OTE VoIP PJSIP Trunk

[georgep138]

Παρολα αυτα, ξεπεταγονται διαφορα issabel απο το πουθενα, μελι εχει?

Ποιά είναι η τελευταία έκδοση του issabel ?
Είναι ενημερωμένη ή έχει κενά ασφαλείας ?

[Magnum]

Η τελευταία έκδοση που είναι η 4.0 της Issabel.
Δεν υποστηρίζει προς το παρόν Pjsip μόνο chain_sip.
Και εγκαθιστάτε με asterisk 13 ή 16 (διαλέγεις κατά την εγκατάσταση)

(Λένε ότι έχει συμπληρωθεί το ποσό μέσω patreon για το Pjsip και λέγανε ότι θα το βγάζανε στις τέλη του 20 αρχές του 21, ακόμη όμως ΔΕΝ...)

Από κάτω είναι Centos με τα πακέτα του Astrerisk οπότε εκεί γίνονται κανονικά Updates...

[netblues]

Ποια εινα τα πακετα του asterisk? To centos δεν εχει σιγουρα τετοια.

To fpbx τρεχει ηδη 18.3 και συντομα 18.4, με αυτοματοποιημενα updates.

[Magnum]

Ναι δίκιο έχεις είναι από το repos του Issabel !

Αυτό που έχει τώρα και παίζει είναι : Asterisk (Ver. 16.16.1)

[barrudo]

Hello Friends!
Who have expirience in OTE SIP Setup?

Can you help me please to setup CHAN_PJSIP trunk to Cosmote OTE,
The support gave me credentials, but dont want to support

I use FreePBX15 and this settings, i see that trunk is online,
on logs i see, that it sometimes reached and sometimes unreached
but anyway dont works:-)

I trid all configurations, but don't work
Some of then on pictures.

Where is mistake?

Thank you!

ang logs is:

11375 [2021-05-27 17:55:56] VERBOSE[23834] res_pjsip/pjsip_configuration.c: Endpoint OTE_+302377022XXX is now Reachable
11376 [2021-05-27 17:55:56] VERBOSE[23834] res_pjsip/pjsip_options.c: Contact OTE_+302377022XXX/sip:+302377022XXX@ims.otenet.gr:5060 is now Reachable. RTT: 91.116 msec
11377 [2021-05-27 17:57:25] WARNING[10948] res_pjsip_outbound_registration.c: No response received from 'sip:ims.otenet.gr:5060' on registration attempt to 'sip:+302377022XXX@ims.otenet.gr:5060', retrying in '60'
11378 [2021-05-27 17:58:57] WARNING[16463] res_pjsip_outbound_registration.c: No response received from 'sip:ims.otenet.gr:5060' on registration attempt to 'sip:+302377022XXX@ims.otenet.gr:5060', retrying in '60'
11379 [2021-05-27 18:00:29] WARNING[23834] res_pjsip_outbound_registration.c: No response received from 'sip:ims.otenet.gr:5060' on registration attempt to 'sip:+302377022XXX@ims.otenet.gr:5060', retrying in '60'
11380 [2021-05-27 18:02:01] WARNING[16463] res_pjsip_outbound_registration.c: No response received from 'sip:ims.otenet.gr:5060' on registration attempt to 'sip:+302377022XXX@ims.otenet.gr:5060', retrying in '60'
11381 [2021-05-27 18:02:59] VERBOSE[23834] res_pjsip/pjsip_configuration.c: Endpoint OTE_+302377022XXX is now Unreachable
11382 [2021-05-27 18:02:59] VERBOSE[23834] res_pjsip/pjsip_options.c: Contact OTE_+302377022XXX/sip:+302377022XXX@ims.otenet.gr:5060 is now Unreachable. RTT: 0.000 msec
11383 [2021-05-27 18:03:01] WARNING[16463] res_pjsip_outbound_registration.c: 403 Forbidden fatal response received from 'sip:ims.otenet.gr:5060' on registration attempt to 'sip:+302377022XXX@ims.otenet.gr:5060', retrying in '30' seconds
11384 [2021-05-27 18:03:56] VERBOSE[23834] res_pjsip/pjsip_configuration.c: Endpoint OTE_+302377022XXX is now Reachable
11385 [2021-05-27 18:03:56] VERBOSE[23834] res_pjsip/pjsip_options.c: Contact OTE_+302377022XXX/sip:+302377022XXX@ims.otenet.gr:5060 is now Reachable. RTT: 91.772 msec

[netblues]

You haven't done things exactly as needed
MAke it like this
https://www.adslgr.com/forum/threads...86#post7026686
and it will work
And post complete screenshots. Critiical things such as match outbound auth isn't posted.

