On Tue, Jul 6, 2021 at 2:14 AM Jean Aunis wrote:
> Le 30/06/2021 à 16:10, Ryan Press a écrit :
>
> [...]
> [from-internal-custom] ; Doorbell video bridge
> exten => doorbell_rtsp,1,Answer() same => n,RTSP-SIP(rtsp://
> admin:12345@192.168.24.53:554/live/sub,0,asterisk,506
7;m not sure this is something I can easily
configure without writing a bunch of new code.
Thanks,
Ryan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: htt
I've found the SPA525G2's much easier to deal with than the 7960's. Probably
more money, but worth it in my opinion.
From: asterisk-users on behalf of
Turritopsis Dohrnii Teo En Ming
Sent: Friday, December 18, 2020 10:36 AM
To: jnov...@stromberg-carlson.org
Cc
I can't speak to the other items, but it's always better to have a dedicated
FXS to answer the modem calls on an analog line. I've had to do this for ATT
network router for their own management and it's always been fine.
From: asterisk-users On Behalf Of
John T. Bittner
Sent: Tuesday, February
"core show channels" and
can't be hung up.
Any ideas? This just started a month or so ago when we started forwarding all
calls as stated below. /shrug
From: asterisk-users On Behalf Of
Ryan, Travis
Sent: Friday, July 26, 2019 11:01 AM
To: Asterisk Users Mailing List - Non-C
Ok, so this might seem weird, but hang with me on this. I have two sites, Indy
and Lafayette that each have their own Asterisk server. They each have their
own outside PRI line. They are also trunked internally via and IAX tunnel over
a private fiber line.
I've recently been asked to have the c
You need more than an ATA. You need something with an FSO and FXO. I've used
Linksys/SPA3102-3.3.6 and been happy with it.
From: asterisk-users On Behalf Of
Sebastian Nielsen
Sent: Thursday, March 21, 2019 3:01 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [a
Can't you just reference everything in IPs? If not, then hardcode the IPs in
your /etc/hosts file. I think that's a bad idea, but that's one way to ensure
you always have the Ip of a domain name.
From: asterisk-users On Behalf Of
John T. Bittner
Sent: Wednesday, February 20, 2019 11:30 AM
To:
Yes it's very easy. Mine is using a simulated PRI over an ATT Flex line. I just
followed the many tutorials out there. I answer the call, then it takes 6-7
seconds (you can add a wait if you want) and then it detects it and drops it to
the fax extension in the same context.
Also, until recentl
Any weirdness with realtime has almost always gone back to schema issue for me.
Just my experience…
On 9/15/17, 10:48 AM, "asterisk-users-boun...@lists.digium.com on behalf of
Joshua Colp" wrote:
On Fri, Sep 15, 2017, at 11:38 AM, Bryant Zimmerman wrote:
> Joshua
>
> We ha
-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan, Travis
Sent: Wednesday, July 19, 2017 3:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] corosync and Asterisk 13
I have an 14.04 server with Asterisk
13.13.
I REALLY need some help figuring this out. 😊
Thanks!
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan, Travis
Sent: Wednesday, July 19, 2017 1:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
Anyone else using corosync with Asterisk 13 and Ubuntu 16.04 or higher?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan, Travis
Sent: Wednesday, July 19, 2017 10:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
19 July 2017 at 14:46, Ryan, Travis
mailto:ry...@oscarwinski.com>> wrote:
I want to use corosync and installed it via ubuntu repository. I guess there is
a version 1 and 2 of corosync. For some reason ./configure for Asterisk (13)
isn’t recognizing I have corosync installed. I can’t enabl
I want to use corosync and installed it via ubuntu repository. I guess there is
a version 1 and 2 of corosync. For some reason ./configure for Asterisk (13)
isn't recognizing I have corosync installed. I can't enable the res_corosync
module in menuselect.
Any ideas?
Thanks!
Travis
--
___
bject: Re: [asterisk-users] BLF sharing between Asterisk 11 and 13
On Sun, Jul 16, 2017, at 02:38 PM, Ryan, Travis wrote:
> So any phone that wants just state information needs to have an
> account on all the servers it needs that information from? Guess I can
> do that, but seems
with pjsip is a little bit different.
