08-16-2013 05:00 PM - edited 03-16-2019 06:54 PM
Hello,
I've got some expierience with the CUBE, but now I'm setting up a proof of concept where it's SIP to FXS. I'm new to FXS.
Nothing fancy, a SIP call comes into the 2921 with FXS, the dial peer routes to the FXS voice card port, the analog phone rings, answers, and RTP flows both ways normally.
When the analog phone performs a hookflash, I get the dialtone at the alanog phone and a new INVITE is generated by the 2921 back up to to the original SIP UA. That INVITE includes "Media Attribute (a): sendonly", but the RTP stops from the 2921. This is causing the far side to time out with the call because it's not getting the RTP it's looking for.
This doesn't look like normal behaviour to me, but before I start posting configs, I wanted to ask if this is normal and/or is there a simple setting to correct my problem?
Thanks in advance.
Solved! Go to Solution.
08-17-2013 01:10 AM
Todd, this is normal, the far end should send media attribute a=receive only, upon receipt of a send only. In Sip when a call is on hold, it's not a two way traffic, hence why you see send only
08-17-2013 01:10 AM
Todd, this is normal, the far end should send media attribute a=receive only, upon receipt of a send only. In Sip when a call is on hold, it's not a two way traffic, hence why you see send only
08-18-2013 04:06 PM
Thanks. Is there a way to get the 2921 to send a "a=inactive" instead?
08-19-2013 01:09 AM
I guess you can use sip profiles to modify the attributes, but that may break something else..
08-19-2013 12:47 PM
Okay, thanks for the heads up.
Discover and save your favorite ideas. Come back to expert answers, step-by-step guides, recent topics, and more.
New here? Get started with these tips. How to use Community New member guide