VoIP Mailing List Archives

[shawnl at waterwheelne...]

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:

[imthiyaz at peopletech...]

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

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/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>

[saigop at gmail.com]

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) <imthiyaz@peopletech.co.in (imthiyaz@peopletech.co.in)> wrote: