Hi,
I've looked around the archives and have spent a while on voip-info.org
but not found an answer so forgive me if this is in a FAQ somewhere.
We've got several Asterisk servers with E1 cards (some Digium, some
Sangoma). We provide non geographic numbers for customers and route
calls to their
Hi,
I'm trying to receive faxes using hylafax / iaxmodem but I just can't
get it to work. We're using Sangoma E1 cards and have calls coming in
over PSTN. I've tried turning hardware echo cancellation off but it
makes no difference. This is what I get in /var/spool/hylafax/log:
[r...@faxhost
On Thu, 2010-01-14 at 15:25 +, Jeff LaCoursiere wrote:
Actually it is fairly clear that his dialplan is correctly routing the
calls to iaxmodem, and that iaxmodem is simply not completing the
training. I would say that the fax machine you are testing with is either
on a horribly noisy
On Sun, 2010-01-17 at 23:28 -0800, Lee Howard wrote:
Kingsley Tart wrote:
Jan 14 12:44:49.39: [ 3403]: -- [9:AT+FRH=3\r]
Jan 14 12:44:56.39: [ 3403]: -- [0:]
Jan 14 12:44:56.39: [ 3403]: MODEM Empty line
Jan 14 12:44:56.39: [ 3403]: MODEM TIMEOUT: waiting for v.21 carrier
Jan 14 12:44
On Mon, 2010-01-18 at 07:03 -0500, Doug Lytle wrote:
Kingsley Tart wrote:
Do you know what I should look at next, or how to get more diagnostics
somehow?
Record the fax using the record option in your iaxmodem config file.
The files will be put into the /tmp or /root/tmp folder
On Wed, 2010-01-20 at 23:41 +0100, Michiel van Baak wrote:
Forget about virtualization!
This system is running linux as base os (I conclude by the tone of your
mail)
Just install asterisk on it besides the monitoring software and be done
with it.
What do you gain by running virtualisation on
Hi,
Does anyone know what it means when I've got an incoming fax routed
through to iaxmodem+hylafax and then I see this in the asterisk log:
DEBUG[18902] chan_dahdi.c: Detected digit 'f'
This happens just after the initial fax negotiation has started and
seems to correspond with the sending fax
On Mon, 2010-01-25 at 07:50 -0800, Lee Howard wrote:
Kingsley Tart wrote:
DEBUG[18902] chan_dahdi.c: Detected digit 'f'
This happens just after the initial fax negotiation has started and
seems to correspond with the sending fax machine giving up.
Turn off fax detection.
Hi Lee
On Mon, 2010-01-25 at 05:49 -0500, Doug Lytle wrote:
Kingsley Tart wrote:
Hi,
Does anyone know what it means when I've got an incoming fax routed
through to iaxmodem+hylafax and then I see this in the asterisk log:
DEBUG[18902] chan_dahdi.c: Detected digit 'f'
This may
On Tue, 2010-01-26 at 07:46 -0500, Doug Lytle wrote:
Kingsley Tart wrote:
Thanks for the link. I looked at that page but couldn't see how it
helped with my specific issue, unfortunately, though I admit I'm fairly
new to asterisk so I don't fully understand what's going
On Tue, 2010-01-26 at 13:17 -0600, Kevin P. Fleming wrote:
Jeff Brower wrote:
How do you know for sure fax detection is turned off? It sounds to me like
your changes to the dahdi config file are
being ignored. Maybe put something in there that should cause an error or
something
On Wed, 2010-04-28 at 11:07 -0500, Danny Nicholas wrote:
FWIW, I would take your STDERR references and give them another handle,
since you're not really trying to produce a CLI/Console output.
The symptoms you have described in this thread are 100% compliant with AGI
protocol violation
On Wed, 2010-04-28 at 11:47 -0400, Fred Posner wrote:
For a AGI that is called repeatedly, maybe you should consider
implementing it in a compiled language.
You can execute XXX AGIs written in C in the time it takes to load the
Perl interpreter and parse your script.
Yes agreed but I
Hi,
I still think we've either got a bug in Asterisk or a bug in the
Asterisk::AGI module.
In a separate part of the dialplan we have a call to a (much simpler)
script that begins with the below code.
