Re: [asterisk-users] Voice "broken" during calls

[Luca Bertoncello]

Am 15.06.2020 23:15, schrieb Jeff LaCoursiere:

Hi again,

just a question, to be sure...


sudo tcpdump -i eth0 -s 0 -w /tmp/test0.pcap &
sudo tcpdump -i eth1 -s 0 -w /tmp/test1.pcap &


eth0 is my DSL interface and eth1 my phone interface?


Try to limit the traffic to just your phone call tests (to reduce the
size of the capture files).  Make all your tests, then:


Well, assuming eth0 is the DSL interface and eth1 the phone interface, I 
can so that:


tcpdump -i dsl0 -s 0 -w /tmp/test0.pcap host tel.t-online.de &
tcpdump -i phone0 -s 0 -w /tmp/test1.pcap host 192.168.200.xx (IP of my 
phone) &


is it correct?

Thanks
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Any api (agi/ari/ami) equivalent of "core show calls"?

[Bruce Ferrell]

On 6/14/20 4:52 PM, Antony Stone wrote:

On Monday 15 June 2020 at 00:41:14, Bruce Ferrell wrote:


Way back in the mists of time, I built my asterisk installation with SNMP
support.

Heh... I never even knew that was possible :)


That said, I actually prefer ARA/ARI to flat file configuration of endpoints
and dialplans.  Changes are more or less instantaneous and easily shared
between instances.

Agreed - ARA is a great system, and I really like that it can be combined with
flatfile configs on a single server.


The ODBC way is a pain, so I tend to just use the native MySQL method

Oh?

What makes you say ODBC is a pain?  I have two files (/etc/odbc.ini and
/etc/idbcinst.ini, which are 8 lines and 3 lines in size respectively) and I
had to install one file /usr/local/lib/libmaodbc.so to make it work with
MariaDB.


ODBC can get fussy about language definitions between the DB and ODBC.  ODBC make's it so you can use ANY db, but try Informix or Oracle... Just LOADS of fun getting the language 
code page settings right (Yes, I actually had a customer want Informix - SHUDDER).


The MySQL connection has it's own configuration fuss, but I understand it and 
HATE trying to debug ODBC configs.



What's easier about the native MySQL method?


for ARA configuration as well as CDR collection.  CDR reports are just a SQL
query away.

Yes:

a) efficient

b) can be done on a machien remote from the call processing

c) is realtime - call ends, CDRs are immediately available for analysis

d) can even include triggers on DB updates, for example to raise anti-fraud
alerts.


YES!  and most non-MySQL/Marias type don't realize that those triggers and 
stored procedures can actually execute routines external to the DB... At least 
in MySQL/MariaDB



Regards,


Antony.





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Re: [asterisk-users] Voice "broken" during calls

[Luca Bertoncello]

Am 15.06.2020 um 23:15 schrieb Jeff LaCoursiere:

Hi

> Yes, sure, please use (replace with correct interface names):
> 
> sudo tcpdump -i eth0 -s 0 -w /tmp/test0.pcap &
> sudo tcpdump -i eth1 -s 0 -w /tmp/test1.pcap &

OK, I'll do it this evening (german time), since now I must go to the
office...

Thanks
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] includes with time and timezone.

[Antony Stone]

On Monday 15 June 2020 at 23:30:28, John T. Bittner wrote:

> Hello,
> 
> I cannot find much on examples but I did find one in Russian that shows
> this to use + or - the time difference from GMT. I have been testing and
> it does not work.
> 
> 1st question do includes work with timezone
> 
> include =>  day,08:00-17:00,mon-fri,*,*,[+5]

Omit the square brackets.

> What is the formatting for timezone in gotoiftime.
> 
> GotoIfTime(times,weekdays,mdays,months,[timezone]?[labeliftrue:[labeliffals
> e]])

The square brackets simply identify optional items - they are not included in 
themselves.  This is a fairly common convention in software documentation, I 
think.

So, just as you would not put a [ after the ? sign, you do not put [ ] around 
the timezone.


