Re: [asterisk-users] Change by Deutsche Telekom end of februar. Can someone help me?
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 got it... I tested the configuration with my Fax number and it seems to work (= I can call the fax and can call my mobile phone from the fax with "originate..."). Congrats! So, it seems it does NOT work as expected... I tried to activate the FAX and it works, then I activated my number and it works, too. Finally I activated the number of my wife and it does not work anymore... If I call the number I can only see (verbose 42): [Feb 18 19:57:12] NOTICE[19379] res_pjsip/pjsip_distributor.c: Request 'INVITE' from '' failed for '217.0.21.64:5060' (callid: p65550t1613674632m753568c93349s2) - No matching endpoint found You have to do all of the configuration mentioned here[1] for *each* number. Afterwards, you have to route the incoming call to an internal device. As I'm using FreePBX, I don't know how to do it *correctly*. and no phone rings... After that, even if I restore the single number to SIP I only get the error and nothing work, until I restored _ALL_ numbers to SIP. Do someone has an explanation and (better!) a solution to the problem? Solution: You have the choice between: programming your PBX yourself (and have the struggle and pain) or let this pretty difficile job do others for you - they provide extremely good solutions for a lot of telephony features - it makes no sense to reinvent those features without having the required knowledge - so, use FreePBX. But it's of course your decision. [1] https://www.ip-phone-forum.de/threads/hilfe-f%C3%BCr-grundeinstellung-asterisk-telekom-ben%C3%B6tigt.307115/post-2374234 -- _ -- 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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Change by Deutsche Telekom end of februar. Can someone help me?
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 got it... >> I tested the configuration with my Fax number and it seems to work (= I >> can call the fax and can call my mobile phone from the fax with >> "originate..."). > > Congrats! So, it seems it does NOT work as expected... I tried to activate the FAX and it works, then I activated my number and it works, too. Finally I activated the number of my wife and it does not work anymore... If I call the number I can only see (verbose 42): [Feb 18 19:57:12] NOTICE[19379] res_pjsip/pjsip_distributor.c: Request 'INVITE' from '' failed for '217.0.21.64:5060' (callid: p65550t1613674632m753568c93349s2) - No matching endpoint found and no phone rings... After that, even if I restore the single number to SIP I only get the error and nothing work, until I restored _ALL_ numbers to SIP. Do someone has an explanation and (better!) a solution to the problem? 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 to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Change by Deutsche Telekom end of februar. Can someone help me?
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 configuration with my Fax number and it seems to work (= I
> can call the fax and can call my mobile phone from the fax with
> "originate...").
Congrats!
> On the registration I have:
>
> [pbxfax]
> type = registration
> retry_interval = 20
> max_retries = 10
> contact_user = 00493514977291
> expiration = 120
> transport = transport-udp
> outbound_auth = pbxfax
> client_uri = sip:03514977...@tel.t-online.de
> server_uri = sip:tel.t-online.de
>
> First: can I use tel.t-online.de or _MUST_ I change it?
No, you mustn't change it. You must use tel.t-online.de.
> If I understand
> your previous E-Mail, I'd say that I can leave tel.t-online.de...
Correctly!
> Then I have a question by the Dialplan... Currently I have:
>
> [fax-out]
> exten => _X.,1,NoOp()
> exten => _X.,n,Verbose(2,Call from FAX)
> exten => _X.,n,Dial(SIP/pbxfax/${EXTEN},,R)
>
> And I'll replace it with:
>
> [fax-out]
> exten => _X.,1,NoOp()
> exten => _X.,n,Verbose(2,Call from FAX)
> exten => _X.,n,Dial(PJSIP/pbxfax/sip:${EXTEN}@tel.t-online.de,,R)
>
> Is it correct? I tried with
> "PJSIP/pbxfax/pjsip:${EXTEN}@tel.t-online.de,,R" and it does NOT work...
> Is it correct, that I have to leave "sip:..."?
Don't know - I don't care about dialplan - I'm using FreePBX :-)
Thanks
Michael
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Re: [asterisk-users] Change by Deutsche Telekom end of februar. Can someone help me?
Am 16.02.2021 um 22:32 schrieb Michael Maier:
Hi Michael
>> Maybe could you send me an abstract of your configuration?
>
> Take a look here [1]
So, maybe I got it...
