similar to: Asterisk on FreeBSD 4.10 dies

I am not having any luck placing out going calls I dial the number 08 82420173 ( our outside line ) But all I get is engaged signal and log this. Oct 14 08:40:14 DEBUG[16401]: File pbx_wilcalu.c, Line 65 (autodial): Entered Wil-Calu fd=20 Oct 14 08:40:14 DEBUG[8201]: File chan_sip.c, Line 657 (create_addr): Setting NAT on RTP to 0 Oct 14 08:40:14 DEBUG[8201]: File chan_sip.c, Line 548

V1 -> V2: Move kmalloc() before mutex_lock(), suggested by Arjan. file_remove_suid() should be called with i_mutex held, file_update_time() too. So move them after mutex_lock(). Plus, check the return value of kmalloc(). Signed-off-by: WANG Cong <amwang@redhat.com> Cc: Arjan <arjan@infradead.org> Cc: Chris Mason <chris.mason@oracle.com> Cc: Yan Zheng

Hello, I'm using X-web lite in a webpage to connect to one of our asterisk server. But now I have a problem, when you are connected to a voice script the voice will not be heard after a couple of seconds. When you press or say something that the voice will come back for a couple of seconds. When I thy X-Lite (stand-alone version) I had the same problem, but when I turned off the

V2 -> V3: set ''err'' to -ENOMEM when kmalloc() fails. Thanks to Tao. V1 -> V2: Move kmalloc() before mutex_lock(), suggested by Arjan. file_remove_suid() should be called with i_mutex held, file_update_time() too. So move them after mutex_lock(). Plus, check the return value of kmalloc(). Signed-off-by: WANG Cong <amwang@redhat.com> Cc: Arjan

Hi, Does anybody know if RAMI (Ruby Ami) is still functional? And is this still compatible with asterisk 1.8 Best Regards, Arjan Kroon Mobillion BV

Arjan, I am forwarding your email to the nut-upsuser mailing list. I am not an expert on the Powermust driver; hopefully somebody on the mailing list will be able to help you! -- Peter Arjan DJ wrote: > > Dear Peter, > Some days ago I bought a Sweex UPS (500VA) and now I have set up NUT (using > the Powermust driver) and everything is working fine. Invoking upsmon -c fsd > gives

Hi all, I have installed an asterisk server at my office, the server is behind a firewall. On the firewall I've set NAT a rule for incoming traffic on port 5060 to be forwarded to the server. Connecting from home with my sip client doesn't work at all. The asterisk server itself is ok, when I make a local connection at my office, 10.0.0.129 (client) to 10.0.0.6 (asterisk server) it

I need help! I installed Winebottler, and installed the first disc of a 4 disc windows program on my mac book pro. But when I went to install disc 2 when prompted, I had to quit Winebottler to put the disc in. But I need Winebottler running to do the install. How do I do it? David Murphey

I downloaded the latest CVS a few days ago. It all compiled nicely on my new AAH platform. However, it won't start up. Investigation of my log files produces this; Jul 26 22:59:18 VERBOSE[31473] logger.c: [pbx_wilcalu.so] Jul 26 22:59:18 VERBOSE[31473] logger.c: [pbx_wilcalu.so] Jul 26 22:59:18 WARNING[31473] loader.c: /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined symbol:

Hello, I'm a newbie in connection several asterisk servers with each others. I've got the following situation. I've got 9 asterisk servers (asterisk00 till asterisk08). When I call to asterisk08 then I want to redirect an application which runs on asterisk00. But how can I redirect in an application on asterisk08 to an application on asterisk00? Or isn't this possible?

Hi, In my application I want to have the sysdate + 5 minutes. I know that the sysdate is in the variable ${DATTIME} But now I want to now how I get the sysdate + 5 minutes into a variable? Doe's anybody knows the answer? Kind Regards Arjan Kroon -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, How do you all handle the situation when a centrale fastagi server process(es) are down? AGI(..) prints "Unable to locate host" and the dailplan jumps to extension h. I'd like to handle the return value and keeping the caller in the dailplan and not to the hangup extension. Any tips about how to handle a AGI(..) returns -1 condition? thx Arjan Kroon Mobillion BV

Hi, We encounter a problem with the variable CALLERID(dnid) We use E1 lines where we can make an inbound call or an outbound call on the same channel (not at the same time) If the CALLERID(dnid) is not used, than the CALLERID(dnid) will be the CALLERID(dnid) of the previous call For example: - First we get a inbound call on channel DAHDI/11-1 with CALLERID(dnid) = '655871460' We read

Hai, Does anybody have problems with a wrong Connected Line ID with asterisk version 1.6 The following bug was for version 1.4, but I cannot make up if this bug is still in version 1.6 http://forums.digium.com/viewtopic.php?t=7780 In version 1.8 it is possible to change the Connected Line ID, but this isn't the case in version 1.6 Regards, Arjan Kroon Mobillion BV

Hi, I'm using the Read command the read a DTMF tone. In this read command I play a voice-file. But now when I press one off they keys of my telephone the voice-file will stop playing a the program go the next priority. Is it possible to play the voice-file until the right DTMF tone is pressed? (say for instance the Zero). Kind regards Arjan Kroon Mobillion B.V.

hi ! i have installed asterisk-1.2.9.1 but am unable to run it i am getting this error "[pbx_wilcalu.so]Jun 11 16:43:00 WARNING[8968]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined symbol: ast_pthread_create Jun 11 16:43:00 WARNING[8968]: loader.c:554 load_modules: Loading module pbx_wilcalu.so failed!" can anyone help me i have redhat linux

I'm porting the most recent version of Neil Provos' systrace to FreeBSD 5.1. I'm sending him the diffs to integrate into his distribution. I'd also like to submit them to someone with FreeBSD for consideration, and hopefully inclusion as a port or whatever you prefer. Who could I send them to, or what would you prefer me to do with regard to FreeBSD? Thanks, Rich Murphey

Hi - after hearing others rave about * I thought I'd have a go - extract from a 'make' on a stock debian system as follows... (I tried to post the whole make up to this point but it was too big for the list) make[1]: Leaving directory `/usr/src/asterisk/channels' make[1]: Entering directory `/usr/src/asterisk/pbx' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes

On Tue, Feb 6, 2018 at 4:46 PM David Woodhouse <dwmw2 at infradead.org> wrote: > On Wed, 2018-02-07 at 00:36 +0000, Chandler Carruth wrote: > > > > > > > That would be __x86_indirect_thunk but the kernel doesn't use it. > > > We use -mindirect-branch-register and only ever expect the compiler > > > to use the register versions which are

Steve, 1) go to /etc/asterisk 2) open modules.conf for editing using vi 3) add this line: noload=pbx_wilcalu.so 4) Save the file 5) Restart asterisk Lightup the candles, open the Cabernet Savignon ( or whatever your prefernce) and call your girlfriend. ;) Seshu -----Original Message----- From: asterisk-users-bounces@lists.digium.com