similar to: What does pbx-wilcalu.so do and why does it keep crashing my * box?

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

Mark, Thanks a lot for the feedback. It's reassuring to say the least Mike Message: 18 Date: Sat, 21 Jan 2006 15:36:18 -0500 From: Mark Phillips <g7ltt@g7ltt.com> Subject: Re: [Asterisk-Users] SIP and NAT - best practices? To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <43D29B42.3060705@g7ltt.com> Content-Type:

So I go into a new Apple store on Sat to buy some stuff for my Mini, and I notice some Snom 360's on the sales counter. Venturing a question, I ask, are they using Asterisk? Guys says yes. Cool! I said: What kind of box are you using. He points to a Mini sitting on the counter! 2 X cool! He's using a SIP-FX0 converter. Plug: http://www.mymacdealer.com great store in Alberta. Anyone else

So it was all working well and then suddenly I'm unable to get incoming calls from BV. Outgoing is fine. I'm using AAH. I have the following settings; register=9738281625@sip.broadvoice.com:PASSWORD-GOES-HERE:9738281625@sip.broadvoice.com/2208 [broadvoice] username=9738281625 user=phone type=peer secret=PASSWORD-GOES-HERE qualify=1000 port=5060 nat=yes insecure=very

One of my buddies signed up with GV yesterday with a view to using them on his * server. Problem is that the settings they gave him don't work with asterisk. They do however work with X-Lite. Any ideas? He's using the settings outlined on my web page. Mark -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

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

Before you all reply that its available via Cisco, I'm not qualified to be a tech member according to Cisco. I just bought 4 7960's with which to use with * and I want to load up the SIP image into them. Does anyone have it that they can make available to me please? Thanks -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/

I've been banging my head with this all day. I today switched from a very old CVS build to AAH1.3 and so far everything has been easy. However I cannot accept calls from a previously working IAX trunk. I've set up an trunk with all the same credentials as before and can call the folks at the other pbx. However whenever they call me I tell them that I don't have an

I was at VON in Boston today and saw on the Digium stand a Cisco 7960 with a picture of Tux and the Asterisk log on its display. I WANT IT!!!!! Anyone know where I can download this file please? -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

Hi, > chan_oh323.so: undefined > symbol: __use_ast_pthread_create_instead__ is not a bug, it's a hint: use "ast_pthread_create" instead [what your were using] and means: replace in asterisk-oh/asterisk-driver/chan_oh323.c at line 3764 "pthread_create" by "ast_pthread_create" Roger.

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

Hi Folks, OK, I'm running CVS Head as of about 3 weeks ago. I want to roll back to V1.09. Other than downloading the code, how do I do it? I thought someone once said that I have to delete all my modules or something? Thanks Mark -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

Does anyone have a collection of stupid hold music? Y'know, the sort of thing that would drive a person mad? Silly songs, repetative tunes etc? The best I can come up with is; I know a song that drives everyone up the wall I know a song that drives everyone up the wall I know a song that drives everyone up the wall And this is how it goes I know a song that drives everyone up the wall I

I thought this had been around before but I can't seem to find anything about it. I have a customer whom prior to upgrading to Asterisk invested in one of those boxes that plays your company sales campaign into the MOH port on your key system. For reasons of message maintenance he wants to keep the box as part of the new system. Can I couple this to the sound card in the Asterisk server

Hi Folks, Prior to upgrading my Zaptel sources everything was working fine. I have a X100P connected to my analogue line. The handset port of the X100P is connected to my desk phone's line 2 input. When the analogue line rings I see the CID on my line 2 but not from Asterisk on line 1 via the Cicso ATA. This used to work fine until I upgraded the sources. I get this when watching the

Hi Yawl, After about 6 months of prattting about I've convinced my boss that we should be installing * into our currently under constuction Data Center in Somerset NJ. There will be 10 permanent people and DR space for another 50. My plan is as follows; ATAComm dual XEON server with quad T1 board. A handfull of ATA's for fax machines, job lot of X-Pro softphones for the DR bit, Polycom

That's great.....it's a virus I tell you * is everywhere :) Viva la asterisk. Cheers, Dean > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Mark Phillips > Sent: Friday, 3 June 2005 6:46 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject:

Hi Folks, I have an IAX trunk link to a collegues house. I'm using AAH and he's got the latest CVS as of last Tuesday. Problem we're having is this; when I dial his extension 7201 (Pulver WiSIP phone) his * box sends me 1 ring and then Alison's busy message. If I call his 7202 extension (X-Ten Pro on a Win2K laptop) I get through but with only 1 way audio (me to him). Until

I just built it and now can no longer get incoming or outgoing service. It was working with CVS Head prior to my "downgrade". Mark -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

Hi all, Anyone got any VoIP traffic shaping rules for m0n0wall that they could let me look at please? Thanks -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com