with the one from the source tar fixed it.
Thank you!
On Mon, Dec 5, 2022 at 3:43 PM Joshua C. Colp wrote:
> On Mon, Dec 5, 2022 at 4:31 PM Nick Olsen wrote:
>
>> Hello,
>>
>> On a fresh install of 18.9 Cert2 (Or the latest 19 if I recall the
>> previous version I tr
Hello,
On a fresh install of 18.9 Cert2 (Or the latest 19 if I recall the previous
version I tried.
PJSIP fails to load properly. It seems that the new res_geolocation module
fails to load. But I can't seem to figure out why. And being that it's a
fairly new module (So it seems) google-fo isn't
Thanks Joshua!
I reviewed the voicemail.conf for an option just like this and somehow
missed it. Thank you sir!
On Fri, Nov 18, 2022 at 9:20 AM Joshua C. Colp wrote:
> On Fri, Nov 18, 2022 at 9:53 AM Nick Olsen wrote:
>
>> Hello,
>>
>> I've got a handful of ser
Hello,
I've got a handful of servers running asterisk 16 currently with voicemail
stored in a database via ODBC.
Some users register to multiple servers and I'm having an issue with MWI.
Specifically Polycom phones seem to not be able to use two different MWI
sources. So in my immediate
> What is your queue/member config?
> What is in your /var/log/asterisk/queue_log log file?
>
> On 2/7/2022 4:23 PM, Nick Olsen wrote:
> > Hello, We're running asterisk 16 with Realtime.
> >
> > We have queues configured in realtime.
> >
> > The "Timeout&q
Hello, We're running asterisk 16 with Realtime.
We have queues configured in realtime.
The "Timeout" setting appears to have an upper 2 minute limit. Even when
setting the timeout in the queue to 600 seconds, the agent is no longer
rung after exactly 120 seconds. The asterisk CLI claims "Exiting
That did it! I had missed that option. Thanks for the assistance!
On Thu, Aug 26, 2021 at 9:50 AM Doug Lytle wrote:
> According to the wiki, you can disable the timestamp
>
> record_file_timestamp
>
> Append the start time to the record_file name so that it is unique.
>
>
>
Hello, I'm attempting to enable conference bridge recording.
I have it working, and I'm dynamically pushing the filename onto the bridge
via the set CONFBRIDGE commands. But it seems regardless of what name I
set, the actual filename is written as WHATIPROVIDED-uniqueid.wav.
Example, I use the
at 5:21 PM Joshua C. Colp wrote:
> On Wed, Mar 3, 2021 at 5:55 PM Nick Olsen wrote:
>
>>
>> SDP for the first 183
>> Session Description Protocol
>> Session Description Protocol Version (v): 0
>> Owner/Creator, Session Id (o): San
Hello!
I've got a number of asterisk systems running asterisk 16.12.0 currently.
They're configured with PJSIP.
Some of them are behind NAT, some aren't.
All systems have SIP trunks to a Sansay INX.
I've had one-way audio issues calling a particular number. After some
investigation, It seems that
Thanks Joshua, I assume by query asterisk you mean I'll need to query it
via AMI? Is that information available via AMI?
*Nick Olsen*
Network Engineer
Office: 321-408-5000 x103
Mobile: 321-794-0763
On Tue, May 26, 2020 at 2:57 PM Joshua C. Colp wrote:
> On Tue, May 26, 2020 at 10:48 AM N
sers/pjsip-and-rtt-in-realtime.html
[2], sourcery.conf
[res_pjsip]
**stuff***
contact_status=realtime,ps_contact_status (Which I've also defined in
extconfig)
*Nick Olsen*
Network Engineer
Office: 321-408-5000 x103
Mobile: 32
base fixes the issue, And
subsequent recordings for the same mailbox have no issue.
*Nick Olsen*
Network Engineer
Office: 321-408-5000 x103
Mobile: 321-794-0763
On Wed, Apr 1, 2020 at 9:04 PM Paddy Grice wrote:
> Hi All
>
> This sounds just like a problem I have had and still i
goes wrong. It would be nice if asterisk handled that more
gracefully.
I post this mostly just for internet history. To hopefully help the next
guy out who has this same issue.
*Nick Olsen*
Network Engineer
Office: 321-408-5000 x103
Mobile: 321-794-0763
On Mon, Mar 2, 2020 at 8:29 PM Joshua C. C
of the database on the same physical asterisk
instance and have the system reference it. Just to "throw everything at the
wall".
*Nick Olsen*
Network Engineer
Office: 321-408-5000 x103
Mobile: 321-794-0763
On Mon, Mar 2, 2020 at 1:58 PM Joshua C. Colp wrote:
> On Mon, Mar 2, 2020 at 2:52
. And the number just increases. Normally when all is fine.
They're all at 0.
Google-foo hasn't produced anything for me outside issues from 13.x that
claim to be resolved. Since asterisk isn't fully crashing, I don't think I
can get backtrace. Someone please correct me if I'm wrong.
Any ideas? Ti
Anyone have any decent ways to handle post dial delay on asterisk? Doesn't
seem like there's a timeout I can set for it. I'd love a "PROGRESSTIMEOUT="
field in PJSIP. Basically something to bring down a call attempt if the
delay between 100 and 18X is >30sec and route advance.
