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?
Thanks a lot
Luca Bertoncello
(lucab...@lucabert.de
ug 16 18:34:37.51: [26819]: <-- [16:ATDT0177yyy\r]
Aug 16 18:34:52.66: [26819]: --> [10:NO CARRIER]
Aug 16 18:34:52.66: [26819]: SEND FAILED: JOB 39 DEST 0177yyy ERR
[2] No carrier detected
Aug 16 18:34:53.66: [26819]: <-- [5:ATH0\r]
Aug 16 18:34:53
Am 31.12.2021 um 16:04 schrieb Antony Stone:
Hi Antony
> Check the Dial() command which places the call to the phone. Does it contain
> the "c" option?
So, I tested it right now and it works... Just removing the "c"...
Thanks a lot for your help and of course happy
Am 31.12.2021 um 16:07 schrieb Luca Bertoncello:
> I'll try to remove it, but I can't test it today...
>
> I'll let you know if it works.
At least a call without anser does not contain the Header anymore...
I'll ask if the number is shown in the missed calls.
Regards
Luca Bertoncel
E-Mail der AB nicht den Namen steht
exten => _529874,n,VoiceMail(74,us)
exten => _529874,n,Hangup
I'll try to remove it, but I can't test it today...
I'll let you know if it works.
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
--
__
${UNIQUEID} / DATE:
${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}))
exten => s,n,System(echo "Verpasster Anruf vom ${CALLERID(NUM)} um
${STRFTIME(${EPOCH},,%H:%M)}" | mail -s "Verpasster Anruf" i...@xxx.de)
exten => h,1,GotoIf($[“${DIALSTATUS}” = “ANSWER”]?done)
r does not change anymore.
Last very strange problem is, that the list of missed calls on the phone
is always empty...
But it can be a problem of the phone hisself...
Maybe has someone an idea?
The phone is a Snom 821-SIP
Thanks and happy new year!
Luca Bertoncello
(lucab...@lucab
s being the source
> of the problem, which I think is good.
Well, this means, that the problem is in the Asterisk... Very huge part
of the infrastructure... :(
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth a
one_ "Ringing" and
sends the phone _two_ "Ringing", the second one with the
P-Asserted-Identity...
Maybe help it to identify the problem?
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth an
to call me from the phone when I sniff the traffic...
I hope, I find someone tomorrow.
Regards
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the ne
and what you mean...
You mean that I should compare what the "180 ringing" in the internal
network (phone to asterisk) and the external one (asterisk to Telekom)?
If so, then I have to check again, since I only sniffed the internal
traffic...
If not, I didn't understand what you m
LAN for the phones.
All traffic captured.
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.as
L, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces
Contact:
Content-Length: 0
So, I see, there is a "P-Asserted-Identity"... But I can't understand why...
Any idea?
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
--
___
I don't see anything strange...
Btw, what do you mean with "180 response"?
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new As
Am 28.12.2021 14:30, schrieb Luca Bertoncello:
Hi again,
If I call a number I can see in the display the called number, after a
few seconds the number changes to the own numer.
After hangup I just see my own number in the call log.
The same if I receive a call.
Very very strange
seconds the number changes to the own numer.
After hangup I just see my own number in the call log.
The same if I receive a call.
On the old Server (with Asterisk 11.7.0) with the same phones there was
no problem.
Do someone have any idea what can be the problem?
Thanks a lot
Luca Bertoncello
Am 06.11.2021 um 21:15 schrieb Łukasz Grzywański:
Hi Łukasz,
Dziękuję
> two legs in this same context
> ( exten => _03529123456,n,Dial(local/123456@main_incoming,,xX) )
>
> PJSIP/pbxmichael_in-0418
> and
> Local/123456@main_incoming-0268
>
> [main_incoming]
> exten =>
e") in new stack
-- Executing [s@noanswer:1]
NoOp("Local/123456@main_incoming-0268;2", "UID CALL: 1636222382.6032
/ DATE: 20211106-191306)") in new stack
-- Executing [s@noanswer:2]
System("Local/123456@main_incoming-0268;2", "echo "
Am 06.11.2021 um 14:43 schrieb Frank Vanoni:
Hi Frank
> On Fri, 2021-11-05 at 10:50 +0100, Luca Bertoncello wrote:
>
>> 1) The E-Mails will be sent "double"
>
> It sends the first mail by executing "noanswer,2" and a second mail
> because because
)
exten => h,n(done),NoOp()
exten => h,n,HangUp()
...
It works, but I have two problems:
1) The E-Mails will be sent "double"
2) The E-Mails will be sent for outgoing unanswered calls, too.
