https://issues.asterisk.org/jira/browse/DAHLIN-379
Le mer. 20 janv. 2021 à 19:50, Jerry Geis a écrit :
> When might there be a new dahdi complete to support the 5.4 kernel?
> Thanks,
>
> Jerry
> --
> _
> -- Bandwidth and
May I add that, to me, I would expect Asterisk to use CALLERID vlaues (name
and num) to set P-Asserted-Id.
Maybe in a couple of days, I'll report my findings here if can find some
time to experiment with Asterisk 17 or Asterisk 18 and compare behaviours..
Le mar. 8 déc. 2020 à 16:41, Olivier
Hello,
With Debian Buster's Asterisk 16.2.1, please consider the following dialplan
;Case A
;exten = 29,1,Dial(PJSIP/${EXTEN})
;Case B
;exten = 29,1,Gosub(foo,${EXTEN},1)
;Case C
exten = 29,1,Gosub(bar,s,1(${EXTEN}))
[foo]
exten = _X.,1,Dial(PJSIP/${EXTEN})
same = n,Return()
[bar]
exten =
, Olivier a écrit :
> Hello,
>
> What is the most FHS-esque (see [1]) way to run several Asterisk
> instances on a single (Debian) host ?
>
> What would you recommend ?
> Would gather each instance directories (etc/, run/, lib/, ...) in
> something like /srv/instance1/
>
Hello,
What is the most FHS-esque (see [1]) way to run several Asterisk instances
on a single (Debian) host ?
What would you recommend ?
Would gather each instance directories (etc/, run/, lib/, ...) in
something like /srv/instance1/
(it doesn't please me as I like to put variable data in /var
Hello,
How would you test how a PJSIP-powered Asterisk 13 instance resist to
hostile REGISTRATION attempts ?
Would you use SIPp ? Any example scenario ?
Would you go with an alternative tool ? Which one would you pick ?
Best regards
--
Hello,
The other day, a 13.14.1 server suddenly stopped working correctly.
First, it printed:
Oct 23 21:53:40 FOOBAR asterisk[2377]: WARNING[27942]: db.c:332 in
ast_db_put: Couldn't execute statment: SQL logic error or missing database
This occurred while this server received a lot incoming
Hi,
What if some fail2ban magic could keep OpenSIPs response from hitting
Asterisk after N attempts ?
Le mer. 28 oct. 2020 à 18:32, Kingsley Tart - Barritel Ltd <
kingsley.t...@barritel.com> a écrit :
> Hi,
>
> We're using Asterisk 13.17.0 with PJSIP 2.8 bundled.
>
> I've found an issue when
53, Joshua C. Colp a écrit :
> On Tue, Oct 27, 2020 at 5:35 AM Olivier wrote:
>
>> Hello,
>>
>> Where can I find doc about PJSIP's ice_support parameter ?
>>
>> Do you need to configure things elsewhere in Asterisk config files
>> (rtp.conf, PJSI
Hello,
Where can I find doc about PJSIP's ice_support parameter ?
Do you need to configure things elsewhere in Asterisk config files
(rtp.conf, PJSIP transport sections, ...) to make ICE work properly ?
I'm asking because, if I'm not mistaken, STUN requires setting a STUN
server so I think ICE
Hello,
Is it possible to set different features.conf dialing sequences (atxfer,
pickup, ...) for different users ?
For instance, what if I want Alice to dial *8 to pickup a call and Bob to
dial ** to pickup calls ?
I can see that features.conf includes application maps but can these be
used for
Hello,
In project, a customer has two WAN access. More precisely:
Internet - --- Router1 --- FortiGate Firewall Router
-- Asterisk
| |
- --- Router2 --
Both WAN
Le sam. 18 juil. 2020 à 08:02, Andrew Yager a écrit :
> Hi,
>
> I realise this is an old question, but I’m struggling to get my head
> around it.
>
> The ERD suggests that endpoints can link to multiple AoRs
>
> In what situation would you actually use this? Given that mapping of
> inbound calls
[3] https://searchcode.com/codesearch/view/50276540/
Le mar. 21 juil. 2020 à 12:00, Sylvain Boily a écrit :
> Hello,
>
> On 2020-07-21 3:57 a.m., Olivier wrote:
>
> Hi,
> Le ven. 10 juil. 2020 à 16:56, Sylvain Boily a écrit :
>
>>
>> It probably can help you:
&g
Hi,
Le ven. 10 juil. 2020 à 16:56, Sylvain Boily a écrit :
>
> It probably can help you:
>
> https://github.com/wazo-platform/wazo-provd-plugins/blob/master/plugins/xivo-jitsi/1/templates/base.tpl
>
> Sylvain
>
> Yes, provided example was exactly what I was after !
