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-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi i've installed asterisk by last cvs and i note res_parking.c is not anymore there; chan_capi-0.3.4b INSTALL file require: in /etc/asterisk/modules.conf insert the line: load => res_parking.so load => chan_capi.so running asterisk i get: [app_capiCD.so]Jul 21 15:32:26 WARNING[1076988448]: loader.c:242 ast_load_resource:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Ddoes it feasible with * to capture a call? when arrives a call, floor bells ring and everyone can hear them in the company, then everyone can answer 'capturing' the call m. - -- Maurizio Marini GSM +39-335-8259739 Work: +39-0721-855285 Fax +39-0721-859609 Home: +39-0721-950396 IAXTel: (700) 350-1234 -----BEGIN PGP SIGNATURE----- Version:

Hi there asterisk goes to 90% cpu usage when trying to authenticate a sip friend using realtime mysql, no other message does appear at cli and asterisk hungs; here some info: *CLI> realtime load sipfriends name 104 Jan 13 11:52:21 DEBUG[8928]: res_config_mysql.c:109 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sipfriends WHERE name = '104' Jan 13 11:52:21 DEBUG[8928]:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 i fixed wrong capi.conf (there was [controller1] after [interfaces]) now capi.conf is: ; ; CAPI config ; [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] msn=855285,859609 incomingmsn=* controller=1,2,3,4 softdtmf=0 accountcode= context=local ;echosquelch=1 ;echocancel=yes ;echotail=64 ;callgroup=1

Hi, sorry for annoying question; i read http://www.voip-info.org/tiki-index.php?page=PBX%20Call%20Pickup without understanding: 1. how to add an ext. to a pickup group (ie:. how to populate pickup group) 2. how 'Directed pickup' does work? "You dial the pickup number and your extension, and the call will only transfer if it is your extension" should i digit something like

i have a traditional pbx attached to one line of NT1, and asterisk with hfc-s to the other one; when a call comes in, it is like asterisk captures it, passing it to the channel configured on dialplan; in the facts, the call is not answered, but NT1 doesn't ring pbx, as it would do if call had been answered. What is happening? Can i avoid this and have pbx and asterisk ringing

Hi i've an hfc-s card with last bristuff installed at cli> i'm receiving: Sep 8 12:35:20 WARNING[1109552048]: chan_zap.c:6902 zt_pri_error: PRI: !! Got a UA, but i'm in state 1 received TEI check request for TEI = 77 what is causing them? 10x Maurizio

bri-stuff.0.1.0-RC4a on debian sid with 2.6.7 kernel isdn sk: Bus 0, device 18, function 0: Network controller: Digi International Datafire Micro V (Europe) (rev 2). IRQ 17. Master Capable. Latency=16. Max Lat=16. I/O at 0xe400 [0xe407]. Non-prefetchable 32 bit memory at 0xd8021000 [0xd80210ff]. i still receive error messages continuosly (every few seconds) i

For whom which may be interested: Here in Italy we have GSM #numbers without leading zero PSTN instead has prefix starting with '0' to have '0' recognized by * i need to insert nationalprefix=0 as Jason Williams suggested me in irc; now, you cannot have: pridialplan=natonal otherwise * will not be able to call GSM phones you need to setup: pridialplan=local

After that my php script has changed passwords successfully: Apr 30 19:22:20 pdc chgpwd.php[26372]: change password Apr 30 19:22:20 pdc chgpwd.php[26372]: change password: prova - something Apr 30 19:22:20 pdc chgpwd.php[26372]: password changed something happen and message "The User name or old password is incorrect. Letters in passwords must be typed using the correct case." this

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi there, i'm in debian sid 3.1 with kernel 2.6.7, * last cvs & chan_capi 0.3.4b; nt1+ with 2 bri in ptmp (http://www.voip-info.org/tiki-index.php?page=DDI) i tried to install avm c4 following step by step http://www.voip-info.org/tiki-index.php?page=Asterisk%20How%20to%20connect%20with%20CAPI step 1. i compiled capi 2.0 support in kernel

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 hi, i'm new to * I've installed an hfc-s card (DIGI Micro V) with bristuff 0.0.2; when i try to call outside i get: -- Accepting AUTHENTICATED call from 192.168.1.110, requested format = 1024, actual format = 1024 -- Executing Dial("IAX2[pippo@pippo]/2", "Zap/g1/0123456") in new stack Jul 13 13:42:49

Hi i have an issue getting agentLogin working /etc/asterisk/queues.conf member => Agent/1001 member => Agent/1002 extension.conf exten => 110,1,Wait,1 exten => 110,2,AgentLogin() now, i call 110 by a firefly client, trying to login in as 1001 agent: Aug 12 16:31:36 DEBUG[1103408048]: chan_sip.c:4423 build_route: build_route: Contact hop: <sip:sip3@192.168.1.151:5060> --

more detailed output Sep 21 15:54:31 DEBUG[1120357296]: pbx.c:1255 pbx_extension_helper: Launching 'RxFAX' Sep 21 15:54:31 DEBUG[1120357296]: channel.c:1699 ast_set_read_format: Set channel Zap/1-1 to read format SLINR Sep 21 15:54:31 DEBUG[1120357296]: channel.c:1666 ast_set_write_format: Set channel Zap/1-1 to write format SLINR Changed from phase 0 to 1 Slow carrier up Slow carrier