ok it works i had a problem with a syntax: i had to wrire: exten =>_!X.,n(external),Dial(SIP/011212664800...@pstn2,,S(20))
thanks ________________________________ De : Adil Zaaraoui <[email protected]> À : Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Envoyé le : Jeu 8 juillet 2010, 19h 41min 15s Objet : Re : [asterisk-users] Re : Communication IAX2 >SIP>IAX2 Yes i agree; ok here the output of verbosity at level 3: -- Executing [00212664800...@pstn2:1] GotoIf("SIP/100-081e3648", "0?internal:external") in new stack -- Goto (pstn2,00212664800450,2) -- Executing [00212664800...@pstn2:2] Dial("SIP/100-081e3648", "SIP/lo...@pstn2/011212664800450||S(20)") in new stack -- Setting call duration limit to 20 seconds. [Jul 8 17:31:14] WARNING[2960]: chan_sip.c:2952 create_addr: No such host: pstn2/011212664800450 [Jul 8 17:31:14] WARNING[2960]: app_dial.c:1286 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/100-081e3648' status is 'CHANUNAVAIL' -- Executing [...@pstn2:1] DeadAGI("SIP/100-081e3648", "agi://localhost/ManageCalls.agi?when=after") in new stack [Jul 8 17:31:14] ERROR[2960]: utils.c:966 ast_carefulwrite: write() returned error: Connection refused [Jul 8 17:31:14] WARNING[2960]: res_agi.c:222 launch_netscript: Connect to 'agi://localhost/ManageCalls.agi?when=after' failed: Connection refused -- Executing [...@pstn2:2] Dial("SIP/100-081e3648", "SIP/lo...@pstn2/011212664800450||S(20)") in new stack -- Setting call duration limit to 20 seconds. [Jul 8 17:31:14] WARNING[2960]: chan_sip.c:2952 create_addr: No such host: pstn2/011212664800450 [Jul 8 17:31:14] WARNING[2960]: app_dial.c:1286 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) my extention.conf: [pstn2] exten => h,1,DeadAGI(agi://localhost/ManageCalls.agi?when=after) exten=>_!X.,1,GotoIf($["${EXTEN:0:1}"="1"]?internal:external) exten =>_!X.,n(external),Dial(SIP/lo...@pstn2/011212664800450,,S(20)) my sip.conf [general] register=>login:p...@host [pstn2] type=peer host=hostname insecure=invite nat=yes qualify=yes secret=secret username=username canreinvite=no disallow=all allow=ulaw allow=gsm allow=alaw fromdomain=domaineName [100] secret=100 username=100 type=friend context=pstn2 nat=yes disallow=all allow=ulaw allow=gsm allow=alaw host=dynamic i do not know why it prints No such host: pstn2/011212664800450?? Any suggestion ________________________________ De : Paul Belanger <[email protected]> À : Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Envoyé le : Jeu 8 juillet 2010, 12h 10min 14s Objet : Re: [asterisk-users] Re : Communication IAX2 >SIP>IAX2 On Thu, Jul 8, 2010 at 6:29 AM, Adil Zaaraoui <[email protected]> wrote: > But it does not work. > Any suggestion > Without posting a debug log it makes it hard to troubleshoot. http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: [email protected] | IRC: pabelanger (Freenode) blog.polybeacon.com -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
