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 new year!
Luca Bertoncello
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 Bertoncello
(lucab...
Am 31.12.2021 um 16:04 schrieb Antony Stone:
Hi
> Check the Dial() command which places the call to the phone. Does it contain
> the "c" option?
Jup...
exten => _529874,1,Verbose(2,Call for Main - [${CALLERID(num)}])
exten => _529874,n,Set(CALLERID(num)=${IF($[ "${CALLERID(num):0:3}" =
"+49"
On Friday 31 December 2021 at 15:54:01, Luca Bertoncello wrote:
> Am 31.12.2021 um 14:39 schrieb Antony Stone:
>
> Hi Antony
>
> >> Last very strange problem is, that the list of missed calls on the phone
> >> is always empty...
> >
> > Check the SIP notifications which are being sent to the te
Am 31.12.2021 um 14:39 schrieb Antony Stone:
Hi Antony
>> Last very strange problem is, that the list of missed calls on the phone
>> is always empty...
>
> Check the SIP notifications which are being sent to the telephone for these
> calls, and whether any of them contain a "Reason" code for "
On Friday 31 December 2021 at 12:29:19, Luca Bertoncello wrote:
> Am 28.12.2021 um 21:21 schrieb Antony Stone:
>
> Hi
>
> > However, at least you've got as far as ruling out Telekom as being the
> > source of the problem, which I think is good.
>
> So, I setted:
>
> sendrpid=rpid
>
> in
Am 28.12.2021 um 21:21 schrieb Antony Stone:
Hi
> However, at least you've got as far as ruling out Telekom as being the source
> of the problem, which I think is good.
So, I setted:
sendrpid=rpid
instead of:
sendrpid=pai
and now it seems to work. The called number does not
Am 28.12.2021 um 21:21 schrieb Antony Stone:
> I would look at whatever part of the dial plan is responsible for inserting
> "Sekretariat", and also check whether you have "sendrpid=yes" in sip.conf.
I find "Sekretariat" in the Ringing sent from Asterisk:
Via: SIP/2.0/UDP
192.168.60.53:3072;bran
On Tuesday 28 December 2021 at 20:39:37, Luca Bertoncello wrote:
> After about 6 seconds I get from the Telekom:
>
> Via: SIP/2.0/UDP
> 87.191.224.158:5060;received=87.191.224.158;rport=5060;branch=z9hG4bKPj43be
> 873a-cf55-4348-8867-5c2bb97bd76a
> To: ;
> tag=h7g4Esbg_p65544t1640719676m169304c93
Am 28.12.2021 um 20:24 schrieb Antony Stone:
> No, you want to look at the "180 Ringing" response in both cases - what goes
> in to Asterisk, and what comes out of it.
OK
> No, data FROM Deutsche Telekom. They are the ones sending the "180 Ringing"
> back to you once they think the external t
On Tuesday 28 December 2021 at 20:07:22, Luca Bertoncello wrote:
> Am 28.12.2021 um 20:00 schrieb Antony Stone:
> >
> > From your earlier packet capture, it looked to me like you were dialling
> > an external number from an internal telephone.
>
> This is correct!
> I called my mobile phone usin
Am 28.12.2021 um 20:00 schrieb Antony Stone:
> Which way round are you making the telephone call?
>
> From your earlier packet capture, it looked to me like you were dialling an
> external number from an internal telephone.
This is correct!
I called my mobile phone using a VoIP phone connected
On Tuesday 28 December 2021 at 19:52:46, Luca Bertoncello wrote:
> Am 28.12.2021 um 19:41 schrieb Antony Stone:
>
> Hi Antony,
>
> > Okay, so, returning to my question, do you see any difference between the
> > packet inbound to Asterisk from the called telephone, and the packet
> > outbound fro
Am 28.12.2021 um 19:41 schrieb Antony Stone:
Hi Antony,
> Okay, so, returning to my question, do yu see any difference between the
> packet
> inbound to Asterisk from the called telephone, and the packet outbound from
> Asterisk to the calling telephone?
I'm trying to understand what you mean
On Tuesday 28 December 2021 at 18:17:00, Luca Bertoncello wrote:
> Am 28.12.2021 um 17:35 schrieb Antony Stone:
> >
> > Where exactly were those packets captured?
>
> tcpdump on the Asterisk-Server on the interface of the VLAN for the phones.
> All traffic captured.
Okay, so, returning to my qu
Am 28.12.2021 um 17:35 schrieb Antony Stone:
>> So, I see, there is a "P-Asserted-Identity"... But I can't understand
>> why...
>>
>> Any idea?
>
> Where exactly were those packets captured?
tcpdump on the Asterisk-Server on the interface of the VLAN for the phones.
All traffic captured.
Thanks
On Tuesday 28 December 2021 at 17:28:47, Luca Bertoncello wrote:
> Am 28.12.2021 um 17:22 schrieb Antony Stone:
>
> Hi Antony
>
> > I mean the response from the called telephone in reply to the INVITE,
> > which contains the SIP code "180 Ringing" and may optionally have an
> > RPID header.
>
>
Am 28.12.2021 um 17:22 schrieb Antony Stone:
Hi Antony
> I mean the response from the called telephone in reply to the INVITE, which
> contains the SIP code "180 Ringing" and may optionally have an RPID header.
OK, I see something strange...
Here what I see if I call my mobile phone (then the
On Tuesday 28 December 2021 at 16:58:01, Luca Bertoncello wrote:
> Am 28.12.2021 um 15:42 schrieb Antony Stone:
>
> Hi Antony,
>
> > Sounds like something strange is happening with Remote-Party-ID.>
> > Do a packet capture and see whether the 180 response from the callee's
> > phone contains an
Am 28.12.2021 um 15:42 schrieb Antony Stone:
Hi Antony,
> Sounds like something strange is happening with Remote-Party-ID.>
> Do a packet capture and see whether the 180 response from the callee's phone
> contains an RPID header with silly content.
I captured the packet but I don't see anything
On Tuesday 28 December 2021 at 14:30:17, Luca Bertoncello wrote:
> I have a Debian Server with Asterisk 16.2.1 from Debian repos and some
> SNOM phones (SNOM 821, last firmware snom821-SIP 8.7.5.35).
> If I call a number I can see in the display the called number, after a
> few seconds the number
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...
The proble
Dear list,
I have a very strange problem and no idea where the problem can be...
I have a Debian Server with Asterisk 16.2.1 from Debian repos and some
SNOM phones (SNOM 821, last firmware snom821-SIP 8.7.5.35).
If I call a number I can see in the display the called number, after a
few second
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