On Monday 04 December 2023 at 13:39:51, Joshua C. Colp wrote:
> The mailing list will not receive emails from the forums. What I was
> referring to is that Discourse does provide the ability to receive emails
> for posts or threads you're interested in, and you are able to respond over
> email to
On Monday 20 November 2023 at 12:14:11, Tahir Almas Dhesi wrote:
> Interested to know good wholesale SIP providers for 15k concurrent calls
You might want to specify a bit more detail, such as:
- which country are you located in
- do you require inbound DDIs (if so, in which region/s)?
-
On Thursday 26 October 2023 at 19:11:45, Carlos Chavez wrote:
> Does anyone know of a good solution to integrate Asterisk and MS
> Teams? Something where you can use the MS Teams client as a regular
> extension?
Kamailio is the usual intermediary I have seen for doing this.
Antony.
--
On Monday 09 October 2023 at 21:05:55, Mike Diehl wrote:
> Hi all,
>
> I need to be able to delete a voicemail message using a program.
>
> Is is sufficient to simply delete the .wav and .txt files in the spool
> directory? Or do I need to also renumber the remaining files?
My approach in a
On Tuesday 11 July 2023 at 10:00:22, Fourhundred Thecat wrote:
> Hello,
>
> my asterisk is working fine, I am just confused why, on the server I see
> private IP address of an endpoint
SIP is rather like FTP in that it embeds IP addresses (layer 3 of the OSI
network model) in the protocol
On Wednesday 21 June 2023 at 17:52:16, TTT wrote:
> Ok I've got multiple phone sets registered with the same extension/secret.
>
> However, this causes a strange problem. If I have 3 phone sets registered
> on extension 123, and I then call extension 123 (from extension 456), only
> a SINGLE
On Monday 19 June 2023 at 16:26:05, TTT wrote:
> That begs another interesting question...with analog phones picking up two
> extensions on the same "line" allow multiple people to participate on the
> call (without a "conference" feature)
>
> Does this become possible with multiple phones on
On Monday 19 June 2023 at 15:09:44, TTT wrote:
> I am creating a dialplan where a single user (Alice) has two offices. Both
> of her phones should ring if her extension is called.
>
> I could use a ring group, but I'm wondering can both phones use the same
> PJSIP extension account
On Thursday 11 May 2023 at 21:15:50, Jerry Geis wrote:
> I have 4 devices that I connect here local and there is no issue.
> I have those same 4 devices connecting from another location across the
> internet.
>
> They all boot up, connect and register I can send audio to them and they
> play.
>
On Thursday 06 April 2023 at 18:29:43, Jeff LaCoursiere wrote:
> If you just want something easy to use out of the box, install the
> FreePBX distro.
Given that Steve originally said "I've been using Asterisk, including
administering and maintaining it, in some aspect since 2003, but this is
On Thursday 06 April 2023 at 15:48:24, Steve Matzura wrote:
> this is the first time I have attempted a
> from-scratch installation and setup on my own.
..
> Then the weeds started to appear, and I was off into them.
>
> The first was the mention of Alembic.
> Reading on, I found this,
? That's connected to MariaDB.
> On Mon, Feb 20, 2023 at 11:12 PM Antony Stone wrote:
> > Hi.
> >
> > I have a strange problem and I'm looking for suggestions on how to
> > investigate it.
> >
> > I have a dialplan which is processing a call, and Asterisk s
Hi.
I have a strange problem and I'm looking for suggestions on how to investigate
it.
I have a dialplan which is processing a call, and Asterisk simply stops doing
anything for that call.
I have verbose and debug logging turned on.
There are two steps at a particular point in the dialplan:
On Thursday 26 January 2023 at 21:58:45, Sean Bright wrote:
> On 1/26/2023 5:16 AM, Antony Stone wrote:
> > It does not work if it's written in AEL - assigning global variables
> > works, but the above does not.