Do someone has an idea what is
Am 03.11.2021 um 21:34 schrieb Antony Stone:
> On Wednesday 03 November 2021 at 21:29:46, Luca Bertoncello wrote:
>
>> I tried so:
>>
>> exten => h,n(hang),Gosub(noanswer,s,1)
>
> The n there should be 1, surely?
Ach, you're right!
Now it works!
Tha
n,Dial(SIP/74,39,RcxX)
exten => _xx,n,Verbose(2,Voicemail for Main)
exten => _xx,n,Set(CALLERID(name)=)
exten => _xx,n,Gosub(noanswer,s,1)
exten => _xx,n,VoiceMail(74,us)
exten => _xx,n,Ha
email, the
Subrouting "noanswer" will not called...
Any ideas?
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community
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?
>>>
>>> Tak
.de,,R" and it does NOT work...
Is it correct, that I have to leave "sip:..."?
Thank you very much for your help!!
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth and Colocation Provided by http://www.ap
is it correct?
> The script unregisters and registers the telekom trunks, if a change is
> detected. This is done as long as there is no call active. This works
> for me - but may not wort for others - feel free to change the code.
OK, I'll check it...
>
tel.t-online.de in my Bind with
these settings? Looks like dangerous, if they changes something...
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out
tion?
Thanks a lot
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to
Hi!
I have a little problem with the given phone...
Do someone know it? My problem is that I'd like to display the name of
the caller (if it is saved in the address book, of course), but it
always display just the number...
Thanks
Luca Bertoncello
(lucab...@lucabert.de
Hi list!
Am 22.06.2020 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
to understand/learn how to check
the involved parts and search for the problem?
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk
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. Probably there will be another problem. I expect
>>>> packet
genet/5
register => lucabertoncello:x@rebvoice/lucabertoncello
jbenable = no
jbmaxsize = 200
jbresyncthreshold = 1000
jbimpl = fixed
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth and Colocation Provided by
and a
peer connected via LTE and the other in LAN, then maybe it's possible to
find the problem...
But if you have any other idea, I'm very happy to hear it! ;)
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth
he list is an expert with iptables and can check it?
I know this program, but I'm not really an expert...
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check
, communication with
both peers in the same interface work correct, but maybe my firewall
script...
If you can reproduce this can you send me a few more packet traces,
from each of the VLAN interfaces involved?
Of course, I can do that!
Maybe I get it this evening.
Regards
Luca Bertoncello
(lucab
akets in the internal
networks...
Thanks a lot
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community
... :(
Everyway: you think, my network works as expected? At least the part
using DSL?
Any idea, where could be the problem?
Thanks a lot
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth and Colocation Provided by http://www.api
transmitted, 0 received, +4 errors, 100% packet loss, time
3965ms
pipe 2
With paket size of 1464 it works...
You know MTU is a size of l2 frame, so using ipv6 you are able to use
higher payload sizes because of ip header size.
OK, thanks!
Luca Bertoncello
(lucab...@lucabert.de
-flags SYN,RST SYN -j TCPMSS --set-mss 128
?
Or I just have to use:
iptables -A FORWARD -p tcp --tcp-flags SYN,RST SYN -j TCPMSS --set-mss
128
instead of:
iptables -A FORWARD -p tcp --tcp-flags SYN,RST SYN -j TCPMSS
--clamp-mss-to-pmtu
?
Thanks
Luca Bertoncello
(lucab...@lucabert.de
Am 23.06.2020 08:43, schrieb Luca Bertoncello:
And another thing, I discovered right now...
Could you suggest me something to restrict the problem?
Currently, I think the problem can be:
1) on Asterisk
2) on my Gateway/Firewall
A couple of years ago I added this entry in my firewall:
/sbin
Am 22.06.2020 20:09, schrieb Luca Bertoncello:
A couple of other ideas...
Conclusion (maybe!): it can *not* be a problem in the DSL connection
and
*maybe* it is not a problem in the communication with the Server of
Deutsche Telekom, since I have many problems to communicate between two
peers
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
I checked it, and I see, that the maximum I can use is a paket size of
1464 with all hosts via IPv4
gt; ping.
I don't understand what you mean, could you explain?
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk communi
hould I reduce the MTU?!?
Maybe I didn't understood what you mean...
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new A
A thing I forgot to report...
My Asterisk listen on an high port (*not* 5060), since I had many
problems in the past with someone trying to use my Asterisk with brute
force attack...
I really don't think, this can be the problem, but better to report all...
Regards
Luca Bertoncello
(lucab
too...
Regarding the ping time: wich line do you have? I have a DSL 50Mbps.