Not to divert the thread from the original question. But have you
considered switching the trunk to SIP? It's highly personal preference. But
I haven't used IAX2 between asterisk boxes in years simply because of
weird™ issues. SIP being more widely accepted protocol from all points of
reference
. I can also confirm that simple
memory_cache was not enough. full_backend_cache=yes was indeed required as you
suspected. Thanks!
Nick Olsen
Network Engineer
Office: 321-408-5000
Mobile: 321-794-0763
From: "Joshua C. Colp"
Sent: 5/16/
this when using realtime but NOT when
using flatfiles?
2. Is there not a way to cache this in Asterisk without having to do a DNS
lookup every time?
3. Any other tweaks I can make to asterisk to speed this up (Not really looking
to match based on other objects, like header or username).
Nick
it on
whatever the G100 is serving with said PRI.
If it's all one number anyways, You can just blanket overwrite it from the G100
dialplan (I think it was in outbound routes). Or ultimately, In the asterisk
instance during receive before shooting it upstream.
Nick Olsen
Network Engineer
Office
it on
whatever the G100 is serving with said PRI.
If it's all one number anyways, You can just blanket overwrite it from the G100
dialplan (I think it was in outbound routes). Or ultimately, In the asterisk
instance during receive before shooting it upstream.
Nick Olsen
Network Engineer
Office
: Yes
Callerid : "Nick" <321XXX>
Nick Olsen
Network Operations (855) FLSPEED x106
From: "Stefan Tichy" <asteri...@pi4tel.de>
Sent: Sunday, September 20, 2015 9:28 AM
To: asterisk-users@lists.digium.com
S
Yes, They are.
Nick Olsen
Network Operations (855) FLSPEED x106
From: "Michele Pinassi" <michele.pina...@unisi.it>
Sent: Thursday, September 17, 2015 3:07 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Realtime
king after a few minutes.
Any ideas? Thanks
Nick Olsen
Network Operations (855) FLSPEED x106
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory web
is as CHANUNAVAIL?
The obvious quick fix is to change my Busy option to attempt another
carrier before finally returning BUSY to the customer. But I was hoping to
not have to do that. Any ideas?
Nick Olsen
Network Operations (855) FLSPEED x106
=g722
allow=ulaw
callerid=FLHSI Nick 321-205-1100
nat=no
call-limit=100
limitonpeer=yes
Nick Olsen
Network Operations (855) FLSPEED x106
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
Not sure if it'll work for your specific use. But I always use app nocdr.
exten = 1,1,NoCDR
exten = 1,2,Dial(SIP/test,30)
Nick Olsen
Network Operations (855) FLSPEED x106
From: janusz_1942 janusz_1...@op.pl
Sent: Tuesday, September 16, 2014
now they just all bounce CHANUNAVAIL which is
expected.
Thanks!
Nick Olsen
Network Operations (855) FLSPEED x106
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
Catchall
1777outbound_1321NXX1Gotooutbound-cocoa,${EXTEN},1
Outbound 1321 Catchall
Thanks
Nick Olsen
Network Operations (855) FLSPEED x106
--
_
-- Bandwidth and Colocation Provided by http
. Letting _NXX555 handle the
rest.
Thanks to everyone that replied.
Nick Olsen
Network Operations (855) FLSPEED x106
From: Josh Metzger joshdmetz...@gmail.com
Sent: Monday, May 12, 2014 1:43 PM
To: n...@flhsi.com, Asterisk Users Mailing
HDLCFCS (In use)
Nick Olsen
Network Operations
(855) FLSPEED x106
From: Mitesh Thakkar mail.mthak...@gmail.com
Sent: Tuesday, January 07, 2014 2:55 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Lot of voice cut
I have added some
Make sure you have nat=yes in your sip.conf either under globals or
individual sip peer settings.
Nick Olsen
Network Operations
(855) FLSPEED x106
From: John Millican j...@millican.us
Sent: Thursday, January 02, 2014 10:50 AM
To: Asterisk Users
not expecting. Or, The NAT device doesn't have a mapping for and being
dropped at one of your routing devices.
Nick Olsen
Network Operations
(855) FLSPEED x106
From: John Millican j...@millican.us
Sent: Thursday, January 02, 2014 11:07 AM
To: asterisk-users
@10.65.6.10;tag=1470823868
To: sip:*8@10.65.6.10;tag=as65ceb9be
Call-ID: 695101044@172.16.10.101
CSeq: 1 ACK
Content-Length: 0
-
Nick Olsen
Network Operations
(855) FLSPEED x106
--
_
-- Bandwidth
. So I'm hoping
not to cause them a re-print.
Nick Olsen
Network Operations
(855) FLSPEED x106
From: Adrian Serafini wealwild...@wombit.com
Sent: Tuesday, December 31, 2013 12:51 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] *8
That did it.
For some reason, Even commented out. Pick up was still *8. And persisted
even after an asterisk service restart. Changed the feature to *7, Rebooted
the whole PBX and it finally took effect.
Nick Olsen
Network Operations
(855) FLSPEED x106
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