I tested the configuration with my Fax number and it seems to work (= I
can call the fax and can call my mobile phone from the fax with
"originate...").
On the registration I have:
[pbxfax]
type = registration
retry_interval = 20
max_retries = 10
contact_user = 00493514977291
expiration = 120
transport = transport-udp
outbound_auth = pbxfax
client_uri = sip:03514977...@tel.t-online.de
server_uri = sip:tel.t-online.de
First: can I use tel.t-online.de or _MUST_ I change it? If I understand
your previous E-Mail, I'd say that I can leave tel.t-online.de...
Then I have a question by the Dialplan... Currently I have:
[fax-out]
exten => _X.,1,NoOp()
exten => _X.,n,Verbose(2,Call from FAX)
exten => _X.,n,Dial(SIP/pbxfax/${EXTEN},,R)
And I'll replace it with:
[fax-out]
exten => _X.,1,NoOp()
exten => _X.,n,Verbose(2,Call from FAX)
exten => _X.,n,Dial(PJSIP/pbxfax/sip:${EXTEN}@tel.t-online.de,,R)
Is it correct? I tried with
"PJSIP/pbxfax/pjsip:${EXTEN}@tel.t-online.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)
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Re: [asterisk-users] Change by Deutsche Telekom end of februar. Can someone help me?
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 your configuration?
Take a look here [1]
You mean, I have to create a "fake" Zone tel.t-online.de in my Bind with
these settings? Looks like dangerous, if they changes something...
If you do that statically -> yes, you're right. You have to do it
dynamically. I attached a script, which can be used to dynamically build
a rpz each 15 minutes e.g. It directly asks the telekom nameserver for
naptr and srv entries. It looks like this:
server 192.168.62.13
zone rpz-tonline
update delete tel.t-online.de.rpz-tonline.
update delete _sips._tcp.tel.t-online.de.rpz-tonline.
update delete _sip._tcp.tel.t-online.de.rpz-tonline.
update add tel.t-online.de.rpz-tonline. 60 NAPTR 10 0 "s"
"SIPS+D2T" "" _sips._tcp.tel.t-online.de.
update add tel.t-online.de.rpz-tonline. 60 NAPTR 30 0 "s"
"SIP+D2T" "" _sip._tcp.tel.t-online.de.
update add _sips._tcp.tel.t-online.de.rpz-tonline. 60 SRV 10 0
5061 s-eps-110.edns.t-ipnet.de.
update add _sip._tcp.tel.t-online.de.rpz-tonline. 60 SRV 10 0
5060 s-epp-110.edns.t-ipnet.de.
send
So if I undestand what you mean, you check the NAPTR and SRV für
_sips._tcp.tel.t-online.de and save the record in a "virtual domain"
rpz-tonline, is it correct?
Then I suppose you use this domain instead of tel.t-online.de in the SIP
configuratione as "host", "outboundproxy" and "fromdomain", is it correct?
No - you have to use the correct domain name in asterisk. Only bind knows about
the fake domain. You have to configure bind correctly.
You have to create the fake domain in the bind config like this:
options {
...
response-policy {
zone "rpz-tonline";
};
};
...
zone "rpz-tonline" {
type master;
file "/var/named/rpz-tonline-override";
allow-query { any; };
allow-transfer { any; };
allow-update { any; };
};
All other things: take a look at the script! It's not that complicated.
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...
Independently you have to add your own trunk names to get it working
(telekomPJSIP-a, ...).
Could you explain me that? I'm not an expert of Asterisk... :(
Well, if you want to use it, you really should engage yourself a bit more to get
it solved. It's not that easy. Or you may forget about the DNS fake and live with
the problem, that asterisk could partly switch sometimes to another server -
breaking the telephony. I don't think it would happen that often, because Telekom
usually is extremely stable. Try at first to get a running pjsip configuration.
The DNS theme could be done later on.
Regards
Michael
[1]
https://www.ip-phone-forum.de/threads/hilfe-f%C3%BCr-grundeinstellung-asterisk-telekom-ben%C3%B6tigt.307115/post-2374234
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Re: [asterisk-users] Change by Deutsche Telekom end of februar. Can someone help me?
