search for: ast_unregister_indication_countri

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

Hi all, what is the meaning of this message: Nov 17 19:18:27 NOTICE[1111514032]: indications.c:397 ast_unregister_indication_country: Removed default indication country 'us' Regards Bastian

I have one TDM2404E digium card on asterisk box, after configuring the zaptel and zapata configuration files, I am getting these errors when reloading asterisk: ast_unregister_indication_country: Removed default indication country 'us' setup_zap: Ignoring signalling setup_zap: Ignoring answeronpolarityswitch unable to recognize channel 13-5 what is the reason for that? Thanks,

Hello, I am trying to REGISTER asterisk to a SIP server, which is listening on Port 6060 (not 5060). The sip.conf file contains register=18474201111:quintum@192.168.2.94:6060/18474201111 maxexpirey=3600 defaultexpirey=120 But the REGISTER message is sent to Port 6060, but the Request-URI still contains, 5060. This is being rejected by SIP server. REGISTER sip:192.168.2.94

I can cut and paste the log file from a reload right now, and provide you with the other information when I get home after work: tmp*CLI> sip debug SIP Debugging Enabled tmp*CLI> reload Mar 21 14:52:42 NOTICE[23231]: indications.c:397 ast_unregister_indication_country: Removed default indication country 'us' 11 headers, 0 lines Reliably Transmitting: REGISTER

hi new user here cant seem to get fwd running, got asterisk from download site as tarball, did the readln and openssl start. Also configured the sip.conf and extensions.conf but an error with the chan_sip.c shows up? any ideas...somebody...anybody! thanx jai

I have a an Asterisk server running asterisk 1.0.3 and a TDM04b card. I'm having a problem with my setup. Incoming and outgoing calls are working to 95%. When the other party hangs up their phone after I've hang up mine it starts ringing in my phone. example: 1. I get an incoming call 2. I answer and talk a bit 3. We say goodbye and I hang up the phone 4. The person at the other end hangs

Hi, Using Zack's -shared replacement posted earlier, addons now compiles. For some reason though, when trying to load it cannot find cdr_mysql.conf even though it's in the /etc/asterisk directory as it should be. Anyone with any ideas? There's still references to _i386 files that are probably incorrect as well. Thanks Rob console messages: apsvr1*CLI> reload Mar 23

hi, I've been running a pure sip asterisk box for a while now with no problems, and i've recently added an isdn2e line from bt in the uk. everything is hooked up and i've got it ringing my sip extensions, but the logs don't quite look perfect and i can't see any description of what i should consider to be normal behaviour. would someone be able to look this over and tell me

Anyone actually got this working with asterisk ? I have read posts that it is possible with capi and the diva server cards. Regular diva and diva pro both claim to support NI-1 and NI-2 and CAPI - will they work as well ? Has anyone actually got it working ? Forget the should and could part, I only care about the does/doesn't and why. If you have it working, please tell me - telco,

Hello I am trying to connect linphone 0.12.2 to an * 0.9.1 box over a LAN using the console version of linphone. both boxs are using the latest alsa drivers on a LFS kernal 2.4. I am running into errors with codec compatability between linphone and *. A point to note is that I am able to connect to asterisk using other sip phones noteably sjphone however linephone is giving me