Regards,


Antony.

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[asterisk-users] includes with time and timezone.

[John T. Bittner]

Hello,

I cannot find much on examples but I did find one in Russian that shows this to 
use + or - the time difference from GMT.
I have been testing and it does not work.

1st question do includes work with timezone

include =>  day,08:00-17:00,mon-fri,*,*,[+5]
Not sure on the formatting, is it correct ? ... I tried without the brackets... 
that also doesn't work.

If not supported in includes

What is the formatting for timezone in gotoiftime.

GotoIfTime(times,weekdays,mdays,months,[timezone]?[labeliftrue:[labeliffalse]])


Any helps is much appreciated.

Thanks

John Bittner
CTO
[xaccellogoemail]
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
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Re: [asterisk-users] Voice "broken" during calls

[Jeff LaCoursiere]

On 6/15/20 2:19 PM, Luca Bertoncello wrote:

Am 15.06.2020 um 20:15 schrieb Jeff LaCoursiere:

Hi Jeff,


We are working on a product to analyze pcap files of VoIP calls.  So far
it does a reasonable job of analyzing the frequency distribution of
packets in both directions, pointing out which direction packet loss /
bad jitter occurs.  If you can trap the traffic on the outside and the
inside of your Banana Pi and send me the pcap files, I would be happy to
run it through our analyzer as further information for you.  If it shows
DTK isn't sending packets when it should, that will be obvious, and you
can send to them as solid evidence of their guilt :)

Thank you for your offer.
Could you say me which options I should pass to tcpdump to get all
information you need?

Yes, sure, please use (replace with correct interface names):

   sudo tcpdump -i eth0 -s 0 -w /tmp/test0.pcap &
   sudo tcpdump -i eth1 -s 0 -w /tmp/test1.pcap &


Try to limit the traffic to just your phone call tests (to reduce the 
size of the capture files).  Make all your tests, then:


   sudo killall tcpdump
   tar cvzf /tmp/tests.tgz /tmp/test?.pcap


Send /tmp/tests.tgz to me by email, or leave somewhere I can download.  
I'll run the analysis tonight and send the results to the list.


Cheers,

--

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Mobile: *+1 815.546.6599*
Email:  *j...@stratustalk.com* 
Website:*https://www.stratustalk.com*
Address:*One Freedom Square
13th Floor
Reston, VA 20190*

 
 




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Re: [asterisk-users] Voice "broken" during calls

[Luca Bertoncello]

Am 15.06.2020 um 22:30 schrieb Antony Stone:

>> What do you mean? In which sense this is "significant"?
> 
> Because if/when the Thomson is registered directly to DT, then Asterisk is 
> not 
> doing anything on the Banana.

Yes, I think it too

>> What do you mean now? If I can use the full available band or if I can
>> download exactly 50Mbs?
> 
> I don't see the difference - I'm asking whether the Banana can manage to 
> route 
> 50Mbps of traffic.

I think, it should be possible... the BananaPI has a Gbit ethernet...

>> The answer to the first question is: YES! That's why I use a traffic
>> shaper... ;)
> 
> So, you're saying that your Banana Pi *can* provide the full 50Mbps 
> throughput 
> if you don't enable the traffic shaper?

If I try to transfer a big file between two devices in different VLANs
(through the BananaPI as Gateway), there is no problem.
I don't really measured it, but it's bigger than 50Mbps...

>> The answer to the second question is: NO. I made a speedtest right now
>> and I get only ~18Mbps download.
> 
> So, is it the traffic shaper which is imposing this limit?

No, I tried the test disabling the traffic shaper, too... no changes...

> I'm very much agreeing with you here, that DT appears to be the problem, and 
> I 
> think Jeff's suggestion / offer to capture the audio data and do an analysis 
> on 
> it would likely show that.

OK!