Thank you very much !
--
Though ASTDB use is very flexible with DB() function, abandoning this
flexibility to gain REDIS features (clustering, speed, ..) has benefits.
Still, being able to "implement ASTDB with REDIS" would allow a lot
currently impossible setups (synchronizing SIP registrations on multiples
Asterisk
Hello,
1. I'm looking for an (anomized) example of a Jitsi Desktop provisioning
file compliant with Asterisk ?
Jitsi Doc mentions it should adhere to Java properties file syntax (see
[1]) but a working example would help.
If this example file included the following settings, it would be
It seems a new Linphone 4.2 is to be published next week !
Hopefully, ...
Le ven. 5 juin 2020 à 13:34, John Hughes a écrit :
> On 26/05/2020 15:33, Olivier wrote:
>
> Hi John,
>
> 1. Could you get any further, in your quest for working BLF with linphone ?
>
> The patches
Lately, I read [1].
So it seems both Jitsi desktop and Linphone are on par, on this ;-)))
[1] https://community.jitsi.org/t/busy-lamp-field-bug/15931
Le ven. 5 juin 2020 à 13:34, John Hughes a écrit :
> On 26/05/2020 15:33, Olivier wrote:
>
> Hi John,
>
> 1. Could you get any f
Hi John,
1. Could you get any further, in your quest for working BLF with linphone ?
2. Have you tried with a different Linphone version (4.12 is pending on
Linux, packaged as an AppImage, or 4.11 exists on iOS/Android/Win10) ?
Best regards
Le mer. 25 mars 2020 à 15:06, John Hughes a écrit :
Hello,
I've seen that Asterisk 17 supports Prometheus but beside [1], I've not
much about how to use this.
Can someone shed some light on this ?
1. If I'm not mistaken, Prometheus favors "a pull model over HTTP".
So basically, a Prometheus instance should be able to query Asterisk "core
Hello,
What is the recommended way to build a language selection menu like "For
english, press 1, Pour le français, tapez 2, ..." ?
Is possible to rely on CORE-SOUNDS-EN, CORE-SOUNDS-IT, ... to find needed
audio files ?
Best regards
--
Hello,
Hard to tell but from [1], " Exceptionally long voice queue length
queuing" should not happen.
1. Are you seeing WARNING or ERROR instances in log before this
"Exceptionally long voice" WARNING ?
2. If possible, I would try to load test a similar setup with SIPp and see
if I can reach or
Hello,
I'm using an Asterisk 17 dialplan that currently includes:
1. many "DB gets" calls (ie statements like Set(FOO=${DB(Foo/Bar)})
2. and a couple of "DB puts" (ie statements like Set(DB(Foo/Bar)=Foo) or
DB_DELETE(Foo/bar))
I would like to add an HTTP Provisionning API that would allow an
+Asterisk+for+WebRTC+Clients
Le mer. 8 janv. 2020 à 10:04, Olivier a écrit :
> Hello,
>
> Le lun. 6 janv. 2020 à 19:01, Olivier a écrit :
>
>> May I add I could successfully (if pjsip show transports has any meaning)
>> add a PJSIP TLS-transport with:
>>
>&g
2020 à 15:11, Olivier a écrit :
> Hello,
>
> On a Debian Buster instance, I compiled Asterisk 17.3.0 from source.
>
> I enables all 3 File, IMAP and ODBC voicemail modules but I'm still using
> classical File module (in modules;conf and voicemail.conf):
> cd asteris
Hello,
On a Debian Buster instance, I compiled Asterisk 17.3.0 from source.
I enables all 3 File, IMAP and ODBC voicemail modules but I'm still using
classical File module (in modules;conf and voicemail.conf):
cd asterisk-17.3.0
...
make menuselect.makeopts
menuselect/menuselect --enable
Hello,
For the very first time, I tried the command bellow on a newly build Debian
Buster box on which I successfully built Asterisk 17.2.0 before. I got :
# contrib/scripts/install_prereq install-unpackaged
*** Installing NBS (Network Broadcast Sound) ***
Anbs-trunk/LICENSE
A
Hi,
As mentioned in [1], a common pattern is to let everyone monitor everyone
except oneself.