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 > > Mmm... I'm using tel.t-online.de, but I'm not sure I'm using pjsip... > > module show say me: > > res_pjsip.so Basic SIP resource > 46 Running core > > Do I use pjsip? pjsip show registrations > You mean, I have to create a "fake" Zone tel.t-online.de in my Bind with > these settings? Looks like dangerous, if they changes something... If you do that statically -> yes, you're right. You have to do it dynamically. I attached a script, which can be used to dynamically build a rpz each 15 minutes e.g. It directly asks the telekom nameserver for naptr and srv entries. It looks like this: server 192.168.62.13 zone rpz-tonline update delete tel.t-online.de.rpz-tonline. update delete _sips._tcp.tel.t-online.de.rpz-tonline. update delete _sip._tcp.tel.t-online.de.rpz-tonline. update add tel.t-online.de.rpz-tonline. 60 NAPTR 10 0 "s" "SIPS+D2T" "" _sips._tcp.tel.t-online.de. update add tel.t-online.de.rpz-tonline. 60 NAPTR 30 0 "s" "SIP+D2T" "" _sip._tcp.tel.t-online.de. update add _sips._tcp.tel.t-online.de.rpz-tonline. 60 SRV 10 0 5061 s-eps-110.edns.t-ipnet.de. update add _sip._tcp.tel.t-online.de.rpz-tonline. 60 SRV 10 0 5060 s-epp-110.edns.t-ipnet.de. send You have to configure bind to use the rpz for all lookup calls resolving *.tel.t-online.de. I assume that the individual t-ipnet.de entries are "statically" and therefore resolved directly (w/o rpz). But this could be added to the script, too (would be a new rpz). At the moment, I'm using only one DNS server for digging the NAPTR and SRV entries - this could be enhanced to use 2 servers if the first fails. If the first fails, the script currently stops and does nothing. I assume, that the DNS server is stable. 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. Independently you have to add your own trunk names to get it working (telekomPJSIP-a, ...). You can verify if it's working by checking for entries like this in journalctl: Feb 16 19:35:46 myfw named[1516]: client @0x7ff574027bd0 192.168.62.13#25869 (tel.t-online.de): rpz QNAME NODATA rewrite tel.t-online.de via tel.t-online.de.rpz-tonline They are appearing at the moment asterisk starts a lookup. Hope this helps! Thanks Michael check-t-online.pl Description: Perl program -- _ -- 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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Change by Deutsche Telekom end of februar. Can someone help me?
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... :( Maybe could you send me an abstract of your configuration? >> You mean, I have to create a "fake" Zone tel.t-online.de in my Bind with >> these settings? Looks like dangerous, if they changes something... > > If you do that statically -> yes, you're right. You have to do it > dynamically. I attached a script, which can be used to dynamically build > a rpz each 15 minutes e.g. It directly asks the telekom nameserver for > naptr and srv entries. It looks like this: > > server 192.168.62.13 > zone rpz-tonline > update delete tel.t-online.de.rpz-tonline. > update delete _sips._tcp.tel.t-online.de.rpz-tonline. > update delete _sip._tcp.tel.t-online.de.rpz-tonline. > update add tel.t-online.de.rpz-tonline. 60 NAPTR 10 0 "s" > "SIPS+D2T" "" _sips._tcp.tel.t-online.de. > update add tel.t-online.de.rpz-tonline. 60 NAPTR 30 0 "s" > "SIP+D2T" "" _sip._tcp.tel.t-online.de. > update add _sips._tcp.tel.t-online.de.rpz-tonline. 60 SRV 10 0 > 5061 s-eps-110.edns.t-ipnet.de. > update add _sip._tcp.tel.t-online.de.rpz-tonline. 60 SRV 10 0 > 5060 s-epp-110.edns.t-ipnet.de. > send So if I undestand what you mean, you check the NAPTR and SRV für _sips._tcp.tel.t-online.de and save the record in a "virtual domain" rpz-tonline, is it correct? Then I suppose you use this domain instead of tel.t-online.de in the SIP configuratione as "host", "outboundproxy" and "fromdomain", 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... > Independently you have to add your own trunk names to get it working > (telekomPJSIP-a, ...). Could you explain me that? I'm not an expert of Asterisk... :( 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 Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Change by Deutsche Telekom end of februar. Can someone help me?