Thanks
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Voice "broken" during calls

[Antony Stone]

On Monday 15 June 2020 at 22:27:51, Luca Bertoncello wrote:

> I checked the xDSL-statistics of my DSL-Modem (which use the BananaPI to
> establish the PPPoE connection):
> 
>   adsl: ADSL driver and PHY status
> Status: Showtime
> Last Retrain Reason:  2
> Last initialization procedure status: 0
> Max:  Upstream rate = 1709 Kbps, Downstream rate = 19888 Kbps
> Bearer:   0, Upstream rate = 1626 Kbps, Downstream rate = 20113 Kbps

Oh!

> So it seems, that my connection is about the half of the theorical one...

Right.

> I think, I must call Deutsche Telekom, but since I'll change my contract
> at 18.06., I'll wait some days. Then I'll have a "business" contract,
> and I hope I don't must speak with someone that can just say "you have
> to reboot your Fritzbox. What? You don't have a Fritzbox? That's not
> possible. Please check your Fritzbox, I can't reach it"... ;)

Hehe :)

Good luck,


Antony.

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Re: [asterisk-users] Voice "broken" during calls

[Antony Stone]

On Monday 15 June 2020 at 21:50:36, Luca Bertoncello wrote:

> Am 15.06.2020 um 21:28 schrieb Antony Stone:
> > On Monday 15 June 2020 at 21:19:51, Luca Bertoncello wrote:
> >> But I'm not really sure, that Asterisk could be the problem, since, as I
> >> said, the problem happens even if I connect the phone direct to the
> >> server of Telekom...
> > 
> > I think that is significant, even if the routing is still going through
> > the Banana.
> 
> What do you mean? In which sense this is "significant"?

Because if/when the Thomson is registered directly to DT, then Asterisk is not 
doing anything on the Banana.

The fact that you get the problem with the Thomson registered directly to DT, 
but with the traffic still going through the Banana, indicates to me that 
"Asterisk running on the Banana" is not the problem.

> > Can you get the full 50Mbps through the Banana when you're doing a
> > download of something biggish?
> 
> What do you mean now? If I can use the full available band or if I can
> download exactly 50Mbs?

I don't see the difference - I'm asking whether the Banana can manage to route 
50Mbps of traffic.

> The answer to the first question is: YES! That's why I use a traffic
> shaper... ;)

So, you're saying that your Banana Pi *can* provide the full 50Mbps throughput 
if you don't enable the traffic shaper?

> The answer to the second question is: NO. I made a speedtest right now
> and I get only ~18Mbps download.

So, is it the traffic shaper which is imposing this limit?

> >> Last but not least: I tried calls via Skype and WhatsApp (with my phone
> >> in my WLAN). No problem and very good quality, so the BananaPI does not
> >> have any problem to manage the data transfer, isn't it?
> > 
> > The big difference there, though, is that Asterisk (running on the
> > Banana) is not handling the call, so you have the traffic being routed
> > through the Banana, but Asterisk is not being asked to do anything with
> > it in the middle.
> 
> Yes, and the voice quality is excellent, but *just* if I'm *not* using
> DT... As soon as I use DT (configuring the phone to connect directly to
> the server) both partners can hear the "interruptions"...

Right.

I'm very much agreeing with you here, that DT appears to be the problem, and I 
think Jeff's suggestion / offer to capture the audio data and do an analysis on 
it would likely show that.

Just one suggestion regarding that: I recommend doing two packet captures; one 
with Asterisk registered to DT and the Thomson registered to Asterisk, and the 
other with the Thomson registered to DT but the traffic still going through the 
Banana.


Regards,


Antony.


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Re: [asterisk-users] Voice "broken" during calls

[Luca Bertoncello]

Am 15.06.2020 um 21:50 schrieb Luca Bertoncello:

> What do you mean now? If I can use the full available band or if I can
> download exactly 50Mbs?
> The answer to the first question is: YES! That's why I use a traffic
> shaper... ;)
> The answer to the second question is: NO. I made a speedtest right now
> and I get only ~18Mbps download.

And some other information, too.

