Hi Mike,
New AMI actions were recently added to app_voicemail to let you remotely
manipulate a mailbox:
https://github.com/asterisk/asterisk/issues/181
Hope this helps.
BR,
-Mike
On Mon, Oct 9, 2023 at 1:06 PM Mike Diehl wrote:
> Hi all,
>
> I need to be able to delete a voicemail message
At the moment, I can't see any differences here. sha512sum is identical.
Regards
Michael
On 08.07.23 at 01:50 Jean-Denis Girard wrote:
Le 07/07/2023 à 12:49, Joshua C. Colp a écrit :
On Fri, Jul 7, 2023 at 6:40 PM Jean-Denis Girard <mailto:jd.gir...@sysnux.pf>> wrote:
Th
FYI: i've created a feature request to add SIP_CODEC_INBOUND equivalent
functionality to chan_pjsip:
https://github.com/asterisk/asterisk-feature-requests/issues/9
Let's see where it goes
*Michael Ulitskiy*
Ace Innovative Networks, Inc.
Main/SMS: 212-868-2366
Direct/SMS: 212-812-1203
https
Oh, that's great. It wasn't clear from that page, at least not for me. :-(
Having it clearly stated on the document would save me (and probably
others) lots of time.
Thanks for clarifying it. Any idea on the timeframe of implementation?
*Michael Ulitskiy*
Ace Innovative Networks, Inc.
Main
in both call legs, thus avoiding
transcoding, but actual asterisk behavior differs.
Am I missing something. What are your thoughts?
Thanks,
*Michael Ulitskiy*
Ace Innovative Networks, Inc.
Main/SMS: 212-868-2366
Direct/SMS: 212-812-1203
https://www.aceinnovative.com
On 7/5/23 11:58, M
Well, I'm trying to migrate to chan_pjsip so that I don't have to do that.
It's so surprising that the issue so seemingly obvious and trivial
hasn't been addressed yet that I wanted to query the collective wisdom
of this list to verify my observations.
Thanks for github pointer.
Michael
Hi Michael,
Thanks for the reply.
I was referring to the scenario you named as 'outbound broken'. I didn't
get to look at inbound call behavior yet, as I got stuck with inability
to avoid transcoding on outbound calls.
To be more specific the scenario is as follows:
1. a phone initiates
Hello Michael,
you are referring to the following behavior - did I get it correctly?:
outbound broken: asterisk offers g722 / g711 to provider (callee),
callee answers g711. Asterisk now transcodes between caller and callee
(g722 <-> g711).
inbound works: call from provider
t up using ulaw end-to-end
Can somebody please advise how to achieve the same with chan_pjsip?
Thanks,
Michael
On 6/30/23 09:30, Michael Ulitskiy wrote:
Hello,
I finally got to look at chan_sip to chan_pjsip migration again. This
time I’m having problems with influencing codec selection on
o
of ${SIP_CODEC_INBOUND}? Or, in other words, what are we supposed to do
to influence /calling/ channel codec selection from dialplan?
I’m working with asterisk 20.3.0.
Thank you,
Michael
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On 09.04.23 at 19:55 Steve Matzura wrote:
Thanks, Michael. A few questions:
Is [transport_name] a reserved word, or am I supposed to replace it with a name of
my own, like '[did-transport]'?
Yes. You are free.
Some of the keywords I haven't seen before. Is ca_list_file supposed
external_media_address=your.ext.host.name ; hostname pointing to your
ext. IP
external_signaling_address=your.ext.host.name ; hostname pointing to
your ext. IP
local_net=192.168.0.0/24 # your local net
Regards
Michael
On 07.04.23 at 17:25 Steve Matzura wrote:
I want to configure communication
les.
I use `audit2why` and `audit2allow` in policycoreutils-devel (on CentOS) to
generate SELinux policy modules.
-Michael Englehorn
‐‐‐ Original Message ‐‐‐
On Monday, January 10th, 2022 at 1:03 PM, Jerry Geis
wrote:
> I am trying to run this command:
> exten => _4XX,n,Syst
delete=yes" as option per mailbox account.
100 => 1234,Test,,,delete=yes
The global setting is only an example.
Michael
http://www.mksolutions.info
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ki.asterisk.org/wiki/display/AST/Configuring+res_pjsip+to+work+through+NAT
--
Michael L. Young
(elguero)
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On 30.06.21 at 23:17 Joshua C. Colp wrote:
> On Wed, Jun 30, 2021 at 1:36 PM Michael Maier wrote:
>
>>
>> Hello!
