Re: [Asterisk-Users] TDM card loses Dial tone
Same here... Usually after several of these show up in my system log: Power alarm on module 1, resetting! Need to unload/reload module wcfxs in order to get the dial tone back. Happens several times a week, sometimes more frequently. John - NetRom Internet Services973-208-1339 voice [EMAIL PROTECTED] 973-208-0942 fax http://www.netrom.com - On Thu, 12 Feb 2004, Youness El Andaloussi wrote: I experienced similar problems too with a 4 chan tdm400. This seems to especially happen when you make configuration changes. It has nothing to do with runing X or no, it does not even have to do with redhat... I experienced the same problem on mandrake. One thing you have to be extra careful is when restarting, make sure that all the modules have entirely reloaded before expecting a dialtone with an asterisk debug console asterisk -r... many of the times I thought there was no dialtone and the asterisk process had gone cukoo, I noticed that configuration was not entirely reload. Yet, reloading many times seems to get some of the TDM400 channels hung. On the other hand, this problem does not seem to happen as extensively when no reloads are made ___ 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] FXO cards
On Tue, 9 Dec 2003, Barton Hodges wrote: [EMAIL PROTECTED] wrote: On Tue, 2003-12-09 at 15:18, Michael Rowley wrote: Hey guys, has anyone put 6 of the wildcat X100P cards in one machine? I am thinking about putting 6 in one machine, what is everyone elses experience Read the docs. 2 card maximum sane install. Can you point me to the documentation that states this? If I need to connect 3 or 4 pstn lines, are my only choices to add another box and connect them via IAX trunking, or to wait for the 4-port FXO card? Does anyone know when the 4-port card will be released? It is possible but not recommended to put more then 2 x100p's in a box. I have a system with a TDM400 and 4 X100p's. Key is to get a motherbrd that lets you assign IRQ resources since you do not want the above cards to share IRQ's (That said the TDM and an X100 do share an IRQ without a problem but this is a 2.4GHZ machine) Using more then one box is best. As for the FXO modules I have passed out many many times holding my breath! :) John - NetRom Internet Services973-208-1339 voice [EMAIL PROTECTED] 973-208-0942 fax http://www.netrom.com - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX connected to a TDM400 card port
I have a similar setup and have found that faxing, sending or receiving works at best 50% of the time. I finally hooked the fax machine to the second RJ-11 jack of the x100p and set it to listen for fax tones and to grab the line from asterisk if it hears a fax. (Actually I may have a Y adapter going into the x100p either way the idea is the incoming line goes to both the x100p and the fax machine) I spent many hours and days trying to get it to work reliably but no dice... Obviously your mileage WILL vary on this issue... John - NetRom Internet Services973-208-1339 voice [EMAIL PROTECTED] 973-208-0942 fax http://www.netrom.com - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mute button in Grandstream?
On Fri, 28 Nov 2003, Roy Sigurd Karlsbakk wrote: On Fri, 2003-11-28 at 13:24, John Vozza wrote: On Fri, 28 Nov 2003, Anton Yurchenko wrote: Hello, Has anybody been able to get the Mute button work on grandstream? it simply does nothing. Only Hold is avalable, which is not that good. Does the GS even HAVE a mute button? The 101's appear not to. bottom right: MUTE/DEL ahhh... on my units that button is labeled only DEL and it does not seem to do anything... John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distintive Ring on x100p
On Wed, 12 Nov 2003, Brian West wrote: http://bugs.digium.com/bug_view_page.php?bug_id=504 I have been testing this patch today. Works great. Just wondered if anyone else was intrested in such a beast. YES, very! John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfer from Grandstream BT100?