Maybe your times are better due to a faster line?
What is your opinion about the tests I did today with the friend and his
phone as VoIP-peer?
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
--
_
ons?
2) assuming are my conclusions correct, can someone suggest me where can
I search the problem?
Thanks a lot
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check o
e "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)
--
_
-- Bandwidth and Colocation Provided by ht
I can do...
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New
s contract).
The quality is disturbed from the first second...
I had the problem, that the connection will be *dropped* after 15
minutes, and I solved it with "session-timers = refuse"
Bye
Luca Bertoncello
(
Am 16.06.2020 10:48, schrieb Antony Stone:
On Tuesday 16 June 2020 at 08:18:51, Luca Bertoncello wrote:
> 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?
Well, one is i
xx (IP of my
phone) &
is it correct?
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://commun
since now I must go to the
office...
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.o
mit?
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
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 m
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
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
> ha
nage the data transfer, isn't it?
Regards
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.
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 Telek
erisk 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
Thanks a lot for your help
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth and Colocation Provided
it does not depend
> on your DSL account (as it is standard with most other VoIP providers).
OK, I really don't think I want to subscribe this option just to check
if the problem is in my account... :D
Any other suggestion how to find *where* the problem is?
Thanks
Luca Bertoncello
, but the problem exists an almost all calls, incoming or outgoing,
no matter from/to which network provider...
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.c
son connected on the same network to the same Asterisk server,
> or is it somewhere else altogether?
Yes, both telefons are in the same VLAN and Asterisk, too.
> Why do you have:
>
>> allow=ilbc
>
> in sip.conf?
I can't really remember why I a
What's the call quality like then?
The quality is terrible. It is not possible to understand any word...
BUT: if I call my wife using the Thomson (she uses a Thomsons, same
model, too!) the quality is excellent...
> In regard to:
>
> On Saturday 13 June 2020 at 18:25:32, Luca Bertoncello wrote:
le that there is a problem on Telekom-side,
but it does not explain why I have the same problems, altought not often
as by Telekom, by MessageNet, too...
Thanks a lot
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth and Col
lly puzzled...
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? St
gt; That looks a little more standard.
The questions are:
1) why the mobile phone, with "too many things" has a better quality
2) where can I change these settings?
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- B
ACK
Promiscuous Redir: No
Route:
DTMF Mode: rfc2833
SIP Options:(none)
Session-Timer: Inactive
Transport: UDP
Media: RTP
So, I'd say, the codecs are the same...
Do you see something strange
Am 13.06.2020 09:30, schrieb Luca Bertoncello:
Hi again (again)
I noticed right now another strange detail...
I made a call using my mobile phone (connected to the Asterisk). The
quality was top...
Maybe is the problem in a codec used from our phones at homes?
Could someone suggest me how
Am 13.06.2020 um 08:28 schrieb Luca Bertoncello:
> Hi!
>
> I have a Asterisk installation to manage my phones at home (provider is
> Deutsche Telekom).
> It works, but very often the voice is "broken"...
> Yesterday during a call it was very difficult to under
if needed.
Thanks a lot
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Aster
ccepting this offer!
So, back to alaw... :(
> Ah, but SIP is not RTP :)
OK, I forgot it...
I privilege RTP, too... ;)
Regards
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth and Colocation Provided by http://www.ap
(
> 2. What is the bandwidth (upstream is more important than downstream) of your
> Internet connection?
Down 50Mbps
Up 10Mbps
On my Router (Debian 9) I configured a traffic shaper that privileges
the SIP-Packets.
Thanks
Luca Bertoncello
(lucab...
Am 03.12.2019 um 19:28 schrieb Luca Bertoncello:
Hi again
> This delay happens on every peer, Deutsche Telekom and Messagenet, so I
> think the problem is NOT by the Provider, but in my configuration...
Maybe I got the solution...
I see, that I had the jitter buffer active. As I deact
SIP Options:timer
Session-Timer: Inactive
Transport: UDP
Media: RTP
Maybe it helps to find the problem?
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth and Colocation
but in my configuration...
Can someone suggest me where can I search the problem?
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk
useful to know.
As I said, I have a BananaPI with a Debian 9, minimal installed from me
with some scripts to manage the DSL.
Asterisk was installed from Debian Repositories.
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
--
__
if you can identify where the latency comes in?
I must say, that I'm not an expert in VoIP, so I really don't know this
tool and don't have any idea how to analyze the problem...
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
--
___
he other party use VoIP, too, since they are in Germany (and Italy) and
here there are just VoIP... Sigh!
