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

Hi,

I have a problem with my IAX softphones. After a call, when the softphone
hangup, it remains unavailable for the other softphones. It can call anybody,
but can not be reached… For example, if A call B, B answer, then A or B
hangup, and C won’t be able to call A or B after that (but A or B would be able
to call C). The Dial function returns that the chan is unavailable. That is very
annoying, the only solution till now is to restart my softphones…. I must say
that sometimes it works fine and i do not encounter this problem. But it happens
very regularly, too often i would say.

I would like to know if some of you noticed that problem ? And if there is a
solution of course…
I tried with different sofphones : iaxlite, or other i do not remember the
name… I also have the sources of iaxComm (an open source softphone from
iaxclient project), do you think it could be a softphone problem, and if i can
do something to change the code….

To illustrate my problem, here is the full output of a failed call-back:
http://pastebin.com/FMeL3mYz

For this test, the extensions.conf was really minimum :
[default]
exten => 222,1,Dial(IAX2/bob.test.org)
exten => 222,2,Hangup()
exten => 111,1,Dial(IAX2/alice.test.org)
exten => 111,2,Hangup()

I use Asterisk 1.6.2.11 (i had the same problem with 1.6.2.6).

Thanks for you help.

Hello list,

I’m using asterisk 1.6.2.11 with realtime SIP (mysql DB).

I notice that when the SIP peer registers, the fields ‘fullcontact’,
‘ipaddr’, ‘port’, ‘regserver’, ‘regseconds’, ‘lastms’ are filled with
values.

But after a while, these fields become empty.

Asterisk CLI shows :

asterisk*CLI> sip show peers
Name/username Host Dyn Nat ACL Port
Status Realtime
sterk1/sterk1 192.168.1.102 D N 5062 OK (30 ms) Cached RT
test13/test13 192.168.1.100 D N 5060 OK (4 ms) Cached RT
test2/test2 192.168.1.102 D N 5061 OK (30 ms) Cached RT
test6/test6 192.168.1.104 D N 5063 OK (12 ms) Cached RT

So Asterisk is still aware of the realtime SIP peers, and they are still
reachable and working (IP-phones).

But why do the fields in the mysql DB become empty ?!

Is this a bug ?! (in asterisk 1.4.30 the fields remain filled until
unregister)

Kind regards,
Jonas.