sh, so the instructions for making from
git should include running that.
like most packages do.
-JimC
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now it worked.
too much malicious ecmascript...
-JimC
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r the entire
GJ> organization. So use...
GJ> https://oss-cla.sangoma.com/asterisk/asterisk
GJ> If that doesn't work, I'm going home. :)
nothing occurs when hitting the green "button".
-JimC
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gpg:
Remember libera not libra
On Wed, May 26, 2021, 4:50 AM Joshua C. Colp wrote:
> Greetings all,
>
> For those who may not have seen the blog post[1] or the comments on IRC
> itself, we've moved the Asterisk IRC channels to the new Libera Chat
> network[2]. You can find the #asterisk-dev channel o
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On 30 May 2020 16:12:29 "Joshua C. Colp" wrote:
On Sat, May 30, 2020 at 11:25 AM Roger James
wrote:
Hi Joshua,
.
I am running master. It was me who submitted that change. There is an
anomaly I am seeing at the moment in that a registration request is sent
out over the ipv4 tr
going on. I
need to trap very early on the registration process
Roger
On 30 May 2020 10:42:38 "Joshua C. Colp" wrote:
On Sat, May 30, 2020 at 6:21 AM Roger James
wrote:
I am looking into the generation of unwanted ipv4 registration requests
when registration is configured for
I am looking into the generation of unwanted ipv4 registration requests
when registration is configured for ipv6 only.
[AAPJSIP]
type=registration
transport=ipv6-udp
outbound_auth=AAPJSIP
retry_interval=60
fatal_retry_interval=30
forbidden_retry_interval=30
max_retries=1
expiration=3600
auth
On 14 May 2020 14:04:19 "Joshua C. Colp" wrote:
On Thu, May 14, 2020 at 9:49 AM roger wrote:
On 13/05/2020 17:04, roger wrote:
On 13/05/2020 13:44, roger wrote:
I will strip strip everything out of the conf files except the ipv6
stuff and see what happens.
The stripped out version reta
Hi Joshua,
Thanks for your prompt response. I will comment inline.
On 13 May 2020 11:10:52 "Joshua C. Colp" wrote:
Does "pjsip show endpoint" continue to show the correct transport?
No. By the time asterisk gets to a point where you can interact with the
cli the endpoint shows the name o
I have an endpoint defined in the conf that has the 'transport' field set
to a named ipv6 transport. I had this working on a test system. This
required the application of the gerrit patch 14404 (currently in code
review). I have moved this on to a more realistic test setup. This setup
also has
On 11 May 2020 14:46:55 George Joseph wrote:
On Sat, May 9, 2020 at 3:50 PM roger wrote:
Hi,
I am debugging a problem witj ipv6 and endpoints. I would like to be
able to examine a particular named endpoint when a breakpnt has been
hit. So basically I need a single line of c code that cal
Oops that was someone else (Kevin Harwell) not you.
On 28 April 2020 10:19:29 "Joshua C. Colp" wrote:
On Tue, Apr 28, 2020 at 4:36 AM Roger James
wrote:
I have working login to Jira but I cannot log in to Gerrit. If I go through
the Crowd Id pasword reset process it all seens to wo
"Joshua C. Colp" wrote:
On Tue, Apr 28, 2020 at 4:36 AM Roger James
wrote:
I have working login to Jira but I cannot log in to Gerrit. If I go through
the Crowd Id pasword reset process it all seens to work, but it still gives
me an invalid authentication error when I try to log in.
I have working login to Jira but I cannot log in to Gerrit. If I go through
the Crowd Id pasword reset process it all seens to work, but it still gives
me an invalid authentication error when I try to log in. I am just going
round in circles. Clicking on the contact link just gives me a 404. Do
On 22 April 2020 20:34:21 "Joshua C. Colp" wrote:
On Wed, Apr 22, 2020 at 4:24 PM Roger James
wrote:
Hi,
A very long time since I Iast posted on here. I have recently moved to
Asterisk 16 and have encountered a problem trying to connect via ipv6-udp
transports. Looking at the
Hi,
A very long time since I Iast posted on here. I have recently moved to
Asterisk 16 and have encountered a problem trying to connect via ipv6-udp
transports. Looking at the traffic it seems that Asterisk only ever queries
for DNS A records. Stepping through sip_resolve in pjsip_resolve.c I
ill try on the asterisk-users
> list.