>
> I've created a JIRA issue[1] for this as well as a proposed
On Wednesday 25 January 2023 at 19:17:04, Daniel wrote:
> Asterisk 20.1.0
>
> [globals]
> Sphones=SIP/SYealinkT38G/SGC610IP
> Kphones=SIP/KC470IP/KSnom870
> Allphones=${Sphones}&${Kphones}
>
> -s*CLI> dialplan show globals
> Allphones=SIP/KC470IP/KSnom870/SYealinkT38G/SGC610IP
>
On Tuesday 24 January 2023 at 18:03:58, Joel Serrano wrote:
> I believe that EVAL might be able to help you here:
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_EVAL
>
> Example:
>
> Allphones=${EVAL(Kphones)}&${EVAL(Sphones)}
>
> I'm not sure if in the globals it
On Wednesday 25 January 2023 at 16:46:14, Daniel wrote:
On Sunday 01 January 2023 at 17:30:03, Antony Stone wrote:
> > The [globals] section of that dialplan includes:
> >
> > Kphones=SIP/KC470IP/KSnom870
> > Sphones=SIP/SYealinkT38G/SGC610IP
> >
On Wednesday 25 January 2023 at 00:38:25, John Novack wrote:
> You have posted the same message several times in the last few days!!
I think it has become clear that I did this not because I was getting no
answers, but because my question was not appearing on the list.
> I would assume no one
On Monday 16 January 2023 at 13:08:50, marek wrote:
> there are new versions of Asterisk but mailing list is empty
>
> http://lists.digium.com/pipermail/asterisk-users/
There's also something very odd going on with incoming messages.
I sent one on January 1st - arrived January 23rd
Another
Hi.
I have a very old dialplan (ie: a dialplan for a very old version of Asterisk)
which I've just transferred to Asterisk 16.28.0
The [globals] section of that dialplan includes:
Kphones=SIP/KC470IP/KSnom870
Sphones=SIP/SYealinkT38G/SGC610IP
Hi.
I have a very old dialplan (ie: a dialplan for a very old version of Asterisk)
which I've just transferred to Asterisk 16.28.0
The [globals] section of that dialplan includes:
Kphones=SIP/KC470IP/KSnom870
Sphones=SIP/SYealinkT38G/SGC610IP
Hi.
I have a very old dialplan (ie: a dialplan for a very old version of Asterisk)
which I've just transferred to Asterisk 16.28.0
The [globals] section of that dialplan includes:
Kphones=SIP/KC470IP/KSnom870
Sphones=SIP/SYealinkT38G/SGC610IP
Hi.
I have a very old dialplan (ie: a dialplan for a very old version of Asterisk)
which I've just transferred to Asterisk 16.28.0
The [globals] section of that dialplan includes:
Kphones=SIP/KC470IP/KSnom870
Sphones=SIP/SYealinkT38G/SGC610IP
On Friday 11 November 2022 at 17:11:26, Joshua C. Colp wrote:
> On Fri, Nov 11, 2022 at 12:09 PM Antony Stone wrote:
> >
> > https://wiki.asterisk.org/wiki/display/AST/Application_Answer tells me
> > that the Answer() application takes an optional parameter which causes
On Friday 11 November 2022 at 17:08:42, Antony Stone wrote:
> Hi.
>
> Asterisk 16.2.1
>
> I have a dialplan where one context (named "inbound") performs:
>
> Originate(Local/${Target}@inOrig,exten,inbound,${EXTEN},208)
>
> The idea is that this comma
Hi.
Asterisk 16.2.1
I have a dialplan where one context (named "inbound") performs:
Originate(Local/${Target}@inOrig,exten,inbound,${EXTEN},208)
The idea is that this command will spawn a "call" to the context "inOrig" on
the same machine, and then return to the "inbound" context at priority
On Thursday 06 October 2022 at 15:24:22, Jerry Geis wrote:
> I added:
>
> externip=xxx
> nat=force_rport,comedia
>
> to the general section of sip.conf
>
> its still sending to the local IP.
Does your local router (the one connecting Linphone to the Internet) have a
"SIP helper" or "SIP ALG"
On Monday 03 October 2022 at 14:14:54, Joshua C. Colp wrote:
> On Mon, Oct 3, 2022 at 9:11 AM Antony Stone <
>
> antony.st...@asterisk.open.source.it> wrote:
> > Hi.
> >
> > I have a dialplan which calls the VoiceMail() application, and I'm
&g
Hi.