I checked the xDSL-statistics of my DSL-Modem (which use the BananaPI to
establish the PPPoE connection):

adsl: ADSL driver and PHY status
Status: Showtime
Last Retrain Reason:2
Last initialization procedure status:   0
Max:Upstream rate = 1709 Kbps, Downstream rate = 19888 Kbps
Bearer: 0, Upstream rate = 1626 Kbps, Downstream rate = 20113 Kbps
Bearer: 1, Upstream rate = 0 Kbps, Downstream rate = 0 Kbps

So it seems, that my connection is about the half of the theorical one...

I think, I must call Deutsche Telekom, but since I'll change my contract
at 18.06., I'll wait some days. Then I'll have a "business" contract,
and I hope I don't must speak with someone that can just say "you have
to reboot your Fritzbox. What? You don't have a Fritzbox? That's not
possible. Please check your Fritbox, I can't reach it"... ;)

Bye
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Voice "broken" during calls

[Luca Bertoncello]

Am 15.06.2020 um 21:28 schrieb Antony Stone:
> On Monday 15 June 2020 at 21:19:51, Luca Bertoncello wrote:
> 
>> But I'm not really sure, that Asterisk could be the problem, since, as I
>> said, the problem happens even if I connect the phone direct to the
>> server of Telekom...
> 
> I think that is significant, even if the routing is still going through the 
> Banana.

What do you mean? In which sense this is "significant"?

>> Well, during the calls, the BananaPI has a load of max 1, and it have 2
>> cores...
> 
> Multi-core CPUs are only a benefit if you can run separate applications (or 
> at 
> least separate threads of an application) on the separate cores.

Yes, of course.

> I'm not sure Asterisk can do this for a single call.

But of course the BananaPI can handle that Asterisk uses a core and
reserve the rest (or part of it) to manage the PPPoE connection...

>> The LAN interface is Gbps, and my DSL is only 50Mbps, so it is not
>> possible to get it full of band...
> 
> Can you get the full 50Mbps through the Banana when you're doing a download 
> of 
> something biggish?

What do you mean now? If I can use the full available band or if I can
download exactly 50Mbs?
The answer to the first question is: YES! That's why I use a traffic
shaper... ;)
The answer to the second question is: NO. I made a speedtest right now
and I get only ~18Mbps download.

>> Last but not least: I tried calls via Skype and WhatsApp (with my phone
>> in my WLAN). No problem and very good quality, so the BananaPI does not
>> have any problem to manage the data transfer, isn't it?
> 
> The big difference there, though, is that Asterisk (running on the Banana) is 
> not handling the call, so you have the traffic being routed through the 
> Banana, 
> but Asterisk is not being asked to do anything with it in the middle.

Yes, and the voice quality is excellent, but *just* if I'm *not* using DT...
As soon as I use DT (configuring the phone to connect directly to the
server) both partners can hear the "interruptions"...

Thanks
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Voice "broken" during calls

[Luca Bertoncello]

Am 15.06.2020 um 21:24 schrieb Antony Stone:
> On Monday 15 June 2020 at 18:55:23, Luca Bertoncello wrote:
> 
>> Absolutly *no changes* on the behaviour compared with my Thomsons...
> 
> Okay, I'm glad we can rule out the specific make / model of phone - that 
> would 
> have been bizarre.

Yes, I really didn't believe, it could be the problem, but know is
better than believe... ;)

>> 2) Asterisk seems not to be the problem, too, since I have the same
>> behaviour if I connect to phone directly to the server of Deutsche Telekom.
> 
> Is that also via the Banana, or with the phone directly on a DSL modem?

Always via the BananaPI
I cannot connect the phone directly to a DSL modem, since the phone does
not have any program to establish a PPPoE

>> 4) The problem happens *only* on active call, not by voicemail.
> 
> So, only when there are two SIP clients active on each side of the Asterisk 
> server...

Yes, you can say it, too... I think, this is the same with other words...

>> 4a) To test it I read a text and my partner just listen it, and then he
>> read a text and I listen it. *No* simulaneously speak!
> 
> But, what were the results - each of you could hear the other perfectly well?

