[asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds

Andrew Martin amartin at xes-inc.com
Wed May 13 13:19:44 CDT 2015

----- Original Message -----
> From: "Steve Davies" <davies147 at gmail.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com>
> Sent: Wednesday, May 13, 2015 11:39:29 AM
> Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds
> Hi,
> In my experience, all Yealink phones work just fine with Asterisk, we have
> hundreds (perhaps even low-thousands) out there with customers on Asterisk
> 1.2, 1.6.2, 1.8 and 11.
> If you are accurately representing the SIP trace on the phone and the SIP
> trace on Asterisk, then I would strongly suggest a SIP ALG exists in the
> network between the two devices and that SIP ALG does not understand SIP
> properly. The two halves simply do not match, so something must surely be
> interfering.
> In my experience it is often an innocent looking Cisco router. Cisco's SIP
> implementation is "SIP By Cisco" rather than "RFC compliant SIP". If that is
> the case Cisco call it a "SIP fixup" and you just need to disable it.
> Hope that helps,
> Steve

That is an interesting point - the server and the phone are both connected to
Netgear switches where I have enabled their "Auto-VoIP" feature, which remarks
packets based on protocol (SIP, SCCP, etc) for better QoS:

I wonder if this remarking process is modifying another part of the packet too?
Both devices are on the same subnet, so although these switches do route 
traffic as well, that shouldn't be coming into play here.


More information about the asterisk-users mailing list