I have a dialplan which calls the VoiceMail() application, and I'm getting the
following behaviour:
- if the inbound caller leaves a message, then presses #, and then presses 1
to accept the recording, everything works as expected and the dialplan
continues processing after the line
On Wednesday 21 September 2022 at 16:20:19, Jerry Geis wrote:
> hi All
>
> How do I restart logging in /var/log/asterisk/messages ?
logger reload
Antony.
--
What do you call a dinosaur with only one eye? A Doyouthinkesaurus.
Please reply
On Tuesday 13 September 2022 at 15:52:43, Joshua C. Colp wrote:
> On Tue, Sep 13, 2022 at 11:49 AM Antony Stone wrote:
> > However, if the calleE hangs up instead, no hangup extension is called at
> > all, so my dialplan cannot tell that the call has ended (which is
> > impo
Hi.
I have a dialplan which accepts an inbound call and dials out to another
number, automatically bridging the channels together when the second call is
answered.
I then have a facility for the caller to put the call on hold (which uses
ChannelRedirect() in the dialplan to play music on hold
On Wednesday 07 September 2022 at 15:32:50, Thomas Ray wrote:
> From https://wiki.asterisk.org/wiki/display/AST/Channels
>
> "The primary exception is with Local Channels. In the case of local
> channels, you'll typically have two local channel legs, one that is
> treated as outbound and the
On Wednesday 07 September 2022 at 15:21:59, Joshua C. Colp wrote:
> On Wed, Sep 7, 2022 at 11:17 AM Antony Stone wrote:
> >
> > This is a follow-up to an email I posted earlier today to the list,
> There's nothing in the moderator queue that I can see.
Thanks, sent agai
Hi.
I'm trying to deal with a problem regarding putting a call on hold and then
later resuming it. I am using chan_sip throughout, and Asterisk 16.
I have two scenarios:
First (works):
1. An inbound call arrives, the dialplan does not Answer() it.
2. The dialplan performs a Dial() to an
On Wednesday 07 September 2022 at 11:44:54, Antony Stone wrote:
> Hi.
This is a follow-up to an email I posted earlier today to the list, although I
haven't seen it come back yet. If it's under moderation for some reason, I
hope some kindly admin will release it :)
> I'm trying t
On Wednesday 31 August 2022 at 15:01:57, Mark Murawski wrote:
> On 8/31/22 05:29, Antony Stone wrote:
> >
> > I realise that a better solution might be to wrap assignments (inside
> > Set() or MSet(), no matter) with $[..] *only* if the expressions contain
>
Replying to list this time...
On Wednesday 31 August 2022 at 12:47:13, aster...@phreaknet.org wrote:
> On 8/31/2022 6:32 AM, Antony Stone wrote:
> > Hi.
> >
> > I think I've discovered a bug in either the implementation or the
> > documentation of the AEL swi
Hi.
I think I've discovered a bug in either the implementation of the
documentation of the AEL switch command.
https://wiki.asterisk.org/wiki/display/AST/AEL+Conditionals gives an example
of using switch, and states at the bottom:
"Neither the switch nor case values are wrapped in $[ ]; they
On Tuesday 30 August 2022 at 23:51:34, Mark Murawski wrote:
> On 8/30/22 12:34, Antony Stone wrote:
> >>> Tracker=${CDR(uniqueid)};
> >>>
> >>> results in:
> >>> MSet(Tracker=-1661872057.2349)
> >>>
> >>> sys
On Tuesday 30 August 2022 at 18:17:08, Mark Murawski wrote:
> On 8/30/22 11:16, Antony Stone wrote:
> > If I write in my AEL dialplan:
> > Set(Tracker=${CDR(uniqueid)});
> >
> > this results in executing:
> > Set(Tracker=eagle.domain.com-166187205
On Monday 29 August 2022 at 16:29:42, Antony Stone wrote:
> On Monday 29 August 2022 at 16:19:17, Mark Murawski wrote:
>
> > what specific situation prevents you from using a=1; style syntax? Why are
> > you feeling the need to use Set(a=1) instead of a=1. What are specific
On Monday 29 August 2022 at 17:00:23, Mark Murawski wrote:
> On 8/29/22 09:30, Antony Stone wrote:
> > It is, although there are ways I think it can be improved - I'm wondering
> > how best to go about proposing these.