In the last 1000 calls, I've had a couple of extension not returned by
AGI errors from the
On Mon, 2010-05-24 at 15:09 +0200, Sasa wrote:
HI, I have in 'inbound route' a IVR, with press 1 or 2 the destination call
is always a ring group called '600', my problem is that after press 1 (but
this problem is present also with press 2) before that the inbound call is
transfer to
On Mon, 2010-05-24 at 14:41 +0100, Kingsley Tart wrote:
On Mon, 2010-05-24 at 15:09 +0200, Sasa wrote:
HI, I have in 'inbound route' a IVR, with press 1 or 2 the destination call
is always a ring group called '600', my problem is that after press 1 (but
this problem is present also
Hi,
We're trying to time-limit some calls by specifying L(x:y:z) as an
option to the Dial command.
If we set the limit to a fairly short duration (eg 120 seconds) then
Asterisk seems to issue the hangup at about the right time.
However, for longish calls we're seeing quite a bit of overspill.
On Thu, 2011-11-03 at 18:50 +0530, amit anand wrote:
Hi you can use Absoulte timeout to set the time limit feature for the channel
Hi,
Thanks for the suggestion. It's good to know that absolute timeout
exists (I'd not noticed that before). However, it won't help here
because we're setting up
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kingsley Tart
Sent: Thursday, November 03, 2011 5:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] duration limits in Dial() not being enforced at
correct time
Hi,
We're
-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kingsley
Tart
Sent: Thursday, November 03, 2011 5:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] duration limits in Dial() not being enforced
at
correct
Hi,
I'm not sure whether this is possible but if it is, I'm sure someone on
here might know ...
Is it possible to use Monitor() to record a conversation[1], but make it
start a new pair of wav files at intervals (eg every 15 minutes) if the
calls go on for a long time?
We already have this
...@lists.digium.com] On Behalf Of Kingsley Tart
Sent: Monday, November 14, 2011 7:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Monitor() - splitting long calls into several
sound files
Hi,
I'm not sure whether this is possible but if it is, I'm
Hi,
We're using it here. As Ido asked, is there an alternative way of
getting the SIP response in the event a Dial() fails?
Cheers,
Kingsley.
On Thu, 2011-08-18 at 07:42 -0500, Matthew Nicholson wrote:
Greetings,
Recently a performance regression in chan_sip was discovered in Asterisk
1.8.
What sort of problem with Read() are you expecting to encounter, and
what do you mean by keep going?
Cheers,
Kingsley.
On Thu, 2011-11-17 at 10:10 -0600, Danny Nicholas wrote:
Hello again list,
Did the following: (on 1.4.42
installation)
asterisk -rx core
because Read() failed and didn't
say anything. Perfection doesn't seem necessary until somebody complains
because you don't have it.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kingsley Tart
Sent
Hi,
Did you get a workaround for this? I sent you a message offlist but you
didn't reply so I don't know whether you saw it.
Cheers,
Kingsley.
On Fri, 2011-11-18 at 13:15 -0600, Danny Nicholas wrote:
My IVR wouldn't sound right if I allowed 2 or 3 times before it was
considered a failure.
We do that with the F option in Dial().
From http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial :
F(context^exten^pri): When the caller hangs up, transfer the called
party to the specified context and extension and continue execution.
Cheers,
Kingsley.
On Mon, 2011-11-21 at 17:38 +1100,
require starting a new AGI.
On 21 November 2011 22:22, Kingsley Tart kings...@skymarket.co.uk
wrote:
We do that with the F option in Dial().
From http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial :
F(context^exten^pri): When the caller
Of Kingsley Tart
Sent: Monday, November 21, 2011 7:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Continue AGI after Dial() following caller
hang up?
Yeah I think I slightly misread your original question, which I realised
when I saw Thorsten's
, and for some reason
the AGI always exits when the caller hangs up - even when I set HUP to
IGNORE. If I set HUP to a subroutine that just logs a message, that
message is never logged.
Thanks for all the help.
On 22 November 2011 05:23, Kingsley Tart kings...@skymarket.co.uk
wrote
Alternatively, if you don't have that extension defined anywhere,
Asterisk will jump to the i extension, where you can then read the
actual entered digits from the INVALID_EXTEN variable and jump back to
the main part of the dialplan.
Note that if they enter digits that *could* match a defined
Of Kingsley Tart
Sent: Saturday, November 19, 2011 4:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question about Read() application
Hi,
Did you get a workaround for this? I sent you a message offlist but you
didn't reply so I don't know whether you
Hi. Aside from converting spaces to plus signs, you don't encode any
special characters before putting them in the URL. It might be safer to
run $line through some sort of encoding before calling Google with it,
even if most special characters probably don't result in any sound.