[barrudo]

Hello! Thank you for your reply!
I have done everything like on your pictures, and the same problem...

You haven't done things exactly as needed
MAke it like this
https://www.adslgr.com/forum/threads...86#post7026686
and it will work
And post complete screenshots. Critiical things such as match outbound auth isn't posted.

[netblues]

Try contact user without +30. I can't recall why, but it had to be like this.
Whats the issue? Do you have outbound? Inbound?
Is pjsip accepting the configuration? Check console logs while reloading.
Output of pjsip show endpoints
and
pjsip show endpoint <name from above>

Looks that you are registering (from rtt time) and then nada. Anything fancy on the network side? Such as speed booster?
And as a more general test, does the connection work as sip?

[barrudo]

Thank you!
As i showed on screenshot, the PJSIP Trunk status is online.
I setuped outbound route, it's easy, and try to call, i see that call goes by this trunk,
but don't works, the typical voice in phone: all lines are busy.

this sip number used in speedbooster.
I asked cosmote to use it on my pbx, they acccepted and sent me credentials.
of course i switched off the viip account on speedbooster.

Nothing fancy, i tryed to use connection on the same speedbooster line and on LTE line - noothing works
on the same router i use anothe sip accont from sip proveder and it works.

Tried to change contact user and the same problem.
all enpoint and other thing i see in conf's.

Do you know correct string for connection to cosmote sip server?
mybe better to use classin chan_sip?

4062 [2021-05-28 11:36:04] VERBOSE[2565] res_pjsip/pjsip_configuration.c: Endpoint OTE_+302377022984 is now Reachable
4063 [2021-05-28 11:36:04] VERBOSE[2565] res_pjsip/pjsip_options.c: Contact OTE_+302377022984/sip:+30237702298@ims.otenet.gr:5060 is now Reachable. RTT: 56.866 msec

4077 [2021-05-28 11:38:11] WARNING[2565] res_pjsip_outbound_registration.c: No response received from 'sip:ims.otenet.gr:5060' on registration attempt to 'sip:+30237702298@ims.otenet.gr:5060', retrying in '60'
4078 [2021-05-28 11:39:43] WARNING[2565] res_pjsip_outbound_registration.c: No response received from 'sip:ims.otenet.gr:5060' on registration attempt to 'sip:+30237702298@ims.otenet.gr:5060', retrying in '60'
4079 [2021-05-28 11:41:15] WARNING[2565] res_pjsip_outbound_registration.c: No response received from 'sip:ims.otenet.gr:5060' on registration attempt to 'sip:+30237702298@ims.otenet.gr:5060', retrying in '60'
4080 [2021-05-28 11:42:47] WARNING[2565] res_pjsip_outbound_registration.c: No response received from 'sip:ims.otenet.gr:5060' on registration attempt to 'sip:+30237702298@ims.otenet.gr:5060', retrying in '60'
4081 [2021-05-28 11:44:19] WARNING[2565] res_pjsip_outbound_registration.c: No response received from 'sip:ims.otenet.gr:5060' on registration attempt to 'sip:+30237702298@ims.otenet.gr:5060', retrying in '60'

Try contact user without +30. I can't recall why, but it had to be like this.
Whats the issue? Do you have outbound? Inbound?
Is pjsip accepting the configuration? Check console logs while reloading.
Output of pjsip show endpoints
and
pjsip show endpoint <name from above>

Looks that you are registering (from rtt time) and then nada. Anything fancy on the network side? Such as speed booster?
And as a more general test, does the connection work as sip?

[netblues]

The settings are correct, and it should work.
I believe the problem is speedbooster.
Try disabling it all together and see how it goes.
Sip settings are a bit easier, but if it is a speedbooster thing, it will also fail.

[Fermat]

Όχι βρε παιδιά, τι βαρεμάρα, αγνή ασχετοσύνη ήταν.

Λοιπόν, να δώσω κι εγώ κάτι πίσω στην κοινότητα (έστω στοιχειώδες μια και απλώς είναι σε ένα σημείο έτοιμο όλο το pjsip.conf), ως ευγνωμοσύνη για την πολύτιμη βοήθειά σας.