> >
> > The most obvious for now are:
> >
On Monday 29 August 2022 at 16:19:17, Mark Murawski wrote:
> On 8/29/22 10:15, Antony Stone wrote:
> >> But! What specific reason do you have for wanting Set() instead of
> >> MSet() for all assignments that can't be otherwise just written as an
> >> in-line
On Monday 29 August 2022 at 15:35:09, Joshua C. Colp wrote:
> MSet is not deprecated.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_MSet
includes the sentence "MSet behaves in a similar fashion to the way Set worked
in 1.2/1.4 and is thus prone to doing things that you may
On Monday 29 August 2022 at 14:51:27, Mark Murawski wrote:
> On 8/29/22 08:48, Mark Murawski wrote:
> >
> > Hi Antony,
> >
> > I love to hear about AEL use-cases. I'm happy that AEL is working out
> > for you.
It is, although there are ways I think it can be improved - I'm wondering how
best
Hi.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Application_Originate
I need to use Originate() in a dialplan, pointing to another location in the
same extension of the same context, so for example:
Originate(Local/${Dest}@Dialout,exten,${CONTEXT},${EXTEN},158);
I don't seem to be
On Monday 22 August 2022 at 18:55:08, List Support wrote:
> Hi
>
> Le 22/08/2022 à 17:16, Antony Stone a écrit :
> > Hi.
> >
> > Is there any way to find out the values of variables set in the [General]
> > section of extensions.conf from the Asterisk CLI (no
Hi.
Is there any way to find out the values of variables set in the [General]
section of extensions.conf from the Asterisk CLI (not from inside the
dialplan, I just mean at the "hostname*CLI>" prompt)?
https://www.voip-info.org/asterisk-dialplan-general/
Thanks,
Antony.
--
If you can
On Tuesday 05 July 2022 at 11:30:35, Joshua C. Colp wrote:
> Kia ora,
>
> Kind of a random email here but thought I'd remind everyone of the
> community forums at https://community.asterisk.org/ which see more activity
> than the mailing list. If you've got questions/issues, you may find an
>
Hi.
I'm wondering where the current documentation for AEL is.
I've found https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=4620445
but there's hardly any content there and it's over 10 years old.
Where should I be looking, please?
Antony.
--
I don't know, maybe if we all waited
On Wednesday 25 May 2022 at 16:54:43, aster...@phreaknet.org wrote:
> On 5/25/2022 10:41 AM, Antony Stone wrote:
> > On Wednesday 25 May 2022 at 15:27:38, aster...@phreaknet.org wrote:
> >>
> >> If I want to log something from the dialplan, I generally send it
On Wednesday 25 May 2022 at 15:27:38, aster...@phreaknet.org wrote:
> On 5/25/2022 8:11 AM, Antony Stone wrote:
> > On Tuesday 24 May 2022 at 01:12:46, Kevin Harwell wrote:
> >> So this turned out more complicated than I originally thought!
> >
>
On Tuesday 24 May 2022 at 01:12:46, Kevin Harwell wrote:
> So this turned out more complicated than I originally thought!
Wow, thank you very much for:
a) such a comprehensive answer
b) confirming my findings
c) most of all, working out why and how all this stuff works (or, perhaps,
Hi.
Does no-one else know either? I thought this was a simple question, and it
was just me being unable to find the appropriate documentation to explain how
these logging levels work.
Please, can anyone help?
On Friday 20 May 2022 at 15:33:45, Antony Stone wrote:
> Hi.
>
> I
On Friday 20 May 2022 at 15:33:45, Antony Stone wrote:
> Hi.
>
> I'm trying to use different logging verbosity levels to get dialplan output
> into different log files, and there's clearly something I haven't
> understood about how Asterisk does this...
>
>
> I ha
Hi.
I'm trying to use different logging verbosity levels to get dialplan output
into different log files, and there's clearly something I haven't understood
about how Asterisk does this...
I have the following in /etc/asterisk/logger.conf:
[logfiles]
logtest.verbose.0 => verbose(0)
> -Original Message-
> From: asterisk-users On Behalf Of
> Antony Stone Sent: Wednesday, March 16, 2022 8:20 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Decimal
> seconds?
>
> Hi.
