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Atif Rasheed
Community Beginner
Atif Rasheed
Community Beginner
Activity
‎Jul 08, 2014
12:40 AM
Sahil, please elaborate the complete scenario and share more details. 1) I am assuming that 17.12 is your AMG and 17.29 is your asterisk. 2) Please mention which user id is registered on Asterisk and which id is registered on AMG 3) Please share dialplan (relevant section only) snapshot from asterisk and sip.xml and workflow.xml of AMG Thanks.
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‎Jul 07, 2014
02:10 AM
Hi Sahil, can you please share wireshark traces as well ? they will help in isolating the issue. Best regards, -- Atif Rasheed | Technology Consultant mobile: +92 (301) 5500958 | skype: atif.79 | gtalk: atifrasheed On Mon, Jul 7, 2014 at 1:54 PM, Crimson Sahil <forums_noreply@adobe.com>
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‎Jul 03, 2014
04:38 AM
I haven't used any Admin console for AMG and probably none is available. However, API's are available which I have used extensively to get SIP/RTMP leg status. You should also watch AMG logs while you attempt your call. Better if you can pastebin AMG logs as well along with wireshark traces. Best regards, -- Atif Rasheed | Technology Consultant mobile: +92 (301) 5500958 | skype: atif.79 | gtalk: atifrasheed On Thu, Jul 3, 2014 at 4:01 PM, Crimson Sahil <forums_noreply@adobe.com>
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‎Jul 03, 2014
03:34 AM
Can you share SIP trace (wireshark) from your Asterisk and AMG servers ? It will help in root-cause analysis.
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‎May 24, 2014
05:05 AM
1 Upvote
If each stream belongs to a different user then perhaps you can try getUserStats().
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‎May 24, 2014
04:25 AM
Hello there, I trying to integrate AMG with AMS so that a SIP Phone user can join a virtual class in AMS. Now my AMG leg service is connected with AMS but when I dial AMS extension from my sip phone, AMG doesn't answer or bridge the RTMP/SIP-RTP audio stream. Please note that a new stream (from AMG) is created and published in AMS as I can see from the wireshark dump and AMS admin console but some call goes un-answered. Can some one please advise what exactly I am missing out here ? I have pasted AMG logs, RTMP capture and SIP capture. Please note that both AMG and AMS are installed on the same server and I am testing it on a local LAN. I can dial to AMG and can listen to the playback so SIP between my phone and AMG is working. My setup is as below. SIP-UA (Zoiper) -----> AMG ---------> AMS In SIP traces we can see that AMG is not answering the call which means I am not getting 200-OK after INVITE and 100-TRYING. AMG logs 2014-05-20::17:09:11.755 DEBUG 1073742997 CORE 23060 num matches: 1 2014-05-20::17:09:11.755 DEBUG 1073742997 CALLLEG 23060 [LEG ID:1] - Execute bridge(rtmp|${destNum}@t2ams) 2014-05-20::17:09:11.755 DEBUG 1073742997 CALLLEG 23060 [LEG ID:1] - Expanded String bridge(rtmp|9999@t2ams) (ID:1) 2014-05-20::17:09:11.755 DEBUG 1073742997 SIPLEG 23060 [LEG ID:1] - Call in progress 2014-05-20::17:09:11.755 INFO 1073742997 CALLLEG 23060 [LEG ID:1] - Ringing 2014-05-20::17:09:11.755 