On 06/06/2016 at 04:40 PM Richard Mudgett wrote:
> On Sun, Jun 5, 2016 at 3:48 AM, Michael Maier <m1278...@allmail.net> wrote:
>
>> Hello!
>>
>> I occasionally can see warnings like these during *idle* times in
>> asterisk log (asterisk 13.7.2):
>>
>
On 06/06/2016 at 04:40 PM, Richard Mudgett wrote:
> On Sun, Jun 5, 2016 at 3:48 AM, Michael Maier <m1278...@allmail.net> wrote:
>
>> Hello!
>>
>> I occasionally can see warnings like these during *idle* times in
>> asterisk log (asterisk 13.7.2):
>>
>
Hello!
I occasionally can see warnings like these during *idle* times in
asterisk log (asterisk 13.7.2):
[2016-06-05 06:11:51] WARNING[27817] pjsip: sip_transactio Unable to
register REGISTER transaction (key xists)
[2016-06-05 06:11:51] WARNING[27817] pjsip: sip_transactio Unable to
Hello Joshua,
On 04/25/2016 at 12:35 PM, Joshua Colp wrote:
> Michael Maier wrote:
>> Hello!
>>
>> I encounter the following problem (asterisk 11 and 13) with Teconisy as
>> trunk provider with enabled qualify and default t1min (100ms):
>>
>> Teconisy mos
value of 100, which can cause much trouble and which
creates totally unnecessary network overhead.
Or is there another solution I overlooked?
Thanks,
Michael Maier
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ion):
[2016-05-12 09:09:58] VERBOSE[4406] res_pjsip/pjsip_configuration.c:
Contact 107/sip:107@192.168.15.73:5060 is now Reachable. RTT: 23.332 msec
-> nothing more!
What's going on in asterisk 13.9? Why does it suddenly behave completely
different?
Thanks,
Michael
On 05/12/2016 at 11:54 AM Joshua Colp wrote:
> Michael Maier wrote:
>> Hello!
>>
>> Today, I tried to switch from asterisk 13.7.2 to 13.9, but I'm getting
>> strange problem w/ the registering of all of my extensions. It looks
>> like that:
>
> This has alr
On 05/03/2016 at 05:43 PM Joshua Colp wrote:
> Michael Maier wrote:
>> On 05/03/2016 at 04:50 PM Joshua Colp wrote:
>>> Michael Maier wrote:
>>>> Hello Joshua!
>>>>
>>>>
>>>> I attached the sip debug without the progressinband=
ng a trunk to a ring
group or an extension?
Puzzled,
regards,
Michael Maier
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On 05/03/2016 at 09:16 PM Joshua Colp wrote:
> Eric Wieling wrote:
>> I don't know the default setting for progressinband in the code, but it
>> is documented in Asterisk 11's sip.conf.sample as defaulting to never.
>> Maybe the docs were fixed since Asterisk 11.
>
> The behavior change to
Hello!
Sometimes, I can see here the following scene:
Asterisk sends 11 SIP OPTIONS-packages (qualify=120) and they are all
ignored by the peers - but the 12. package is answered immediately as
expected (I'm sure there is no network related problem).
I can see this on trunks via Internet and
On 01/30/2017 at 05:55 PM Motty Cruz wrote:
> Fresh installed CentOS 7.3 and Asterisk 13.13.1. Download Asterisk from here:
> http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar.gz
>
>
>
>
> I continue to see errors like this:
>
> [2017-01-30 08:37:17]
Derek Bolichowski wrote:
HI Michael,
You can set this in sip.conf:
session-timers=refuse
I know of this option - it doesn't help, because the provider ignores it
(on some calls) and the call is dropped anyway.
Normally, there is no problem with the timers. And the problem which
occurred
Hello all!
I can see a strange problem during invite in dialog in the context of
timer handling.
Given is the following incoming call from provider at 8.195.88.234 (2@2)
to my asterisk at 28.19.57.152 (1@1):
After 900s suddenly *asterisk* starts the timer reinvite - I would have
expected the
On 12/14/2016 at 06:22 PM, Luca Bertoncello wrote:
> Hi list!
>
> I already had the problem last year, then it would be solved (surely from
> some technician by Deutsche Telekom on their servers), and now I have the
> problem again (but I didn't changed my Asterisk configuration).
>
> The
On 12/28/2016 at 05:36 PM Michael Maier wrote:
> On 12/27/2016 at 07:54 PM Michael Maier wrote:
>> Hello!