>
> Has nobody got a cl
Hi.
Has nobody got a clue for me about this?
It must be possible somehow, otherwise the %3q parameter wouldn't exist...
On Friday 11 March 2022 at 17:31:54, Antony Stone wrote:
> Hi.
>
> I'm looking at
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_STRFTIME
Hi.
I'm looking at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_STRFTIME and
trying to work out how to obtain an Epoch timestamp for "now" containing
fractional / decimal seconds so that the %3q format parameter works.
On Friday 28 January 2022 at 02:43:17, John Covici wrote:
> I have been using system commands in my dialplan for years and the &
> goes through and puts the process in background like it should,
> asterisk does not do anything, so you are left with what the shell
> does.
That's completely
On Thursday 27 January 2022 at 21:31:35, Kingsley Tart wrote:
> Does asterisk follow HTTP redirects? If so can you use something like
> tinyurl.com to produce an alternative URL?
I'm (pretty) sure that that would work.
The other similar idea I had was to use a reverse proxy server to accept an
On Thursday 13 January 2022 at 15:45:02, Jerry Geis wrote:
> > Hi Josh
> >
> >chan_sip did not add a video stream. What is the actual configuration for
> >
> > it? What is the actual call file used for it?
>
> sip.conf has videosupport in the general section.
>
> I did find that where I am
On Tuesday 11 January 2022 at 17:20:44, Michael Englehorn wrote:
> If you're on RHEL or CentOS or one of its descendants,
Oh, now that reminds me that those systems also tend to alias "rm" to "rm -i",
so they won't delete files without confirmation.
Irritating in general IMHO, but it might be
On Monday 10 January 2022 at 20:03:55, Jerry Geis wrote:
> I am trying to run this command:
> exten => _4XX,n,System(/usr/bin/rm /tmp/test.incoming.txt)
>
> From the log:
> Executing [402@smvoice-sip:7] System("SIP/103-0018", "/usr/bin/rm
> /tmp/test.incoming.txt") in new stack
>
>
> Is
On Sunday 09 January 2022 at 00:50:27, John Covici wrote:
> Hi. I am using asterisk 18.3 and freepbx.
Hm, which version of FreePBX uses Asterisk 18.3?
> How can both sip and pjsip be listening at port 5060 at the same time
They can't.
One might be on TCP and the other on UDP, but you can't
On Friday 31 December 2021 at 15:54:01, Luca Bertoncello wrote:
> Am 31.12.2021 um 14:39 schrieb Antony Stone:
>
> Hi Antony
>
> >> Last very strange problem is, that the list of missed calls on the phone
> >> is always empty...
> >
> > Check
On Friday 31 December 2021 at 12:29:19, Luca Bertoncello wrote:
> Am 28.12.2021 um 21:21 schrieb Antony Stone:
>
> Hi
>
> > However, at least you've got as far as ruling out Telekom as being the
> > source of the problem, which I think is good.
>
> So, I se
On Tuesday 28 December 2021 at 20:39:37, Luca Bertoncello wrote:
> After about 6 seconds I get from the Telekom:
>
> Via: SIP/2.0/UDP
> 87.191.224.158:5060;received=87.191.224.158;rport=5060;branch=z9hG4bKPj43be
> 873a-cf55-4348-8867-5c2bb97bd76a
> To: ;
>
On Tuesday 28 December 2021 at 20:07:22, Luca Bertoncello wrote:
> Am 28.12.2021 um 20:00 schrieb Antony Stone:
> >
> > From your earlier packet capture, it looked to me like you were dialling
> > an external number from an internal telephone.
>
> This is correct!
On Tuesday 28 December 2021 at 19:52:46, Luca Bertoncello wrote:
> Am 28.12.2021 um 19:41 schrieb Antony Stone:
>
> Hi Antony,
>
> > Okay, so, returning to my question, do you see any difference between the
> > packet inbound to Asterisk from the called telephone, and
On Tuesday 28 December 2021 at 18:17:00, Luca Bertoncello wrote:
> Am 28.12.2021 um 17:35 schrieb Antony Stone:
> >
> > Where exactly were those packets captured?
>
> tcpdump on the Asterisk-Server on the interface of the VLAN for the phones.
> All traffic captured.