How do implement this ?
Is there something like this:
[alice_list]
list_item = full_list
list_exclude_item = alice
[bob_list]
list_item = full_list
list_exclude_item = bob
If negative, would it
Hello,
My Asterisk 16.2 instance (Debian Buster package) has:
same = n,Verbose(0,CHANNEL is ${CHANNEL})
same = n,Verbose(0,CHANNEL(accountcode) is ${CHANNEL(accountcode)})
same = n,Verbose(0,CHANNEL(contact) is ${CHANNEL(contact)})
same = n,Verbose(0,CHANNEL(endpoint) is ${CHANNEL(endpoint)})
How could I miss this blog post ?
Thank you very much, Ben, for replying !
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
Hello,
>From Astricon 2019 notes [1], you can read "[a]ll 3 app_voicemail variants
can now be built".
What does it mean ?
Is this change tied with a specific Asterisk version ?
Is possible to change from ODBC to IMAP without re-compilation ?
Is it also possible to mix mailbox types on a single
Hello,
Days ago, I banged on a similar issue with Debian Buster's asterisk (16.2):
my box had two interfaces (one North and one South) both with private
addresses
when relaying calls from South to North, my box used South Address for
media handling.
Upgrading to 16.7.0 without changing
Hello,
Le lun. 6 janv. 2020 à 19:01, Olivier a écrit :
> May I add I could successfully (if pjsip show transports has any meaning)
> add a PJSIP TLS-transport with:
>
> [transport-tls]
> type=transport
> protocol=tls
> bind=0.0.0.0:5061
> cert_file=/etc/as
Hello,
Reading [1], I would be happy to discuss here, the changes bellow.
1. In "Create certificate" section, instead of 'ls -w 1
/etc/asterisk/keys', could a 'ls -l /etc/asterisk/keys' be used ?
This would help to check file permissions.
If possible, having those file permissions shown when
:33, Olivier a écrit :
> Hello,
>
> On a newly re-installed Asterisk 16.7.0 on Debian Buster, I can't find a
> way to enable HTTPS.
> Asterisk is running as asterisk:asterisk:
>
> asterisk 11097 0.3 6.7 741352 67984 ?Ssl 17:53 0:06
> /usr/sbin/asterisk -g -f
Hello,
On a newly re-installed Asterisk 16.7.0 on Debian Buster, I can't find a
way to enable HTTPS.
Asterisk is running as asterisk:asterisk:
asterisk 11097 0.3 6.7 741352 67984 ?Ssl 17:53 0:06
/usr/sbin/asterisk -g -f -p -U asterisk
# cat /etc/asterisk/http.conf
[general]
Hello,
In /etc/asterisk/users.conf, you can set hassip=yes to declare a chansip
entity.
Is there any equivalent for PJSIP ?
Best regards
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the
Hello,
I'm thinking about using a single (long) integer value as a compact way to
store user privileges.
Do bitwise operations as in [1], exist in Asterisk's dialplan ?
Any workaround ?
[1] https://wiki.python.org/moin/BitwiseOperators
Best regards
--
Hello,
I've just discovered jigasi : "a server-side application acting as a
gateway to Jitsi Meet conferences. Currently allows regular SIP clients to
join meetings and provides transcription capabilities"
Have someone used it with Asterisk ?
How does it work ?
[1]
Hello, I need some advice:
I use 2 different suppliers of trunk SIP in my infrastructure, both send me
regularly prices in a .csv format.
So I have two SQL tables that contain the prefix and the tariff.
For now, I generate a dialplan with a Perl script that allows me to select
the prefix trunk
Hello,
Have you tried with ACL (acl.conf) ?
Cheers
Le lun. 18 nov. 2019 à 13:22, Benoit Panizzon a
écrit :
> Hi Gang
>
> To increase security against phished passwords and similar attacks, we
> consider offering customers to define IP ranges (or GeoIP locations)
> from which their dynamic
Hello,
Reading [1], I would be very curious to read about WebRTC on MacOS, either
for Voice or Voice and Video calls.
How does MacOS compare today to Windows or Linux regarding WebRTC support ?