On Mon, 3 Nov 2003, John Brown (CV) wrote: what version of GS firmware are you running ? I call from PSTN to GS, GS does xfer to XTEN, hang up GS call continues if you aren't running 1.0.3.81 or newer, then upgrade :) Or NEWER Latest I can find is 1.0.3.81. Care to give us all an early holiday gift? :) john brown chagres ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream 102
I haven't checked in a few months but while the info below is correct the 102 limits the PC Lan port to 10mb even if using a 100mb NIC card. Can anyone else confirm or deny this? John - NetRom Internet Services973-208-1339 voice [EMAIL PROTECTED] 973-208-0942 fax http://www.netrom.com - On Sun, 5 Oct 2003, Michael T Farnworth wrote: Typically you run a cable into the phone, then a cable out of the phone into the computer, it appears to just be a bridge. It also works fine to run the cable into the phone and then a cable into another phone and then into a computer. Michael On Sun, 5 Oct 2003, Nicolas Gudino wrote: Sorry about this off-topic question... I want to know if the second ethernet port on the Grandstream 102 phone works as a bridge to connect from there to a PC. Do I need two ethernet jacks to connect a phone and a PC, or this phone let me connect both with only one? Thanks in advance! Nicolas Gudino ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Source?
BS! :) Take the time to read and learn as much as you can from what's available and believe it or not you may just learn something. Even if that something is what to ask/search for. All those that get paid to answer questions on this list please raise your hand. I know my hand is still on the keyboard. I always amazes me how so many EXPECT so much for nothing... Regards John - NetRom Internet Services973-208-1339 voice [EMAIL PROTECTED] 973-208-0942 fax http://www.netrom.com - On Thu, 18 Sep 2003, PJ Welsh wrote: I have to defend us newbies on this. This environment does not facilitate sequential knowledge building! Based on my entry to Asterisk, I should have already known T1/E1/VOIP/SIP/FreeWorld/H.232/X100P/PBX/FXO/FXS/channel bank etc you get the idea (still trying to figure out skinny...cisco something, I know). Heck, I'm struggling to get a grip on what and how to use/confiure SIP for linux and keep my hair. You don't start off with a prerequisite of knowledge to join like a class/school. You don't have the you-must-have-asterisk-101-before going to asterisk-102 before you can join this list. You have a forum that is GENERAL. I would like to a better effort to provide a more sensible way to start helping us newbies. I have to say that the Digium handbook helped a little, but not much. I have googled till I couldn't see straight. I just don't yet have the big picture that most of you do. I couldn't even tell you if I need a channel bank or a channel changer ;) at this point. A group of you seem to expect people to have a knowledge base that allows for entering keywords to google. I don't know those keywords. You know the context to search for when someone says I'm having a problem with insert-thing-here. Instead of the usual, Search the archives. It would be more helpfull to give a hint on what to search for. I could search for SIP and get back several hundred answers. Then I have to figure out where that answer lies in the series of possible answers. Then I have to somehow figure out if it works. As most of you teachers (past and present) should know, not all of us learn the same. Some people just get written material. Some NEED the spoon to make it to the next level. Some need the hands-on experience and other's just can't learn any more than they have already know(those people are not likely on this list, however). You do realize that the http://www.asterisk.org/index.php?menu=support lists the mailing list first for support, don't you. In fact, you have to go to the second page before you even see the google reference. More a few people tend to look for the FIRST way to get help not ALL ways to get help... flame suit on On Thu, Sep 18, 2003 at 08:31:59PM +0200, Dave Cotton wrote: ... Absolutely agree with you Steve. I left teachers training college in 1970. I shock some teachers when I said that in all the years since I haven't taught anyone anything. I've just enabled them to learn. The problem is that in most national education systems the teacher is expected to provide the answers to pass some test at the end of the course. Thinking is not part of the curriculum. -- Dave Cotton [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
[Asterisk-Users] Grandstream firmware update.