O
On Tuesday 28 December 2021 at 17:28:47, Luca Bertoncello wrote:
> Am 28.12.2021 um 17:22 schrieb Antony Stone:
>
> Hi Antony
>
> > I mean the response from the called telephone in reply to the INVITE,
> > which contains the SIP code "180 Ringing" and may
On Tuesday 28 December 2021 at 16:58:01, Luca Bertoncello wrote:
> Am 28.12.2021 um 15:42 schrieb Antony Stone:
>
> Hi Antony,
>
> > Sounds like something strange is happening with Remote-Party-ID.>
> > Do a packet capture and see whether the 180 response from the
On Tuesday 28 December 2021 at 14:30:17, Luca Bertoncello wrote:
> I have a Debian Server with Asterisk 16.2.1 from Debian repos and some
> SNOM phones (SNOM 821, last firmware snom821-SIP 8.7.5.35).
> If I call a number I can see in the display the called number, after a
> few seconds the
On Thursday 23 December 2021 at 22:16:26, John Harragin wrote:
> gotoif accomplishes exactly what you want (except the one line part).
Goto() and GotoIf() always remind me of programming in BASIC in the 1980s.
Antony.
--
"In fact I wanted to be John Cleese and it took me some time to realise
On Thursday 23 December 2021 at 18:31:38, aster...@phreaknet.org wrote:
> > -Original Message-
> > From: asterisk-users On Behalf
> > Of Dovid Bender
> > Sent: Thursday, December 23, 2021 12:11 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion > us...@lists.digium.com>
On Wednesday 01 December 2021 at 22:43:47, Kingsley Tart wrote:
> On Wed, 2021-12-01 at 21:49 +0100, Antony Stone wrote:
> >
> > What is the exact "complaint"?
> [Nov 29 16:44:08] ERROR[25803] pjproject: tlsc0x7f1c74246778 RFC
> 5922 (section 7
On Wednesday 01 December 2021 at 21:39:52, Kingsley Tart wrote:
> Hi,
>
> I can't get Asterisk to send a SIP call to Twilio over TLS because it
> complains about Twilio's wildcard certificate.
What is the exact "complaint"?
> Is there a way round this?
Maybe, once we know what the error
On Friday 12 November 2021 at 23:34:25, Steve Edwards wrote:
> On Fri, 12 Nov 2021, Antony Stone wrote:
> > I've never used AGI, so what would your suggested solution involve?
>
> If all you need is to update/insert/delete some rows in a database, ODBC
> could be a solution.
On Friday 12 November 2021 at 18:08:11, aster...@phreaknet.org wrote:
> On 11/12/2021 12:39 PM, Antony Stone wrote:
> > On Friday 12 November 2021 at 17:36:07, Eric Wieling wrote:
> >> Create a spool file from the 'h' extension to generate the call.
> >
> > Yes, I t
On Friday 12 November 2021 at 18:18:03, Eric Wieling wrote:
> On 11/12/21 12:39, Antony Stone wrote:
> > On Friday 12 November 2021 at 17:36:07, Eric Wieling wrote:
> >> Create a spool file from the 'h' extension to generate the call.
> >
> > Yes, I thought of th
On Friday 12 November 2021 at 17:36:07, Eric Wieling wrote:
> Create a spool file from the 'h' extension to generate the call.
Yes, I thought of that, but it somehow feels a bit clunky, and was hoping for
a neater solution :)
Antony.
--
Software development can be quick, high quality, or
On Friday 12 November 2021 at 17:20:39, Frank Vanoni wrote:
> On Fri, 2021-11-12 at 16:56 +0000, Antony Stone wrote:
> > I use Dial() commands with custom SIP headers to pass information
> > (eg: about the current state of a call) between the front-end and back-end
> > mac
Hi.
I have a setup which comprises some "front-end" Asterisk servers which have
SIP trunks to external providers, and very simple dial plans, and some "back-
end" servers which only talk to the front-end machines, and have the majority
of my dialplan logic on them.
I use Dial() commands with
On Thursday 11 November 2021 at 22:29:34, David Cunningham wrote:
> Hello,
>
> We have a commercial client
> If anyone has ideas for other places to advertise this request let me know!
