Not using the CDR for billing, but I do use it to see usage and to
know if it's cheaper to purchase a provider with unlimited incoming
and pay-per-minute outgoing. I disabled 'SIP Transformation' in the
SonicWall and so far so good (10/10 calls worked, more testing to be
had, stay tuned.)
On Sat,
Specs: Asterisk 1.4.22 running behind a SonicWall (transparent mode)
with a public IP address. We have our phone system setup as 172.16.2.x
that connect through the SonicWall to Asterisk. Incoming calls work
flawlessly and we no longer get one-way audio. We are only using SIP
(3 trunks now,
Ok, recompiling it now with a 1 instead of XMIT_CRITICAL. Will check
back to see if it worked. Would be nice if it did :)
Thanks,
Kurt
On Fri, Nov 7, 2008 at 3:38 PM, Doug [EMAIL PROTECTED] wrote:
At 14:15 11/7/2008, SIP wrote:
Kurt Knudsen wrote:
Specs: Asterisk 1.4.22 running behind
That seems to have sort of worked. It seems the phone decided to end
the call this time, instead of Asterisk and now the call is dangling
inside of 'sip show channels'.
So that solution didn't work :(
On Fri, Nov 7, 2008 at 4:28 PM, Kurt Knudsen [EMAIL PROTECTED] wrote:
Ok, recompiling it now
A previous issue has popped up and once again I'm out of ideas. During
the evenings it seems that the TDM channels will spike (dahdi_monitor)
and will refuse to listen for audio of any type, this includes DTMF.
The only resolution I know of is to stop Asterisk and restart the
dahdi service, but
Any updates? It still seems to happen, though not as often as it used to.
We're using Polycom 320 phones, if that makes a difference, though we did do
it with X-Lite as well.
On Sat, Oct 11, 2008 at 3:03 PM, Kurt Knudsen [EMAIL PROTECTED]wrote:
Thanks, Steve,
That's what I am unsure of. I
them together.
Or some other math logic to check the result.
On incoming Set(DIALSTATUS=CHANUNAVAIL) and it'll ring busy to bandwidth(or
out of service, you can tweak this).
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kurt Knudsen
Sent: Monday, October 20, 2008 10
Bandwidth.com?
Thanks.
On Mon, Oct 20, 2008 at 8:30 PM, [EMAIL PROTECTED] wrote:
-- Kurt Knudsen wrote :
Hello,
We have 2 SIP trunks from Bandwidth.com and if both are in use and someone
tries to dial out, they cause another call to get one-way audio (the caller
hears us, we cannot hear them
:[EMAIL PROTECTED] On Behalf Of Kurt Knudsen
Sent: Monday, October 20, 2008 1:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent
one-way audio
The GotoIf works, because it does failover sometimes, just
Now that I have a new card and my echo problems are 'mostly' solved, I
have another major issue to deal with. After about an hour or so the
card will stop detecting DTMF tones on incoming calls. dahdi_monitor
shows the following:
[EMAIL PROTECTED] wctdm24xxp]# dahdi_monitor 1 -v
Visual Audio
Here's some freaky stuff coming from Areski CDR tool:
101. 2008-10-13 03:41:23 DAHDI/1... 000 unknown 000 BackGround
silence/5 s
ANSWERED 00:20
102. 2008-10-13 03:11:30 DAHDI/1... 000 unknown 000 BackGround
silence/5 s
ANSWERED 00:21
103. 2008-10-13 02:41:23 DAHDI/1...
I use the 'generic' file in Postfix to map an email address that is not in
use to someone's text messaging address. It'd be [EMAIL PROTECTED]
ie: [EMAIL PROTECTED] Then, any email that gets sent to
[EMAIL PROTECTED], will get automatically sent to that person's phone.
On Mon, Oct 13, 2008 at 3:14
since you are setting up two
separate trunks with Bandwidth, you need to limit each trunk to one call,
rather than two.
Thanks,
Steve Totaro
On Fri, Oct 10, 2008 at 9:47 PM, Kurt Knudsen [EMAIL PROTECTED]wrote:
externip messes up DTMF detection, and by messes up I mean it doesn't
Hello,
We have 2 SIP trunks from Bandwidth.com and if both are in use and someone
tries to dial out, they cause another call to get one-way audio (the caller
hears us, we cannot hear them). This happens 100% of the time and
Bandwidth.com doesn't offer any support. I don't see any setting that
channels are
in use, it tries to connect to the 2nd trunk and thus kills the audio.
Nothing strange came up in Wireshark or the firewall logs.
Thanks.
On Fri, Oct 10, 2008 at 5:40 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Fri, Oct 10, 2008 at 5:17 PM, Kurt Knudsen [EMAIL PROTECTED]wrote
problems.
Thanks,
Steve Totaro
On Fri, Oct 10, 2008 at 6:32 PM, Kurt Knudsen [EMAIL PROTECTED]wrote:
Hi Steve,
It's behind a NAT/Firewall but SIP translation is enabled and removing it
from behind the firewall did nothing, it still dropped calls. The calls
connect and everything works
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