similar to: Current CVS problem

Today's CVS gives me an error while compiling: chan_zap.c: In function `zt_train_ec': chan_zap.c:1076: `ZT_ECHOTRAIN' undeclared (first use in this function) chan_zap.c:1076: (Each undeclared identifier is reported only once chan_zap.c:1076: for each function it appears in.) make[1]: *** [chan_zap.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/channels' make: ***

I am getting the following error when I try to build asterisk. gcc -c -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-11/13/03-12:48:12\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\"

Hi, Recently I did a fresh CVS checkout of asterisk and am getting the following errors on compile: chan_zap.c: In function `zt_train_ec': chan_zap.c:1078: `ZT_ECHOTRAIN' undeclared (first use in this function) chan_zap.c:1078: (Each undeclared identifier is reported only once chan_zap.c:1078: for each function it appears in.) make[1]: *** [chan_zap.o] Error 1 make[1]: Leaving directory

Hi, I have a deltathree account and I can place calls but I can't receive calls. I use Grandstram sip phones. When I call my deltathree phone # the voicemail is answer :(( I need some help and solutions from the guys who allready are using deltathree. I search on Internet and I try all types of configurations... :( This is my configurations files: - sip.conf - [general] port = 5060

Hi all, I must admit that I am stuck. I have a TDM400P card with two FXS and two FXO modules which I had set up and configured so that it was working beautifully. The only problem was that occasionally it would get itself into a state where outgoing calls would simply be met with a very loud static. A reboot would fix this issue and everything would work fine for a while. Recently however,

Hi, VoiceGlo is comercial version of Asterisk? :))) loooooooooollllllllllllllllllll Take a loock on http://www.voiceglo.com/ The softphone is IAX :) Best regards, Chris HARIGA Techselesta Inc. http://www.techselesta.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031201/307c10e9/attachment.htm

Hi friends, I've noticed that first register message sent by * always get rejected by the destination sip server. Then * sends a second registration message ( with Autherization section, and that get accepted by the destination host). Why is this ? Isnt there a way to tell * to send with Autothorization message the first attempt ? Asterisk sends this first 9 headers, 0 lines 11 headers,

Hi, i have diffirent provider example(3 single account in deltathree, 4 account in packet8 and so on) . How this possible to make the three individual sip account in deltathree act as trunk so that i cannot get a busy call. If line one fail goto line 2 then line 3 or another trunk line 1 then line 2 then line3....I read it in asterisk at home but the script i am copying is not working .

This morning, I received a very strange error message on the Asterisk console. The error occurs when I try to access iconnect WARNING[196621]: File chan_sip.c, Line 386 (__sip_xmit): sip_xmit of 0x80d0854 (len 649) to 213.137.73.178 returned -1: Bad file descriptor I also get this error when I try to reload: WARNING[16384]: File chan_sip.c, Line 5355 (reload_config): Unable to get IP address for

Hi, Can U tell me the Vonage ATA 186 settings? I would like to try to have a web interface on my adapter :-)) Best regards, Chris Hariga

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Hi guys i signed up to iconnect a few hours ago to try do some cool stuff. but im having a few problems. Im running asterisk and xp/xlite softphone.. both xp box and asterisk box are on public ips. The problem is that when i ring anyone in the world it'll ring they'll pickup and i can hear them 100% perfectly/clearly.. but they cant hear me.. occasionaly they can hear something like a

Hi all, i use asterisk 1.2.7 and i have a problem with callerid. i use sangoma a200 cards. one fxo one fxs module i have these scenario - bob calls adam, where bob calls into my asterisk and adam picks up "from" my asterisk - bob and adam are speaking to each other - meanwhile eve calls adam, adam hears a beep, and knows there is an other caller on line. - bob and adam stop seaking

Hi, si posible on SIP phones to have the dial tone after 9 like on the FXS card? I set ignorepat => 9 on my extensions.conf... Best regards, Chris HARIGA

Hi, I have Fedora Core 2 running with a T1 card. I try to put the log on db but I get the error: Aug 20 15:17:47 ERROR[262160]: cdr_addon_mysql.c:378 my_load_module: Failed to connect to mysql database asteriskcdrdb on localhost. The database exists and I try with "mysqlaccess localhost asteriskcdrdb" and I get: Access-rights for USER 'localhost', from HOST

I have this line of code: new Effect.toggle(document.getElementsByClassName(''modarea'', Event.findElement(ev,''li'')), ''blind''); I''ve confirmed that the Effect.toggle is indeed getting the proper element, however, I''m getting this in my javascript console: element.visible is not a function (effects.js: line 153) I

I''m using prototype 1.4.0 in a big web application I''ve been working on for months and today a bug was raised by one of our testers. After investigation it turns out that something done 0 times will actually fire once. So, for example, if I had (0).times( function(i) { alert(i); } ); it''d fire once rather than not at all. Is this a bug in prototype or the

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Hi folks. I''m working on a project where I need some drag and drop, but I need to implement what happens in a very different way, so I''m creating my own class for it. I don''t want to reinvent the wheel and redo Thomas'' excellent drag/drop functionality, nor do I want to copy/paste his code over into mine. So, I was wondering, has anyone tried to extract