[Freeswitch-users] Freeswitch with Audiocode Mediant 2000

[Gopal krishnan]

Hi,

  Once the trunk group is set the calls are getting thru. Thanks for the
kind help.

On Tue, Sep 23, 2008 at 11:56 PM, imthiyaz at peopletech.co.in <
imthiyaz at peopletech.co.in> wrote:

> it can be a codec issue , make sure to use g711 at both ends /updating
> latest firmware can help
>
> Original Message:
> -----------------
> From: Gopal krishnan saigop at gmail.com
> Date: Tue, 23 Sep 2008 19:31:07 +0530
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Freeswitch with Audiocode Mediant 2000
>
>
> 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 <sip%3A9894929942 at 172.20.176.254><
> sip%3A9894929942 at 172.20.176.254 <sip%253A9894929942 at 172.20.176.254>
> >SIP/2.0
> Via: SIP/2.0/UDP 172.20.176.31;rport;branch=z9hG4bKKmB9HrNr22HZQ
> Max-Forwards: 69
> From: "Extension 1002"
> <sip:9894929942 at 172.20.176.31 <sip%3A9894929942 at 172.20.176.31><
> sip%3A9894929942 at 172.20.176.31 <sip%253A9894929942 at 172.20.176.31>>
> >;tag=j9a4e9Q4ycvtr
> To: <sip:9894929942 at 172.20.176.254 <sip%3A9894929942 at 172.20.176.254> <
> sip%3A9894929942 at 172.20.176.254 <sip%253A9894929942 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>
> 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 <sip%3A9894929942 at 172.20.176.31><
> sip%3A9894929942 at 172.20.176.31 <sip%253A9894929942 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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-- 
Thank you with regards,
Gopal,
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