>
> Kind regards,
> Sean
>
> --
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; libogg-devel
> libsndfile-devel
> libsrtp-devel
> libvorbis-devel
> lua-devel
> neon-devel
> speex-devel
> speexdsp-devel
>
> Why RedHat decided to hide these devel packages in a repository that is
> disabled by default it is beyond me.
>
>
> On 10/17/19 2:24 P
:25 AM George Joseph wrote:
>
>
> On Thu, Oct 17, 2019 at 12:03 PM James Finstrom
> wrote:
>
>> I started writing
>> https://github.com/jfinstrom/just-a-wiki/wiki/FreePBX-on-Centos-8 when
>> Centos 8 launched. It actually 90% works.
>> FreePBX itse
e options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-dev
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This email was sent from a personal email account. The content of this
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the
J> pjmedia-sdp code doesn't like it.
Ah. That bug in the sdp should have jumped out at me. [SIGH]
Still it would be better to ignore any dynamic payload from m= which lack
a matching a=. Liberal in a
obviously the
attacker doesn't want audio. But I'd still like to waste some of their
cycles dealing with a series of 180s.
Chan_sip would send the call to the dialplan.
Should pj not also?
-JimC
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to get proper
qualified attention.
On Tue, Oct 23, 2018, 6:57 AM hasan malik wrote:
> james finstrom,
> i find your response insulting.
> i suggest you pick up a copy of how to win friends and influence people or
> some similar title.
>
>
>
> On Mon, Oct 22, 2018 at
ently this reads as, someone remake facebook for me and I
will pay you some random amount of money.
On Mon, Oct 22, 2018 at 12:54 AM hasan malik wrote:
>
> hello james,
> comsys.net has many big telecom providers buying their platform.
> they are not cheap, but they know what they&#x
What feature(s) do you feel are missing?
What is the bounty you are willing to pay?
I am thinking an email this vague will yeild no real return.
On Wed, Oct 17, 2018, 6:17 PM hasan malik wrote:
> Hello,
>
> Bounty will be paid.
>
> Looking for voice messaging platform like comsys.net
>
> Please
I personally am offering to sign GPG keys for the #WebOfTrust at Astricon.
Note this is not an official venture and is not related to my employer or
related projects. Submit a request at: https://t.co/0ti9v9rpMr I will
setup several meeting times to do physical verification including devcon
breaks
t;
>>
>> On 12 Sep 2018, at 16:11, Gaston Draque wrote:
>>
>> From the Asterisk side, I would start by looking into the different logging
>> facilities provided[1] but as stated, which Asterisk API you are using will
>> determine which logging facility to look
i-digital.com --
>
> Astricon is coming up October 9-11! Signup is available at:
> https://www.asterisk.org/community/astricon-user-conference
>
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m's decode routine
truncates the output once it sees a non-gsm 33-octet packet.
Ie one which does not start with the nybble 0xD. So all of the
data after that is lost.
-JimC
--
James Cloos OpenPGP: 0x9
them. Just because you
> don't understand or see the reason doesn't mean it's not valid.
>
> On Mon, Nov 13, 2017 at 4:47 PM, Jean Aunis wrote:
>
>> Le 13/11/2017 à 17:58, Steve Edwards a écrit :
>>
>> On Mon, 13 Nov 2017, James Finstrom wrote:
>&
merged upstream ?
>
> Regards
>
> Jean Aunis
>
>
> --
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elp you. Documentation was mentioned at devcon and in
this post.
Again if there is something NOT on the wiki, or something that needs to be
stripped down to simpler terms bring it up so someone can write it.
On Tue, Oct 10, 2017 at 2:40 PM, Matt Fredrickson
wrote:
>
>
> On Sun, Oct 8, 20
pdate options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-dev
>
> --
> Seán C McCord
> CyCore Systems, Inc
> +1 888 240 0308 <(888)%20240-0308>
> PGP/GPG: http://cycoresys.com/scm.asc
>
> --
> _
One does not simply depricate a sip stack.
Ok so at devcon there was a discussion of depricating chan_sip. This may
sound a lot worse than it actually is. Chan_sip has been essentially
untouched in 4ish years. It does not receive bug fixes. It is just sort of
a barge floating in the ocean.
So one
specify static mappings for otherwise dynamic
codecs. It should provide commented out examples which match the static
numbers used by pre-dynamic versions of asterisk to show how to set
them.