>>
>> I'm facing ReInvites as caller from UAS despite configured
>> session-timers=refuse (which can be seen in the SIP trace) always after
>> 900s. (The behav
On 12/27/2016 at 07:54 PM Michael Maier wrote:
> Hello!
>
> I'm facing ReInvites as caller from UAS despite configured
> session-timers=refuse (which can be seen in the SIP trace) always after
> 900s. (The behavior is the same if session-timers is set to accept).
>
> This
Hello!
I'm facing ReInvites as caller from UAS despite configured
session-timers=refuse (which can be seen in the SIP trace) always after
900s. (The behavior is the same if session-timers is set to accept).
This just happens with one provider (German Telekom to callee at kabelbw).
- The
On 12/27/2016 at 07:54 PM, Michael Maier wrote:
> Hello!
>
> I'm facing ReInvites as caller from UAS despite configured
> session-timers=refuse (which can be seen in the SIP trace) always after
> 900s. (The behavior is the same if session-timers is set to accept).
>
> This
Hello!
I'm trying to send a fax via T.38 to a destination, which should be T.38
capable. My provider supports T.38, too. Unfortunately, it doesn't work.
This means:
Call is started and SDP is negotiated w/ alaw. Callee sends reinvite -
for alaw again (and not for T.38)!! After about 30s, callee
On 04/06/2017 at 08:33 PM, Joshua Colp wrote:
> On Thu, Apr 6, 2017, at 03:15 PM, Michael Maier wrote:
>> Hello!
>>
>> I'm trying to send a fax via T.38 to a destination, which should be T.38
>> capable. My provider supports T.38, too. Unfortunately, it doesn't work
On 04/06/2017 at 08:33 PM Joshua Colp wrote:
> On Thu, Apr 6, 2017, at 03:15 PM, Michael Maier wrote:
>> Hello!
>>
>> I'm trying to send a fax via T.38 to a destination, which should be T.38
>> capable. My provider supports T.38, too. Unfortunately, it doesn't work
On 05/12/2017 at 08:49 PM, Joshua Colp wrote:
> On Fri, May 12, 2017, at 02:46 PM, Michael Maier wrote:
>
>
>
>>
>> If I'm doing exactly the same call originated with another extension,
>> there can't be seen these frequent changes. But the strange thing is,
Hello!
I'm facing completely choppy sound. The wireshark trace shows, that
there are a lot of codec changes without any trigger (means no options
or reinvite or any other package).
Background:
The call is initiated by asterisk and is received by the same asterisk
conference room via
Phone
On 05/12/2017 at 07:46 PM, Michael Maier wrote:
Forgot to mention: It's actual asterisk 13 branch from today (version
before I tested, which has the same problem, was 13.15).
Regards,
Michael
Hello!
I'm facing completely choppy sound. The wireshark trace shows, that
there are a lot
On 06/11/2017 at 04:34 PM Michael Maier wrote:
> On 06/11/2017 at 01:29 PM Joshua Colp wrote:
>> On Sun, Jun 11, 2017, at 08:16 AM, Michael Maier wrote:
>>
>>
>>
>>> I did some further investigations and fixed a local problem. Now,
>>> a
On 06/11/2017 at 11:35 PM Daniel Tryba wrote:
> On Sun, Jun 11, 2017 at 01:16:10PM +0200, Michael Maier wrote:
>> Let's go into details:
>> I'm starting at the point where asterisk / fax client receives the T.38
>> reininvite from the server from the provider 195.185.37.6
Hello!
I'm still trying to get a working t.38 configuration w/ pjsip.
I'm now able to send t.38 faxes to my own extension:
hylafax -> t38modem -> extension -> extension -> t38modem -> hylafax.
The fax is sent by t38modem. The receiving part of t38modem accepts the
call, sends ReInvite for
On 06/05/2017 at 11:30 AM, Joshua Colp wrote:
> On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote:
>> On 06/04/2017 at 01:41 PM Telium Technical Support wrote:
>>> Just a guess (without knowing about your network), but are the two ends
>>> points on public networks
On 06/05/2017 at 05:00 PM, Joshua Colp wrote:
> On Mon, Jun 5, 2017, at 11:49 AM, Michael Maier wrote:
>> On 06/05/2017 at 11:30 AM, Joshua Colp wrote:
>>> On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote:
>>>> On 06/04/2017 at 01:41 PM Telium Technical Sup
On 06/05/2017 at 06:10 PM, Joshua Colp wrote:
> On Mon, Jun 5, 2017, at 12:00 PM, Joshua Colp wrote:
>> On Mon, Jun 5, 2017, at 11:49 AM, Michael Maier wrote:
>>> On 06/05/2017 at 11:30 AM, Joshua Colp wrote:
>>>> On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote
Hello!