Did you read the
https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip+for+Presence+Subscriptions?
On 16 Jul 2017 3:38 am, "Ryan, Travis"
mailto:ry...@oscarwinski.com>> wrote:
I have servers setup in versions 11 and 13. Between t
...@lists.digium.com] On Behalf Of Joshua Colp
Sent: Sunday, July 16, 2017 5:34 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] BLF sharing between Asterisk 11 and 13
On Sat, Jul 15, 2017, at 11:37 PM, Ryan, Travis wrote:
> I have servers setup in versions 11 and 13. Betw
res_pjsip/pjsip_distributor.c:347
log_unidentified_request: Request from '"Travis Ryan" '
failed for '10.1.2.XXX:5060' (callid: 8c79c540-c0710...@10.1.2.xxx) - No
matching endpoint found
How do I make a server allow another extensi
Ok, so a few years ago, when 13 first came out, I was having a core dump
(crash) issue with asterisk 13. I worked with Josh some and even used my Digium
subscription for support. Never was able to get it fixed at that time so let it
go. Well now I am trying on the same server, after a completely
is likely the cause of the delay
(looking for caller ID).
All the best,
David
On 27 April 2017 at 12:48, Ryan, Travis
mailto:ry...@oscarwinski.com>> wrote:
Hey all,
I have a setup with two analog lines coming into and Asterisk 13 box with a
TDM400P and it takes a lot of rings before as
Hey all,
I have a setup with two analog lines coming into and Asterisk 13 box with a
TDM400P and it takes a lot of rings before asterisk takes over. I've traced
this same box on two different analog providers so it probably isn't a problem
with them.
I DO have callerid enabled and not sure I c
My go to phones are Polycom VVX series or
X-Lite / Bria softphones. The key is to make sure you have configured
Asterisk sip.conf with the externip= and nat=yes settings. Additionally on
the NAT routers that the outside phones are behind SIP ALG should be
disabled.
Ryan
--
__
So if someone has their own hardware and infrastructure but wants a software
(not FreePBX but perhaps similar) what options do we have? Would like to
virtualize it and not stuck with any one virtualization technology.
Discuss... :)
Travis
--
What is the best virtual server tech (and most stable, etc) to use for a
asterisk virtual hosting environment?
I have a client that wants to do virtual hosting of Asterisk (only SIP or IAX,
no PRI, etc) and I'm wondering if Xen or something else would be best? We'd
like to stay away from the co
Is there any way to have a meeting request in Outlook allow someone to
attach/setup a conference bridge, time, etc for Asterisk?
Travis Ryan
Director of Information Technologies
Oscar Winski Company
2407 North Ninth Street
Lafayette, IN 47905
ry...@oscarwinski.com<mailto:ry...@oscarwinski.
Travis Ryan
Director of Information Technologies
Oscar Winski Company
2407 North Ninth Street
Lafayette, IN 47905
ry...@oscarwinski.com
(765) 742-1102
We're not the IT departmentWe're the I-TEAM department!
> -Original Message-
> From: asterisk-users-boun...@
ne (not service script) here is what happens.
http://pastebin.com/3GFe6fG9
Travis Ryan
Director of Information Technologies
Oscar Winski Company
2407 North Ninth Street
Lafayette, IN 47905
ry...@oscarwinski.com<mailto:ry...@oscarwinski.com>
(765) 742-1102
We're not the IT departmentW
change Enterprise setup in which case I would suggest exploring unified
messaging
Thanks,
Neeraj
On Thu, Mar 3, 2016 at 8:22 AM, Ryan, Travis
mailto:ry...@oscarwinski.com>> wrote:
I am wondering what the best solution is for initiating a call from Outlook
Contacts. I imagine something that wou
like this?
Travis Ryan
Director of Information Technologies
Oscar Winski Company
2407 North Ninth Street
Lafayette, IN 47905
ry...@oscarwinski.com<mailto:ry...@oscarwinski.com>
(765) 742-1102
We're not the IT departmentWe
Getting the some errors making dahdi 2.11.0.