This would allow one to only force static mappings when one has a peer
which requires them.
-JimC
erever appropriate. This should help googlers get back to current
information.
On Wed, Oct 5, 2016 at 4:29 AM, Eric Klein
wrote:
> James,
>
> You missed a few points:
> 1. There needs to be a move in the training materials, and DCAP exam away
> from (the soon to be deprici
t be cleaned up with a little bleach and some elbow grease
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ring groups, this is a FreePBX
> construc. Asterisk itself does allow the option to be set on an individual
> basis for the entries in sip.conf.
>
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
>
> --
> __
for not completing my thought.
I meant to add: [Given all of that,] where is the code which uses
alembic to generate the sql files during the release process?
I don't see any explicit calls when running git grep alembic. And th
wiki page ignores anything other than using alembic to access a
>>>>> "JC" == Joshua Colp writes:
JC> Schemas are now managed using alembic[1]. They exist within the
JC> contrib/ast-db-manage directory.
There are no sql files therein.
-JimC
--
James Cloos
I see that the contrib/realtime directory seen in the tar src does not
exist in the 13 or master branch on gerrit.
Where is it's scm home?
-JimC
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7;s ruri like
chan_sip's CHAN_START event does, even though the subsequent events do.
I thought I had explained that clearly; apologies for missing any
ambiguity in my note.
-JimC
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___
ug? Or an expected behavorial difference beteen chan_sip and
res_pjsip?
For reference, this particular box does not have a context named
default. And I do not see any way to tell res_pjsip what the default
context should be, like sip.conf's [general] section.
-JimC
--
James Cloos
} else if (process_sdp_a_audio(value,
p, &newaudiortp, &last_rtpmap_codec)) {
But, as I wrote, it still failed w/o rtp crypto and forcing avpf works.
-JimC
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_
r tls, and that worked w/o any
force_avp setting in sip.conf.
TCP vs TLS was the only difference. Until I added force_avp=yes, then
the both worked.
-JimC
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hen sip is
secure?
-JimC
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ffer
quality support for it. And Asterisk should use them.
-JimC
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To UN
should be > 0 (the remote sends 1476, which asterisk
dislikes and replaces with 400, which with fec should result in max_ifp
185), so that shouldn't be the reason for the 488.
But I don't see why else it would 488.
-JimC
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a few other endpoints that will be happier if
that RTP bit passes though properly as well.
Thanks!
-James
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Thanks for getting this finished!
-JimC
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to try the two ip version semi-sequentially. (Ie, abandon
the lower pref one as soon as the socket to the higher-pref one is
established.)
-JimC
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.
Any, it is generated:
https://reviewboard.asterisk.org/r/4230/
I don't see a way there to request a review of a proposed patch for
https://github.com/asterisk/pjproject, though.
The patch is posted on the jira issue (ASTERISK-24575).
Is that enough?
-JimC
--
James Cloos
My attempts to start a review of my patch on bug 24575 fail with a 500
error.
I do not see anywhere in jira or on the reviewboard site to submit a bug
report about https://reviewboard.asterisk.org/, so I am posting here.
-JimC
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/jira/browse/ASTERISK-24575
-JimC
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parallel for cafile was trivial:
Twelve added and five changed lines for https://github.com/asterisk/pjproject
and twelve added lines for asterisk trunk.
I still need to test, though.
Should the pjproject diff go via github or via the asterisk bug tracker?
-Ji
On Wed, Nov 26, 2014 at 4:54 PM, James Lamanna wrote:
> Hi,
> I have filed a bug for this here:
> https://issues.asterisk.org/jira/browse/ASTERISK-24555
>
> It appears that something in Asterisk 11 is continuously allocating frames
> and not freeing them.
> This cause
.
This PBX is primarily used for faxing (Asterisk + IAXModem & T.38 Gateway)
so I'm wondering if the issue is in the SLIN codec or the T.38 gateway code.
I've taken a quick glance and I haven't seen any mismatched ast_frdup() /
ast_frfree() yet.
nk?
-JimC
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I
presume there isn't anything lacking in spandsp.
-JimC
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To UNSUBS
s not in a nat. And it certainly did use the
same src port for the t38 as it had been using for the rtp. The dst
ports were different -- each was correctly sent to the other side's
advertised port -- but both came from the s
the initial INVITE.)
