Hi Everyone,

I have a provider whose DID used to come into the box just fine but recently
stopped working. Nothing has been changed on our end.

Here is what I get when doing “sip set debug peer PROVIDER”:

Sending to 123.123.123.123 : 5060 (no NAT)

^^^^ That is ALL I am getting with sip debug turned on.

With Allow Anonymous SIP set to YES, then the call comes in properly and you
see the ACK, REQUEST and ACCEPT of sip debug just fine.

This is Elastix with Asterisk 1.4.33.1

Any thoughts?

Thanks

The past few days I started having a problem with a small call center
setup. All agents use Eyebeam 1.5 to receive calls from a queue. Eyebeam is
configured to auto answer the call. The problem is that the agents claim that
they get a call but no audio. From the logs I can see that it is calling the
agent phone but after 10 seconds (the queue timeout for pickup) I get the
message that nobody answered and the call is sent to the next available agent.
This can happen with up to three agents (the third finally answers the call).
This has happened at least 20 times in the past two days. At first the
supervisor thought that the same call was ringing on three different agents at
once but the logs say that the first two do not answer and the third does.

Here is an extract from the log file: http://pastebin.com/sB9JxJFm

We are using Asterisk 1.4.35 (upgraded from 1.4.32 just in case) on a
CentOS 5.5 x64 server with DAHDI 2.3.0.1 and a TE220B card. Could this be a
problem with chan_agent, the SIP phones or the queue? Any ideas where to
begin debugging?

Does anyone have a suggestion on how to handle this? For example, if I
have a list of numbers that I want to go out a certain sip channel and
another that I want to go out the dahdi device, is there a way to do
this? None of the numbers will fit into a pattern, so just plain
pattern matching won’t do.

The most straightforward way would be to just define explicit patterns.
Obviously that works, but doesn’t seem scalable in terms of maintenance.
Ideally there should be a variable or list of numbers, and the dialplan
logic jumps into a subroutine that checks if the dialed number is on the
list, then routes accordingly. Does anyone have any suggestions as to
how to approach that, or if they have a entirely different way in mind?

hose

Hi Everyone,

I see one long post on Cisco community forum where everyone including ISPs
are complaining about silence on FXS port, reboots, frozen state, etc….of
WRP400. This is the a wireless router + 2 FXS combo box. I am looking to use
this for home user to connect to hosted Asterisk PBX.

I am looking for some feedback from the community as to how stable the unit
is – or if it is stable at all?

Thanks for your input.

Hi Everyone,

Wondering if any of you folks ever had trouble using *Thomson
TG784
*DSL/Wireless Router/FXS ATA combo? I am looking to use this to connect
users from home to a hosted Asterisk PBX.

Any and all inputs are appreciated.

Thanks