>>
>> Short question: Is it possible to set
>>
>> a=silenceSupp:off
>>
>> in the SDP for alaw / ulaw for fax calls?
Hello!
Short question: Is it possible to set
a=silenceSupp:off
in the SDP for alaw / ulaw for fax calls?
Thanks
Michael
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On 02.05.21 at 17:24 Michael Maier wrote:
On 02.05.21 at 10:08 Michael Maier wrote:
Hello!
I've just playing around some time to get NAT and pjsip running with asterisk
18.3 and 18.4 (w/o any patches added). NAT should be used for connection to the
trunk.
I wasn't able to get it working
On 02.05.21 at 10:08 Michael Maier wrote:
Hello!
I've just playing around some time to get NAT and pjsip running with asterisk 18.3
and 18.4 (w/o any patches added). NAT should be used for connection to the trunk.
I wasn't able to get it working, because SDP address rewriting just doesn't
a reference configuration for a working NAT
configuration?
Thanks
Michael
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On 01.05.21 at 07:13 Michael Maier wrote:
Hello all!
I'm actually wondering about how to achieve fast fail handling for the pjsip
transport if underlying WAN IP address changes.
Forgot to mention:
- I'm using TLS!
- pjsip tries every 92s to send the Registration until the timeout comes up
restart of the individual transport? Maybe in pjsip (connection timeout
detection)?
Another idea would be to restart the relevant transports based on dnsmgr detecting
new external IP.
Thanks
Michael
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ssumption that the variable would be
inherited into subsequent channels, but that does not work either.
What am I missing?
Asterisk: 13.14.1~dfsg-2+deb9u4
OS: Debian 9.13 (Stretch).
--
Michael Munger, dCAP, MCPS, MCNPS, MBSS
*Microsoft Certified Professional*
*Microsoft Certified Small Business S
On 18.02.21 at 20:01 Luca Bertoncello wrote:
Am 18.02.2021 um 18:59 schrieb Michael Maier:
On 17.02.21 at 21:46 Luca Bertoncello wrote:
Am 16.02.2021 um 22:32 schrieb Michael Maier:
Hi Michael
Maybe could you send me an abstract of your configuration?
Take a look here [1]
So, maybe I
On 17.02.21 at 21:46 Luca Bertoncello wrote:
> Am 16.02.2021 um 22:32 schrieb Michael Maier:
>
> Hi Michael
>
>>> Maybe could you send me an abstract of your configuration?
>>
>> Take a look here [1]
>
> So, maybe I got it...
> I tested the configurati
On 16.02.21 at 20:33 Luca Bertoncello wrote:
Am 16.02.2021 um 19:56 schrieb Michael Maier:
Hi Michael,
Do I use pjsip?
pjsip show registrations
gw*CLI> pjsip show registrations
No objects found.
So I don't use pjsip... :(
Yes.
Maybe could you send me an abstract of y
Hi Luca,
On 15.02.21 at 21:48 Luca Bertoncello wrote:
> Am 15.02.2021 um 21:40 schrieb Michael Maier:
>
> Hi Michael,
>
>> They're switching to DNS NAPTR / SRV[1]. If you are using Asterisk /
>> pjsip and hostnames (tel.t-online.de e.g. for the AllIP service), you
>
s-epp-110.edns.t-ipnet.de.
_sip._tcp.tel.t-online.de. 3600 IN SRV 20 0 5060
h2-epp-100.edns.t-ipnet.de.
Asterisk now must use always the same server for all activities to
Telekom - like register, invite, options - but that's not yet supported
by Asterisk - therefore you have to ensure
- On Feb 5, 2021, at 11:18 AM, Michael L. Young wrote:
> - On Feb 4, 2021, at 4:26 PM, Social Boh wrote:
>> The problem is with this CentOS 7 glibc version:
>> 2.17-317.el7
>> After the library update and system reboog,
>> gotoif Asterisk applicati
a 'yum history' and note the transaction ID of
the update. Then try running 'yum history undo [transaction id]'. That should
roll you back to the previous glibc.
Looks like Red Hat is already working on it:
https://access.redhat.com/solutions/5778071
--
Michael
On 29.01.21 at 22:33 Ruisheng Peng wrote:
Thanks for the detailed explanation Michael.