DEBUG 1073742997 CALLLEG 23060 [LEG ID:2] - Allocated new Call Leg(rtmp) 2014-05-20::17:09:11.756 INFO 1073742997 RTMPLEG 23060 [LEG ID:2] - setting silence level to 0 for Leg 2014-05-20::17:09:11.756 INFO 1073742997 RTMPLEG 23060 [LEG ID:2] - setting silence Timeout to 5000 for Leg 2014-05-20::17:09:11.756 DEBUG 1073742997 CALLLEG 23060 [LEG ID:2] - Going For State 0 2014-05-20::17:09:11.756 DEBUG 1073742997 CALLLEG 23060 [LEG ID:2] - State ALLOC 2014-05-20::17:09:11.792 INFO 1073742997 RTMP 23060 Received onStatus <Success> code <status> classType <StreamCreated> description <> details 2014-05-20::17:09:11.794 INFO 1073742997 RTMP 23060 Received onStatus <NetStream.Publish.Start> code <status> classType <fmg/AutoConnect/1 is now published.> description <(null)> details 2014-05-20::17:09:11.816 DEBUG 1073742997 CALLLEG 23060 [LEG ID:2] - State Change ALLOC -> START 2014-05-20::17:09:11.816 DEBUG 1073742997 CALLLEG 23060 [LEG ID:2] - Going For State 1 2014-05-20::17:09:11.817 DEBUG 1073742997 RTMPLEG 23060 [LEG ID:2] - Inside start handler 2014-05-20::17:09:11.817 DEBUG 1073742997 CALLLEG 23060 [LEG ID:2] - Call Leg START 2014-05-20::17:09:25.701 ERROR 1073742997 RTMP 23060 Received Invalid Command on NetConnection:_OnServerHeartBeat, Message Dropped 2014-05-20::17:09:45.700 ERROR 1073742997 RTMP 23060 Received Invalid Command on NetConnection:_OnServerHeartBeat, Message Dropped 2014-05-20::17:09:51.606 DEBUG 1073742997 SIP 23060 Local Interface : 192.168.1.109 will be used 2014-05-20::17:09:51.606 DEBUG 1073742997 SIP 23060 Registration successful for profile sipPhone2 2014-05-20::17:10:05.699 ERROR 1073742997 RTMP 23060 Received Invalid Command on NetConnection:_OnServerHeartBeat, Message Dropped 2014-05-20::17:10:11.388 DEBUG 1073742997 SIP 23060 Remote SIP address not initialize yet for Profile sipPhone 2014-05-20::17:10:11.762 DEBUG 1073742997 CALLLEG 23060 [LEG ID:2] - Timeout waiting for Answer on Call 2014-05-20::17:10:11.762 DEBUG 1073742997 CALLLEG 23060 [LEG ID:2] - Hangup Call (cause 408), from FMSMGAppNodesHelper.cpp:529 2014-05-20::17:10:11.762 INFO 1073742997 CALLLEG 23060 [LEG ID:2] - Hangup [START] [Request_timeout] 2014-05-20::17:10:11.762 ERROR 1073742997 CALLLEG 23060 [LEG ID:1] - Error Creating Call Leg for destination rtmp|9999@t2ams 2014-05-20::17:10:11.762 DEBUG 1073742997 CALLLEG 23060 [LEG ID:1] - App bridge(rtmp|9999@t2ams) Returns 2 (next ID:1) 2014-05-20::17:10:11.762 DEBUG 1073742997 CALLLEG 23060 [LEG ID:2] - Going For State 7 2014-05-20::17:10:11.762 DEBUG 1073742997 CALLLEG 23060 [LEG ID:1] - Execute hangup(null) 2014-05-20::17:10:11.762 DEBUG 1073742997 CALLLEG 23060 [LEG ID:2] - Call Leg HANGUP, cause: Request_timeout 2014-05-20::17:10:11.762 INFO 1073742997 CALLLEG 23060 [LEG ID:2] - CallLeg Ended RTMP network capture (using tshark) 277.991307 127.0.0.1 -> 127.0.0.1 RTMP 312 createStream() | MGCallLegServiceMessage() 277.991475 127.0.0.1 -> 127.0.0.1 RTMP 109 _result() 277.991616 127.0.0.1 -> 127.0.0.1 RTMP 127 publish('fmg/AutoConnect/1') 277.992596 127.0.0.1 -> 127.0.0.1 RTMP 241 Set Chunk Size 1024 | onStatus('NetStream.Publish.Start') 278.010156 127.0.0.1 -> 127.0.0.1 RTMP 269 MGLegServiceMessage() 278.015654 127.0.0.1 -> 127.0.0.1 RTMP 225 