Το γράφω και με κόκκινα για να φαίνεται σε όποιον στο μέλλον πέσει εδώ: ΕΤΟΙΜΟ pjsip.conf για ΟΤΕ, ΕΛΑ ΠΑΡΕ-ΠΑΡΕ!

[OTE123456]
type = aor
qualify_frequency = 60
contact = sip:+302310123456@ims.otenet.gr

[OTE123456]
type = auth
auth_type = userpass
password = XXXXXXXX
username = +302310123456@ims.otenet.gr

[OTE123456]
type = endpoint
transport = simpletrans
context = from-OTE
disallow = all
allow = alaw,g722
aors = OTE123456
send_connected_line = false
language = en
outbound_auth = OTE123456
from_domain = ims.otenet.gr
contact_user = +302310123456
user_eq_phone = no
t38_udptl = no
t38_udptl_ec = none
fax_detect = no
trust_id_inbound = no
t38_udptl_nat = no
identify_by = auth_username
direct_media = no
rtp_symmetric = yes
dtmf_mode = auto


[OTE123456]
type = identify
endpoint = OTE123456
match = ims.otenet.gr
srv_lookups = true

[OTE123456]
type = registration
transport = simpletrans
outbound_auth = OTE123456
retry_interval = 60
fatal_retry_interval = 30
forbidden_retry_interval = 30
max_retries = 10000
expiration = 3600
auth_rejection_permanent = no
line = yes
endpoint = OTE123456
contact_user = +302310123456
server_uri = sip:ims.otenet.gr
client_uri = sip:+302310123456@ims.otenet.gr

Τώρα είμαι πτώμα, αλλά από αυτά που ξέρω ότι λείπουν από το παραπάνω config είναι να βρω που και ποιο είναι το config για το ότι η γραμμή έχει δύο κανάλια. Θα το συμπληρώσω παραπάνω όταν το βρω.

Ευχαριστώ πολύ! It works like a charm

Το δοκιμασα σε non-gui container andrius/asterisk:latest-data-2. βαζω και εναν sample user για οποιον ψαχνεται που παει στο default context σε pjsip format

[1001]
type = endpoint
context = default
transport=simpletrans
disallow = all
allow = ulaw
aors = 1001
auth = auth1001
 
[1001]
type = aor
max_contacts = 1
 
[auth1001]
type=auth
auth_type=userpass
password=xxxx
username=1001

επισης χρειάστηκα το παρακατω

[simpletrans]
type=transport
protocol=udp
bind=0.0.0.0

[barrudo]

The settings are correct, and it should work.
I believe the problem is speedbooster.
Try disabling it all together and see how it goes.
Sip settings are a bit easier, but if it is a speedbooster thing, it will also fail.

SOLVED! Thank you for your PJSIP configuration, it works now!
The OTE gave me a wrong password.

[daras]

Καλημερα σε ολους και καλο μηνα,

Οταν εχω μια συνδεση με πολλαπλα MSN's στις ρυθμισεις του trunk αλλαζα το context σε from-pstn-toheader το ιδιο κανω και σε PJSIP ?

Αν και το εχω διαβασει ολο το θεμα, δεν εχω καταλαβει πως με 2 συνδεσεις ΟΤΕ οπου εχω απο 1 MSN στην καθε μια, θα πρεπει να κανω ενα PJSIP trunk και απο ποιο router του ΟΤΕ θα φευγει η τηλεφωνια ?

το ΟΤΕ1 ή το ΟΤΕ 2 π.χ ?


Ευχαριστω

[astbox]

Αν έχεις δύο συνδέσεις, δύο router, πρέπει να κάνεις δύο trunk άσχετα με το πόσους αριθμούς έχει η κάθεμια.
MSN είναι ο κάθε extra αριθμός στην ίδια σύνδεση, όχι σε διαφορετική. Εσύ έχεις ένα αριθμό σε κάθε σύνδεση άρα τον κεφαλικό της καθεμίας που θα στον σερβίρει σωστά η Cosmote και θα δουλέψει κατευθείαν το DID inbound rule.
Το from-pstn-to-header το θες για την περίπτωση MSN γιατί τους συγκεκριμένους αριθμούς τους παραδίδει σε άλλο header από αυτόν που διαβάζει το asterisk για να δουλέψουν οι DID κανόνες.

[daras]

Ευχαριστω για την απαντηση

Και απο ποιο router θα φευγει η τηλεφωνια ?
Παιζει ρολο αυτο ?