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]>
To:     [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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-- ======================
    Tony Graziano, Manager
    Telephone: 434.984.8430 <tel:434.984.8430>
    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:
    http://blog.myitdepartment.net

    Linked-In Profile:
    http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4

    Ask about our voip fax services!



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--
======================
Tony Graziano, Manager
Telephone: 434.984.8430
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
sip: [email protected] <mailto:[email protected]>

Helpdesk Contract Customers:
http://support.myitdepartment.net

Blog:
http://blog.myitdepartment.net

Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4

Ask about our voip fax services!



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