<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:times new roman,new york,times,serif;font-size:10pt"><div><br></div><div style="font-family: times new roman,new york,times,serif; font-size: 10pt;">Yes i agree; ok here the output of verbosity at level 3:<br> -- Executing [00212664800450@pstn2:1] GotoIf("SIP/100-081e3648", "0?internal:external") in new stack<br> -- Goto (pstn2,00212664800450,2)<br> -- Executing [00212664800450@pstn2:2] Dial("SIP/100-081e3648", "SIP/login@pstn2/011212664800450||S(20)") in new stack<br> -- Setting call duration limit to 20 seconds.<br>[Jul 8 17:31:14] WARNING[2960]: chan_sip.c:2952 create_addr: No such host: pstn2/011212664800450<br>[Jul 8 17:31:14] WARNING[2960]: app_dial.c:1286 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)<br> == Everyone is busy/congested at
this time (1:0/0/1)<br> == Auto fallthrough, channel 'SIP/100-081e3648' status is 'CHANUNAVAIL'<br> -- Executing [h@pstn2:1] DeadAGI("SIP/100-081e3648", "agi://localhost/ManageCalls.agi?when=after") in new stack<br>[Jul 8 17:31:14] ERROR[2960]: utils.c:966 ast_carefulwrite: write() returned error: Connection refused<br>[Jul 8 17:31:14] WARNING[2960]: res_agi.c:222 launch_netscript: Connect to 'agi://localhost/ManageCalls.agi?when=after' failed: Connection refused<br> -- Executing [h@pstn2:2] Dial("SIP/100-081e3648", "SIP/login@pstn2/011212664800450||S(20)") in new stack<br> -- Setting call duration limit to 20 seconds.<br>[Jul 8 17:31:14] WARNING[2960]: chan_sip.c:2952 create_addr: No such host: pstn2/011212664800450<br>[Jul 8 17:31:14] WARNING[2960]: app_dial.c:1286 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)<br> == Everyone is
busy/congested at this time (1:0/0/1)<br><br>my extention.conf:<br><br>[pstn2]<br><br>exten => h,1,DeadAGI(agi://localhost/ManageCalls.agi?when=after)<br>exten=>_!X.,1,GotoIf($["${EXTEN:0:1}"="1"]?internal:external)<br>exten =>_!X.,n(external),Dial(SIP/login@pstn2/011212664800450,,S(20))<br><br>my sip.conf<br>[general]<br>register=>login:pass@host<br><br><br><br>[pstn2]<br>type=peer<br>host=hostname<br>insecure=invite<br>nat=yes<br>qualify=yes<br>secret=secret<br>username=username<br>canreinvite=no<br>disallow=all<br>allow=ulaw<br>allow=gsm<br>allow=alaw<br>fromdomain=domaineName<br><br><br>[100]<br>secret=100<br>username=100<br>type=friend<br>context=pstn2<br>nat=yes<br>disallow=all<br>allow=ulaw<br>allow=gsm<br>allow=alaw<br>host=dynamic<br><br><br>i do not know why it prints No such host: pstn2/011212664800450??<br>Any suggestion<br> <div style="font-family: arial,helvetica,sans-serif; font-size: 13px;"><font size="2" face="Tahoma"><hr
size="1"><b><span style="font-weight: bold;">De :</span></b> Paul Belanger <paul.belanger@polybeacon.com><br><b><span style="font-weight: bold;">À :</span></b> Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com><br><b><span style="font-weight: bold;">Envoyé le :</span></b> Jeu 8 juillet 2010, 12h 10min 14s<br><b><span style="font-weight: bold;">Objet :</span></b> Re: [asterisk-users] Re : Communication IAX2 >SIP>IAX2<br></font><br>On Thu, Jul 8, 2010 at 6:29 AM, Adil Zaaraoui <<a ymailto="mailto:adilzeaaraoui@yahoo.fr" href="mailto:adilzeaaraoui@yahoo.fr">adilzeaaraoui@yahoo.fr</a>> wrote:<br>> But it does not work.<br>> Any suggestion<br>><br>Without posting a debug log it makes it hard to troubleshoot.<br><br><a href="http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt"
target="_blank">http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt</a><br><br>-- <br>Paul Belanger | dCAP<br>Polybeacon | Consultant<br>Jabber: <a ymailto="mailto:paul.belanger@polybeacon.com" href="mailto:paul.belanger@polybeacon.com">paul.belanger@polybeacon.com</a> | IRC: pabelanger (Freenode)<br>blog.polybeacon.com<br><br>-- <br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users"
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