MGCallLegServiceMessage() SIP network capture (using tshark) 12 31.896848 192.168.1.100 -> 192.168.1.109 SIP/SDP 968 Request: INVITE sip:9999@192.168.1.109;transport=UDP | 31.897834 192.168.1.109 -> 192.168.1.100 SIP 511 Status: 407 Proxy Authentication Required | 31.914622 192.168.1.100 -> 192.168.1.109 SIP 436 Request: ACK sip:9999@192.168.1.109;transport=UDP | 31.914640 192.168.1.100 -> 192.168.1.109 SIP/SDP 1209 Request: INVITE sip:9999@192.168.1.109;transport=UDP | 31.915874 192.168.1.109 -> 192.168.1.100 SIP 387 Status: 100 Trying | 91.926390 192.168.1.109 -> 192.168.1.100 SIP 400 Status: 408 Request Timeout | 92.029868 192.168.1.100 -> 192.168.1.109 SIP 436 Request: ACK sip:9999@192.168.1.109;transport=UDP |
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‎May 23, 2014
07:22 AM
Hi, thanks for your help. My setup is as below. SIP-UA (Zoiper) -----> AMG ---------> AMS So basically AMG is my SIP server and I register my soft phone on AMG and then dial a number defined in workflow.xml. "sipPhone" profile is a different profile which is currently not being used and profile I am currently using is "sipPhone2" so that is definitely not the issue. Please note that I can listen to the IVR/Prompts on AMG, secondly SIP Leg is established and that is why AMG subsequently initiates session with AMS if we follow the sequence. So I believe AMS has to establish session first, RTMP session between AMG and AMS and then in return AMG will answer the SIP call. Issue lies between AMG/AMS over the RTMP. I am probably missing something as AMS doesn't initiate RTMP stream even after a successful CONNECT and PUBLISH. And perhaps AMG waits for the RTMP stream (audio) before it could answer the SIP call. I am just presuming as AMG documentation is very limited and I have no details as how the backend actually works.
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‎May 20, 2014
05:54 AM
Hello there, I trying to integrate AMG with AMS so that a SIP Phone user can join a virtual class in AMS. Now my AMG leg service is connected with AMS but when I dial AMS extension from my sip phone, AMG doesn't answer or bridge the RTMP/SIP-RTP audio stream. Please note that a new stream (from AMG) is created and published in AMS as I can see from the wireshark dump and AMS admin console but some call goes un-answered. Can some one please advise what exactly I am missing out here ? I have pasted AMG logs, RTMP capture and SIP capture. Please note that both AMG and AMS are installed on the same server and I am testing it on a local LAN. In SIP traces we can see that AMG is not answering the call which means I am not getting 200-OK after INVITE and 100-TRYING. AMG logs 2014-05-20::17:09:11.755 DEBUG 1073742997 CORE 23060 num matches: 1 2014-05-20::17:09:11.755 DEBUG 1073742997 CALLLEG 23060 [LEG ID:1] - Execute bridge(rtmp|${destNum}@t2ams) 2014-05-20::17:09:11.755 DEBUG 1073742997 CALLLEG 23060 [LEG ID:1] - Expanded String bridge(rtmp|9999@t2ams) (ID:1) 2014-05-20::17:09:11.755 DEBUG 1073742997 SIPLEG 23060 [LEG ID:1] - Call in progress 2014-05-20::17:09:11.755 INFO 1073742997 CALLLEG 23060 [LEG ID:1] - Ringing 2014-05-20::17:09:11.755 DEBUG 1073742997 CALLLEG 23060 [LEG ID:2] - Allocated new Call Leg(rtmp) 2014-05-20::17:09:11.756 INFO 