Now I disabled the jitter (jbenable = no), and I called my father in
law. He sayd me, the quality is really better, but I hear sometimes
little noises...
Any other suggestion?
Thanks
Luca
what can I check and what can be the problem.
The problem exists since a very long time, but in the last months it got
worse...
Thank you for your help, I can send abstracts of my configuration, if
you say me what should I send.
Luca Bertoncello
(lucab...@lucabert.de
twork "phone0" (192.168.200.0/24) and the
mobile phone in the network "intlan0" (192.168.10.0/24). The BananaPI hat IPs
on bot networks and I configured Asterisk to bind to 0.0.0.0.
And, as I said, the mobile phone CAN register in Asterisk...
Luca Bertoncello <lucab...@lucabert.de> schrieb:
> But if I try to call another VoIP-phone it rings but no voice will be
> transferred...
Got it!
A "little" firewall problem... :(
Regards
Luca Bertoncello
0 left 'simple_bridge' basic-bridge
<0ef9a447-b1b3-45af-a4af-7c4ac4d10546>
== Spawn extension (default, 00493517654321, 1) exited non-zero on
'SIP/00493511234567-'
Where is the error?!?
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
--
code: No such file or directory
Asterisk Ready.
it does not seems to be normal, but I can't understand why /dev/dahdi/channel
does not exists...
I installed the Paket asterisk-dahdi, of course...
Other question: what does the error about res_phoneprov.c means?
Can someone help me?
Thank you very much
Zitat von Tzafrir Cohen <tzafrir.co...@xorcom.com>:
Yes. It is useful if you want to call using a local sound device.
On a Banana PI? ;)
Consider editing /etc/asterisk/modules.conf and disable ('noload =>')
chan_oss.so .
So I did...
Thanks
Luca Bertoncello
(lucab...@lu
d I really don't think, I need this module...
As I undestand, I just need it, if I want to call/answer call using
the console, and I really don't need this...
Or I understood wrong?
Regards
Luca Bertoncello
(luca
t for your help!
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk?
- [25/Oct/2017:19:38:40 +0200] "GET /phonebook.xml HTTP/1.1"
200 36611
Can someone help me?
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com -
nes as its port range, not your
> phone. If you get one way voice (remote hears phone) then you are on the
> right direction. You'll then need to open the incoming ports too for the
> ports that your phone is expecting to get its RTP from.
OK, tomorrow I'll check it..
Luca Bertoncello <lucab...@lucabert.de> schrieb:
Hallo again
> I configured an user for my mobile phone and I can call, but as soon
> as the other party answer, I get this error in Log:
>
> [Oct 24 15:32:41] NOTICE[3731]: channel.c:4280 __ast_read: Dropping
> incompatib
phone with Android 7...
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
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Check out the new Asterisk community forum at: https://community.asterisk.org
ve data sources using the CALLERID(NUM) and change CALLERID(NAME)
> to be the name you set.
Thanks a lot!
I found this page:
http://deepliquid.com/blog/archives/59
and I successfully got it working!
Regards
Luca Bertoncel
to the number in the E-Mail of voicemail.
Is it possible?
I currently use ${VM_CALLERID} in emailbody and it gives, of course, the
phone number...
Thanks a lot!
Luca Bertoncello
(lucab...@lucabert.de)
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e WRT device or just restarted the Asterisk
> service to resolve your problem? Maybe it's less an Asterisk issue but one
> with DNS caching on this device?
I just restarted Asterisk...
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
pgpqpqFm8dZW7.pgp
Description: Digitale Signatur von
our SIP channel driver
> supports it ;-)
Could you say me how can I disable the SRV lookups?
I use Asterisk 1.8.30.0 on an OpenWRT device.
> You should also use the dnsmgr of Asterisk, resp. configuring it to
> reasonable values. In dnsmgr.conf I set:
The version of Asterisk on
ine.de
> 4. Did the IP address of Telekom's end of the connection change?
I really don't know, but I suppose not
> 5. Did the IP address of your end of the connection change?
No.
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
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I think, this should not be normal... Can someone explain me why it happens
and what I have to change in the configuration to avoid this problem?
Thanks a lot
Luca Bertoncello
(lucab...@lucabert.de)
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App I found
always ask if I want to phone via GSM or VoIP...
Thanks for your suggestion and sorry for the OT!
Luca Bertoncello
(lucab...@lucabert.de)
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Check
Luca Bertoncello <lucab...@lucabert.de> schrieb:
Hi again!
> The problem: after 15 minutes will the call dropped, but only if the call is
> to another nation! If I just call another phone in Germany, I can speak
> longer than 15 minutes...
After a long work, and with the huge
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