similar to: Rate Engine Compile Error

Hi all, I compiled rate-engine version rate-engine-0.5.2 with Asterisk CVS-HEAD-07/14/04-18:25:12 with some warnings. When I load the asterisk in debug mode, I got an error: [rate_engine.so]Jul 29 13:51:19 WARNING[1024]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/rate_engine.so: undefined symbol: ast_channel_walk Jul 29 13:51:19 WARNING[1024]: loader.c:423 load_modules: Loading

I'm getting this message when I start Asterisk chan_capi.c:2205 capi_handle_msg: Command.Subcommand = 0x5.0x81 but when I try and recompile I get this chan_capi.c:60: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) any help would be greatly appreciated. -- Dave Cotton <dcotton@linuxautrement.com>

I've got Zaphfc working running Asterisk v. 0.7.2 Then I have tried with Asterisk V. 1.0 and the latest from CVS - with no succes. Has anybody got zaphfc working with newer version than 0.7.2 ? NR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040616/32481fca/attachment.htm

Well, my faxes passes through asterisk successfully, however I still have some problems about fax reception by rxfax. The softfax answers, and negotiates transmission, however then as some stage of communiation something is wrong. But I have nothing more but this log: Jun 2 00:10:21 DEBUG[16900]: chan_zap.c:4242 zt_read: DTMF digit: * on Zap/10-1 Jun 2 00:10:22 DEBUG[16900]: chan_zap.c:4242

I am trying to compile the meetme2 application with the latest CVS head and it fails. Here is the error message that I get. Can someone point me in the right direction? gcc -pipe -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS

Anyone using the rate_engine from TrollPhone? There is absolutly no documentation on how to setup data in the tables. If someone could send sample data, or post it to the wiki, it would be helpful. If any others are successfully using another Least Cost Routing method, please pass it along. THanks, Matthew

I just downloaded chan_capi.0.3.4a.tar.gz but it will not compile on my system (Suse 9.1). I compiled and installed Asterisk and it is running, did I miss a configuration or dependency for Chan_Capi somewhere ? cheers, Mike linux:/usr/src/chan_capi-0.3.4a # make gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DCAPI_ES

-------------- next part -------------- An embedded message was scrubbed... From: Deon Rodden <drodden@webunited.net> Subject: Re: [Asterisk-Users] Strange Quality problems with Asterisk, Gentoo, Redhat and Kernels - /dev/dsp Date: Thu, 30 Sep 2004 09:05:39 -0400 Size: 5509 Url: http://lists.digium.com/pipermail/asterisk-users/attachments/20040930/289c69cc/dsp.eml

>> I'm trying to compile cdr_addon_mysql but keep getting this error. >> Again, searching the Wiki and the mailing list archive didn't bring up >> anything useful. Any help? Yes, I'm using MySQL 4.0. Maybe I have to >> switch back to 3.23? >> >> >> # make >> cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o

Is anyone familiar with the Trollphone's LCR package? There is a field in the egress table labeled substitue. Placing a $1 there results in the correct dial extension being passed. However how is this field used to substitute replacement dial extensions... in other words as an example, lets say my EXTEN as passed to and back from the rate_engine is 01144123456789 and I want to have a

Are you looking for local or Long distance? What will these T1s primarily be used for(inbound/outbound, domestic, local, long distance, international) How important are per minute rates to you? how many minutes do you expect to use per month? We are in Tampa Florida and have 15 T1s from several different providers so I may be able to refer you to one if it's a match to what you're

If I relay through my Cisco 7940/7960, does it do QOS, even with a dumb switch? I know you can set quality/qos but only if you have a layer2/layer3 switch that supports the tagging. A simple little linksys 5 port switch wouldn't know about QOS, it'd give everybody equal priority. If a computer plugged into the phone, and the phone into the dumb 5 port switch and then to the internet,

How do I limit the length of an extension? In my test IVR/Automated Attendant (whatever it's called), at the beginning it plays "if you know your parties 3 digit extension, you may enter it now) and then it gives a list of options. If the caller puts the 3 digit extension, it goes through fine, if they press 1, or 2 it goes to the selected menu option, but if they dial 91235551212 it

Hi, I have read the wiki entries on IAX(2), but I'm afraid, it still have some questions: I have a working connection between two Asterisk-Servers (Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k on Debian 3.1) via IAX. Does this connection work with IAX or IAX2? When I try to load chan_iax2.so, I get the error message chan_iax2.so]Oct 11 10:09:52 WARNING[2288]: loader.c:258 ast_load_resource:

I've compiled Asterisk on Redhat 9 and Fedora Core 1 in the past, generally without any problems. Especially w/ the stock kernel, which I generally loathe. When I tried to upgrade my Redhat 9 Server to the 2.4.27 kernel, doing a manual/clean compile, I had massive quality issues. I was forced to go back down to a stock 2.4.24 kernel. Never figured out why. Now, I've installed Gentoo

Good day all I have figured out most/basics of asterisk.I went with sip and made my own sip.conf and extensions.conf No I have 2 servers running sip in different towns.Both is connected with static ip so thats fine,but now. Lets say someone want to call someone else in the other town.How do I get asterisk to know,for instance sip extension 101 is on another sip server on a different ip. And I

Hi, I have following problems compiling bristuff-0.2.0RC8m. Has anybody seen them before and can point me in the right direction? zaphfc/make all: ---------------------- /usr/src/asterisk/bristuff-0.2.0-RC8m/zaphfc/zaphfc.c: In function 'hfc_findCards': /usr/src/asterisk/bristuff-0.2.0-RC8m/zaphfc/zaphfc.c:1000: error: invalid lvalue in assignment make[2]: *** [

For almost 6 months now I've upgraded Asterisk every couple of weeks or so and I've never had this problem. When I'm at the asterisk console (asterisk -r) it shows me live status. Who called who, what it's playing and when, etc. It logs to the screen. When I type reload, it says "added so and so to so and so context" gives me some long display as it reloads. But

I've been working with Asterisk for about 2 months now and am doing well. However I decided to switch platforms from Fedora Core 1, that my predacessor was using, to Gentoo, for obvious reasons. It just seems faster and less "bloated" everything I need, nothing I don't. Anyways, I've read what the Wiki had to say about it and I was only confused on one thing, putting

Hello, We use all SCSI PCI card hardware RAIDs on all 4 of our production Asterisk servers. They all have Digium quad T1 cards and they all have from 2 to 4 T1s hooked up to them. We have had no noticable problems with dropped calls/poor quality. What are you looking to do with this system? what kind of traffic will be going through these 4 T1s? MATT--- -----Original Message----- From: Deon