Google say and if
Hi,
I want to create a system for incoming calls where, under some
circumstances, callers get routed straight to voicemail (or some other
means of recording a message) but if they enter a valid extension number
then the recorded message would be abandoned and they'd be diverted to
the extension
Hi,
I'm not sure what you mean. Can you elaborate?
Cheers,
Kingsley.
On Thu, 2012-02-02 at 18:13 +0530, virendra bhati wrote:
You may used even capturing in the case... when call is recoding in
conference
On Wed, Feb 1, 2012 at 4:04 PM, Kingsley Tart
kings...@skymarket.co.uk wrote
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.
> &
Hi,
I'm using Asterisk 18 to receive a call via SIP, dial a different SIP
destination and bridge them together.
However, even if the destination indicates that it doesn't support
telephone-event, Asterisk is still sending DTMF as events, not
transcoding to inband.
Asterisk is recognising inband
I forgot to mention that pjsip.conf for this endpoint (that doesn't
support telephone-event) already has this:
dtmf_mode=auto
Cheers,
Kingsley.
On Tue, 2021-10-19 at 15:19 +0100, Kingsley Tart wrote:
> Hi,
>
> I'm using Asterisk 18 to receive a call via SIP, dial a different SIP
>
On Wed, 2021-10-20 at 06:44 -0300, Joshua C. Colp wrote:
> > Should I download and compile this instead?
> >
> > http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18-current.tar.gz
>
> If you want to be running Asterisk 18 and a known released version, yes.
Right OK thanks, I'll do
On Tue, 2021-10-19 at 15:02 -0300, Joshua C. Colp wrote:
> # asterisk -V
> > Asterisk GIT-master-cc127a999cM
> > #
>
> That's the master branch from around March or so, not 18.
Wow, all this time I thought I was running 18! What version would it
be? How can I tell?
Should I download and compile
Hi,
I have built a new Asterisk installation:
root@gw9:/tmp# asterisk -V
Asterisk 18.7.1
It still does the same thing, which is
a. Asterisk receives INVITE containing SDP telephone-event
b. Asterisk uses Dial with pjsip and sends INVITE to destination
including SDP telehone-event
c. Asterisk
On Tue, 2021-12-21 at 10:30 -0400, Joshua C. Colp wrote:
> Allow traffic from specific IP addresses? Others may have better
> input or guidance on such a situation.
Hi,
Thanks.
That's the problem. Customers have automated access to their setup and
may at any point change the SIP destination of
Hi,
It's not the perfect solution but I have found that the AMI has a
PJSIPNotify command which I could periodically call on the relevant
channels. This ought to be enough to keep firewalls open.
My AGI daemon already has a section that periodically runs background
tasks so I can get this to
On Tue, 2021-12-21 at 09:45 -0400, Joshua C. Colp wrote:
> No. Session timers on the endpoint is the closest thing to making
> sure a call is active and keeping things open but does not use
> OPTIONS. Note that if you're sending calls to them, then without
> OPTIONS outside of calls any NAT
Hi,
I see I can set qualify_frequency (for UDP) on an AOR to keep open
holes through firewalls etc, and in [global] I can set
keep_alive_interval for TCP based transports.
However, is it possible to configure it so that these OPTIONS
keepalives only get sent while there's an active call to that
On Mon, 2021-11-08 at 12:01 -0600, Carlos Chavez wrote:
> Just use the something like the following in your script:
>
> make menuselect.makeopts
> menuselect/menuselect --enable codec_opus --enable codec_silk --
> enable
> codec_siren7--enable codec_siren14 menuselect.makeopts
>
> Docs are
my last few emails to this list haven't appeared so I'm just testing
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
This turned out to be a brain fart on my part, not a bug in Asterisk.
Thanks for your help and sorry to waste your time ...
Cheers,
Kingsley.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check
On Mon, 2021-11-08 at 12:01 -0600, Carlos Chavez wrote:
> make menuselect.makeopts
> menuselect/menuselect --enable codec_opus --enable codec_silk --enable
> codec_siren7--enable codec_siren14 menuselect.makeopts
>
> Docs are here:
>
>
Hi,
I realise that this is not really specific to Asterisk, but this seems
as sensible a place to ask as any.
If I want to create a script to automate the build of my chosen
Asterisk setup, what's the best way to automate my selections that I
did interactively when I ran "make menuselect" ?
I
On Mon, 2021-11-08 at 12:01 -0600, Carlos Chavez wrote:
>
>
> Just use the something like the following in your script:
>
> make menuselect.makeopts
> menuselect/menuselect --enable codec_opus --enable codec_silk --enable
> codec_siren7--enable codec_siren14 menuselect.makeopts
>
> Docs
Hi,
I'm using Asterisk 18.7.1.
