This looks like a nat issue to me, please re-test this against latest svn trunk
and if its still not working pastebin a full sip trace and report the link back
here.
Mike
On Nov 21, 2009, at 6:23 PM, RobertT wrote:
Yep, I use proxy media. First it started with 1.0.4 release, then I've
OK, this is what I've got.
First, I've updated FreeSwitch from trunk to version 15630 and deployed it
to my server. Performed a tets and again no magic happened. The link to SIP
trace is below.
Then I've installed 1.0.4 version to another server (virtual hosting), and
performed tha same. And
You know what, guys? I've just made it working be opening ALL tcp trafic in
and out from server by adding two match-all ip filters into local security
policy.
I can't say I like this solution... Why did this problem appeared with
policy matching exact (sofia profiles) ports?
Regards, Robert.
Attached is graphical representation of SIP message flow. You can see that
for some reason FS doesn't resend to callee an ACK message recieved from
caller.
Regards, RobertT
attachment: TCP FS SIP msgs.PNG___
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Well since we aren't a proxy we wouldn't resend the one we receive...
what svn rev and are you using proxy media?
/b
On Nov 21, 2009, at 7:28 AM, RobertT wrote:
Attached is graphical representation of SIP message flow. You can
see that for some reason FS doesn't resend to callee an ACK
Yep, I use proxy media. First it started with 1.0.4 release, then I've
updated a week or two ago with the latest svn trunk, not sure what was the
rev number.
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Well, I start 2 user agents. Each of them successfully registers as 1000
1001 extensions via tcp SIP transport. Then I issue a call, say from 1000 to
1001, and watch it being disconnected in several seconds by recieving client
due to abovementioned conditions (no completing answer from FS). Why
Well depends are you using x-lite 4 beta? you didn't include ANY
logs... I know TCP to TCP works fine I use that daily.
can you include some debug logs or join #freeswitch on irc.freenode.net?
/b
On Nov 20, 2009, at 6:30 AM, RobertT wrote:
Well, I start 2 user agents. Each of them
No, I don't use Xlite. I use my own .Net wrapper around pjsip ua lib.
Foreseeing uncertaincies about it's quality I may say that pjsua reference
implementation yields the same results in this scenario.
Actually I have no doubt that FS is working nicely with tcp and tls as well
because I had it
I've tried to add ;transport=tcp in dialplan bridge application and it has
ended up with error on FS with message can't find registered extension *
called_extension*%external_call;transport=tcp whereas this extension is
registered in FS via tcp. Also I tried to reproduce the same scenario with
How exactly are you doing this?
/b
On Nov 18, 2009, at 7:07 AM, RobertT wrote:
I've tried to add ;transport=tcp in dialplan bridge application and
it has ended up with error on FS with message can't find registered
extension called_extension%external_call;transport=tcp whereas this
Hello everyone!
I'v got strange problem with incomplete call via tcp transport. When I
perform bridged call from one ua (no matter what transport udp or tcp)
through FS this call's leg b message sequence (over tcp) lacks finishing SIP
message what in it's turn cause the call to be disconnected by
tack on a ;transport=tcp
/b
On Nov 12, 2009, at 4:27 PM, RobertT wrote:
action application=bridge data=sofia/external_call/
$1%${domain_name}/
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but FS does use tcp for that call leg - RX 1167 bytes ... from *tcp* ...:
And after all there can be other SIP transports combinations FS should
interconnect...
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