No! Maybe I didn't explained well...
All the tests I done with my father in law, during that we experienced
the "interruptions" were made as I described, one of us spoke, the other
counts the "interruptions".

> This sounds interesting - more ideas below.
> 
>> 5) A *single call* (since I couldn't reproduce it anymore), made using
>> my Android phone as SIP-client connected to my Asterisk, had not the
>> problem. Any other try to call someone using my mobile phone via SIP had
>> the problem.
> 
> You seem to have the problem in general, so a single (or small number of) 
> instances of no problem doesn't mean there isn't something to be resolved.

Yes, this is a general problem, happening using the phone of my wife,
too, btw...

>> I really think, the problem should be by Deutsche Telekom...
> 
> Especially since you say you do not get the problem when you have calls in 
> via 
> Messagenet for your Italian calls.

Sometimes I experienced problem with MessageNet, too, but not so
frequently as with Deutsche Telekom...

> What happens if:
> 
> a) you call someone external, speak for about 30 seconds without them making 
> any sound, then they start speaking *at the same time as you*, then you stop 
> talking and they carry on.

Are these 30 seconds of "silence" important? If not, it happens very
frequently that both partners speak at the same time. In this case, the
quality is a little bit less than normal, but very very little!, and I
hear "interruptions" in this case, too...

> b) exactly the same, except this time they call you, so it's an inbound call.

I experienced these "interruptions" if I call even if I receive the
call, if this was your question...

> Do you get good quality while only one person speaks, and bad while both do?  
> Does the quality return to good when one person stops speaking?

Actually I don't get a good quality at all, expect for some isolated
calls...

Thanks
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Voice "broken" during calls

[Antony Stone]

On Monday 15 June 2020 at 21:19:51, Luca Bertoncello wrote:

> But I'm not really sure, that Asterisk could be the problem, since, as I
> said, the problem happens even if I connect the phone direct to the
> server of Telekom...

I think that is significant, even if the routing is still going through the 
Banana.

> Well, during the calls, the BananaPI has a load of max 1, and it have 2
> cores...

Multi-core CPUs are only a benefit if you can run separate applications (or at 
least separate threads of an application) on the separate cores.

I'm not sure Asterisk can do this for a single call.

> The LAN interface is Gbps, and my DSL is only 50Mbps, so it is not
> possible to get it full of band...

Can you get the full 50Mbps through the Banana when you're doing a download of 
something biggish?

> Last but not least: I tried calls via Skype and WhatsApp (with my phone
> in my WLAN). No problem and very good quality, so the BananaPI does not
> have any problem to manage the data transfer, isn't it?

The big difference there, though, is that Asterisk (running on the Banana) is 
not handling the call, so you have the traffic being routed through the Banana, 
but Asterisk is not being asked to do anything with it in the middle.


Antony.

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Re: [asterisk-users] Voice "broken" during calls

[Antony Stone]

On Monday 15 June 2020 at 18:55:23, Luca Bertoncello wrote:

> Absolutly *no changes* on the behaviour compared with my Thomsons...

Okay, I'm glad we can rule out the specific make / model of phone - that would 
have been bizarre.

> I try to summarize:
> 
> 1) Phones are not the problem, since 3 phones of 2 different
> companies/model have the same issue.

Good (if you see what I mean).

> 2) Asterisk seems not to be the problem, too, since I have the same
> behaviour if I connect to phone directly to the server of Deutsche Telekom.

Is that also via the Banana, or with the phone directly on a DSL modem?

> 3) Traffic shaping seems not to be the problem, too, since I tried to
> deactivate it.

Good test / check.

> 4) The problem happens *only* on active call, not by voicemail.

So, only when there are two SIP clients active on each side of the Asterisk 
server...

> 4a) To test it I read a text and my partner just listen it, and then he
> read a text and I listen it. *No* simulaneously speak!

But, what were the results - each of you could hear the other perfectly well?

This sounds interesting - more ideas below.

> 5) A *single call* (since I couldn't reproduce it anymore), made using
> my Android phone as SIP-client connected to my Asterisk, had not the
> problem. Any other try to call someone using my mobile phone via SIP had
> the problem.