I've got a problem to select the correct trunk if there is one provider
and different numbers with different configurations for this same provider.
Example:
trunk-prov1-2345
trunk-prov1-2346
trunk-prov1-2347
Each trunk registers an own number (at the same provider) and provides
own
Hello Joshua,
thank you very much for your extremely quick answer! I really appreciate
your work and your friendly and your patient support!
On 06/07/2017 at 10:33 PM, Joshua Colp wrote:
> On Wed, Jun 7, 2017, at 05:28 PM, Michael Maier wrote:
>> Hello!
>>
>> I've go
On 06/04/2017 at 01:41 PM Telium Technical Support wrote:
> Just a guess (without knowing about your network), but are the two ends
> points on public networks and visible to one another? If not the reinvite
> may be passing an internal (nat'ed) address to the other and the connection
> will
On 06/14/2017 at 10:17 PM, Joshua Colp wrote:
> On Wed, Jun 14, 2017, at 05:09 PM, Michael Maier wrote:
>
>
>
>>
>> I can now say, that asterisk / pjsip seams to work *mostly* as expected.
>> Just one exception - and that's the package in question, which can't
On 06/16/2017 at 04:00 PM, Joshua Colp wrote:
> On Fri, Jun 16, 2017, at 10:49 AM, Michael Maier wrote:
>
>
>
>>
>> t38modem and asterisk are using
>>
>> m=image 35622 udptl t38
>>^
>>
>> Provider uses
>>
&g
On 06/17/2017 at 02:18 PM, Michael Maier wrote:
> On 06/16/2017 at 04:00 PM, Joshua Colp wrote:
>> On Fri, Jun 16, 2017, at 10:49 AM, Michael Maier wrote:
>>
>>
>>
>>>
>>> t38modem and asterisk are using
>>>
>>> m=image 35622 udp
On 06/11/2017 at 04:39 PM Joshua Colp wrote:
> On Sun, Jun 11, 2017, at 11:34 AM, Michael Maier wrote:
>
>
>
>>>
>>> PJSIP uses a dispatch model. The request is queued up, acted on, and
>>> then that's it. The act of acting on it removes it from the queu
On 06/11/2017 at 06:51 PM Joshua Colp wrote:
> On Sun, Jun 11, 2017, at 01:47 PM, Joshua Colp wrote:
>> The distributor is in res/res_pjsip/pjsip_distributor.c, the distributor
>> function being the entry point. That function returning PJ_TRUE
>> indicates to PJSIP that it has been handled and no
On 06/14/2017 at 05:53 PM Joshua Colp wrote:
> On Wed, Jun 14, 2017, at 12:47 PM, Michael Maier wrote:
>
>
>
>>
>> I added this patch to see, if really all packages are are freed after
>> they have been processed:
>>
>> --- b/res/res_pjsip/pjsip_d
On 06/15/2017 at 08:15 AM Michael Maier wrote:
> On 06/14/2017 at 10:17 PM, Joshua Colp wrote:
>> On Wed, Jun 14, 2017, at 05:09 PM, Michael Maier wrote:
>>
>>
>>
>>>
>>> I can now say, that asterisk / pjsip seams to work *mostly* as expected.
&g
On 05/13/2017 at 07:21 AM Michael Maier wrote:
> On 05/12/2017 at 08:49 PM, Joshua Colp wrote:
>> On Fri, May 12, 2017, at 02:46 PM, Michael Maier wrote:
>>
>>
>>
>>>
>>> If I'm doing exactly the same call originated with another extension,
&
Am 16.06.2017 um 11:12 schrieb Joshua Colp:
On Fri, Jun 16, 2017, at 02:13 AM, Michael Maier wrote:
Has anybody any idea why asterisk drops the media stream in the 200 OK?
The channel has been T38_ENABLED before! Or is it necessary to add more
debug code? Who does the negotiating?
Only
On 06/18/2017 at 12:11 PM, Joshua Colp wrote:
> On Sun, Jun 18, 2017, at 06:00 AM, Michael Maier wrote:
>> Hello!