Do you need to use Chrome or Firefox to get WebRTC ?
[1]
Hello,
I would like to offer end users in a LAN, asking for this (why ? I don't
know) the capability to use a laptop (along or in replacement of
hardphones) to emit and receive PSTN calls.
PSTN pass through a plain SIP trunk which does not support video (nor Opus)
How can I best integrate
Hello,
Following [1], you get precious help for webRTC installation.
Something that is missing there, though, is a note expliciting
/etc/asterisk/keys files ownerships and modes.
As people are either running asterisk as root:root, asterisk:root and
others or as asterisk:asterisk, the number of
.so
noload = res_hep_pjsip.so
noload = res_hep_rtcp.so
Le lun. 18 nov. 2019 à 22:18, Olivier a écrit :
> Unfortunately, changing ownership did not solve the issue:
>
> # ls -al keys/
> total 40
> drwxr-xr-x 2 asterisk asterisk 4096 nov. 18 20:47 .
> drwxr-x--- 3 asterisk asterisk 4096 nov.
> Asterisk HTTP General Status
/static/... => Asterisk HTTP Static Delivery
Enabled Redirects:
Le lun. 18 nov. 2019 à 22:08, Richard Mudgett a
écrit :
>
>
> On Mon, Nov 18, 2019 at 2:53 PM Olivier wrote:
>
>> Hello,
>>
>> I've installed a new Asterisk 17.0.0
Hello,
Reading this old thread, isn't there also an error in [1] as It also
mentions a tlscafile setting.
Cheers
[1]
https://wiki.asterisk.org/wiki/display/AST/Installing+and+Configuring+CyberMegaPhone
Le ven. 7 déc. 2018 à 16:41, Kevin Harwell a écrit :
> On Fri, Dec 7, 2018 at 9:11 AM Dan
Hello,
I've installed a new Asterisk 17.0.0 on a Debian Buster system.
This Asterisk instance is run by asterisk user (and group).
I've got:
# ls -l /etc/asterisk
total 68
-rw-r--r-- 1 asterisk asterisk 501 nov. 18 19:12 asterisk.conf
-rw-r--r-- 1 asterisk asterisk 135 nov. 18 18:57
Hello,
With Debian Buster's asterisk package, what can you use instead of Digium's
contrib/scripts/ast_tls_cert ?
If that matters, this is for using WebRTC and Cyber Mega Phone 2K (both on
the same box) in a private LAN environment.
My intent was to use easy-rsa package but I wouldn't mind
Le lun. 26 août 2019 à 14:21, Joshua C. Colp a écrit :
> On Mon, Aug 26, 2019, at 9:00 AM, Olivier wrote:
> >
> >
> > Le lun. 26 août 2019 à 12:07, Joshua C. Colp a écrit
> :
> > > ...
> > >
> > > libpjnath is the ICE/STUN/TURN library which i
Le lun. 26 août 2019 à 12:07, Joshua C. Colp a écrit :
> ...
>
> libpjnath is the ICE/STUN/TURN library which is used by res_rtp_asterisk
> for that functionality. If you're using WebRTC or ICE/STUN/TURN, then you
> would be using that library.
>
Yes, I'm using ICE/STUN/TURN.
That explains
Hello,
I've got an Asterisk 11.13.1 system running on a Debian Jessie platform.
This system's extensions.conf doesn't include any reference to PJSIP, yet
(only using chan_sip at the moment).
This morning, it failed with:
Aug 26 09:07:33 foobar kernel: [6534231.776418] asterisk[9701]: segfault at
Le ven. 18 janv. 2019 à 17:30, Joshua C. Colp a écrit :
>
>
> >
> > You mean with a softphone you can't select a single (or several) video
> > among those available, can you ?
> > Even with DTMF sequence and some features.conf magic, a user cannot ask
> > to receive a specific video stream ?
>
Thanks for your (fast) reply !
Le ven. 18 janv. 2019 à 16:32, Joshua C. Colp a écrit :
> On Fri, Jan 18, 2019, at 11:22 AM, Olivier wrote:
> > Hello,
> >
> > I've just read in [1] about SIP MESSAGE addition to both chan_pjsip and
> > ConfBridge.
> > I
Hello,
I've just read in [1] about SIP MESSAGE addition to both chan_pjsip and
ConfBridge.
It seems very interesting addition as it brings the capability to mix
voice, video and text in conferencing.