Seems same as listed here http://forums.asterisk.org/viewtopic.php?f=1&t=96455
In that link they say to use 2.10.2 but that's from December. Is there a fix
yet for this?
Travis Ryan
Director of Information Technologies
Oscar Winski Com
That would be cool.
Travis Ryan
Director of Information Technologies
Oscar Winski Company
2407 North Ninth Street
Lafayette, IN 47905
ry...@oscarwinski.com
(765) 742-1102
We're not the IT departmentWe're the I-TEAM department!
> -Original Message-
> From: ast
//alembic.readthedocs.org/en/latest/offline.html
[Ryan, Travis] I’m also very interested. I have tables that are already named
the same as alembic uses, so it causes me issues on upgrades.
--
_
-- Bandwidth and Colocation Provided
e: [asterisk-users] Detected alarm on channel 3: Red Alarm
>
> Humm, if I put a filter in this lines, maybe back?
>
>
>
> 2016-01-05 12:36 GMT-02:00, Ryan, Travis :
> >
> >> -Original Message-
> >> From: asterisk-users-boun...@lists.digium.com [mailto
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Vitor Mazuco
> Sent: Tuesday, January 05, 2016 9:21 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Detected al
http://www.wunderground.com/weather/api/
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
d...@donkelly.biz
Sent: Wednesday, December 16, 2015 9:20 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion
So I am using PJSIP realtime with Asterisk 13. I set the qualify_frequency
column AORS and it now shows the RTT in milliseconds in the console. I want to
be able to display that in a webpage, and was hoping the RTT would be updated
in one of the realtime tables, but I don't see it. The old chan
Sorry, figured out i had to add ulaw to my tables for my realtime PJSIP setup
on the device trying to use it.
Thanks,
Travis
From:
mailto:asterisk-users-boun...@lists.digium.com>>
on behalf of Travis Ryan mailto:ry...@oscarwinski.com>>
Reply-To: Asterisk Users Mailing List - No
I am trying to get my Linksys/Cisco SPA3102 to connect to asterisk 13 PJSIP. It
is registered just fine but when I dial one of my known extensions on the
server. As far as I can tell it should be able to translate as also pasted
below.
Can anyone help me?
res_pjsip_sdp_rtp.c:324 set_caps:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dmitriy Serov
Sent: Tuesday, October 06, 2015 10:06 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] does res_pjsip support ZRTP?
06.10.2015
not pulling from ODBC
On 15-10-05 09:16 AM, Ryan, Travis wrote:
[snip]
>
>
> So should anyone using realtime PJSIP be using the registrations line? Even
> if it's not used for any trunking?
A registrations line in sorcery.conf for res_pjsip would do absolutely nothing.
If
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp
Sent: Sunday, October 04, 2015 12:44 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] pjsip realtime registrations not pulling fr
Of Ryan, Travis
Sent: Friday, September 25, 2015 5:28 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Losing my mind on MWI
Can't get MWI working with PJSIP and my Cisco phones and realtime. I have
"mailboxes" populated in the endpoints and aors tables, with 312@de
Can't get MWI working with PJSIP and my Cisco phones and realtime. I have
"mailboxes" populated in the endpoints and aors tables, with 312@default which
is the voicemail context. I'm not sure what else to try.
Please help! :)
Travis
--
__
I've not used analog for quite some time. It seems it's not possible in
asterisk to spoof a phone number/name on an analog call?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
Does something change with MWI when moving from SIP to PJSIP? Seems my phone
isn't be alerted of its new VM.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introduct
I have a TDM400P analog card in my asterisk server. I haven't used analog for a
while. The caller hears at least two rings before my 312 extension gets rang
internally. Does it usually take that long? Below is my relevant dialplan. Also
callerID isn't working but that might just be the test anal
Yes, the schema can change between versions. Following the instructions on
https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime#SettingupPJSIPRealtime-InstallingandUsingAlembic
will cause alembic to upgrade the tables.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Ja
Did something change DB-wise with PJSIP and realtime between 13.3.2 and 13.5.0?