I can't get res_fax to load. I built it accidentally with app_fax
enabled, and was getting this in the log on startup:
[Nov 3 11:52:31] ERROR[10886] loader.c: Error loading module
'res_fax_spandsp.so', missing dependency: res_fax
Discovering that app_fax and
> Is the app_fax.so module still in /usr/lib/asterisk/modules? If so -
> if you remove it do things work.
> Is app_fax.so explicitly being loaded in modules.conf?
Thanks.
I was already waiting for it to finish recompiling after Doug's
suggestion but yes, app_fax.so was still in there and
Hi,
When dialling a remote SIP host with PJSIP, is it possible either
within the dialplan or via the AMI to find out the IP address of the
remote host?
If for example a remote host has multiple A records, I would like to
know which one Asterisk has connected to.
We have an issue with some
On Wed, 2021-11-03 at 16:25 -0300, Joshua C. Colp wrote:
> On Wed, Nov 3, 2021 at 3:31 PM Kingsley Tart
> wrote:
> > Hi,
> >
> > When dialling a remote SIP host with PJSIP, is it possible either
> > within the dialplan or via the AMI to find out the IP addr
On Thu, 2021-11-04 at 09:45 -0300, Joshua C. Colp wrote:
> The information may not yet be available. Why that would be, I do not
> know.
Right OK, a bit of a mystery then.
I have tried to figure out whether this information is available via
the AMI but I haven't been able to find anything.
Do
On Thu, 2021-11-04 at 10:10 -0300, Joshua C. Colp wrote:
> > Do you know whether it is possible to get the remote_addr from the
> > AMI?
>
> I don't know off the top of my head. AMI actions and events are
> documented on the wiki[1], so you could look there and see.
>
> [1]
Hi,
I can't get Asterisk to send a SIP call to Twilio over TLS because it
complains about Twilio's wildcard certificate.
This is with Asterisk 18.8.0 and PJSIP 2.10
pjsip show transport shows me this:
allow_reload : false
async_operations : 1
bind
On Wed, 2021-12-01 at 22:54 +0100, Antony Stone wrote:
> So, https://datatracker.ietf.org/doc/html/rfc5922#section-7.2 does seem
> pretty
> clear about this. "Implementations MUST NOT match any form of wildcard"
>
> Have you contacted the provider who is using a wildcard certificate in this
>
On Wed, 2021-12-01 at 21:49 +0100, Antony Stone wrote:
> 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
Thank you everyone for your help and comments with this.
I can't explain this but it has now started working. I had no luck with
tlsv1 or tlsv1_2 but using sslv23 does work.
The strange thing is, I tried that before and it DIDN'T work. I'm not
sure why.
Apologies for my delay in responding to
Hi,
I'm using Asterisk 18.8.0 with pjsip version 2.10.
With a database defined endpoint, I can't find a way to define
outbound_proxy with ";lr" (without the quotes) on the end.
It works fine if I configure an endpoint in pjsip.conf, eg:
-- 8<
On Tue, 2021-12-07 at 09:28 -0400, Joshua C. Colp wrote:
> > Is this a bug, or am I doing this wrong?
>
> It's not a bug, it's a result of ";" having special meaning from the
> database - it means multiple values. You have to encode it and use
> ^3B instead of ; in the entry.
Fantastic, thanks
Does asterisk follow HTTP redirects? If so can you use something like
tinyurl.com to produce an alternative URL?
Or, base64 encode the URL, and then set a variable with
Set(url=${BASE64_DECODE(${encodedURL})) ?
Cheers,
Kingsley.
On Wed, 2022-01-26 at 16:56 -0500, Dovid Bender wrote:
> I tried
Hi,
We have instances where we dial multiple destinations simultaneously
and then an answer 'macro' prompts the callee to press 1 to accept the
call or 3 to reject.
Previously if they pressed 3 (or just hung up) the other destinations
would continue to ring.
Now on Asterisk 18.13.0, the first
On Thu, 2022-07-28 at 10:52 -0300, 'Joshua C. Colp' via Kingsley dev
wrote:
> I haven't tested it, but you're calling it a macro when it's not.
> You're invoking a subroutine. As a result MACRO_RESULT won't do
> anything, it should be GOSUB_RESULT. You also shouldn't call Hangup
> in there.
Hi,
We've recently hit an issue with Asterisk 18.8.0 where a call comes in
via SIP (using pjsip) but it can take 5 seconds before starting to
execute the dialplan.
