 Aug 2003, Andrew Joakimsen wrote: Linuxdevices says $400 http://www.linuxdevices.com/articles/AT9406437906.html -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Saturday, August 23, 2003 1:33 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Intresting.. hrm The real question is: How much? On Fri, 2003-08-22 at 23:44, Brian West wrote: And it runs linux. http://www.zip4x4.com/ZIP4x4.htm -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone) ___ 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] zaptel compile problems
Im sorry, but I would try dumping RedHat kernel. I have never been able to compile any kernel code with distro kernels. Use Vanilla kernel from ftp.kernel.org FT On Sat, 23 Aug 2003, John Brown wrote: Hi list, I'm having problems getting zaptel to compile. I'm not a big Linux person and so don't know all the nifty ways RH does things. If this was FreeBSD it wouldn't be an issue :) here is the first few lines from the make gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include /usr/src/linux-2.4/include/linux/modversions.h -DECHO_CAN_MARK2 -DCONFIG_ZAPATA_PPP -DTORMENTA_BASE=0xd -DDEFAULT_TONE_ZONE=0 -DSTANDALONE_ZAPATA -c zaptel.c In file included from zaptel.c:34: /usr/src/linux-2.4/include/linux/kernel.h:60: nondigits in number and not hexadecimal /usr/src/linux-2.4/include/linux/kernel.h:60: floating constant exponent has no digits /usr/src/linux-2.4/include/linux/kernel.h:60: missing white space after number `0d5e' /usr/src/linux-2.4/include/linux/kernel.h:60: parse error before `0d5e' /usr/src/linux-2.4/include/linux/kernel.h:60: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:61: nondigits in number and not hexadecimal /usr/src/linux-2.4/include/linux/kernel.h:61: nondigits in number and not hexadecimal /usr/src/linux-2.4/include/linux/kernel.h:61: parse error before `01075bf0' [EMAIL PROTECTED] zaptel]# output from uname -a Linux pbx.joe.tld 2.4.20-20.7 #1 Mon Aug 18 15:05:54 EDT 2003 i686 unknown ls of /usr/src: [EMAIL PROTECTED] zaptel]# ll /usr/src total 20 drwxr-xr-x5 root root 4096 Aug 22 19:07 . drwxr-xr-x 16 root root 4096 Aug 9 15:58 .. lrwxrwxrwx1 root root 17 Aug 22 19:07 linux-2.4 - linux-2.4.20-20.7 drwxr-xr-x 14 root root 4096 Aug 22 19:07 linux-2.4.20-19.7 drwxr-xr-x 16 root root 4096 Aug 23 12:24 linux-2.4.20-20.7 drwxr-xr-x7 root root 4096 Aug 9 15:57 redhat [EMAIL PROTECTED] zaptel]# not sure whatelse to include. AST compiles fine Mucho thanks ___ 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] [OT] Virus propagation by asterisk user member.
I dont think so, it is unsuspecting windows users (lemmings??). Since they use outlook and your email address is in the postings. The virus gets your email address from the posted emails and blasts you. I do not care what other people chose (windows or linux or whatever) but when their braindead choice comes and haunts me is when I start calling them lemmings. Regards On Tue, 19 Aug 2003 [EMAIL PROTECTED] wrote: I've gotten a lot of unwanted, unsolicited mail today as well. Most probably with the subject line wicked screensaver. I guess the bad guys are mining the asterisk list. Guess I'll have to play with iptables and the mirror arguement. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Oen source IP phone, maybe?