On an other hand, there are some softphones (Zoiper, Bria, ...) that tout
voice, video and chat
Hello,
Is it possible to find real domain names instead of IP addresses in SIP URI
?
For instance, in a book dedicated to SIP (Understanding the Session
Initiation Protocol), I'm reading an example of a SIP INVITE that looks
like:
INVITE sip:4...@salzburg.at;user=phone SIP/2.0
...
In my
Hello,
These questions crossed my mind this morning :
In general, are anonymous international calls allowed (ie calling from one
country to a number in an other country while hiding your own caller id) ?
Are there special rules in Europe for this ?
Best regards
--
Hello,
Page [1] gathers information on how to configure Asterisk CDR Radius
backend.
I'm not familiar at all with Radius in IP Telephony.
1. Would a Radius database and its associated tools allow live call
accounting data displaying of an Asterisk instance powered by such CDR
Radius backend ?
Hello,
There is question that bounces in my mind for quite a long time.
Today, I dare to ask it here:
how do you package and use your custom asterisk .deb package ?
The background is:
- I'm now a long time Debian user and I learned to appreciate Debian's deb
package benefits specially when
A site question: which of the following RFC would describe as-feature-event
?
[1] https://www.iana.org/assignments/sip-events/sip-events.xhtml
Le mer. 1 mars 2017 à 21:03, Trey Hilyard a écrit :
> Is there any "easy" way to add a custom subscribe handler? I have a set of
> users with Polycom
Hi all,
Is there a way with Polycom phones or alternatives, to configure a specific
SIP server for such as-feature-event or call-info events ?
If positive, maybe a third party SIP server (Kamailio, ...) supporting
those events would allow such implementation.
Looking at Yealink phone Admin
Hello,
I've been asked if it is possible or not to set several (10 or so) SIP
trunks between two boxes, one beeing an Avaya IPBX, the other being an
Asterisk 13 or 16 box (with either chan_sip or pjsip).
The reason behind this question come from billing requirements.
I'm not convinced yet
Hello,
I've edited my diaplan to print some data on screen with statements like:
[foobar]
exten = foo,1,Verbose(0,Whatever I need to display)
exten = bar,1,Verbose(0,Some more text)
When using rasterisk and entering "channel originate Local/foo@foobar
application Noop", I can read lines such
Hello,
On a linux desktop, Google Contacts web application ties phone numbers with
URL such as
https://hangouts.google.com/?action=chat=%2B123456789=fr=0.
Have you ever tried to redirect or rewrite such URL and replace with
something like
https://myasteriskdialer.example.com/?tel=%2B123456789 ?
Hello,
I'm setting up a new cluster that must replace several old Asterisk
instances.
For various reasons, this new cluster must use chan_sip (migration to PJSIP
is planned in a later phase).
This new cluster uses VRRP in active/passive mode:
- at any time, only one cluster member is active,
-
Hello,
I'm testing an Asterisk instance.
At the moment, I'm focusing on its capability to receive and challenge
incoming SIP Registrations.
For various reasons, I would prefer to use SIPp instead of Asterisk to act
as SIP Client.
Has someone successfully done this ?
If negative, what explains
Hello,
I'm curently setting a lab environment for load testing an Asterisk
instance.
This environment includes:
- a management workstation where I would like to run scripts and store
test reports
- a box hosting SIPp
- the Asterisk box I'm load testing (System Under Test)
- an other Asterisk
Le mer. 10 oct. 2018 à 12:26, Joshua Colp a écrit :
> On Wed, Oct 10, 2018, at 4:27 AM, Olivier wrote:
> > Hello,
> >
> > I think I met a case similar to the one solved by [1] . Quoting this
> case :
> >
> > * res_pjsip: Handle deferred SDP hold/unhold proper
Hello,
I think I met a case similar to the one solved by [1] . Quoting this case :
* res_pjsip: Handle deferred SDP hold/unhold properly.
Some SIP devices indicate hold/unhold using deferred SDP reinvites. In
other words, they provide no SDP in the reinvite.
A
Hello,
Now that systemd is default init system in several Linux distribution, is
there a Makefile entry to generate a local systemd asterisk.service file ?
Something like "make asterisk-service" just like "make config".
Cheers
--
Hello,
I've just read this [1] blog entry.
I'm completely new with statsd.