You seem to have the problem in general, so a single (or small number of) 
instances of no problem doesn't mean there isn't something to be resolved.

> I could *not* test connecting to the server of Deutsche Telekom using
> the Internet connection of someone other, since Telekom bounds my
> VoIP-login to my IP.

Right.

> I really think, the problem should be by Deutsche Telekom...

Especially since you say you do not get the problem when you have calls in via 
Messagenet for your Italian calls.

> What is your opinion? Do you see some other tests I should try?

Yes.

I'm intrigued by the "only one party speaking at a time" test you did.

What happens if:

a) you call someone external, speak for about 30 seconds without them making 
any sound, then they start speaking *at the same time as you*, then you stop 
talking and they carry on.

b) exactly the same, except this time they call you, so it's an inbound call.

Do you get good quality while only one person speaks, and bad while both do?  
Does the quality return to good when one person stops speaking?


Regards,


Antony.

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Re: [asterisk-users] Voice "broken" during calls

[Luca Bertoncello]

Am 15.06.2020 um 20:15 schrieb Jeff LaCoursiere:

Hi Jeff,

> We are working on a product to analyze pcap files of VoIP calls.  So far
> it does a reasonable job of analyzing the frequency distribution of
> packets in both directions, pointing out which direction packet loss /
> bad jitter occurs.  If you can trap the traffic on the outside and the
> inside of your Banana Pi and send me the pcap files, I would be happy to
> run it through our analyzer as further information for you.  If it shows
> DTK isn't sending packets when it should, that will be obvious, and you
> can send to them as solid evidence of their guilt :)

Thank you for your offer.
Could you say me which options I should pass to tcpdump to get all
information you need?

But I'm not really sure, that Asterisk could be the problem, since, as I
said, the problem happens even if I connect the phone direct to the
server of Telekom...

> Beyond that, are you certain you aren't taxing the Banana Pi?  It really
> *should* be able to handle a single call while handling your LAN's
> routing/firewall tasks, but you are probably skating the edge.  The
> results of the above might point out that the Pi isn't *sending* packets
> it should be, or sending them way late, in which case the issue is
> actually your hardware.

Well, during the calls, the BananaPI has a load of max 1, and it have 2
cores...
The LAN interface is Gbps, and my DSL is only 50Mbps, so it is not
possible to get it full of band...

And during the test as I connected the phone to the Telekom servers the
load of the BananaPI was lower as 1.

Last but not least: I tried calls via Skype and WhatsApp (with my phone
in my WLAN). No problem and very good quality, so the BananaPI does not
have any problem to manage the data transfer, isn't it?

Regards
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Voice "broken" during calls

[Antony Stone]

On Monday 15 June 2020 at 20:15:22, Jeff LaCoursiere wrote:

> Beyond that, are you certain you aren't taxing the Banana Pi?  It really
> *should* be able to handle a single call while handling your LAN's
> routing/firewall tasks, but you are probably skating the edge.

I have a Banana Pi R1 (the version with 5 ethernet ports, actually VLANned off 
a single physical interface), and all my Internet traffic goes through it, 
including SIP to NetCologne, who is my local telephony and connectivity 
provider here.

My only major difference from Luca's situation is that I don't run Asterisk on 
this machine (although I do run it on a Raspberry Pi inside my network).

I don't experience any of the problems Luca reports - the worst I get from 
time to time is some briefly reduced call quality if someone happens to do a 
large file download while I'm on the phone (the download speed here is 'only' 
25Mbps, so it's easy enough to saturate for a short time with a browser).

I certainly agree with the idea of doing a packet analysis on both sides of 
the Banana, though, to see whether DT is causing the problem, or whether the 
hardware's just not up to the job.


