similar to: Removed default indication country 'us'

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

Hello to everybody, does anybody knows how to install Asterisk on a Linksys WRT54G(S)? I had read in the Wiki that it is possible. If somebody has a tip, this would help me very much. Regards Bastian

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 there, I've got a small problem with the zaphfc channel. No MSN of an any incoming call which comes trough the ISDN card (Acer ISDN, with HFC chipset and zaphfc driver) which will be forwarded to the SIP-Phone will be displayed. Always it will be shown "asterisk" an the Display. --- snip (zapata.conf) --- [channels] language=de switchtype = euroisdn signalling =

Hello, I'm searching for a router for our company. Does anybody has a suggestion for a router with a SIP Application Layer Gateway and good working QoS (Upstream AND Downstream). Regards Bastian

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 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,

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

When I do a "reload" in the Asterisk CLI I get a long list "Unregistered indication country" lines during the parsing of the features.conf file. Then, when parsing the indications.conf file, they seem to all get re-registered (lines saying "Registered indication country" are displayed). What do these lines mean and why are they unregistered and then registered?

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

Hello *, I try to establish a Asterisk-Server for internal and external usage. Perfect use case for a DMZ, or not? My configuration: I N T E R N E T | | | E | | X | | T |

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

Anyone noticed the snom 360 indications are incorrect for US zone? menu->preferences->tone scheme->usa indications.conf: [general] country=us extensions.conf: exten => 1111,1,Answer exten => 1111,n,Playtones(dial) exten => 1111,n,Wait(30) exten => 2222,1,Busy exten => 3333,1,Answer exten => 3333,n,Playtones(busy) exten => 3333,n,Wait(30) hit speakerphone on the

Hello list, Hope you all doing fine! I've tried to use the 'alias' directive in the indications.conf file but apparently it doesn't work.... It looks like maybe this feature was removed, because old sample for the indications.conf file have example using the alias parameter, but newer samples don't have it anymore.... also I couldn't find any ticket saying this parameter

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've got problems to install zaptel on a SuSE 9.1 System. The System has got a Linux 2.6.9 Kernel. If I try to load zaptel framework (modprobe zaptel) I get this message: FATAL: Error inserting zaptel (/lib/modules/2.6.9-041214/misc/zaptel.ko): Invalid module format How I can fix this. At compile time, there were no Errors. Regards Bastian

Hello Everybody, I'm using the snom Phones together with Asterisk and I already able to see which Peer is used via "hint" priority. Then a LED on the snom phone is blinking. But I don't see who is calling the other phone. I know that the snom phones are already support this feature. But how I can enable this on Asterisk? Regards Bastian

Hello Asterisk fans, I try to connect an analog modem to Asterisk. The modems are connected e.g. to alarm systems or a cash terminals (POS). As PSTN-Interface I'm using a Wildcard TE110P (E1). Is it possible to connect the modems to an ATA? Which ATA I should use for that scenarios? Cheers Bastian ____________ Virus checked by G DATA AntiVirusKit Version: AVK 17.1806 from 04.01.2007 Virus

Hi, I tried to use the version 0.6 of chan_capi-cm for outgoing calls it works perfectly but for incoming calls it will not work: --- snip --- *CLI> capi debug CAPI Debugging Enabled -- CONNECT_IND (PLCI=0x101,DID=97,CID=0179903xxxx,CIP=0x1,CONTROLLER=0x1) == reventix: Incoming call '0179903xxxx' -> '97' -- reventix: info element CALLED PARTY NUMBER --