Posts in Category "UV"

UV Call drops automatically in Adobe Connect

PROBLEM :- I have configured SIP account on my connect server and the connection to audio bridge drops automatically.

 

REASON :- This could be a issue with the silence level detection on the remote SIP provider where the call is automatically disconnected when no RTP media / SIP signals are been sent by Adobe connect to remote SIP provider.

Flash Media Gateway (FMG) core.00.log indicates the following lines -:

2011-03-22::09:42:51.339 DEBUG SIP 1908 Received Call msg type : -1
2011-03-22::09:42:51.339 DEBUG SIPLEG 1908 [LEG ID:106] – **callLegStateChangedEvHandler** State=RVSIP_CALL_LEG_STATE_DISCONNECTED Reason=RVSIP_CALL_LEG_REASON_REMOTE_DISCONNECTED Dir=RVSIP_CALL_LEG_DIRECTION_OUTGOING
2011-03-22::09:42:51.339 DEBUG SIP 1908 [LEG ID:106] – Callleg disconnect cause was undefined, now set to 200 for reason RVSIP_CALL_LEG_REASON_REMOTE_DISCONNECTED
2011-03-22::09:42:51.339 DEBUG CALLLEG 1908 [LEG ID:106] – State Change SENDRECV -> HANGUP
2011-03-22::09:42:51.339 DEBUG SIPLEG 1908 [LEG ID:0] – **callLegStateChangedEvHandler** State=RVSIP_CALL_LEG_STATE_TERMINATED Reason=RVSIP_CALL_LEG_REASON_CALL_TERMINATED Dir=RVSIP_CALL_LEG_DIRECTION_OUTGOING
2011-03-22::09:42:51.339 DEBUG SIP 1908 [LEG ID:0] – Callleg disconnect cause was undefined, now set to 500 for reason RVSIP_CALL_LEG_REASON_CALL_TERMINATED
2011-03-22::09:42:51.355 DEBUG CALLLEG 4772 [LEG ID:106] – Hangup Call (cause 200), from FMSMGAppNodesHelper.cpp:1153
2011-03-22::09:42:51.355 DEBUG CALLLEG 4772 [LEG ID:105] – Bridging Completed for
2011-03-22::09:42:51.355 DEBUG CALLLEG 4772 [LEG ID:105] – Hangup Call (cause 200), from FMSMGAppNodesHelper.cpp:1251
2011-03-22::09:42:51.355 INFO CALLLEG 4772 [LEG ID:105] – Hangup [EXEC] [OK]
2011-03-22::09:42:51.355 DEBUG CALLLEG 4772 [LEG ID:105] – App bridge(sip|18665463377@sipGateway) Returns 0 (next ID:1)
2011-03-22::09:42:51.355 DEBUG CALLLEG 4772 [LEG ID:105] – Going For State 7
2011-03-22::09:42:51.355 DEBUG CALLLEG 4772 [LEG ID:105] – Call Leg HANGUP, cause: OK
2011-03-22::09:42:51.355 INFO CALLLEG 4772 [LEG ID:105] – CallLeg Ended
2011-03-22::09:42:51.355 DEBUG CALLLEG 4772 [LEG ID:105] – Cleaning up Leg [HANGUP]
2011-03-22::09:42:51.355 INFO RTMP 4328 Received onStatus <NetStream.Unpublish.Success> code <status> classType <fmg/fmg/1 is now unpublished.> description <(null)> details
2011-03-22::09:42:52.339 DEBUG CALLLEG 4480 [LEG ID:105] – Hangup Call (cause 200), from FMSMGAppNodesHelper.cpp:1153
2011-03-22::09:42:52.339 DEBUG CALLLEG 4480 [LEG ID:106] – Bridging Completed for
2011-03-22::09:42:52.339 DEBUG CALLLEG 4480 [LEG ID:106] – Hangup Call (cause 200), from FMSMGAppNodesHelper.cpp:960
2011-03-22::09:42:52.339 DEBUG CALLLEG 4480 [LEG ID:106] – Going For State 7
2011-03-22::09:42:52.339 DEBUG SIPLEG 4480 [LEG ID:106] – hangupHandler called…sipGateway
2011-03-22::09:42:52.339 DEBUG SIP 4480 Closing audio RTP session
2011-03-22::09:42:52.339 DEBUG SIP 4480 closed audio socket
 
 

SOLUTION -:

1. Do not leave the audio conference bridge idle for silence level detection algorithim

2. Login onto the connect server , open the console page > Flash Media Gateway Settings > Server Configurations.

3. Set the “Registration” value as “Required”.

4. Decrease the value of Registration Expiration in steps of 60

5. Save the configuration and re-test the issue.

6. Repeat step 4 and 5 until the issue is fixed.

 

Conference IVR does not accepts the conference codes.

PROBLEM -: I have enabled Universal voice feature on my Adobe connect server , when i dial into the audio conference bridge (using test dial-in steps), pass the audio conference code , the IVR prompt doesnt realize the conference codes and prompts to re-enter the conference code.

When i dial the same conference bridge using my base phone/mobile and pass the audio conference code , IVR accepts it.

SOLUTION -: Try to replace the SIP gateway/provider/vendor with another vendor which supports DTMF mode RFC 2833.

When you pass the conference code Adobe connect FMG (Flash Media Gateway) sends out DTMF tones to the the remote SIP gateway.

Adobe connect supports only RFC 2833 standard mode of DTMF communication. It seems that the remote SIP provider is not supporting this mode and the conference code send out as DTMF tones by FMG to the audio conference bridge is failing due to non supportive DTMF media (RFC 2833 standards) by the SIP gateway/provider.

Configuring Cisco Call Manager with Adobe Connect

PROBLEM -: I want to enable Universal Voice feature on my on-premise Adobe connect server using our internal Cisco call manager server.

SOLUTION -:

There are some basic network configurations needed for this integration as -:

1. SIP port 5060 should be opened between Adobe connect and cisco call manager for SIP communications

2. Port range 5000 – 6000 should be opened between Adobe connect and cisco call manager  for transversing voice packets via RTP.

3. DTMF mode RFC 2833 must be supported by cisco call manager as well as its carrier.

Integration steps -:

1. I need to create username/password based SIP account on cisco call manager

2. Login into cisco call manager and Navigate to Device -> Phone -> Add new ->

Refer to the screenshot below

 Select Third party Basic device

Select Sip profile as standard sip profile

Select Digest user as LDAP ID of a user x

Check mark Media Termination Point Protocol.

 

3. Navigate to User Management > End User .

Search for LDAP ID of user x

The user-id of the LDAP user x would be the username of SIP account

The pin would be the password of the SIP account.

 

4. Configure your adobe connect SIP account settings with username of SIP account , password of the SIP account as per step 3.