I'm getting an unknown column error and unsure where I need that column and the
type it needs to be.
Thanks!
[Sep 24 15:32:41] -- Attempted to remove non-existent contact
'sip:312@10.1.1.201:5060' from AOR '31
something like that and it must have did it¹s own install of
pjsip that sent me down a duplicate path, etc.
Thanks Josh and all!
On 9/24/15, 2:10 PM, "asterisk-users-boun...@lists.digium.com on behalf of
Joshua Colp" wrote:
>Ryan, Travis wrote:
>>>
>> I think i¹m dow
On 9/24/15, 1:30 PM, "asterisk-users-boun...@lists.digium.com on behalf of
Joshua Colp" wrote:
>Ryan, Travis wrote:
>>>
>> That folder doesn¹t have any libpj files in it. How do I make it find
>>the
>> real ones?
>
>You can delete libpjproject.pc
On 9/24/15, 8:10 AM, "asterisk-users-boun...@lists.digium.com on behalf of
Joshua Colp" wrote:
>On 15-09-24 08:54 AM, Ryan, Travis wrote:
>>
>> travis@pcimphone1:~/downloads/asterisk-13.5.0$ pkg-config --libs
>> libpjproject -L/usr/local/lib -lpjsua2-x86
14.04
On 15-09-23 10:25 PM, Ryan, Travis wrote:
> Ok I did all that and it's still crashing. I did find some other areas
> I think that shouldn't have had any of those files, so I thought it
> would work, because I got rid of ALL of them per your instructions and
> compl
:23 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] problems with PJSIP install on UBUNTU 14.04
On 15-09-23 10:25 PM, Ryan, Travis wrote:
> Ok I did all that and it's still crashing. I did find some other areas
> I think that shouldn't have had any of those fil
with PJSIP install on UBUNTU 14.04
On 15-09-23 08:00 PM, Ryan, Travis wrote:
> I ran all the uninstall commands and the rm commands. Made sure that ldconfig
> -p had no pj stuff in it.
>
> That's not enough? What did I miss?
"make uninstall" uses the configure parameters. Y
with PJSIP install on UBUNTU
> 14.04
>
> On 15-09-23 07:53 PM, Ryan, Travis wrote:
> >
> > I'm not sure what that means. I just built it how the wiki says too,
> > and earlier messages in this thread. J
>
> It means not all instances of PJSIP were removed fr
14.04
On Wed, Sep 23, 2015 at 5:43 PM, Ryan, Travis
mailto:ry...@oscarwinski.com>> wrote:
> -Original Message-
> From:
> asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com>
> [mailto:asterisk-users-<mailto:asterisk-users-&
with PJSIP install on UBUNTU
> 14.04
>
> On 15-09-23 07:36 PM, Ryan, Travis wrote:
> > I've got the backtrace, but how much of the info do you want?
>
> Ideally everything.
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan
roblems with PJSIP install on UBUNTU
> 14.04
>
> On 15-09-23 07:17 PM, Ryan, Travis wrote:
> > Getting constant segfaults now...
> >
> > [ 157.894809] asterisk[1424]: segfault at c ip 7f8b2fbcfd04 sp
> > 7f8b91722010 error 4 in res_hep_pjsip.so[7f8b2fba2000+4500
m
> Subject: Re: [asterisk-users] problems with PJSIP install on UBUNTU
> 14.04
>
> On 15-09-23 07:17 PM, Ryan, Travis wrote:
> > Getting constant segfaults now...
> >
> > [ 157.894809] asterisk[1424]: segfault at c ip 7f8b2fbcfd04 sp
> > 7f8b9172
s-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Ryan, Travis
> Sent: Wednesday, September 23, 2015 11:46 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] problems with PJSIP install on UBUNTU
> 14.04
>
>
&
roblems with PJSIP install on UBUNTU
> 14.04
>
> On 15-09-23 12:14 PM, Ryan, Travis wrote:
> > Spoke too soon. Same thing.
> >
> > Josh, any other ideas?