Regards,


Antony

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Re: [asterisk-users] Voice "broken" during calls

[Jeff LaCoursiere]

Hi,

We are working on a product to analyze pcap files of VoIP calls. So far 
it does a reasonable job of analyzing the frequency distribution of 
packets in both directions, pointing out which direction packet loss / 
bad jitter occurs.  If you can trap the traffic on the outside and the 
inside of your Banana Pi and send me the pcap files, I would be happy to 
run it through our analyzer as further information for you.  If it shows 
DTK isn't sending packets when it should, that will be obvious, and you 
can send to them as solid evidence of their guilt :)


Beyond that, are you certain you aren't taxing the Banana Pi?  It really 
*should* be able to handle a single call while handling your LAN's 
routing/firewall tasks, but you are probably skating the edge.  The 
results of the above might point out that the Pi isn't *sending* packets 
it should be, or sending them way late, in which case the issue is 
actually your hardware.


Cheers,

*Jeff LaCoursiere*
STRATUSTALK, INC. / CTO

Phone:  *+1 703.496.4990 x108*
Mobile: *+1 815.546.6599*
Email:  *j...@stratustalk.com* 
Website:*https://www.stratustalk.com*
Address:*One Freedom Square
13th Floor
Reston, VA 20190*

 
 



On 6/15/20 11:55 AM, Luca Bertoncello wrote:

Am 14.06.2020 um 17:33 schrieb Luca Bertoncello:

Hi

So, I got a phone (Elmeg IP290) from a collegue and tested it...


What I'll do tomorrow with a test phone is:

1) connecting it to my Asterisk and try to make a call
2) connecting it directly to the servers of Deutsche Telekom (using my
network) and try to make a call

Absolutly *no changes* on the behaviour compared with my Thomsons...

I try to summarize:

1) Phones are not the problem, since 3 phones of 2 different
companies/model have the same issue.
2) Asterisk seems not to be the problem, too, since I have the same
behaviour if I connect to phone directly to the server of Deutsche Telekom.
3) Traffic shaping seems not to be the problem, too, since I tried to
deactivate it.
4) The problem happens *only* on active call, not by voicemail.
4a) To test it I read a text and my partner just listen it, and then he
read a text and I listen it. *No* simulaneously speak!
5) A *single call* (since I couldn't reproduce it anymore), made using
my Android phone as SIP-client connected to my Asterisk, had not the
problem. Any other try to call someone using my mobile phone via SIP had
the problem.

I could *not* test connecting to the server of Deutsche Telekom using
the Internet connection of someone other, since Telekom bounds my
VoIP-login to my IP.

I really think, the problem should be by Deutsche Telekom...

What is your opinion? Do you see some other tests I should try?

Thanks a lot
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Voice "broken" during calls

[Luca Bertoncello]

Am 14.06.2020 um 17:33 schrieb Luca Bertoncello:

Hi

So, I got a phone (Elmeg IP290) from a collegue and tested it...

> What I'll do tomorrow with a test phone is:
> 
> 1) connecting it to my Asterisk and try to make a call
> 2) connecting it directly to the servers of Deutsche Telekom (using my
> network) and try to make a call

Absolutly *no changes* on the behaviour compared with my Thomsons...

I try to summarize:

1) Phones are not the problem, since 3 phones of 2 different
companies/model have the same issue.
2) Asterisk seems not to be the problem, too, since I have the same
behaviour if I connect to phone directly to the server of Deutsche Telekom.
3) Traffic shaping seems not to be the problem, too, since I tried to
deactivate it.
4) The problem happens *only* on active call, not by voicemail.
4a) To test it I read a text and my partner just listen it, and then he
read a text and I listen it. *No* simulaneously speak!
5) A *single call* (since I couldn't reproduce it anymore), made using
my Android phone as SIP-client connected to my Asterisk, had not the
problem. Any other try to call someone using my mobile phone via SIP had
the problem.

I could *not* test connecting to the server of Deutsche Telekom using
the Internet connection of someone other, since Telekom bounds my
VoIP-login to my IP.

I really think, the problem should be by Deutsche Telekom...

What is your opinion? Do you see some other tests I should try?

Thanks a lot
Luca Bertoncello
(lucab...@lucabert.de)

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