My questions are:
1. This [1] mentions both res_chan_stats and res_endpoint_stats.
I can't find any res_chan_stats.so or res_endpoint_stats.so file in my
debian Stretch asterisk box.
What does it mean ?
2. On a general
Hello,
On a freshly update Debian Stretch packaged-Asterisk (13.14.1) box, I'm
reading this:
asterisktuto*CLI> module load res_statsd.so
Unable to load module res_statsd.so
Command 'module load res_statsd.so' failed.
[Oct 9 12:53:26] WARNING[488]: loader.c:1077 load_resource: Module
roadcast-sp-edition>
> SMS, Fax and Voice broadcasting & Inbound / Outbound Campaigns
> http://www.ictbroadcast.com/
>
>
> On Thu, Sep 27, 2018 at 1:06 AM Carlos Chavez wrote:
>
>> On 9/26/18 10:20 AM, Matthew Fredrickson wrote:
>>
>> > On Wed, Sep 26, 20
Le mer. 26 sept. 2018 à 16:40, Carlos Chavez a écrit :
> On 9/26/2018 4:46 AM, Olivier wrote:
>
> > Hello,
> >
> > This morning, I asked myself if WebRTC could be a viable alternative
> > to softphone deployment.
> >
> > For me, main issue wi
Hello,
This morning, I asked myself if WebRTC could be a viable alternative to
softphone deployment.
For me, main issue with Softphones is the amount of work needed for
installation and configuration.
Also, Softphones must be carefully choosen if Deskphone-like quality is
expected.
Now that
Hello,
For personal lab testing, I would like a mock database, replacing a legacy
ENUM database.
More precisely, I would like to:
- play with Asterisk's ENUMLOOKUP, ENUMQUERY and so on functions
- populate mock db with a couple of fake numbers or ranges of numbers
- test common use-cases
> Cmnd_Alias EDITORS = /bin/nano, /etc/asterisk/[A-z]*, /usr/bin/vim
> /etc/asterisk/[A-z]*
>
> %pbxadmin ALL = (root) NOEXEC: EDITORS, ASTERISK, CAPTAGENT
>
> This prevents my admin users from being able to spawn a shell or
> subprocess from vim, nano, and the asterisk console.
&
Hello,
Is there a way to let someone access to Asterisk CLI and type whatever
command (s)he likes but the shell command (the ones started by !) ?
Ideally, it could be an argument to rasterisk:
rasterisk --no-shell
When done, a session could be like this:
> pjsip show endpoints
...
> core
2018-08-14 15:53 GMT+02:00 Barry Flanagan :
> On Tue, 14 Aug 2018 at 14:34, Olivier wrote:
>
>> Hello,
>>
>> I've got Asterisk installed on a Debian Stretch host.
>> From another Debian Stretch host on which Asterisk is not installed, I
>> want to
Hello,
I've got Asterisk installed on a Debian Stretch host.
>From another Debian Stretch host on which Asterisk is not installed, I want
to run rasterisk over SSH in one step with:
ssh root@foobar rasterisk
The above command "rougly works" but some non-printable characters cause
undesirable
Hello,
I've just discovered chan_sip's ignoresdpversion setting.
Do you use it ?
If positive which kinnd of issue could you solve with it ?
Best regards
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Hello,
In my testing, I saw that Asterisk always included a REFER value in each
INVITE's Allow header, no matter how allowtransfer/allow_tranfer was set.
Is there a way to remove this REFER value entirely either globally or
specifically for a given peer/endpoint ?
Which telephony feature would
2018-06-05 20:29 GMT+02:00 George Joseph :
>
>
> On Tue, Jun 5, 2018 at 10:59 AM Olivier wrote:
>
>>
>>
>> 2018-06-05 15:27 GMT+02:00 George Joseph :
>> Thank you very much, George for replying.
>>
>>>
>>>
>>> On Tue, Jun 5
2018-06-05 15:27 GMT+02:00 George Joseph :
Thank you very much, George for replying.
>
>
> On Tue, Jun 5, 2018 at 3:35 AM Olivier wrote:
>
>> Hi,
>>
>> After a long discussion with a friend, I would like to ask here:
>>
>> 1.According SIP RFCs, is pos
Hi,
After a long discussion with a friend, I would like to ask here:
1.According SIP RFCs, is possible/recommended to have different values in
>From and P-Asserted-Id fields ?