I stop the current asterisk process (started by systemd), and restart it as
asterisk:
[asterisk@voip1 ~]$ strace -f -o /home/asterisk/strace.log asterisk -fmq
-vvv -C /etc/asterisk/asterisk.conf
from
On 29.01.21 at 06:41 Michael Maier wrote:
On 27.01.21 at 22:57 Ruisheng Peng wrote:
Thanks Michael for the suggestion! I've installed strace and assigned one
of the endpoints (SOFTPHONE_B) to use transport-tls. Then run strace (as
user asterisk):
[asterisk@voip1 ~]$ strace asterisk -rx
On 27.01.21 at 22:57 Ruisheng Peng wrote:
Thanks Michael for the suggestion! I've installed strace and assigned one
of the endpoints (SOFTPHONE_B) to use transport-tls. Then run strace (as
user asterisk):
[asterisk@voip1 ~]$ strace asterisk -rx "module reload res_pjsip.so"
You
e, that the file access rights of
the file and the complete path are ok. Do a strace to verify, if the
file is really loaded at all.
Michael
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Re: Get a SHAKEN Identity Token (Alexander Perkins)
Saint Michael
1:06 PM (0 minutes ago)
to Asterisk
Please look at this
https://issues.asterisk.org/jira/browse/ASTERISK-28924
I have a solution that works for any version of Asterisk, if interested
contact me at venefax at the Google mail
Please look at this
https://issues.asterisk.org/jira/browse/ASTERISK-28924
I have a solution that works for any version of Asterisk, if interested
contact me at venefax at the Google mail service.
On Sun, Jan 24, 2021 at 1:00 PM
wrote:
> Send asterisk-users mailing list submissions to
>
Stir Shaken
Asterisk cannot do that, but my company can give you Stir Shaken for
Asterisk, via ODBC, any version.
Please contact me via email venefax at the google mail system
Philip Orleans
On Fri, Jan 8, 2021 at 1:00 PM
wrote:
> Send asterisk-users mailing list submissions to
>
On 21.10.20 at 12:49 Joshua C. Colp wrote:
> On Wed, Oct 21, 2020 at 7:46 AM Michael Maier wrote:
>
>> Hello!
>>
>> On 20.10.20 at 14:00 Asterisk Development Team wrote:
>>> The Asterisk Development Team would like to announce the release of
>> Asteris
you please add the correct configuration you expect to get the expected
result alaw?
Thanks
Kind regards
Michael
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On 16.10.20 at 11:07 sergio wrote:
> On 16/10/2020 10:11, Michael Maier wrote:
>>> Sometimes, linphone shows missed calls as missed.
>> You could try to reproduce it
>
> I can't reproduce it, it happens less than once a month.
Then you should enable the tracing as I wr
is?
You could try to reproduce it while activating pcap traces and analyze
it afterwards - or you could enable pcap traces on asterisk[1] itself
and just wait for the different issues and compare them.
Regards
Michael
[1] activating pcap traces on asterisk / pjsip
pjsip set logger on
pjsip set
advised “hey I
> am not available” and actually show a RED dot.
> thanks
>
> --
Hi Jobst,
there is a setting for the Yealink phones so that the BLF keys are "off"
instead of "green" when idle:
BLF LED Mode = 1
in t
memory vs disk cache
> This is an issue that has plagued Asterisk since day one. Basically there
>> is no solution available because there is no way to set aside memory to be
>> kept from a growing disk cache. I did some research and this looks like a
>> bad design from the Kernel people.
>
> This is an issue that has plagued Asterisk since day one. Basically there
> is no solution available because there is no way to set aside memory to be
> kept from a growing disk cache. I did some research and this looks like a
> bad design from the Kernel people. Meanwhile all you can do us
led pjsip library. Doing
it this way ensures that pjsip and asterisk match for sure (and some additional
patches are applied to pjsip on top regarding usage of pjsip in asterisk).
Greetings
Michael
[1] https://downloads.asterisk.org/pub/telephony
I need to point out the this is factually misleading and materially false:
"I think this, being the basis of your whole argument, is the fallacy.