I don’t see anything in the rfcs or in t.38 which requires that either
send send from the same port it advertises to receive, but did I miss
anything?
Nor do not see any relevant differences between chan_sip in 1.8 vs 11,
so is this expected?
Thanks,
-JimC
--
James Cloos
; n,SIPAddHeader(URL:https://example.com/sip)
same => n,Dial(SIP/target/${EXTEN})
-JimC
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ast
Missed that last night. Thanks.
MJ> The ability to add whatever header you want to your outbound INVITE
MJ> requests is a much more powerful abstraction
Agreed. Heartily.
And it works nicely. I should have noticed that ☹.
Thanks!
-JimC
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-JimC
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the code to detect. Its used to detect answer,
busy, congestion, dialtone, dead, and others on IAX, SIP, ZAP, and
other channels if there is another application that will do the same
please let me know.
Regards
James
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I installed asterisk now.
I have an eicon Diva T1 card installed
but I am not allowed to installed the drivers.
I can not use the su command to get into root.
how do I install the source RPM for the Diva card
also the capi-chan to run it will capi and what is the USA settings for a T1
robbed bit s
Hi Paul I recived my first inbound call on the Tesco IAX2 trunk last
night and it all worked perfect (the CSID was not past through but I
think this might have been as it was a mobile call and that it might
have had it turned off)
I would love to see this code checked in and to find its way in to
Well I throught it was working, but I did a few more tests and after a
few minutes if you call my inbound number I get sent to the providers
voice mail meaning the some how the provider things I am off line.
Doing iax2 show registry shows as Registered
Any ideas guys/girls
On 11/4/06, James
Thanks Paul will give that a shot
On 11/3/06, Paul Hewlett <[EMAIL PROTECTED]> wrote:
On Friday 03 November 2006 12:44, James Trix wrote:
> Hello list
>
> I have been trying to work out why my registration to my provider
> keeps getting droped every few seconds and all my i
Hello list
I have been trying to work out why my registration to my provider
keeps getting droped every few seconds and all my inbound calls go to
the providers voicemail. After some traceing and some googleing I
found the problem.
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Sub
but in the register program it is staticly linked.
Matt Riddell (IT) wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
James Jones wrote:
Does anyone know why the g729 codec module sold by diguim does not
display the OpenSSL copyright information. Do they have an agreement
with
Does anyone know why the g729 codec module sold by diguim does not
display the OpenSSL copyright information. Do they have an agreement
with OpenSSL to not display the Copyright Information that is required
ny their license when distributed as part of a binary that uses OpenSSL.
The registrat
> Hi James,
>
> it looks like you are using the 0.2.1 Version of chan_misdn and
> therefore an old mISDN. I would recommend to upgrade to the latest
> chan_misdn version and get the newest mISDN from the i4l cvs.
>
> Just have a look at http://www.voip-info.org/wiki/view/
Hi there,
Is this the correct list for questions relating to using the manager API?
Cheers
James
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ient to provide adequate protection for the transported data.
I don't think anyone was suggesting using IPSEC tunnel mode, just
transport, although I'm not sure that NAT is a solved problem there.
But I agree with you that VPN's aren't neces
> In article <[EMAIL PROTECTED]>,
> James Harper <[EMAIL PROTECTED]> wrote:
> >
> > > Paul Cadach mentioned something about a jitter buffer for TDMoE,
but
> > > I don't know whether he was talking about an idea or some real
code.
> > >
> &
channel numbering scheme) would self describe
the adapter and span they belong to.
That's my AUD$0.02 fwiw.
James
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..
If you ever get a spare moment, could you write some brief documentation
on the flow of data from ethernet into zaptel and from zaptel out to the
ethernet? Failing that, if I were to look through the code and write
some doco, could you look at it and tell me where I messed up? I'
> Richard Lyman <[EMAIL PROTECTED]> wrote:
> > James Harper wrote:
> >
> > >Is there an RFC or other technical documentation for the zaptel
TDMoE
> > >protocol anywhere?
> > >
> > http://www.dynx.net/ASTERISK/TDMOE/TDMoE-HOWTO
> >
> &
Is there an RFC or other technical documentation for the zaptel TDMoE
protocol anywhere?
Thanks
James
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On 27/01/06, tim panton <[EMAIL PROTECTED]> wrote:
>
> If I read the patch right this was a bug where a signed 32bit quantity was
> treated as if it were unsigned (or the other was around).