I would suggest http://lists.digium.com/mailman/listinfo/asterisk-biz because
that is the commercial list
On Wednesday 03 November 2021 at 21:29:46, Luca Bertoncello wrote:
> I tried so:
>
> exten => h,n(hang),Gosub(noanswer,s,1)
The n there should be 1, surely?
> exten => h,n,Hangup
I would say "remove that line". The call has already been hung up, so calling
Hangup is at best going to go into
On Thursday 14 October 2021 at 19:22:00, hw wrote:
> Hi,
>
> when asterisk registers with the VOIP provider via ipv6 and when
> local phones don't work with ipv6 but only with ipv4, am I to
> expect issues?
Do a SIP packet capture and see what the SDP in the INVITE is telling each end
to
Hi.
I'm using Asterisk 16 with a MySQL realtime DB, containing both outbound
registrations to other PBXs (Asterisk as SIP client) and inbound accounts for
clients to register to (Asterisk as SIP server).
All in all, working well.
However, I just had a requirement to register outbound to a PBX
On Thursday 09 September 2021 at 17:56:10, Marek Greško wrote:
> Hello,
>
> I would not like to open whole range of udp ports for rtp.
Why not? What is the risk?
What would possibly be listening on UDP ports 1 - 2 (the Asterisk
default range) which an external scanner / attacker
On Monday 06 September 2021 at 23:05:27, Duncan Turnbull wrote:
> > On 7/09/2021, at 8:30 AM, Marek Greško wrote:
> >
> > Hello,
> >
> > it is only local nftables with nf_conntrack_sip on the asterisk
> > server. Probably a kernel bug? It did not trigger with previous
> > providers since they
On Sunday 05 September 2021 at 00:54:10, Marek Greško wrote:
> the local provider's router does not possess any ipv4 address on the
> external interface, only ipv6.
So, what do the two addresses which you have labelled in the packet captures
as 198.51.100.1 and 192.0.2.2 correspond to?
I see
On Thursday 08 July 2021 at 20:57:30, Marek Greško wrote:
> Hello,
>
> I have an asterisk setup using pjsip. Everything used to work
> correctly until one remote site changed internet provider and thier
> router does not support sip protocol algorithms.
I'm slightly bothered by the word
On Sunday 05 September 2021 at 00:19:41, Marek Greško wrote:
> Hello,
>
> could you please answer my previous question about anonymizing several
> parameters? I have the data ready, but will post after answer. I have
> no clue whether I could disclose some important data not deleting
> them.
On Saturday 04 September 2021 at 22:13:32, Marek Greško wrote:
> Hello,
>
> I agree my knowledge of SIP itself is poor, but I have quite well
> general tcp/ip understanding. What sip parameters should be
> anonymized? How about tag, branch, call-id, cseq values?
Show us your packet captures
On Saturday 04 September 2021 at 00:34:49, Duncan Turnbull wrote:
> > On 4/09/2021, at 7:53 AM, Marek Greško wrote:
> >
> > So you suspect something is messing up SIP protocol? Maybe the phone
> > itself is not working properly. The phone itself is not aware of the
> > internet address, so is
On Friday 20 August 2021 at 19:06:09, George Joseph wrote:
> On Fri, Aug 20, 2021 at 8:33 AM Antony Stone wrote:
> >
> > So, if I have Asterisk registered as a SIP client to some remote server,
> > how can I get Asterisk to tell that remote server to put the call on hold
On Friday 20 August 2021 at 16:14:44, George Joseph wrote:
> On Wed, Aug 18, 2021 at 3:33 AM Antony Stone wrote:
> > Hi.
> >
> > Just to summarise: I have a SIP client talking to a SIP server, and I
> > need something which can send commands to that server to put ca
On Wednesday 18 August 2021 at 16:47:35, d...@donkelly.biz wrote:
> I think I would start by finding an open source SIP client that can manage
> calls like you want,
I can certainly find those.
> then figure out how to divide the control and audio responsibilities between
> these two SIP
Hi.
I wonder if anyone has some helpful advice or suggestions for me?
I have a very basic SIP client application, which can make and receive phone
calls, and that's about it. Regard it as a pretty dumb softphone.
Unfortunately I cannot change it for a smarter one.
This client is talking to
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