On 7/02/2014 3:38 AM, Tech Support wrote:
All;
I’m running Asterisk 1.8.15-cert3 with the newest version of spandsp.
I’ve even tried unloading that and using Digium’s FFA module but I
receive the same error on an outbound transmission:
[2014-02-06 14:35:14] ERROR[19066]: udptl.c:294 encode_open
You could have the call immediately return to the transferer
-Original Message-
From: John Kiniston
Sender: asterisk-users-bounces@lists.digium.comDate: Thu, 6 Feb 2014 17:14:02
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Reply-To: Asterisk Users Mailing List - Non-Comme
I'm trying to address a problem with users transferring to invalid
destinations.
In my sip peer I'm setting both __FORWARD_CONTEXT and __TRANSFER_CONTEXT to
a context with a extension defined below to set some CDR variables before
the call is transferred.
[customer-forward]
exten => _X.,1,Progres
Even if they don't support another modem, you can usually use another
router. Just put the Ubee in bridge mode and slap a router behind it.
This is our typical best practice install.
You just can't trust the piece of shit routers that ISP's send out these days..
dw
On Thu, Feb 6, 2014 at 12:54 P
On 2/6/14, 11:18 AM, Mike Diehl wrote:
Hi all,
I have an SPA112 that in sitting behind a Ubee cable modem. The
internet link is solid, but the device becomes unreachable within a
day or so of being rebooted. Then the customer goes to reboot the
device, they report that all 4 lights are lit.
All;
I'm running Asterisk 1.8.15-cert3 with the newest version of spandsp.
I've even tried unloading that and using Digium's FFA module but I receive
the same error on an outbound transmission:
[2014-02-06 14:35:14] ERROR[19066]: udptl.c:294 encode_open_type: UDPTL
(SIP/XXX_outboun
Unfortunately, we plug straight into the Ubee and the ISP will not support
any other modem.
GRRr..
Mike.
On Thu, Feb 6, 2014 at 12:34 PM, David Wessell wrote:
> Is there another router in the mix? Put the cable modem in bridge mode and
> attAch a real router.
>
> http://401stblow.
On 02/06/2014 09:25 AM, Mike Diehl wrote:
I've got the registration period set to 15 minutes. However, I've got
similar devices all over the place that don't seem to have this
unreliability issue. The way I solved it with the SPA303 that I had in the
office was to replace the Ubee modem with a
Is there another router in the mix? Put the cable modem in bridge mode and
attAch a real router.
http://401stblow.wordpress.com/2012/10/21/fixing-time-warner-cable-ubee-modem-connectivity-issues/
On Thursday, February 6, 2014, Mike Diehl wrote:
> I've got the registration period set to 15 minut
I've got the registration period set to 15 minutes. However, I've got
similar devices all over the place that don't seem to have this
unreliability issue. The way I solved it with the SPA303 that I had in the
office was to replace the Ubee modem with a different make/model. That's
not an option
How long is the registration timeout? If the device is behind a
router/firewall, then you need to set a registration timeout lower than the
state table "life" in the router/firewall. I usually set my devices to just
2 minutes and it works almost all the time. Most Cisco devices have a very
long tim
Hi all,
I have an SPA112 that in sitting behind a Ubee cable modem. The internet
link is solid, but the device becomes unreachable within a day or so of
being rebooted. Then the customer goes to reboot the device, they report
that all 4 lights are lit. The ISP reports that the device does respo
I have an asterisk 11.4.0 server with two interfaces, eth0 and eth1.
Eth0 has a default gateway on it, eth1 is connected the subnet that has my
phones registered.
I'd like to use the multicastRTP driver to do paging. However, when a
phone dials an extension with multicastRTP, the multicast
hi
when i try to this with page()
exten => 506,1,SIPAddHeader("Call-Info:__\; answer-after=0")
exten => 506,n,page(SIP/105)
CLI>Accepting call from '0661xx' to '506' on channel 1/13, span 1
-- Executing [506@default:1] SIPAddHeader("DAHDI/13-1", ""Call-Info:__;
answer-after=0"") in new s
We run a multi node, multi tenanted hosted VoIP service using centralised
databases for sip/extensions/voicemail configuration allowing resellers and
end users to make updates to their walled garden themselves. We're using
asterisk 1.8 but Realtime is no different on asterisk 12 (with the
exception
On 14-02-06 11:10 AM, James Wystead wrote:
> Hi - I figured this was probably the best place to ask this question
>
> Is there anyone that has done anything practical with the API and/or
> Real Time Database config?
We actually have a new mailing list[1] that is focused on discussion
about the ne
Hi - I figured this was probably the best place to ask this question
Is there anyone that has done anything practical with the API and/or Real
Time Database config?
If so, I would like to pick your brains if I may.
Thanks - G
--
__
Il 05/02/2014 8.42, Olivier ha scritto:
channel then it depends upon what you have the priindication option
set to. With
priindication=outofband then a busy cause code is sent to the
network and the call
is hung up. With priindication=inband then a busy tone is sent
afte
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