S/S is forcing people to take responsibility, for sure, but carriers
won't just let their customers leave because they don't want to sign
calls. It
>
> There is a big confusion here about Stir Shaken. It is NOT a provider
> issue. Un fact, all providers are whasing their hands and modifying their
> swihtches to pass-through the Signature. They cannot sign the call because
> then the become the responsible party for the call before the FCC,
On 13.07.20 at 10:54 Joshua C. Colp wrote:
> On Sun, Jul 12, 2020 at 11:37 PM Michael Maier wrote:
>> One more question,
>> what about the pjsip pcap support? Will it be backported to Asterisk 16,
>> too? Would be absolutely cool! Debugging encrypted SIP is really a p
absolutely cool! Debugging encrypted SIP is really a pain.
BTW: what about the (encrypted) RTP packets? Will they be dumped, too?
Thanks
Michael
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C
WORLDWIDE EMERGENCY
The code below needs to be executed before any SIP or PJSIP call destined
to the US network, or soon no call will terminate. This is called
Stir-Shaken, a new law from the FCC.
If this is not working the whole Asterisk industry will crash, vanish, be
gone. I am assuming that
er and reliable rtp
stream - even over hours.
Regards
Michael
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N
Am 24.06.20 um 08:10 schrieb Luca Bertoncello:
Am 24.06.2020 05:05, schrieb Michael Maier:
Hi
Your basic architecture looks good to me - now you have to start the
Nice to hear it...
analysis of the problem with pcapsipdump and wireshark as I wrote
before to get an idea what actually
On 23.06.20 at 21:10 Luca Bertoncello wrote:
> Am 23.06.2020 um 21:08 schrieb Michael Maier:
>> On 23.06.20 at 08:05 Luca Bertoncello wrote:
>>> Am 23.06.2020 07:27, schrieb Luca Bertoncello:
>>>
>>> I again
>>>
>>>>> Do not change MTU. P
-packets never reach this size. Size is about 214 bytes.
Regards
Michael
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you (regarding distance). You're RTT of ping (18 ms) is pretty bad. I'm having here 5 ms to the
primary server (Telekom provides 3). See
dig +noall +answer _sip._udp.tel.t-online.de SRV
e.g. (don't know the hostname for the business infrastructure)
Regards,
Michael
[1] h
"internal number") the quality is excellent.
>>
>> If I call my wife using the "external number", the quality is very bad...
>>
>> Thanks
>> Luca Bertoncello
>> (lucab...@lucabert.de)
Michael
http://www.mksolutions.info
--
_
eally do *not* expect any change in the situation... I think, the
> problem should be somewhere by Deutsche Telekom...
>
> What is your opinion?
>
> Btw: I did all tests with my father in law, since he had time for me
> today, but the problem exists an almost all calls, incomin
So the call used Alaw as Codec.
> Am 13.06.2020 um 17:23 schrieb Luca Bertoncello :
>
> Am 13.06.2020 um 13:47 schrieb Michael Keuter:
>
> Hi
>
>> Try "sip show peer " for a phone.
>
> So:
>
> mobile phone:
> bpi*CLI> sip show peer 0049177
a phone.
Then "sip show channels" during an existing call.
And "sip show channel " for more info.
Michael
http://www.mksolutions.info
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ut like
core set debug 5
Maybe some link time functionality not enabled? Don't know ... . Or some other
additional asterisk switch needed?
Thanks
Regards
Michael
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I got the response below from a provider. How do I extract the Identity
header and apply it to the next INVITE? Is it possible at all with PJSIP?
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 172.16.7.254:52169
;rport=52169;received=XX.205.172.89;branch=z9hG4bK-524287-1---129f4244aaba9f04
https://wiki.asterisk.org/wiki/display/AST/STIR+and+SHAKEN
The Wiki above is misleading in what Stir-Shaken means and how it works.
End users cannot get a certificate, they cannot self-certify their calls.
Somebody completely misunderstood the model. I am afraid the moment will
come and thousands
>
> My company is one if the six service providers approved. We are not ready
> yet, probbably next week, since the certificate needs to be issued by the
> Certification Authority. As I said, we are the ONLY provider that you may
> use with Asterisk remotely, via UnixODBC. The rest of the other
In a few weeks, no SIP call is going to terminate unless they are signed
properly, as mandated by law. We are in the business of Stir-Shaken,
signing calls, as an FCC-approved provider. A big differentiator between
our service and the rest: we are the only ones who don't need to receive
the calls
About this case: the old SIP channel behaves correctly.
On Sun, May 17, 2020 at 2:44 AM Saint Michael wrote:
> My phone is located behind a NAT, 172.16.0.0/21.
> Asterisk 16 is on a public IP.