This was intermittent, but frequent (eg approx half of the calls).
We have verbose logging on, but I didn't see any errors.
Hi David,
I chanced upon your question while I was looking for the same thing
myself. I don't know whether this is still relevant to you, given that
it's over 2 years ago since you asked the question.
There's an option in the global section of pjsip.conf that defaults to
"no", but if you set it
On Tue, 2023-02-28 at 09:50 -0400, Joshua C. Colp wrote:
> Is the local hostname configured in /etc/hosts and not reliant on an
> outside DNS server? Are you using ICE or STUN at all?
Hi,
thanks for responding.
No ICE or STUN.
Some of the servers have entries for themselves in /etc/hosts and
Hi,
This is using Asterisk certified/13.21-cert2, FWIW.
I have a hangup handler on an outgoing SIP channel that grabs the SIP status
like this:
NoOp(keys=${HANGUPCAUSE_KEYS()} sipmsg=${HANGUPCAUSE(${CHANNEL},tech)})
This works fine if the call connects to the other end but the caller for
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
Hi,
We're using Asterisk 13.17.0 with PJSIP 2.8 bundled.
I've found an issue when Asterisk tries to make a SIP call out using
auth, but has the wrong credentials and keeps getting returned a SIP
407, in this example to an OpenSIPs server requiring user auth.
Basically this happens:
1.
On Wed, 2020-10-28 at 14:40 -0300, Joshua C. Colp wrote:
> This is not yet fixed, but is being worked on. I have it as a
> security issue currently out of caution (although I don't think we'll
> treat it as one after further investigation).
Right OK, thanks.
Do you have any idea of the sort of
wrote:
> On Tue, Oct 19, 2021 at 11:46 AM Kingsley Tart
> wrote:
> > I forgot to mention that pjsip.conf for this endpoint (that doesn't
> > support telephone-event) already has this:
> >
> > dtmf_mode=auto
>
> What version of 18? Have you enabled A
On Tue, 2021-10-19 at 15:02 -0300, Joshua C. Colp wrote:
> # asterisk -V
> > Asterisk GIT-master-cc127a999cM
> > #
>
> That's the master branch from around March or so, not 18.
Wow, all this time I thought I was running 18! What version would it
be? How can I tell?
Should I download and compile
On Fri, 2021-10-22 at 11:11 -0300, Joshua C. Colp wrote:
> I don't provide direct support like that. As there seems to be a bug
> and you have a case that reproduces it with logs, then you can file
> an issue[1] and the current individual doing bug triage will look. If
> it is accepted there is no
On Mon, 2021-11-08 at 12:01 -0600, Carlos Chavez wrote:
>
>
> Just use the something like the following in your script:
>
> make menuselect.makeopts
> menuselect/menuselect --enable codec_opus --enable codec_silk --enable
> codec_siren7--enable codec_siren14 menuselect.makeopts
>
> Docs
On Mon, 2021-11-08 at 12:01 -0600, Carlos Chavez wrote:
> make menuselect.makeopts
> menuselect/menuselect --enable codec_opus --enable codec_silk --enable
> codec_siren7--enable codec_siren14 menuselect.makeopts
>
> Docs are here:
>
>
> Is the app_fax.so module still in /usr/lib/asterisk/modules? If so -
> if you remove it do things work.
> Is app_fax.so explicitly being loaded in modules.conf?
Thanks.
I was already waiting for it to finish recompiling after Doug's
suggestion but yes, app_fax.so was still in there and
On Thu, 2021-11-04 at 08:52 -0300, Joshua C. Colp wrote:
> > Thanks, that looks perfect. What is the syntax? I have tried a few
> > things but none work:
> >
>
> ${CHANNEL(pjsip,remote_addr)}
Hmm, I can't get this to work. This dialplan code:
exten => s,n,NoOp(### state=${CHANNEL(state)} ##)
Does asterisk follow HTTP redirects? If so can you use something like
tinyurl.com to produce an alternative URL?
Or, base64 encode the URL, and then set a variable with
Set(url=${BASE64_DECODE(${encodedURL})) ?
Cheers,
Kingsley.
On Wed, 2022-01-26 at 16:56 -0500, Dovid Bender wrote:
> I tried
On Tue, 2023-11-07 at 08:42 +0100, Luca Bertoncello wrote:
> The best will be a free service, but if not, I don't want to pay too
> much...
> As said: I need a SIP Provider to have an italian number (better if I
> can choose the prefix) only to receive calls.
>
> Any suggestion?
Assuming that
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