invalid media description

[dlegault]

1. Brekeke Product Name and version:2.2.1.6

2. Java version:1.5

3. OS type and the version:win 2003

4. UA (phone), gateway or other hardware/software involved:
speech server 2007

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/networkpatterns.html :pattern 8

6. Your problem:
PSTN to PBX -- works fine
PBX to PTSTN -- works fine
PSTN to PBX to SpeechServerApp -- works fine
SpeechServerApp to PBX to PSTN (or any IP phone) -- dial initiates but once answered immediately terminates with the following error.


Event Type: Warning
Event Source: Office Communications Server 2007 Speech Server
Event Category: Telephony Application Host
Event ID: 32771
Date: 8/12/2008
Time: 12:13:38 PM
User: N/A
Computer: MSSDEV
Description:
The requested operation 'OpenAsync()' on the Telephony Session failed for the following reason: 'An invalid media description was received from the remote end.'.

Further trace information for support personnel follows:

Microsoft.SpeechServer.TelephonyException: An invalid media description was received from the remote end. ---> System.ArgumentException: The requested DTMF payload type (101) has been refused by the remote end.
at Microsoft.SpeechServer.Core.SpeechSession.OpenAsync(MediaDescription mediaDescription, IPAddress localAddress, Boolean useSecureRtp, MediaRelayAuthenticationData mrasData, Boolean receiveRTAudio)
at Microsoft.SpeechServer.Core.TelephonySessionOutbound.SerializedOnMediaResponseReceived(Exception exception, MediaDescription mediaDescription, MediaDescription oldMediaDescription)

[Haddas]

Hi.

Are you using a FXO gateway for connecting to PSTN?
If you are using a gateway, you need to set the same audio codecs as a speech server at the gateway .

Also..let me know your DialPlan settings.

For testing, try a softphone (e.g. X-Lite) instead of PSTN.

[dlegault]

using brekeke pbx ARS to my sip trunk provider

ARS in match:
To -- sip:mynumber@

ARS in deploy:
To -- 01000

rtp relay on
send rtcp on

ARS out match:
To -- sip:9(.+)@

ARS out deploy:
From ="somthing" <sip:123456789@nexvortex.com>
To=sip:$1@nexvortex.com

--------------------------------------------------

sip proxy dial plan added
$request=^INVITE
$port=^15062
To=(.*nexvortex\.com)

deploy
$auth=false
$target=proxy_of_nexvortex
$rtp=on

the above dial plan was added above the "from PBX" rule....

tested going softphone x-lite, and same issue.... from speech server that is......

softphone to pstn
pst to softphone
pstn to speech server
softphone to speech server

all work fine....
speech server to softphone or pstn do not work

[Haddas]

Hi,

The Speech Server's error message indicated the following..
>> The requested DTMF payload type (101) has been refused by the remote end.

So let you try the below.
- go to PBX Admintool > [Options] > [Advanced]　page.
- add
com.brekeke.tel.sip.stack.InviteHandlerBase.SDP_ADD_TELEPHONE_EVENT = true
- push [save] and restart the PBX.

Try it.

[dlegault]

it worked,

can you explain what this does?

thanks again.

[Haddas]

If "com.brekeke.tel.sip.stack.InviteHandlerBase.SDP_ADD_TELEPHONE_EVENT" is "true", the Brekeke PBX adds the DTMF payload type (101) in a response packet such as 200 OK to the Speech Server.
The default value is "false".