@rabdallah wrote:
I am in the process of setting up a freepbx server to join NRENum. I started by experimenting in a lab environment where I installed two freepbx servers each with a single network interface. The first server has IP 193.1.2.3 and the second server has IP 192.168.1.252. On both servers I have created an ENUM tunnel and a default outbound route with
a dial match pattern X. On server one I have created extension 2252 and on server 2 extension 2059. When I dial from server one 9613763630 the server does a DNS NAPTR lookup and route the call to the second freepbx server however on server 2 I get No matching endpoint foundlogs from server 1 where the call is initiated from extension 2252:
-- Launched AGI Script /var/lib/asterisk/agi-bin/enumlookup.agi
-- enumlookup.agi: Looking up 9613763630 on e164.org via dns_get_record
-- enumlookup.agi: Looking up 9613763630 on e164.arpa via dns_get_record
-- enumlookup.agi: Looking up 9613763630 on e164.info via dns_get_record
-- enumlookup.agi: Looking up 9613763630 on nrenum.net via dns_get_record
-- enumlookup.agi: Setting DIALARR to sip/2092@192.168.1.252%
-- AGI Script enumlookup.agi completed, returning 0
-- Executing [s@macro-dialout-enum:14] ExecIf("PJSIP/2252-00000049", "1?Set(CONNECTEDLINE(num,i)=9613763630)") in new stack
-- Executing [s@macro-dialout-enum:15] ExecIf("PJSIP/2252-00000049", "1?Set(CONNECTEDLINE(name,i)=CID:2252)") in new stack
-- Executing [s@macro-dialout-enum:16] GotoIf("PJSIP/2252-00000049", "0?s-,1") in new stack
-- Executing [s@macro-dialout-enum:17] ExecIf("PJSIP/2252-00000049", "1?Set(DIAL_TRUNK_OPTIONS=M(setmusic^none)Ttr)") in new stack
-- Executing [s@macro-dialout-enum:18] ExecIf("PJSIP/2252-00000049", "0?Set(DIAL_TRUNK_OPTIONS=M(confirm)M(setmusic^none)Ttr)") in new stack
-- Executing [s@macro-dialout-enum:19] Set("PJSIP/2252-00000049", "TRYDIAL=sip/2092@192.168.1.252") in new stack
-- Executing [s@macro-dialout-enum:20] Set("PJSIP/2252-00000049", "DIALARR=") in new stack
-- Executing [s@macro-dialout-enum:21] Dial("PJSIP/2252-00000049", "sip/2092@192.168.1.252,300,M(setmusic^none)Ttr") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called sip/2092@192.168.1.252
[2016-09-26 15:33:16] NOTICE[1960][C-00000044]: chan_sip.c:23827 handle_response_invite: Failed to authenticate on INVITE to ';tag=as75b8ce9b'
-- SIP/192.168.1.252-0000002f is circuit-busyLogs from server 2:
log_failed_request: Request 'INVITE' from '' failed for '193.1.2.3:5160' (callid: 49a998b00b7cfd177a8a7a893a3ab951@193.1.2.3:5160) - No matching endpoint found
freepbx-vm*CLI>I appreciate any help
Regards,
Ramzi
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