1073742997 RTMPLEG 23060 [LEG ID:2] - setting silence level to 0 for Leg 2014-05-20::17:09:11.756 INFO 1073742997 RTMPLEG 23060 [LEG ID:2] - setting silence Timeout to 5000 for Leg 2014-05-20::17:09:11.756 DEBUG 1073742997 CALLLEG 23060 [LEG ID:2] - Going For State 0 2014-05-20::17:09:11.756 DEBUG 1073742997 CALLLEG 23060 [LEG ID:2] - State ALLOC 2014-05-20::17:09:11.792 INFO 1073742997 RTMP 23060 Received onStatus <Success> code <status> classType <StreamCreated> description <> details 2014-05-20::17:09:11.794 INFO 1073742997 RTMP 23060 Received onStatus <NetStream.Publish.Start> code <status> classType <fmg/AutoConnect/1 is now published.> description <(null)> details 2014-05-20::17:09:11.816 DEBUG 1073742997 CALLLEG 23060 [LEG ID:2] - State Change ALLOC -> START 2014-05-20::17:09:11.816 DEBUG 1073742997 CALLLEG 23060 [LEG ID:2] - Going For State 1 2014-05-20::17:09:11.817 DEBUG 1073742997 RTMPLEG 23060 [LEG ID:2] - Inside start handler 2014-05-20::17:09:11.817 DEBUG 1073742997 CALLLEG 23060 [LEG ID:2] - Call Leg START 2014-05-20::17:09:25.701 ERROR 1073742997 RTMP 23060 Received Invalid Command on NetConnection:_OnServerHeartBeat, Message Dropped 2014-05-20::17:09:45.700 ERROR 1073742997 RTMP 23060 Received Invalid Command on NetConnection:_OnServerHeartBeat, Message Dropped 2014-05-20::17:09:51.606 DEBUG 1073742997 SIP 23060 Local Interface : 192.168.1.109 will be used 2014-05-20::17:09:51.606 DEBUG 1073742997 SIP 23060 Registration successful for profile sipPhone2 2014-05-20::17:10:05.699 ERROR 1073742997 RTMP 23060 Received Invalid Command on NetConnection:_OnServerHeartBeat, Message Dropped 2014-05-20::17:10:11.388 DEBUG 1073742997 SIP 23060 Remote SIP address not initialize yet for Profile sipPhone 2014-05-20::17:10:11.762 DEBUG 1073742997 CALLLEG 23060 [LEG ID:2] - Timeout waiting for Answer on Call 2014-05-20::17:10:11.762 DEBUG 1073742997 CALLLEG 23060 [LEG ID:2] - Hangup Call (cause 408), from FMSMGAppNodesHelper.cpp:529 2014-05-20::17:10:11.762 INFO 1073742997 CALLLEG 23060 [LEG ID:2] - Hangup [START] [Request_timeout] 2014-05-20::17:10:11.762 ERROR 1073742997 CALLLEG 23060 [LEG ID:1] - Error Creating Call Leg for destination rtmp|9999@t2ams 2014-05-20::17:10:11.762 DEBUG 1073742997 CALLLEG 23060 [LEG ID:1] - App bridge(rtmp|9999@t2ams) Returns 2 (next ID:1) 2014-05-20::17:10:11.762 DEBUG 1073742997 CALLLEG 23060 [LEG ID:2] - Going For State 7 2014-05-20::17:10:11.762 DEBUG 1073742997 CALLLEG 23060 [LEG ID:1] - Execute hangup(null) 2014-05-20::17:10:11.762 DEBUG 1073742997 CALLLEG 23060 [LEG ID:2] - Call Leg HANGUP, cause: Request_timeout 2014-05-20::17:10:11.762 INFO 1073742997 CALLLEG 23060 [LEG ID:2] - CallLeg Ended RTMP network capture (using tshark) 277.991307 127.0.0.1 -> 127.0.0.1 RTMP 312 createStream() | MGCallLegServiceMessage() 277.991475 127.0.0.1 -> 127.0.0.1 RTMP 109 _result() 277.991616 127.0.0.1 -> 127.0.0.1 RTMP 127 publish('fmg/AutoConnect/1') 277.992596 127.0.0.1 -> 127.0.0.1 RTMP 241 Set Chunk Size 1024 | onStatus('NetStream.Publish.Start') 278.010156 127.0.0.1 -> 127.0.0.1 RTMP 269 MGLegServiceMessage() 278.015654 127.0.0.1 -> 127.0.0.1 RTMP 225 MGCallLegServiceMessage() SIP network capture (using tshark) 12 31.896848 192.168.1.100 -> 192.168.1.109 SIP/SDP 968 Request: INVITE sip:9999@192.168.1.109;transport=UDP | 31.897834 192.168.1.109 -> 192.168.1.100 SIP 511 Status: 407 Proxy Authentication Required | 31.914622 192.168.1.100 -> 192.168.1.109 SIP 436 Request: ACK sip:9999@192.168.1.109;transport=UDP | 31.914640 192.168.1.100 -> 192.168.1.109 SIP/SDP 1209 Request: INVITE sip:9999@192.168.1.109;transport=UDP | 31.915874 192.168.1.109 -> 192.168.1.100 SIP 387 Status: 100 Trying | 91.926390 192.168.1.109 -> 192.168.1.100 SIP 400 Status: 408 Request Timeout | 92.029868 192.168.1.100 -> 192.168.1.109 SIP 436 Request: ACK sip:9999@192.168.1.109;transport=UDP |
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‎May 20, 2014
05:33 AM
Hi, I embedded/merged your script with our SSAS. Now my leg service is connected but when I dial AMS from my sip phone AMG doesn't answer or bridge the RTMP/SIP-RTP audio stream. Stream is created and published in AMS as I can see from the wireshark dump and AMS admin console. Can you please advise what exactly I am missing out here ? I have pasted AMG logs, RTMP capture and SIP capture. Please note that both AMG and AMS are installed on the same server and I am testing it on a local LAN. In SIP traces I can see that AMG is not answering the call i.e. I am not getting 200-OK after INVITE and 100-TRYING. AMG logs 2014-05-20::17:09:11.755 DEBUG 1073742997 CORE 23060 num matches: 1 2014-05-20::17:09:11.755 DEBUG 1073742997 CALLLEG 23060 [LEG ID:1] - Execute bridge(rtmp|${destNum}@t2ams) 2014-05-20::17:09:11.755 DEBUG 1073742997 CALLLEG 23060 [LEG ID:1] - Expanded String bridge(rtmp|9999@t2ams) (ID:1) 2014-05-20::17:09:11.755 DEBUG 1073742997 SIPLEG 23060 [LEG ID:1] - Call in progress 2014-05-20::17:09:11.755 INFO 1073742997 CALLLEG 23060 [LEG ID:1] - Ringing 2014-05-20::17:09:11.755 DEBUG 1073742997 CALLLEG 23060 [LEG ID:2] - Allocated new Call Leg(rtmp) 2014-05-20::17:09:11.756 INFO 1073742997 RTMPLEG 23060 [LEG ID:2] - setting silence level to 0 for Leg 2014-05-20::17:09:11.756 INFO 1073742997 RTMPLEG 23060 [LEG ID:2] - setting silence Timeout to 5000 for Leg 2014-05-20::17:09:11.756 DEBUG 1073742997 CALLLEG 23060 [LEG ID:2] - Going For State 0 2014-05-20::17:09:11.756 DEBUG 1073742997 CALLLEG 23060 [LEG ID:2] - State ALLOC 2014-05-20::17:09:11.792 INFO 1073742997 RTMP 23060 Received onStatus <Success> code <status> classType <StreamCreated> description <> details 2014-05-20::17:09:11.794 INFO 1073742997 RTMP 23060 Received onStatus <NetStream.Publish.Start> code <status> classType <fmg/AutoConnect/1 is now published.> description <(null)> details 2014-05-20::17:09:11.816 DEBUG 1073742997 CALLLEG 23060 [LEG ID:2] - State Change ALLOC -> START 2014-05-20::17:09:11.816 DEBUG 1073742997 CALLLEG 23060 [LEG ID:2] - Going For State 1 2014-05-20::17:09:11.817 DEBUG 1073742997 RTMPLEG 23060 [LEG ID:2] - Inside start handler 2014-05-20::17:09:11.817 DEBUG 1073742997 CALLLEG 23060 [LEG ID:2] - Call Leg START 2014-05-20::17:09:25.701 ERROR 1073742997 RTMP 23060 Received Invalid Command on NetConnection:_OnServerHeartBeat, Message Dropped 2014-05-20::17:09:45.700 ERROR 1073742997 RTMP 23060 Received Invalid Command on NetConnection:_OnServerHeartBeat, Message Dropped 2014-05-20::17:09:51.606 DEBUG 1073742997 SIP 23060 Local Interface : 192.168.1.109 