>
> You'll only catch this kind of thing with lint, and/or strict use of
> macros/functions to do time compar
tive'
cards... can zaptel cope with different levels of processing being done
on the card, or does it like to do it all itself? Or none of it itself?
I've had a look around for this sort of documentation, and it doesn't
appear to exist except in the code, which is fine, i
Tilghman Lesher wrote:
On Tuesday 27 December 2005 14:15, James Sizemore wrote:
I think I found what is munging up the peer lookup:
This call from another Asterisk box starts:
<-- SIP read from 192.168.69.254:5060:
The peer lookup that fail reads:
<-- SIP read from 192.168.7.250
Doubling the value to 500 did not seem to effect the length of the
tone played at allhm. Back to the drawing board for me.
Anyone know what this value is supposed to effect?
James Sizemore wrote:
I did a bit of searching around and found this class in chan_sip.c:
I am going to test the
quot;, "application/dtmf-relay");
add_header(req, "Content-Length", clen);
add_line(req, tmp);
return 0;
}
==
James Sizemore wrote:
I have a gateway using a Digium card to convert a PRI
call to a sip call then I transport the sip call to a Cisco
IAD w
I have a gateway using a Digium card to convert a PRI
call to a sip call then I transport the sip call to a Cisco
IAD where it is converted back to a PRI. This all works
well except DTMF is sent with a duration of .25sec.
PRI specs says this should be .25sec to .5sec so this
is with in spec, howev
If I had to guess you also log cdr's to a database and your database
server is slow for some reason, Asterisk will not hang up a call till
the database query finished, the telco will only wait so long for an
acknowledgment from a hang up and disconnects it's end and tried to use
the same channe
Is there anyon here currently in New Zealand that use asterisk, I need to help getting voice and internet services. I will be moving in a week. Any help would be great. Please use the details below to get ahold of me.
Thanks in advance.
James Jones
Signate, LLC
[EMAIL PROTECTED
igium want to sell the product like it is rightnow, and have no plan to
do masive change to fix any core problems. They think that if they
start redesign this, it will bring back asterisk to be unstable again.
Marc O.
James Jones wrote:
> I know of good way to solve this problem. I h
I know of good way to solve this problem. I have been authorize by my company to try to a group of people and businesses to give donations to get Digium to fix this issue. We will start the pot at $200. Are there any takers?
On Sat, 2005-08-27 at 10:08 -0400, [EMAIL PROTECTED] wrote:
> So
ple rates -- then stuff like
this could be thrown in easily as an add on.
-JimC
--
James H. Cloos, Jr. <[EMAIL PROTECTED]>
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on of Speex installed too.
In exchange for this you'll have the possibility of lowered bandwidth,
automatic gain control, denoiser and dereverb. The downside is slightly
higher CPU consumption and silence periods sounding "dead" when Speex
runs out of frames to interpolate during si
> Are there any SCCP feature requests out there? I had a quick look on
> Mantis but couldn't really see anything.
One of the requests I have from my people is to add server side
conferencing. We currently have virtual PBX from ICG, and one of the
things everyone likes is the ability to add as ma
We have tested using the current CVS. I would not post this bounty if it was such a simple solution.
On Thu, 2005-04-28 at 12:35 -0500, Steven Critchfield wrote:
On Thu, 2005-04-28 at 12:46 -0400, James Jones wrote:
> My company needs a stable version of app_chanspy. The current vers
My company needs a stable version of app_chanspy. The current version crashes asterisk on our systems. We will pay $250 if it done by the end business on May 6, 2005, After that it will we pay $100. We need it run on asterisk-1.0.2. Please contact me at [EMAIL PROTECTED] if you complete this ta
Stephan A. Edelman wrote:
Hello John,
I'll see if I can create a few diffs and post this on the site you
suggested.
The integration wasn't that straightforward: Sphinx's real-time decoder
uses 16KHz 16-bit PCM, whereas Asterisk provides a feed through the EAGI
interface at 8KHz. I just used a c
I doubt this will be useful for GPUs behind AGP busses. It may be OK
for those directly on a PCI bus, but it ought -- in theory ;/ -- to
work well over a PCI Express bus.
PCI Express should have enough bandwidth and low enough latency in
both directions to help.
AGP defines two-way communication
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