[Spce-user] Help with upgrade...

[Derrick Bradbury]

Something else to add.. on the inbound call. I don't get an "answer" in the RTP log:

(outbound call)
Jan 21 14:53:31 sipce01 rtpengine[4391]: Got valid command from 127.0.0.1:41852: answer - { "sdp":  .....

On the inbound call, that doesn't show up.. I get:
Jan 21 14:54:37 sipce01 rtpengine[4391]: [edd375f522f07c59] Call branch '533EF9AE-54C003B4000BA0EF-971C8700' deleted, no more branches remaining


________________________________
From: Spce-user [spce-user-bounces at lists.sipwise.com] on behalf of Derrick Bradbury [derrickb at halex.com]
Sent: Wednesday, January 21, 2015 2:16 PM
To: spce-user ‎[spce-user at lists.sipwise.com]‎
Subject: [Spce-user] Help with upgrade...

Hi there,

A couple things... first a simple request for the next release..

Can you add a configurable port for SSH:

# What ports, IPs and protocols we listen for
[% IF sshd.port -%]
Port [% sshd.port %]
[% ELSE -%]
Port 22
[% END -%]

or something like that...

Now the main problem...

I went from 3.2.1 -> 3.7.1 over multiple upgrades.....

Now, I have outbound calls working no problems, but my inbound has no audio, and drops after 32 seconds.

My first issue was I couldn't compile the RTPENGINE module, because the header files were missing, that was solved, its now there...

But now when I get an inbound call, there is a disconnect on the audio stream! I can see rtp packets coming in on both sides, but nothing makes it through.

Then after 32 seconds, the call hangs up.

AAA.XXX.XXX - external server
BBB.XXX.XXX - SipCE servers
CCC.XXX.XXX - testing client logged into sipce

config.yml isn't done yet.. still a few defaults...

Any help is appreciated... Thanks!

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20150121/6badb4d4/attachment-0001.html>