> PJSIP has the config below:
> force_rport=yes
> direct_media=yes
> disable_direct_
On 17.05.20 at 01:28 Joshua C. Colp wrote:
> On Sat, May 16, 2020 at 10:58 AM Michael Maier wrote:
>
>> => How are the RTT values exactly calculated? Which values are actually
>> used for?
>>
>
> The value is calculated according to the logic in the RFC[
I want to see the help when I type core show application , and it's not
available. This is asterisk 16 from sources. I have libxml2-dev installed.
Ubuntu 19
What am I missing?
Philip
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Endpoint sends an INVITE
Asterisk send an INVITE to the Carrier
Carrier is down, does not even sends ACK
PJSIP sends several INVITES
End point sends
<--- Received SIP request (397 bytes) from UDP ::50187 --->
CANCEL sip:xxx@xxx SIP/2.0
Via: SIP/2.0/UDP xxx
g
points located?
=> How are the RTT values exactly calculated? Which values are actually used
for?
=> What about the processing time between the inbound leg and the outbound leg
(transcoding, ...)?
Thanks
Michael
--
___
03:22:42 alaw 608K 00 0.000
608K 00 0.000 0.023
Thanks
Michael
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owse/ASTERISK-26143 |
https://issues.asterisk.org/jira/browse/ASTERISK-26143 ]
Not sure if this is the answer to your problem but thought that I would throw
that out there.
Michael L. Young
(elguero)
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>
> Asterisk needs urgently to push the RTP engine to the Kernel, away from
> userland, like professional and commercial softwares do. I measured the
> cost of passing call from a public IP to a private IP, like typically a
> Session Border Controller may do. In Asterisk, ulaw, no transcoding, it
>
> I have no control over the SIP calls I receive. PJSIP should log a warting
> and continue. It is causing the CPU usage to spike dramatically.
>
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Check
PJISP cannot handle the From field when it does not contain a number.
Can this be fixed?
[Feb 25 12:35:43] ERROR[7143]: pjproject: : sip_transport.c
Error processing 400 bytes packet from UDP 8.38.43.67:5060 : PJSIP syntax
error exception when parsing 'From' header on line 4 col 40:
Is there a guide on how to use PJSIP and never have the media travel inside
Asterisk? No matter what I do, I cannot make this work.
Philip Orleans
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as user asterisk work, but fail using
> systemd ?
>
Have you checked SELinux? After creating the configuration files, did you run
'restorecon' on the appropriate asterisk directories? If not, the files are
not labeled correctly and SELinux might be denying
I have a customer who wants me to send a DTMF on the calling channel if the
called channel says any word. So I am using
[my_gosub_routine]
exten => s,1,NoOp(ARG1=${ARG1} ARG2=${ARG2})
same => n,Playback(hello)
same => n,Return()
[default]
exten => _X.,1,NoOp()
same =>
Michael
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New to Asterisk? Start here:
https://wiki.asterisk.org/wiki
* wi-mke1-dc2-c5-47-haf-7578.cyberlynk.net (66.185.29.26) 118.165 ms *
15 199.102.239.92 (199.102.239.92) 122.352 ms 126.759 ms 123.944 ms
16 199.102.239.92 (199.102.239.92) 123.890 ms 124.069 ms 126.670 ms
Thanks
Michael
--
e steadily
growing up and using more and more open files (and memory) until asterisk can't
do
anything anymore because some limits are exceeded as a result?
Could there be a problem with the retry interval 60s and the real ReRegister
done each 60s, too (performing a "for
systems have been correctly unmounted and all processes have
been stopped before unmounting)?
Is the startup slow even if it's done w/o reboot before?
Regards
Michael
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240] NOTICE[1385] loader.c: 286 modules will be loaded.
> [08/12 15:33:23.844] VERBOSE[1385] loader.c: Loading extconfig.
>
>
> Loading the modules is taking 20 seconds after this incident occurred.
> Looking at the debug logs, I see the modules loadi
Hello!
Does anybody by chance know of a softphone which additionally has a management
suite to fully configure it userID based for Windows clients? Any idea is
appreciated!
Thanks
Michael
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ent users).
Besides the possibility to use different IP addresses (aliases) - is there a
generic way to define on trunk base which local port to use for each user /
number?
Thanks
Michael
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le expect / want to have.
