I will try and open a ticket with them again. I got them to work with me
and make a decent amount of progress, so we will see if I have any luck.
On 8/1/2011 2:08 PM, Tony Graziano wrote:
(not trying to hijack matthew's thread, but we answer both of the last
2 posts on this thread with this)
the constant being the spectralink
look at frame 6 in the merged file matthew sent in. what is missing is
that there is no ACK back. noone knows there is any media "really
established" because the UA (spectralink) wont ack it.After seeing
session progress, there are no ack's FROM the UA at 10.81.3.5) back
to sipxbridge or the proxy at 172.30.216.62.
OK, I keep saying OK to myself long enough and you dont say anything
back I'm guessing you ain't there. It's a bad idea to leave this
nailed up if you are there because if you were you's ACK me back. BYE
(frame 18).
it's a bug that needs to be fixed by polycom in the spectralink (IMO).
Getting them to acknowledge and do anything about it is another thing
altogether.
On Mon, Aug 1, 2011 at 2:56 PM, McIlvin, Don
<[email protected] <mailto:[email protected]>>
wrote:
Matthew,
Anything new!
I am getting the external call drop at 15 seconds too, multiple
scenarios. For us it is after dialing inbound.
This is on a test environment.
We tried a desktop Bria3 (just updated) calling extension to
extension with a remotely registered 3G iPhone using the Bria
iPhone client. Call dropped at 15 seconds.
The goal of the test was to take Polycom and our carrier out of
the equation. So we still get the exactly 15 sec drop with no
Polycom phone or ITSP carrier in the mix.
In contrast the same two end points, with the iPhone (using a Bria
SIP client) registered over the local WiFi in our building. Worked
like a charm, calls connect, two way audio, did not drop on call
exceeding 2 minutes.
We have a new SBC being tested (Audiocodes MSBG 6.2), but your
configuration seems to be using just SipX bridge.
We also have a one way and sometimes zero way audio problem we are
chasing as well. On this, when we have one way audio it is the UDP
RTP stream coming in from the external device that does not get
through. We have identified that the BGP routers are not getting
the udp/rtp stream sent to them, and the FW sees no packets coming
in (nor dropped).
Don
*From:*[email protected]
<mailto:[email protected]>
[mailto:[email protected]]
<mailto:[mailto:[email protected]]> *On
Behalf Of *Matthew Kitchin (public/usenet)
*Sent:* Monday, August 01, 2011 11:37 AM
*To:* sipx-users
*Subject:* [sipx-users] Fwd: Re: Polycom Spectralink 8020 - drops
at 15 seconds
Polite bump request.
Does anyone see anything obvious in the sip trace?
I'm 99.9% positive this has not been occurring all the time. If it
has, my testing is absolutely awful.
Outbound calls drop at 15 seconds.
This is the latest version of Sipx 4.4 with all updates. The
server in this example is virtual, but it is happening on all my
physical boxes as well.
8020 = 10.81.3.253
Sipx = 10.81.3.5
Verizon VoIP = 172.30.216.x
There is no IP address NAT, but there is an inbound 5060=>5080 NAT.
-------- Original Message --------
*Subject: *
Re: [sipx-users] Polycom Spectralink 8020 - drops at 15 seconds
*Date: *
Thu, 28 Jul 2011 14:46:17 -0500
*From: *
Matthew Kitchin (public/usenet) <[email protected]>
<mailto:[email protected]>
*To: *
[email protected] <mailto:[email protected]>
1 way audio was a false alarm. The phone was fine after a rebbot.
The dropped call at 15 second issue still exists on external
calls. I am using sipxbride. The sip trace is attached. The test
call is the only call in the logfiles. Any ideas?
On 7/28/2011 2:41 PM, Tony Graziano wrote:
"verizon is not seeing an ack on the call and thinks rtp is
not actually established."
is now "sipxecs" is not verizon is not seeing an ack on the
call and thinks rtp is not actually established.
On Thu, Jul 28, 2011 at 3:33 PM, Matthew Kitchin (public/usenet)
<[email protected] <mailto:[email protected]>> wrote:
I just realized they are getting 1 way audio even on internal
calls too. Something has to be wrong on our end. I will keep
investigating.
On 7/28/2011 2:21 PM, Tony Graziano wrote:
verizon is not seeing an ack on the call and thinks rtp is not
actually established.
so then either the proxy is not sending the ack or the phone is not.
got a siptrace?
On Thu, Jul 28, 2011 at 3:15 PM, Matthew Kitchin (public/usenet)
<[email protected] <mailto:[email protected]>> wrote:
Can anyone help me interpret the attached wireshark?
I'm 99.9% positive this has not been occurring all the time. If it
has, my testing is absolutely awful.
Outbound calls drop at 15 seconds.
This is the latest version of Sipx 4.4 with all updates. The
server in this example is virtual, but it is happening on all my
physical boxes as well.
8020 = 10.81.3.253
Sipx = 10.81.3.5
Verizon VoIP = 172.30.216.x
There is no IP address NAT, but there is an inbound 5060=>5080 NAT.
I'm not an expert at reading these, but I believe the BYE is
coming from Verizon and I do not know why. I think I am going to
have to open a ticket with them, but I wanted to check here first
and see if I am missing something.
Thanks,
Matthew
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sip: [email protected]
<mailto:[email protected]>
Fax: 434.326.5325 <tel:434.326.5325>
Email: [email protected]
<mailto:[email protected]>
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426 <tel:434.984.8426>
sip: [email protected]
<mailto:[email protected]>
Helpdesk Contract Customers:
http://support.myitdepartment.net
Blog:
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sip: [email protected]
<mailto:[email protected]>
Fax: 434.326.5325
Email: [email protected] <mailto:[email protected]>
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
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Helpdesk Contract Customers:
http://support.myitdepartment.net
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