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Apologies around for posting HTML. My bad.
--Paul
Paul Dugas
Computer Engineer
Dugas Enterprises,
LLC
522 Black Canyon
Park
Canton, GA 30114
phone:
404.932.1355
fax:
866.751.6494
paul@dugas.cc
http://DugasEnterprises.com
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Check the handset cords. They can get loose and cause this exact issue.
Bill
________________________________
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Paul Dugas
Sent: Thursday, September 28, 2006 10:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Polycom 501 One-way Audio
I have a site running an up-to-date version of Asterisk from the 1.2
trunk. We have a dozen Polycom 501 units and one of them (none of the
others) is having recurring one-way-audio problems. As Murphy's Law
dictates, it's the bosses phone!
The user gets a few calls a day where the caller can hear her fine but
she hears dead silence. It happens when she calls out sometimes too.
Even internal voicemail and extension-to-extension calls are affected.
I just called her three times from another extension; the first two were
affected, the third got through. None of the other units seem to have
the problem. They're all running the same firmware and are loading
central configs that are identical except for line-button text and
registration info.
I've been running * with lots of debug/verbose logging enabled and have
yet to see it complain about anything when she reports the problem. I'm
about to replace her phone with a spare to see if that fixes it.
Wondering if anyone has seen something like this and might be able to
tell me what to look for as a potential cause.
TIA,
Paul
Paul Dugas
Computer Engineer
Dugas Enterprises, LLC
522 Black Canyon Park
Canton, GA 30114
phone:
404.932.1355
fax:
866.751.6494
paul@dugas.cc
http://DugasEnterprises.com <http://dugasenterprises.com/>
This e-mail and any attachments are confidential. If you receive this
message in error or are not the intended recipient, you should not
retain, distribute, disclose or use any of this information and you
should destroy the e-mail and any attachments or copies.
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It looks like a NAT issue on RTP. How many ports are u running on rtp.conf? Are you using static or dynamic nat translation for rtp? Redouane ________________________________ De : Paul Dugas [mailto:paul@dugas.cc] Envoy? : jeudi 28 septembre 2006 16:32 ? : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [asterisk-users] Polycom 501 One-way Audio I have a site running an up-to-date version of Asterisk from the 1.2 trunk. We have a dozen Polycom 501 units and one of them (none of the others) is having recurring one-way-audio problems. As Murphy's Law dictates, it's the bosses phone! The user gets a few calls a day where the caller can hear her fine but she hears dead silence. It happens when she calls out sometimes too. Even internal voicemail and extension-to-extension calls are affected. I just called her three times from another extension; the first two were affected, the third got through. None of the other units seem to have the problem. They're all running the same firmware and are loading central configs that are identical except for line-button text and registration info. I've been running * with lots of debug/verbose logging enabled and have yet to see it complain about anything when she reports the problem. I'm about to replace her phone with a spare to see if that fixes it. Wondering if anyone has seen something like this and might be able to tell me what to look for as a potential cause. TIA, Paul Paul Dugas Computer Engineer Dugas Enterprises, LLC 522 Black Canyon Park Canton, GA 30114 phone: 404.932.1355 fax: 866.751.6494 paul@dugas.cc http://DugasEnterprises.com <http://dugasenterprises.com/> This e-mail and any attachments are confidential. If you receive this message in error or are not the intended recipient, you should not retain, distribute, disclose or use any of this information and you should destroy the e-mail and any attachments or copies. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060928/f81577ca/attachment.htm
On Thu, Sep 28, 2006 at 10:32:12AM -0400, Paul Dugas wrote:> I have a site running an up-to-date version of Asterisk from the > 1.2 trunk. We have a dozen Polycom 501 units and one of them (none > of the others) is having recurring one-way-audio problems. As > Murphy's Law dictates, it's the bosses phone!Does swapping the physical locations of the phones move the problem? Swapping IP addresses in the original locations? Did you mention whether they've all got identical FW versions? Cheers, -- jra -- Jay R. Ashworth jra@baylink.com Designer Baylink RFC 2100 Ashworth & Associates The Things I Think '87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 "That's women for you; you divorce them, and 10 years later, they stop having sex with you." -- Jennifer Crusie; _Fast_Women_
On Thu, 2006-09-28 at 16:34 +0100, Redouane Doumer wrote:> Are you using static or dynamic nat translation for rtp?No NAT. Everyone on the same switch and subnet. -- Paul Dugas, Computer Engineer Dugas Enterprises, LLC paul@dugas.cc phone: 404-932-1355 522 Black Canyon Park http://dugas.cc fax: 866-751-6494 Canton, GA 30114 USA -- This e-mail and any attachments are confidential. If you receive this message in error or are not the intended recipient, you should not retain, distribute, disclose or use any of this information and you should destroy the e-mail and any attachments or copies. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060928/84ff515b/attachment.pgp