>>
>> unchanged asterisk crashes during udptl / t.38 negotiation with telekom
>> - they do not support t.38 / udptl.
>
> All Asterisk issues need
On 06/11/2017 at 01:29 PM Joshua Colp wrote:
> On Sun, Jun 11, 2017, at 08:16 AM, Michael Maier wrote:
>
>
>
>> I did some further investigations and fixed a local problem. Now,
>> asterisk is able to successfully activate T.38 - unfortunately just for
>> very
On 06/09/2017 at 08:44 PM Michael Maier wrote:
> On 06/08/2017 at 10:22 PM Michael Maier wrote:
>> Hello Joshua,
>>
>> thank you very much for your extremely quick answer! I really appreciate
>> your work and your friendly and your patient support!
>>
>>
&g
On 06/05/2017 at 09:32 PM Joshua Colp wrote:
> On Mon, Jun 5, 2017, at 04:26 PM, Michael Maier wrote:
>> On 06/05/2017 at 06:29 PM, Joshua Colp wrote:
>>> On Mon, Jun 5, 2017, at 01:22 PM, Michael Maier wrote:
>>>>
>>>> Do you have any idea where to
On 06/08/2017 at 10:22 PM Michael Maier wrote:
> Hello Joshua,
>
> thank you very much for your extremely quick answer! I really appreciate
> your work and your friendly and your patient support!
>
>
> On 06/07/2017 at 10:33 PM, Joshua Colp wrote:
>> On Wed, Jun 7,
Hello!
unchanged asterisk crashes during udptl / t.38 negotiation with telekom
- they do not support t.38 / udptl.
In detail:
fax client -> asterisk -> telekom -> easybell -> asterisk -> fax server
Fax server sends t.38 reinvite via asterisk to easybell.
Session Description Protocol
Hello!
I just implemented a jitterbuffer for pjsip in the dialplan in a SBC:
[fromtrunk]
exten => _[+0-9]!,1,Set(JITTERBUFFER(fixed)=default)
This jitterbuffer catches all calls coming from ISP.
My understanding is, that the incoming rtp stream in leg1a is now
buffered and handed out
On 06/05/2017 at 06:29 PM, Joshua Colp wrote:
> On Mon, Jun 5, 2017, at 01:22 PM, Michael Maier wrote:
>>
>> Do you have any idea where to start to look at? Adding additional output
>> in the source code? Which functions could be interesting? I may add own
>> d
Hello!
I'm facing the following scenario:
- Initial call opened to asterisk: SDP g722,alaw,ulaw
- Outgoing call to provider started with Invite / SDP alaw, g726 and
g729.
- Provider sends 183 Session progress SDP: g729, alaw
- Provider sends g729 rtp packages
But: there is no license to
On 11/01/2017 at 10:14 AM Antony Stone wrote:
> On Wednesday 01 November 2017 at 08:10:36, Michael Maier wrote:
>
>> Hello!
>>
>> I'm facing the following scenario:
>>
>> - Initial call opened to asterisk: SDP g722,alaw,ulaw
>>
>> - Outgoing call
Hello!
I'm working on a fax solution, which reports the result of each sent or
received fax to the database.
When using option "F" or not using option "f", it's clear which protocol
has been used. But if "f" is used, it could be g711 or t38. Is there any
variable which contains this information
On 05/21/2018 at 06:46 PM Andre Gronwald wrote:
> after completion you find ${FAXMODE} filled with audio or T38, depending
> on what has been used.
> hope that is what you are looking for.
Yes, that's exactly what I've been looking for!
Thanks,
Michael
--
641K 8809 0.000 0.026
=> This doesn't sound reliable to me: there are 188K packets and 16641K
of them are lost?! The Pct value is fluctuating between about 6009 and 9009.
Thanks,
Michael
>
>
> Am 11.02.2018 19:27 schrieb "Michael Maier" <m1278...@mailbox.org>:
>
&g
Hello,
could somebody please tell me the meaning of "Pct" as seen in asterisk cli:
...Receive. .Transmit..
CountLost Pct Jitter CountLost Pct Jitter RTT
Thanks,
Michael
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>
> -Ursprüngliche Nachricht-
> Von: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Michael Maier
> Gesendet: Montag, 12. Februar 2018 17:46
> An: asterisk-users@lists.digium.com
> Betreff: Re: [
Hello!