Hi The BOM that you have for a 8 bit uC based system comes to be high for the processing power you would need. I would do it this way (I have already done it, but since I did it for someone else I cannot give it away). Hardware 1) A mips or arm based DSP/network processor 2) Telephony interface (Slic/Slac, A/D etc) or Voice interface (Mic. Seakers D/A ,etc) phone voice is 4KHZ so you dont need any fancy stuff. 3) Network interface (ethernet/usb or serial) you can build on that Software 1) uClinux (or mips-linux or arm-linux) 2) DSP control/RTP abstraction 3) Signalling stack (h323, mgcp are already available, maybe SIP) A system like this would be able to drive 2 lines snd also do some fancy stuff. Regards On Tue, 19 Aug 2003, James Sharp wrote: On Tue, 19 Aug 2003, Michael Sandee wrote: I guess you will need some software/mem/cpu/flash too? getting it on a cicuitboard etc? Software would be opensource...get a couple of people together to write it RAM I missed, I thought the C400 had onboard ram, but it doesn't...so add another $10. CPU is what the DS80C400 is. Its a 8051-based microcontroller with a built in ethernet controller and IPV4/V6 stack. Its fairly easy to write code for. There's also no need for much flash...you can DHCP/TFTP the C400. You would be more looking at 200$+ for a full board... the thing is you need something with drivers, or open standards hardware that you can write drivers for. I've not seen much available boards with dsp etc... one at broadcom iirc Looking through the C400's documentation again, they indicate that its got some Particularly Beefy(tm) math ability, which would eliminate the need for an external DSP...just need an 8 bit DAC/ADC. Case is probably not something you make just like that, you usually get a design, and people make it... and thats very expensive... but it gets For a phone, you are correct. For an ATA-type thingie...a $10 black plastic box from Radio Shack would do just fine. I've been dreaming about this aswell though ;) there are just many hooks to it.. overcome audio problems, speakerphones, echo, echo cancelling... You don't really need all of that for an ATA-type adapter, which is what I was thinking of. Of course a phone based on the same technology would be much more. Oof. Forgot about DTMF decoding from the phone through the adapter. Shouldn't be that hard... Hmmm. Price may be a bit higher than $40...but I'm positive it would be less than an ATA-186 and you'd have full control over the code. ___ 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] Can Asterisk use hardware codecs?
Hi I am rephrasing my quastion. If I have a Quicknet lineJack, can I use the hardware codecs provided by lineJack. It would save a lot of CPU if I did not have to use its cycles for RTP generation. Regards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323/SIP gatekeeper
Hi George You probably will need to run a local Gatekeeper which registers to the outside gatekeepers. So Asterisk registers to your local Gatekeeper and the Local Gatekeeper registers to Germany and UK Gatekeepers. Not too many answers on this mailing list unless you have a non-related qustion. FT On Sat, 16 Aug 2003, George Lin wrote: Hello List, Does asterisk H323/SIP allowes me to conditionally use diff gatekeeper to route the call ? e.g. for the call to germany, I want to use gatekeeper1, and for the call to UK, I want to use gatekeeper2. if yes, where and how to specify in these .confs file ? Thanks, George Lin ___ 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] Can I runAsterisk remotely from telnet session?
Please explain, what do you mean by does not work. It looks like a path problem. Instead of using su use su -. su does not initialize the environment for the user you are suing as, but su - will. /sbin and /usr/sbin a special path's that are only in root's environment. Hope this help, I will recommend reading a book on UNIX, this is UNIX 101 stuff. On Fri, 15 Aug 2003, Steve Lane wrote: I am having problems trying to run asterisk from a telnet session. I am able to su to root and the command asterisk does not work. Any ideas why this may be occurring? I thought Asterisk could be configured remotely as well as run remotely? Thanks in advance. Steve Lane ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk, quicknet and codecs (G729, 7231 Question)
Hi I am using asterisk with a Quicknet lineJack card. I am trying to get a proof of concept demo together before real deployment. I have a couple of qustions. 1) Can I use the codecs that are on Quicknet card (G.723, etc, etc). I tested it and I cannot use any codec in hardware, the only codecs that work are G711 and GSM. Since the lineJack provides G.723 in hardware, it should be available (other oh323 software can use it) 2) What would it take to get G729 for my demo. I cannot seem to find any link for this. I was told that I can use G729 as a paid option. I am willing to pay for it, but I do not know who to pay. Is trying to get 729 and 7231 working on asterisk worth it. Regards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Astrisk admin and webmin question
Hi Has anyone ever been able to use webmin module that is in asterisk directory. I will appreciate if someone quickly lays it out. What is the admin interface that anyone uses, which one is the best. I have seen refrence to different managers like Astmanagerm and Gastman. Are the useful, is there any documentation available for these. Regards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
