[Freeswitch-users] Freeswitch with Audiocode Mediant 2000

[Shawn Lewis]

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&printable=yes 
> <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 at 172.20.176.254 
> <mailto:sip%3A9894929942 at 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 at 172.20.176.31 
> <mailto:sip%3A9894929942 at 172.20.176.31>>;tag=j9a4e9Q4ycvtr
> To: <sip:9894929942 at 172.20.176.254 
> <mailto:sip%3A9894929942 at 172.20.176.254>>
> Call-ID: 7702517d-0413-122c-efab-0019d150d051
> CSeq: 104969298 INVITE
> Contact: <sip:mod_sofia at 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 at 172.20.176.31 
> <mailto:sip%3A9894929942 at 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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