will be used 2014-05-20::17:09:51.606 DEBUG 1073742997 SIP 23060 Registration successful for profile sipPhone2 2014-05-20::17:10:05.699 ERROR 1073742997 RTMP 23060 Received Invalid Command on NetConnection:_OnServerHeartBeat, Message Dropped 2014-05-20::17:10:11.388 DEBUG 1073742997 SIP 23060 Remote SIP address not initialize yet for Profile sipPhone 2014-05-20::17:10:11.762 DEBUG 1073742997 CALLLEG 23060 [LEG ID:2] - Timeout waiting for Answer on Call 2014-05-20::17:10:11.762 DEBUG 1073742997 CALLLEG 23060 [LEG ID:2] - Hangup Call (cause 408), from FMSMGAppNodesHelper.cpp:529 2014-05-20::17:10:11.762 INFO 1073742997 CALLLEG 23060 [LEG ID:2] - Hangup [START] [Request_timeout] 2014-05-20::17:10:11.762 ERROR 1073742997 CALLLEG 23060 [LEG ID:1] - Error Creating Call Leg for destination rtmp|9999@t2ams 2014-05-20::17:10:11.762 DEBUG 1073742997 CALLLEG 23060 [LEG ID:1] - App bridge(rtmp|9999@t2ams) Returns 2 (next ID:1) 2014-05-20::17:10:11.762 DEBUG 1073742997 CALLLEG 23060 [LEG ID:2] - Going For State 7 2014-05-20::17:10:11.762 DEBUG 1073742997 CALLLEG 23060 [LEG ID:1] - Execute hangup(null) 2014-05-20::17:10:11.762 DEBUG 1073742997 CALLLEG 23060 [LEG ID:2] - Call Leg HANGUP, cause: Request_timeout 2014-05-20::17:10:11.762 INFO 1073742997 CALLLEG 23060 [LEG ID:2] - CallLeg Ended RTMP network capture (using tshark) 277.991307 127.0.0.1 -> 127.0.0.1 RTMP 312 createStream() | MGCallLegServiceMessage() 277.991475 127.0.0.1 -> 127.0.0.1 RTMP 109 _result() 277.991616 127.0.0.1 -> 127.0.0.1 RTMP 127 publish('fmg/AutoConnect/1') 277.992596 127.0.0.1 -> 127.0.0.1 RTMP 241 Set Chunk Size 1024 | onStatus('NetStream.Publish.Start') 278.010156 127.0.0.1 -> 127.0.0.1 RTMP 269 MGLegServiceMessage() 278.015654 127.0.0.1 -> 127.0.0.1 RTMP 225 MGCallLegServiceMessage() SIP network capture (using tshark) 12 31.896848 192.168.1.100 -> 192.168.1.109 SIP/SDP 968 Request: INVITE sip:9999@192.168.1.109;transport=UDP | 31.897834 192.168.1.109 -> 192.168.1.100 SIP 511 Status: 407 Proxy Authentication Required | 31.914622 192.168.1.100 -> 192.168.1.109 SIP 436 Request: ACK sip:9999@192.168.1.109;transport=UDP | 31.914640 192.168.1.100 -> 192.168.1.109 SIP/SDP 1209 Request: INVITE sip:9999@192.168.1.109;transport=UDP | 31.915874 192.168.1.109 -> 192.168.1.100 SIP 387 Status: 100 Trying | 91.926390 192.168.1.109 -> 192.168.1.100 SIP 400 Status: 408 Request Timeout | 92.029868 192.168.1.100 -> 192.168.1.109 SIP 436 Request: ACK sip:9999@192.168.1.109;transport=UDP |
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‎Apr 02, 2014
10:57 AM
Hi Pranav, please refer to my first email. I am trying to view the PDF in adobe docs url mentioned in my first email. Best regards, -- Atif Rasheed | Technology Consultant mobile: +92 (301) 5500958 | skype: atif.79 | gtalk: atifrasheed
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‎Apr 02, 2014
06:46 AM
Hi Rave, when I try to open it I get a different error. I have attached the screen shot for your reference.
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‎Apr 02, 2014
06:38 AM
Hi, Did Adobe discontinued AMG support or perhaps the product itself? I do not see alot of AMG documentation and activity on AMG forums. If yes then what would be the alternative for SIP to RTMP? Thank you.
