Hi

Any idea why early media is enabled on SIP TRUNKS and what is the impact on the sequence of JTAPI events?

Is there a way to turn this off?

Thanks

Hi

Any response would be much appreciated.

Thanks

I can find Multimedia Early Answer on the station form in CM for H323 stations but not SIP stations. I am not sure what you are referring to with a SIP trunk. Can you please explain further what is happening, and I will see if there is a way to turn it off.

Hi Craig

Thank you for the reply.

Please see the attached image for the setting I am refering to. - Convert 180 to 183 for early media

Thanks

That flag defaults to 'n' on my CM. I was able to change the flag to 'y', and back to 'n' without issue. Do you need help with CM SAT commands to change it or is it not allowing you to submit the change? There is not any document that explains the effects on JTAPI events with each field in CM. The thing to do would be to change the field and see if there are any changes to the events.

Hi Craig

I can change it to y/n without any issues. What I am not sure about is the usage of this setting and under what situations do we enable/disable this.

Thanks

From Communication Manager documentation: When SDP answer is returned by Communication Manager in 18x messages, some entities have a problem receiving the SDP in a 180 Ringing message. If you set the Convert 180 to 183 for Early Media field to y, Communication Manager puts the SDP into a 183 Session Progress message. The default value is n. If you aren't having problems with SDP in the 180 ringing message then you can leave it off. With it on, the codec negotiation is done before the call is setup. As for JTAPI events I can setup and run a test if you open a DevConnect ticket and have support hours available.

Thanks Craig.

I will try to change the flag to 'n' and see how it goes.

Thank you again

Hi Craig

Is this option available on CM 5.2.1?

Thanks

In theory there should be no problem with wanting to send a call from a SIP trunk to another.
I have a SIPTrunk between a predictive dialer and Avaya have another SIPTrunk between two Avaya PBX , then ; I want to send a call from the first to the second SIPTrunk SIPTrunk but short ... Ex: 69065 on the first VDN1 SIPTrunk CUA has a route to VDN2 282274 , and the call is cut with busy tone

Since this is an AE Services JTAPI forum you probably aren't going to get an answer to that question. Further this sounds like an issue that is going to require more in depth troubleshooting. Communication Manager SIP trunk support is available to members at an upgraded membership level. If you have support hours with DevConnect please open a Technical Support ticket (under Communication Manager). If not, please look at upgrading your DevConnect membership.