Hi! On 15.02.21 at 08:43 Luca Bertoncello wrote: > Hi list! > > I received a letter from Deutsche Telekom where they say me, that I need > to change "something" on my router until 28.02.2021, otherwise I cannot > phone anymore. > Since I use Asterisk and I don't have a router, I'm not sure what I need > to do... > In the letter there is an URL to "explain" how to change the > configuration if I use a VoIP-phone, but they only say, that I don't > have to use Port 5060, but Port 0... > > Surely there are in this list someone other using Deutsche Telekom... > Does someone of them understand what I should change in the Asterisk > configuration? 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 won't have any problem (using asterisk 14 or higher), because it's default. But you may have problems with the handling of the calls, because Telekom needs the client always to use the same server for all activities after the register has been done (the SRV entries contain 3 servers and asterisk will use them "randomly" if it detects a problem - regardless which server of the list has been used for registration - this won't work with Telekom and will lead to not working outbound calls / interrupted calls e.g.). This won't happen very often (because they have been extremely stable in the past), but I could see it nevertheless already. If you want to be really sure to not face this problem, you have to create a workaround by adding a rpz zone e.g. with an own bind, which is fed by an own job and presents asterisk just one server when looking up the SRV entries after the NAPTR call. NAPTR / SRV works like this (example for tel.t-online.de): 1. Search for the service names dig noall +answer tel.t-online.de NAPTR tel.t-online.de.5 IN NAPTR 10 0 "s" "SIPS+D2T" "" _sips._tcp.tel.t-online.de. tel.t-online.de.5 IN NAPTR 30 0 "s" "SIP+D2T" "" _sip._tcp.tel.t-online.de. 2. Take the answer of the NAPTR output (TCP/TLS, TCP) dig +noall +answer _sips._tcp.tel.t-online.de SRV _sips._tcp.tel.t-online.de. 2234 IN SRV 10 0 5061 s-eps-110.edns.t-ipnet.de. _sips._tcp.tel.t-online.de. 2234 IN SRV 20 0 5061 h2-eps-100.edns.t-ipnet.de. _sips._tcp.tel.t-online.de. 2234 IN SRV 30 0 5061 d-eps-100.edns.t-ipnet.de. dig +noall +answer _sip._tcp.tel.t-online.de SRV _sip._tcp.tel.t-online.de. 3600 IN SRV 30 0 5060 d-epp-100.edns.t-ipnet.de. _sip._tcp.tel.t-online.de. 3600 IN SRV 10 0 5060 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, that asterisk always uses the same server. Easiest way is to provide just one in the DNS answer ... . Regards Michael [1] https://geschaeftskunden.telekom.de/hilfe-und-service/online-services/hilfe-internetanschluss/telefonieanpassung#telekom -- _ -- 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? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Change by Deutsche Telekom end of februar. Can someone help me?
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 Mmm... I'm using tel.t-online.de, but I'm not sure I'm using pjsip... module show say me: res_pjsip.so Basic SIP resource 46 Running core Do I use pjsip? > won't have any problem (using asterisk 14 or higher), because it's > default. But you may have problems with the handling of the calls, > because Telekom needs the client always to use the same server for all > activities after the register has been done (the SRV entries contain 3 > servers and asterisk will use them "randomly" if it detects a problem - > regardless which server of the list has been used for registration - > this won't work with Telekom and will lead to not working outbound calls > / interrupted calls e.g.). This won't happen very often (because they > have been extremely stable in the past), but I could see it nevertheless > already. If you want to be really sure to not face this problem, you > have to create a workaround by adding a rpz zone e.g. with an own bind, > which is fed by an own job and presents asterisk just one server when > looking up the SRV entries after the NAPTR call. NAPTR / SRV works like > this (example for tel.t-online.de): You mean, I have to create a "fake" Zone 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 the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Change by Deutsche Telekom end of februar. Can someone help me?
Hi list! I received a letter from Deutsche Telekom where they say me, that I need to change "something" on my router until 28.02.2021, otherwise I cannot phone anymore. Since I use Asterisk and I don't have a router, I'm not sure what I need to do... In the letter there is an URL to "explain" how to change the configuration if I use a VoIP-phone, but they only say, that I don't have to use Port 5060, but Port 0... Surely there are in this list someone other using Deutsche Telekom... Does someone of them understand what I should change in the Asterisk configuration? 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 Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