Regards
Michael
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New to Asterisk? Start here:
On 06.07.19 at 12:16 hwilmer wrote:
> On 7/6/19 10:40 AM, Michael Maier wrote:
>> On 05.07.19 at 22:02 hw wrote:
>>>
>>> openssl verify -CAfile ca.pem asterisk.pem
>>> asterisk.pem: OK
>>>
>>>
>>> When I set tlsdontverifyserver=
to connect via tls to the ISP?
To be able to verify the certificate of the ISP, asterisk has to know the local
CA database. For CentOS 7, this is /etc/pki/tls/certs/ca-bundle.crt.
Regards
Michael
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96
00 0.000 0.000
Instead of "pjsip show channelstats" you have to use something like sip show
[press 2 times tab key] to get the possible commands.
Each call generates two entries: one for the call from your local phone to
asterisk and the other from ast
Joshua
Is there a way in PJSIP to send the audio between the parties always,
unless one of the parties is behind a NAT?
A session refresh would work.
That my only problem with PJSIP. This is routine in the old sip channel.
On Sat, May 25, 2019 at 1:03 PM
wrote:
> Send asterisk-users mailing
Hello!
I'm just wondering if it's possible to decrypt sips packages in Wireshark while
asterisk runs as sips client (connecting to the provider w/
tls 1.2)? I don't use an own certificate.
Thanks
Michael
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time I looked into this with PHP was under PHP 5.6, and tests
at that time did not yield the results we wanted. We ultimately moved to Python
when we needed multi-threading, which is extremely elegant and reliable for
this application.
[cid:image001.png@01D4F6B9.8C1499D0]
Michael J. Munger, dCAP
continue. Would need testing so it fails
gracefully.
[cid:image001.png@01D4F6B8.F3CEF8F0]
Michael J. Munger, dCAP, MCPS, MCNPS, MBSS
Microsoft Certified Professional
Microsoft Certified Small Business Specialist
Digium Certified Asterisk Professional
High Powered Help, Inc.
p:
678-905-8569
w
Excellent point.
This is it: https://www.valcom.com/pdf/v-1109rthf.pdf
Get Outlook for Android<https://aka.ms/ghei36>
On Thu, Mar 21, 2019 at 7:22 PM -0400, "John Novack"
mailto:jnov...@comcast.net>> wrote:
Michael Munger wrote:
Does anyone have an (overhead) paging
It worked on the old system.
I am open to suggestions, but don't want (or have the option) to add a
TDM card.
Michael Munger, dCAP, MCPS, MCNPS, MBSS
*Microsoft Certified Professional*
*Microsoft Certified Small Business Specialist*
*Digium Certified Asterisk Professional*
*High
.
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Michael J. Munger, dCAP, MCPS, MCNPS, MBSS
Microsoft Certified Professional
Microsoft Certified Small Business Specialist
Digium Certified Asterisk Professional
High Powered Help, Inc.
p:
678-905-8569
w:
hph.io<https://hph.io> e: m...@hph.io<mailto:m.
compile from source, which uses /var/lib/asterisk by default. First
time using a repo package)
Asterisk version: 13.14.1~dfsg-2+deb9u4
OS: Debian 9.8 (stretch). Fully updated.
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Michael J. Munger, dCAP, MCPS, MCNPS, MBSS
Microsoft Certified Professional
On 17.12.18 at 11:52 Joshua C. Colp wrote:
> On Sun, Dec 16, 2018, at 4:43 AM, Michael Maier wrote:
>
>
>
>>
>> Another question: is there any use case for 183 Session Progress w/o
>> SDP? IOW: Is a 183 Session
>> Progress just a bug of the ISP? If so, pro
990a42e1317f4aa2bad51a3ef9f17
I am using "mailboxes=##@default" and had the issue as well (before).
Michael
http://www.mksolutions.info
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Chec
>
> when compiling the latest version, it fails here
./configure LDFLAGS="-z muldefs" --libdir=/usr/lib64
--with-unixodbc=$(odbc_config --include-prefix)/ --disable-asteriskssl
-enable-xmldoc NOISY_BUILD=yes
> gcc -o res_pjsip/config_transport.o -c res_pjsip/config_transport.c -MD
> -MT
On 28.12.18 at 13:20 Doug Lytle wrote:
>>>> Before I'm opening an issue, I would like to prove my expectations - maybe
>>>> it isn't a problem at all or it's a problem of the phone.
>
> Michael,
>
> Just a side note. I've had reports of MWI not turning off
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