>
> Not really, that's the exact configure line I use. You may have to do a
> "make distclean&quo
Spoke too soon. Same thing.
Josh, any other ideas?
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Ryan, Travis
> Sent: Wednesday, September 23, 2015 10:50 AM
> To: Asterisk Users Mai
, 2015 10:12 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] problems with PJSIP install on UBUNTU
> 14.04
>
> Ryan, Travis wrote:
> > Ok so now I'm getting this when doing a make in asterisk...
> >
> > t
Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Ryan, Travis
> Sent: Wednesday, September 23, 2015 10:01 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [aster
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Ryan, Travis
> Sent: Wednesday, September 23, 2015 9:55 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
>
e: [asterisk-users] problems with PJSIP install on UBUNTU
> 14.04
>
> Ryan, Travis wrote:
> > I've built PJSIP a few months ago on a server that was 12.04 and
> can't
> > remember how I got past this same issue. I've looked at the links
> I'll
>
I've built PJSIP a few months ago on a server that was 12.04 and can't remember
how I got past this same issue. I've looked at the links I'll put below and the
comments section where others had the issue, but those tips aren't helping
either.
Basically everything seems to compile and install co
buntu*CLI> originate SIP/732-xxx-@vonage-out application dial
> == Using SIP RTP CoS mark 5
> [Aug 5 14:16:49] WARNING[32891][C-0006]: chan_sip.c:23160
> handle_response_invite: Received response: "Forbidden" from '"Anonymous"
> >@69.59.234.67<http:/
I¹m not quite sure I understand everything you typed, but I¹ll say the
following.
You can spoof any phone number on any call outbound through your PRI,
except any toll free numbers, etc. Basically if someone else is paying for
the phone call, like a toll free number is, then the payer (called part
Ø From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Oceanet - Cédric
BASSAGET
Sent: Thursday, July 09, 2015 11:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk 13 / realtime voicemai
Asterisk13 can do native tls with each phone? Nice.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ricky gutierrez
Sent: Wednesday, July 08, 2015 3:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discus
I've seen this before. It can be done by calling an AGI script when placing the
outgoing call. You'd then prompt and make sure the code matches and do your
billing logic, etc there. Then place the call if it's valid.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mail
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jamie Rees
Sent: Monday, July 06, 2015 5:54 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] DTMF issue
Hello folks,
We have an issue with several Cisco SPA512G phones co
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan, Travis
Sent: Monday, July 06, 2015 4:20 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] CDR i
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jg
Sent: Monday, July 06, 2015 4:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CDR in an MySQL-Database
> Hi
I hope his mother in law doesn't live with him. That's a support issue for sure.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Larsen
Sent: Thursday, June 25, 2015 2:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discuss
I'm doing an upgrade from Asterisk 11 to 13. I'm following the guide at
https://wiki.asterisk.org/wiki/display/ast/setting+up+PJSIP+REaltime to setup
realtime, as I use realtime on Asterisk 11 too.
I'm getting the following error when trying to connect the peer to the server.
Help? :)
Thanks,
ok at the Patton SmartNode 4110 series devices or a Cisco router with FXO
card and DSP modules. I have deployed both and haven't had any complaints.
They just work once configured.
Ryan
--
_
-- Bandwidth and Colocation Prov
fficially Polycom will fix the issue in 5.3 in a few months..
>
> Thanks
> David
>
Could this be a 5.2.x issue only? I have a hundred of the VVX 400 phones
running 4.1.7 and haven't heard of this issue yet from our users.
Thanks,
Ryan
--
___
the
users handset, once they answer Asterisk then dials the outbound number. No
need for any transferring. You could also look at Asterisk call files to
originate the call.
Ryan
--
_
-- Bandwidth and Colocation Provided by http:
ly using chan_sip with Asterisk. The
result has been rock solid performance.
Ryan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thur
Has there been a change in the way certified Asterisk is being packaged?
Starting with certified Asterisk 11.6 has all the extended options are
checked by default in menuslect? Certified Asterisk 11.2 does not have them
checked and neither does certified Asterisk 1.8.15?
Thanks,
Ryan
ses lsof to check to see if Asterisk is writing
to the file.