For instance, From field showing 123456789 and P-Asserted-Id showing
987654321 (beside privacy considerations) ?
2.
Hello,
Thinking back to my current practices, I would be very curious to share
here about when should applications such as Congestion, Progress or Ringing
be used in today's telephony.
I would define today's telephony with:
- SIP phones
- Asterisk
- a SIP trunk to an ITSP
- fixed or mobile lines
Hello,
I don't know if this list is the best place to ask such question but here
it is, anyway.
In page [1], I can read in PJSIP's endpoint section configuration reference:
identify_by username,location Way(s) for Endpoint to be
identified
Then clicking over identify_by text, you
2018-04-27 14:59 GMT+02:00 Joshua Colp <jc...@digium.com>:
> On Fri, Apr 27, 2018, at 9:57 AM, Olivier wrote:
> > Hello
> >
> > I've just discovered this [1] invaluable blog post (thank you very much
> > Richard for writing it) and its reference to PJSIP's
>
Hello
I've just discovered this [1] invaluable blog post (thank you very much
Richard for writing it) and its reference to PJSIP's
endpoint_identifier_order setting.
On my Debian Stretch box powered with a packaged Asterisk 13.14.1, I edited
a pjsip.conf file with the following content (and
Hello,
>From [1], you can read:
"If you don't have an identify section defined, or else you have
res_pjsip_endpoint_*identifier_ip* loading *after* res_pjsip_endpoint_
*identifier_user*, then ..."
To remove the above uncertainty coming from modules loading order, how can
you either or both :
-
gt;, it
would match foobar with a [foobar] endpoint, and then set CALLERID(name) to
John Doe and CLAARID(num) to 123456789
2018-04-27 12:00 GMT+02:00 Olivier <oza.4...@gmail.com>:
> Hello,
>
> I'm setting an Asterisk 13.14.1 box (Debian Stretch with packaged
> Asterisk) to impl
Hello,
I'm setting an Asterisk 13.14.1 box (Debian Stretch with packaged
Asterisk) to implement SIP trunking services ie to both trunk with carrier
trunks and IPBX trunks from various brands.
For various reasons, I was inclined to implement this services with
pjsip_wizard.conf and I'm realizing
Hello,
Today, one Asterisk instance of mine crashed.
This instance is only providing SIP trunking (from IPBXs to carriers, no
transcoding, playing of voice prompts and fancy dialplan tricks, ).
The instance is built :
- as a VMWare 6.5 guest,
- with Debian Stretch (9)
- and asterisk 13.20.0
Hello,
I'm working on an Asterisk active/passive cluster where the following
applies:
- members are both VM
- /etc/asterisk files are copied from one provisonning server to both VM
- asterisk is running on active member
- asterisk is not running on passive member
- members share floating IP
sfull call out of 700 but
I could succeed to get, even once, 700 successfull calls, even when I tried
with my current 40 maxfiles limit.
2018-03-06 23:35 GMT+01:00 Bruce Ferrell <bferr...@baywinds.org>:
>
> On 03/06/2018 01:58 PM, Olivier wrote:
>
>> Hello,
>>
>&g
Hello,
I'm running load testing sessions.
My System Under Test is an asterisk 13 with 16GB, configured with maxfiles
set to 400 000.
This system is supposed do produce simple SIP trunking services without
transcoding.
The box sending call to my System Under Test is anabled with SIPp.
I'm
Hello,
I'm currently trying to configure a passive Asterisk instance that must
backup an active Asterisk instance.
Each instance is connected this way:
PSTN <---> Gateway <-- SIP --> Asterisk <-- SIP --> endpoints or IPBXs
Most endpoints connect through registration.
With chan_sip, Asterisk
Hello,
Digging a bit further, having a local cdr_custom CSV seems to make
updatings work !
I did have enough time to properly test this and become more affirmative
but it seems to depend on active CDR backend;
2018-02-21 22:19 GMT+01:00 Olivier <oza.4...@gmail.com>:
> As a complem
it also apply to data later found in cdr-scv/Master.csv
file ?
2018-02-22 18:14 GMT+01:00 Richard Mudgett <rmudg...@digium.com>:
>
>
> On Thu, Feb 22, 2018 at 5:23 AM, Olivier <oza.4...@gmail.com> wrote:
>
>> Hello,
>>
>> I'm load testing a new Ast
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