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@peopletech.co.in < imthiyaz@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@gmail.com Date: Tue, 23 Sep 2008 19:31:07 +0530 To: freeswitch-users@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@172.20.176.254 <sip%3A9894929942@172.20.176.254>< sip%3A9894929942@172.20.176.254 <sip%253A9894929942@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@172.20.176.31 <sip%3A9894929942@172.20.176.31>< sip%3A9894929942@172.20.176.31 <sip%253A9894929942@172.20.176.31>> ;tag=j9a4e9Q4ycvtr To: <sip:9894929942@172.20.176.254 <sip%3A9894929942@172.20.176.254> < sip%3A9894929942@172.20.176.254 <sip%253A9894929942@172.20.176.254>>> Call-ID: 7702517d-0413-122c-efab-0019d150d051 CSeq: 104969298 INVITE Contact: <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 <sip%3A9894929942@172.20.176.31>< sip%3A9894929942@172.20.176.31 <sip%253A9894929942@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,