Looks to me that the Audiocodes is reporting that it has no routing. or the routing for TN '9894929942' is not available to the Audiocodes when attempting route the call out. ??? 1d:2h:46m:9s ( lgr_TrnkGrp)(344 ) !! [ERROR] #1:TrunkGroup::AllocateEndPoint- Can't find EndPoint for phone number 9894929942 1d:2h:46m:9s ( lgr_psbrdif)(345 ) !! [ERROR] AcBoard::GetEndPoint- Can't find EndPoint for Dest:9894929942 Source:9894929942 SourceIp:ac14b01f Shawn Gopal krishnan wrote:

Hi,

I followed the below link to configure the Audiocode Mediant 2000 with Freeswitch http://wiki.freeswitch.org/index.php?title=Configuring_AudioCodes_MP-114/118... <http://wiki.freeswitch.org/index.php?title=Configuring_AudioCodes_MP-114/118&printable=yes>

but the above link is for FXO line, where I am using digital PRI line.

when I try to dial I am getting call failed, the traffic from freeswitch were hitting audiocode the log as follows, attached with this email,

*some sample SIP header as follows,* d:2h:17m:7s INVITE sip:9894929942@172.20.176.254 <mailto:sip%3A9894929942@172.20.176.254> SIP/2.0 Via: SIP/2.0/UDP 172.20.176.31 <http://172.20.176.31>;rport;branch=z9hG4bKKmB9HrNr22HZQ Max-Forwards: 69 From: "Extension 1002" <sip:9894929942@172.20.176.31 <mailto:sip%3A9894929942@172.20.176.31>>;tag=j9a4e9Q4ycvtr To: <sip:9894929942@172.20.176.254 <mailto:sip%3A9894929942@172.20.176.254>> Call-ID: 7702517d-0413-122c-efab-0019d150d051 CSeq: 104969298 INVITE Contact: <sip:mod_sofia@172.20.176.31:5060 <http://sip:mod_sofia@172.20.176.31:5060>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9596M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: 100rel, timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 347 Remote-Party-ID: "Extension 1002" <sip:9894929942@172.20.176.31 <mailto:sip%3A9894929942@172.20.176.31>>;screen=yes;privacy=off

1d:2h:17m:7s ( sip_stack)(212 ) ?? [WARNING] AcSIPParser: Unrecognized Header was detected at line: 12

1d:2h:46m:9s ( lgr_TrnkGrp)(344 ) !! [ERROR] #1:TrunkGroup::AllocateEndPoint- Can't find EndPoint for phone number 9894929942

1d:2h:46m:9s ( lgr_psbrdif)(345 ) !! [ERROR] AcBoard::GetEndPoint- Can't find EndPoint for Dest:9894929942 Source:9894929942 SourceIp:ac14b01f

1d:2h:46m:9s ( lgr_psbrdif)(346 ) TrunkBoard::GetEndPoint- Current trunk status:0010

1d:2h:46m:9s ( lgr_call)(347 ) !! [ERROR] Call::GetEndPoint- Can't find endpoint for phone number 9894929942

*Freeswitch log* *as follows* http://pastebin.freeswitch.org/5635

So how to proceed in this stage. -- Thank you with regards, Gopal,

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