I'm just searching for a fax to email / email to fax open source based
complete solution which covers the following core features:
- high availability
- possibly multi tenant
- about 40,000 users
- about 100 lines parallel
- supports G.711 / T.38
- scriptable user management (reading,
On 08/09/2018 at 10:49 PM Michael Maier wrote:
> Hello!
>
> I'm using TOS as shown below with pjsip 13.22.0-rc1 (same with 13.21.1).
> Unfortunately, the TOS isn't set in reality any more (it used to work
> some time ago).
Got the problem.
I'm using freepbx and for the configurat
Hello!
I'm using TOS as shown below with pjsip 13.22.0-rc1 (same with 13.21.1).
Unfortunately, the TOS isn't set in reality any more (it used to work
some time ago).
Transport:
==
Transport:
Hi!
I just want to say, that 13.24.1 doesn't fix the problem described in the posts
above.
Regards,
Michael
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Hello!
An extension registered at asterisk 13.23 initiates an external call (pjsip).
After the Invite, the
callee (-> ISP) sends a
100 Trying
183 Session Progress (*without* SDP)
Asterisk now sends to the extension:
183 Session Progress (*with* SDP)
183
On 12.12.18 at 19:43 Joshua C. Colp wrote:
> On Wed, Dec 12, 2018, at 12:31 PM, Michael Maier wrote:
>
>
>
>>
>> The problem: The extension doesn't create a ringback locally, because
>> it most probably expects it to
>> be sent by the callee -
Hello!
I'm facing a problem concerning MWI. The problem is:
The phone switches off the MWI exactly at the moment the second NOTIFY package
for one voice mail
arrives.
The phone switches off MWI independently if the voice mail has been
acknowledged or not before.
The behavior in detail:
-
On 26.12.18 at 14:48 Michael Maier wrote:
> Hello!
>
> I'm facing a problem concerning MWI. The problem is:
>
> The phone switches off the MWI exactly at the moment the second NOTIFY
> package for one voice mail
> arrives.
>
> The phone switches off MWI independently
On 12.12.18 at 19:43 Joshua C. Colp wrote:
> On Wed, Dec 12, 2018, at 12:31 PM, Michael Maier wrote:
>
>
>
>>
>> The problem: The extension doesn't create a ringback locally, because
>> it most probably expects it to
>> be sent by the callee -
On 27.12.18 at 18:14 Joshua C. Colp wrote:
> On Thu, Dec 27, 2018, at 1:07 PM, Michael Maier wrote:
>> Hi!
>>
>> I just want to say, that 13.24.1 doesn't fix the problem described in
>> the posts above.
>
> You're going to need to file an issue[1] with traces
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 after a message
> has been listened
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
On 11.06.19 at 20:32 Luca Bertoncello wrote:
> Hi list!
>
> I use Asterisk 13.14.1 from Debian repository on a DSL from Deutsche
> Telekom.
>
> Asterisk works well, but I have really often an high delay (I understand
> it since the other party speak some seconds before he hears my question
> and
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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On 05.07.19 at 22:02 hw wrote:
>
> openssl verify -CAfile ca.pem asterisk.pem
> asterisk.pem: OK
>
>
> When I set tlsdontverifyserver=yes, it works (i. e. asterisk registers
> to the SIP provider and there is no error message). Otherwise I'm
> getting the error message and asterisk does not
On 06.07.19 at 22:16 hwilmer wrote:
Is there an advantage to using pjsip? What's needed for easybell with pjsip?
For easybell, I don't know of any advantage. But that's not very reliable, because I'm using easybell for dedicated requirements only. I'm
considering chan_sip legacy. I wouldn't
On 14.08.19 at 16:26 Dan Cropp wrote:
> We have a customer where their VM running Asterisk appears to have crashed.
> Fortunately, we had some debugging enabled.
> The asterisk messages file has this... (in notepad+ the blank line in the
> middle is all [NUL][NUL] [NUL][NUL])
>
> [08/12
On 14.08.19 at 18:12 Dan Cropp wrote:
Maybe because the machine is performing a file system check on some other
partitions in parallel and it's slowed down therefore?
Wouldn't /var/log/syslog show something like this if it's happening in parallel?
Well, it was just speculation. Is it even
Hello!
I encountered an outage of asterisk which showed like that:
- 2019-08-10 07:22:21 Asterisk start
- 2019-08-15 19:39:33 WARNING taskprocessor.c: The
'pjsip/outreg/ispPJSIP-0060' task processor queue reached 500 scheduled
tasks.