/usr/sbin/lsof | grep filename | wc -l
Thanks,
Ryan
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introdu
On Wed, Apr 16, 2014 at 10:20 AM, Kevin Larsen <
kevin.lar...@pioneerballoon.com> wrote:
>
> You are a bit outside of what I have done, but this looks like it might be
> what you want to do with SIP:
> http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP
>
>
I had looked at that g
he trunks and routes is going to become cumbersome.
I wanted to move to DUNDi to simplify the setup. It looks like I need to
switch to IAX trunks to be able to do this.
Thanks,
Ryan
--
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-- Bandwidth and Colocation Provided by http
;Received incoming SIP connection from unknown peer to dundi") in new stack
Is there a way to configure DUNDi to use SIP or does it only work with IAX?
Thanks,
Ryan
--
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-- Bandwidth and Colocation Provided by http://www.api
d version 4.1.5 on a
Polycom VVX 400. Buddies work on all three phones. The firmware is for both
SIP and Lync. You change the base profile option accordingly. Look in the
Polycom UC Software Admin Guide for more information.
Ryan
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t calls for the majority of the day,
and a total of 8-10k calls processed per day. A few times a week I will see
the last minute load at 20 and the 5 min load at 7. This seem to happen
when there are a high volume of new calls as the FreePBX
age to the original UA leaving the first phone stuck in a
holding state. Am I missing something here? Here is the refer sip
message (see attached) Thank you!
--
Ryan Tilton
Seattle Event Disc Jockey
www.DJRT.com
Toll Free: 1-877-411-DJRT
Cell: 206-409-3906
Fax: 206-922-6199
--
Reviews a
scoding mess. Maybe Asterisk 12 with
pjsip will have a better solution.
Ryan
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
c order, however a lower bandwidth codec is chosen in cases where I
would prefer a higher bandwidth codec.
I looked at this a year ago on Asterisk 1.8 and ended up using ulaw for
everything but remote phones. The remote phones end up transcoding g729 to
ulaw for most calls.
Ryan
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and AST_CODEC_CHOOSE would be ulaw this for channel.
However I'm not sure how to make this change as I don't know my way around
the interaction with the Asterisk core and the channels.
Ryan
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-- Bandwidth and Coloc
On Sat, Dec 14, 2013 at 10:31 PM, Ryan Wagoner wrote:
> Let's say I have two devices configured and the follow call scenarios
> occur.
>
> [100]
> disallow=all
> allow=g722&ulaw
>
> Polycom phone with g722,ulaw,alaw,g729
>
> [101]
> disallow=all
> all
in common to prevent transcoding?
Ryan
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
as
success
setting up a VMware ESXi server with Windows VMs for AD and Exchange and
Linux VMs for Asterisk and Web / FTP. Asterisk with Exchange UM for
voicemail is a winning combination and works seamlessly. It is essentially
a private cloud of the customer. Why not use the OS that
o see. Although this might be tricky depending on the OS
version.
Ryan
On Mon, Dec 9, 2013 at 5:56 PM, Jeff LaCoursiere wrote:
> Upgrading an ancient customer installation... was running 1.4.23.1
> (Trixbox) with Zaptel 1.4.12.9 and a Sangoma A102D, which has been running
> fin
I haven't tried it, but the res_corosync module states it will sync device
state across servers.
https://wiki.asterisk.org/wiki/display/AST/Corosync
On Thu, Nov 14, 2013 at 3:54 AM, Leandro Dardini wrote:
> Aligning presence over multiple servers is not simple and require some
> changes on the
;
>
>
You want some form of raid for redundancy. I usually go with two 15K SAS
drives in raid 1 or four 7.2k SATA drives in raid 10. Performance between
the two should be similar. With drives being as cheap as they are skip raid
5.
Ryan
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On Fri, Mar 30, 2012 at 1:16 PM, Bryant Zimmerman wrote:
> What does this patch fix? Why is it not in Jarr?
>
> Thanks
>
> Bryant
>
It looks like the patch is a backport of the t.38 gateway functionality in
As
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