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‎Apr 02, 2014
05:57 AM
We have a virtual class room application (like WizIQ) and I want SIP users to dial in to the class room. I have configured AMG with AMS where our SSAS is hosted. Now if I enable AutoConnect in AMG profiles then I get below error. DEBUG 1073742997 RTMP 32433 Received onStatus <NetConnection.Call.Failed> code <error> classType <Method not found (MGCallLegServiceMessage).> description <(null)> details And when I try to dial my application instance I get an error that RTMP profile is not connected. DEBUG 1073742997 CALLLEG 32433 [LEG ID:3] - Execute bridge(rtmp|${destNum}@ams2_aws) DEBUG 1073742997 CALLLEG 32433 [LEG ID:3] - Expanded String bridge(rtmp|7777@ams2_aws) (ID:1) DEBUG 1073742997 SIPLEG 32433 [LEG ID:3] - Call in progress INFO 1073742997 CALLLEG 32433 [LEG ID:3] - Ringing DEBUG 1073742997 CALLLEG 32433 [LEG ID:4] - Allocated new Call Leg(rtmp) ERROR 1073742997 RTMPLEG 32433 [LEG ID:4] - Destination rtmp profile/URL (rtmp://ams.teletaaleem.com/vod/_definst_) not connected ERROR 1073742997 CALLLEG 32433 [LEG ID:4] - cannot create call Leg DEBUG 1073742997 CALLLEG 32433 [LEG ID:4] - Cleaning up Leg [ALLOC] ERROR 1073742997 CALLLEG 32433 [LEG ID:3] - Error Creating Call Leg for destination rtmp|7777@ams2_aws DEBUG 1073742997 CALLLEG 32433 [LEG ID:3] - App bridge(rtmp|7777@ams2_aws) Returns 4 (next ID:1) DEBUG 1073742997 CALLLEG 32433 [LEG ID:3] - Execute hangup(null) DEBUG 1073742997 CALLLEG 32433 [LEG ID:3] - Hangup Call (cause 200), from FMSMGBasicAppNodes.cpp:235 INFO 1073742997 CALLLEG 32433 [LEG ID:3] - Hangup [EXEC] [Not_found] do I have to manage SIP in AMS SSAS ? if we are handling SIP in AMG then why we need to handle this AMG leg service seperately in AMS SSAS?
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‎Apr 02, 2014
05:41 AM
Hi, I am facing same issue but I think it's because of RTMP profile which is not registered on the AMS. RTMP Profile AutoConnect is disabled in rtmp.xml and this is what I see in the logs. 2014-04-02::17:33:44.018 DEBUG 1073742997 CALLLEG 30201 [LEG ID:5] - Execute bridge(rtmp|${destNum}@ams_aws) 2014-04-02::17:33:44.019 DEBUG 1073742997 CALLLEG 30201 [LEG ID:5] - Expanded String bridge(rtmp|9999@ams_aws) (ID:1) 2014-04-02::17:33:44.019 DEBUG 1073742997 SIPLEG 30201 [LEG ID:5] - Call in progress 2014-04-02::17:33:44.019 INFO 1073742997 CALLLEG 30201 [LEG ID:5] - Ringing 2014-04-02::17:33:44.019 DEBUG 1073742997 CALLLEG 30201 [LEG ID:6] - Allocated new Call Leg(rtmp) 2014-04-02::17:33:44.019 ERROR 1073742997 RTMPLEG 30201 [LEG ID:6] - Destination rtmp profile/URL (rtmp://ams.teletaaleem.com/t2/39/3015/1) not connected 2014-04-02::17:33:44.019 ERROR 1073742997 CALLLEG 30201 [LEG ID:6] - cannot create call Leg 2014-04-02::17:33:44.019 DEBUG 1073742997 CALLLEG 30201 [LEG ID:6] - Cleaning up Leg [ALLOC] 2014-04-02::17:33:44.019 ERROR 1073742997 CALLLEG 30201 [LEG ID:5] - Error Creating Call Leg for destination rtmp|9999@ams_aws 2014-04-02::17:33:44.019 DEBUG 1073742997 CALLLEG 30201 [LEG ID:5] - App bridge(rtmp|9999@ams_aws) Returns 4 (next ID:1) 2014-04-02::17:33:44.019 DEBUG 1073742997 CALLLEG 30201 [LEG ID:5] - Execute hangup(null) 2014-04-02::17:33:44.019 DEBUG 1073742997 CALLLEG 30201 [LEG ID:5] - Hangup Call (cause 200), from FMSMGBasicAppNodes.cpp:235 2014-04-02::17:33:44.019 INFO 1073742997 CALLLEG 30201 [LEG ID:5] - Hangup [EXEC] [Not_found] 2014-04-02::17:33:44.019 DEBUG 1073742997 CALLLEG 30201 [LEG ID:5] - App hangup(null) Returns 0 (next ID:2) 2014-04-02::17:33:44.019 DEBUG 1073742997 CALLLEG 30201 [LEG ID:5] - Going For State 7
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‎May 13, 2013
01:20 AM
Hi there, Below pdf in whitepapers section is not opening. It says corrupt file. Please look into it. http://download.macromedia.com/pub/solutions/downloads/elearning/standards.pdf Thank you.
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