- 2019-08-15 19:39:34 WARNING
On 11.09.19 at 15:24 Joshua C. Colp wrote:
> On Wed, Sep 11, 2019, at 10:18 AM, basti wrote:
>> Hallo,
>> is there a Freepbx mailinglist? or can this be posted here?
>
> FreePBX does not have a mailing list. People use the community forum[1]
> instead.
>
> [1] https://community.freepbx.org/
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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Hello!
Following problem: If there are different trunks (-> different numbers and
users / passwords) to the same destination, asterisk (16.x) always uses the
same local tcp port for each connection. This is a problem with Deutsche
Telekom (they
want to have different local ports for different
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=
Hello all!
I'm reproducibly getting my *own MOH* if I should get the MOH of the Callee
instead. I can see this with asterisk 13 and 16 (and probably before, too). The
reason of the
wrong MOH is an in dialog reInvite received from trunk, which sends a SDP
containing
a=sendonly
After this
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[
Hello!
I'm just wondering what the RTT exactly means. Where are the exact measuring
points located?
> pjsip show channelstats
...Receive.
.Transmit..
BridgeId ChannelId UpTime.. Codec. CountLost Pct
On 15.05.20 at 14:31 Doug Lytle wrote:
> Google says Round Trip Time
>
> https://www.voip-info.org/asterisk-rtcp/
That doesn't answer my question (I know the abbreviation RTT). Therefore I'm
trying again:
I'm just wondering what the RTT *exactly* means. Where are the exact measuring
points
On 05.09.20 at 15:22 sean darcy wrote:
> asterisk-16.13.0-rc2. Fedora 32
>
> pjsip won't load because of undefined symbols:
This means, that your pjsip library doesn't match the asterisk binary. It's
best to remove the independent pjsip library and compile asterisk[1] with the
bundled pjsip
Hi Sergio
On 16.10.20 at 07:54 sergio wrote:
> Sometimes, linphone shows missed calls as missed. Look like asterisk
> replies with cause=487 that time, but I can't understand why.
>
> Grandstream always shows calls as missed ones.
>
> How should I investigate this?
You could try to
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
On 31.05.20 at 19:26 Jeff LaCoursiere wrote:
> Hi,
>
> I had posted this a few hours ago, but got caught in moderation for size. I
> trimmed down the pic and attached.
>
> I am on an Ubuntu 16 workstation, in an Ubuntu terminal window, ssh'ed to the
> PBX (amazon instance). You can see my
On 03.07.20 at 19:57 Luca Bertoncello wrote:
[...]
>> On the Gateway (Banana PI), where the Asterisk server also runs, the
>> load is about 0.50 during calls and it has a Gbps LAN.
>> I can't believe, the problem is here...
>
> So, now I know what was the problem and I solved it...
>
> The
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
On 13.07.20 at 00:17 Joshua C. Colp wrote:
> On Sun, Jul 12, 2020 at 7:12 PM Dan Jenkins wrote:
>
>> Asterisk 18 will have support based on this asterisk update Matt F did for
>> CommCon's sponsor slots
>>
>> https://youtu.be/eas1csaX-wc
>>
>>
> As well support will go into Asterisk 16 and 17 as
Am 22.06.20 um 16:48 schrieb Luca Bertoncello:
Hi list!
So, now I have a business contract and a technician was here to check
the DSL...
Nothing found, except that for 50Mbps I need now vectoring. Really
nice... A couple of years ago I could get 50Mbps without vectoring.
Of course, Deutsche
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. Probably there will be another problem. I expect
>>> packet size 1466 would pass and higher will have the same result. It
RTP-VoIP-packets never reach this size.
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
Hello!
On 20.10.20 at 14:00 Asterisk Development Team wrote:
> The Asterisk Development Team would like to announce the release of Asterisk
> 18.0.0.
> This release is available for immediate download at
> https://downloads.asterisk.org/pub/telephony/asterisk
I just tested the new codec
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
On 26.01.21 at 21:12 Ruisheng Peng wrote:
> Hi,
>
> I'm experimenting with Asterisk-16.14.0 on a CentOS7 box, and run into
> problems loading the SSL certificate to establish transport-tls. Tried
> self-signed certificate generated with ast_tls_cert under contrib/scripts
> and the one issued
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
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