Just noticed that version 1.0.3.81 has been released on the Grandstream website. Have fun... John - NetRom Internet Services973-208-1339 voice [EMAIL PROTECTED] 973-208-0942 fax http://www.netrom.com - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CE certification for Europe
BLONK :0 * [EMAIL PROTECTED] /dev/null Night! - NetRom Internet Services973-208-1339 voice [EMAIL PROTECTED] 973-208-0942 fax http://www.netrom.com - On Sat, 5 Apr 2003, d hinton wrote: who said i was selling to the public??? read people. again i never said i was going to sell, but rather that it could be done for less - Original Message - From: Brian Capouch [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, April 05, 2003 2:31 PM Subject: Re: [Asterisk-Users] CE certification for Europe d hinton wrote: to tilghman: Then contribute already. Don't troll the list you bozoo, before you call someone a troll you should prob read the post. and if that's you opinion arfter that then fine, so be it. but it's always some punk, who got bullied on as a kid that hides behind the internet and slings pot-shots ;-( Lord, Dwayne. Do you think after we sit here and read this sort of thing that there is going to be a single one of us who would be a potential customer for your product? You might be a bright engineer, but your organization needs to hire someone to filter you off from your potential buying public. And of course you should thank Digium for providing you this nice forum that you're using to attack them. B. ___ 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] Compile Problem?
Just did a CVS update and now compiling * fails. app_privacy.c: In function `privacy_exec': app_privacy.c:86: warning: passing arg 1 of `ast_safe_sleep' makes pointer from integer without a cast app_privacy.c:86: too few arguments to function `ast_safe_sleep' make[1]: *** [app_privacy.o] Error 1 John - NetRom Internet Services973-208-1339 voice [EMAIL PROTECTED] 973-208-0942 fax http://www.netrom.com - On Wed, 26 Mar 2003, Michael Manousos wrote: Roy Sigurd Karlsbakk wrote: On Wednesday 26 March 2003 12:57, Michael Manousos wrote: Have you tried asterisk-oh323 (regarding H.323 support)? nope. I heard chan_h323 should be better... Does this work? Where can I find it? You can get it from: http://www.inaccessnetworks.com/projects/asterisk-oh323 I haven't tested it on debian, though. Michael. ___ 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] Speex goes 1.0...
http://www.xiph.org/press/2003/nonprofitspeex1/ John - NetRom Internet Services973-208-1339 voice [EMAIL PROTECTED] 973-208-0942 fax http://www.netrom.com - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No way to send secret...
Thanks to all who set me straight on the codec format stuff... I have a remote asterisk system running on my laptop which sucessfully connects back to my main * server. (Lets me bring my phones to my customer jobsites...) After the last few CVS updates I started seeing; NOTICE[13326]: File chan_iax2.c, Line 2999 (authenticate): No way to send secret to peer 'xxx.xxx.xxx.xxx' (their methods: 4) Everything still works fine but should I be concerned about this or is my iax.conf missing something? Thanks John - NetRom Internet Services973-208-1339 voice [EMAIL PROTECTED] 973-208-0942 fax http://www.netrom.com - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codec Formats
I've been trying to find a list of codec format numbers so I can more clearly understand the following message; Accepting AUTHENTICATED call from xxx.xxx.xxx.xxx, requested format = 4, actual format = 4 I've seen 4, 32, 512 and I think a few others. For example I think format 32 equal ADPCM but what are the others? TIA John - NetRom Internet Services973-208-1339 voice [EMAIL PROTECTED] 973-208-0942 fax http://www.netrom.com - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec Formats
Thanks for the feedback but I'm still lost on this one (Forgive my ignorance please) I don't understand how #define AST_FORMAT_ADPCM(1 5) becomes a format = 32 in the * console display. Regards John - NetRom Internet Services973-208-1339 voice [EMAIL PROTECTED] 973-208-0942 fax http://www.netrom.com - On Fri, 14 Mar 2003, Martin Pycko wrote: The formats that asterisk uses are #define'd in asterisk/include/asterisk/frame.h RTP formats are #define'd in asterisk/rtp.c regards Martin On Fri, 14 Mar 2003, John Vozza wrote: I've been trying to find a list of codec format numbers so I can more clearly understand the following message; Accepting AUTHENTICATED call from xxx.xxx.xxx.xxx, requested format = 4, actual format = 4 I've seen 4, 32, 512 and I think a few others. For example I think format 32 equal ADPCM but what are the others? TIA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
