Thank you for suggestions and sorry for not answering for so long. I’ve tried to sketch a PJSIP config. My current best guess is:
[telegram-endpoint]
type=endpoint
context=telegram-incoming-test
disallow=all
allow=opus ; Set opus coded as in example, but I could see the warning from TG2SIP
auth=telegram-auth
aors=telegram-aors
[telegram-auth]
type=auth
auth_type=userpass
username=telegram ; Is it for inbound or outbound registrations?
password=123 ; There is no place for passwords in TG2SIP settings. I suspect this is not used or wrong..
[telegram-aors]
type=aor
contact=sip:telegram@127.0.0.1:5062
; Do I also need "identify" section?
Now, if I try to dial to this endpoint using same => n,Dial(PJSIP/telegram-endpoint), I see that Asterkis does connect to TG2SIP using UDP, but TG2SIP cancels the call very soon with the following reason:
[23:45:00.765][t:23597][p:23591][pjsip][trace] Processing incoming message: Request msg INVITE/cseq=25833 (rdata0x7f75cc0018f8) [25/394]
[23:45:00.765][t:23597][p:23591][pjsip][debug] RX 898 bytes Request msg INVITE/cseq=25833 (rdata0x7f75cc0018f8) from UDP 127.0.0.1:5060:
INVITE sip:telegram@127.0.0.1:5062 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bKPj73382712-ad0d-4b85-8b5e-3dd12082c2c2
From: "dongle0" <sip:XXXXXXXXXX@127.0.0.1>;tag=2b42e16b-5c98-42f8-bd98-ece203b9170b
To: <sip:telegram@127.0.0.1>
Contact: <sip:asterisk@127.0.0.1:5060>
Call-ID: 7b720240-4b76-44ed-abe9-5b1687a4904e
CSeq: 25833 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 16.1.1
Content-Type: application/sdp
Content-Length: 237
v=0
o=- 1026255045 1026255045 IN IP4 127.0.0.1
s=Asterisk
c=IN IP4 127.0.0.1
t=0 0
m=audio 10390 RTP/AVP 107 101
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv
--end msg--
[23:45:00.765][t:23597][p:23591][pjsip][debug] Incoming Request msg INVITE/cseq=25833 (rdata0x7f75cc0018f8)
[23:45:00.765][t:23597][p:23591][pjsip][trace] Transaction created for Request msg INVITE/cseq=25833 (rdata0x7f75cc0018f8)
[23:45:00.765][t:23597][p:23591][pjsip][trace] Incoming Request msg INVITE/cseq=25833 (rdata0x7f75cc0018f8) in state Null
[23:45:00.765][t:23597][p:23591][pjsip][trace] State changed from Null to Trying, event=RX_MSG
[23:45:00.765][t:23597][p:23591][pjsip][trace] Transaction tsx0x7f75cc004c68 state changed to Trying
[23:45:00.765][t:23597][p:23591][pjsip][trace] UAS dialog created
[23:45:00.765][t:23597][p:23591][pjsip][trace] Module mod-invite added as dialog usage, data=0x7f75cc00ac68
[23:45:00.765][t:23597][p:23591][pjsip][trace] Session count inc to 3 by mod-invite
[23:45:00.766][t:23597][p:23591][pjsip][trace] UAS invite session created for dialog dlg0x7f75cc005aa8
[23:45:00.766][t:23597][p:23591][pjsip][trace] Session count inc to 3 by mod-pjsua
[23:45:00.766][t:23597][p:23591][pjsip][trace] Response msg 488/INVITE/cseq=25833 (tdta0x7f75cc010a38) created
[23:45:00.766][t:23597][p:23591][pjsip][error] Request verification failed: Not Acceptable Here [status=170488]
[23:45:00.766][t:23597][p:23591][pjsip][trace] Sending Response msg 488/INVITE/cseq=25833 (tdta0x7f75cc010a38)
[23:45:00.766][t:23597][p:23591][pjsip][trace] Sending Response msg 488/INVITE/cseq=25833 (tdta0x7f75cc010a38) in state Trying
[23:45:00.766][t:23597][p:23591][pjsip][debug] TX 506 bytes Response msg 488/INVITE/cseq=25833 (tdta0x7f75cc010a38) to UDP 127.0.0.1:5060:
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 127.0.0.1:5060;rport=5060;received=127.0.0.1;branch=z9hG4bKPj73382712-ad0d-4b85-8b5e-3dd12082c2c2
Call-ID: 7b720240-4b76-44ed-abe9-5b1687a4904e
From: "dongle0" <sip:XXXXXXXXXXX@127.0.0.1>;tag=2b42e16b-5c98-42f8-bd98-ece203b9170b
To: <sip:telegram@127.0.0.1>;tag=I8mpIA3eTJop2EuonUdNXCiFZj1-mdfy
CSeq: 25833 INVITE
Warning: 399 darktower "No suitable codec for remote offer (PJMEDIA_SDPNEG_NOANSCODEC)"
Accept: application/sdp
Content-Length: 0
--end msg--
[23:45:00.766][t:23597][p:23591][pjsip][trace] State changed from Trying to Completed, event=TX_MSG
For reference: TG2SIP settings.ini:
[logging]
core=1 ; 0-trace 2-info 4-err 6-off
; 1-debug 3-warn 5-crit
tgvoip=5 ; same as core
pjsip=0 ; same as core
sip_messages=true ; log sip messages if pjsip debug is enabled
console_min_level=0 ; minimal log level that will be written into console
file_min_level=0 ; same but into file
;tdlib=3 ; TDLib is written to file only and has its own log level values
; not affected by other log settings
; 0-fatal 2-warnings 4-debug
; 1-errors 3-info 5-verbose debug
[sip]
public_address=127.0.0.1
port=5062
;port_range=0 ; Specify the port range for socket binding, relative to the start
; port number specified in port.
id_uri=sip:telegram@127.0.0.1
; The Address of Record or AOR, that is full SIP URL that identifies the account.
; The value can take name address or URL format, and will look something
; like "sip:account@serviceprovider".
callback_uri=sip:telegram@127.0.0.1:5060 ; FIXME: unhardcode the port
; SIP URI for TG->SIP incoming calls processing
raw_pcm=false ; use L16@48k codec if true or OPUS@48k otherwise
; keep true for lower CPU consumption
;thread_count=1 ; Specify the number of worker threads to handle incoming RTP
; packets. A value of one is recommended for most applications.
[telegram]
api_id=XXXXX
api_hash=YYYYY
system_language_code=ru-RU ; IETF language tag of the user's operating system language
[other]
extra_wait_time=10 ; If gateway gets temporary blocked with "Too Many Requests" reason,
; then block all outgoing telegram requests for X more seconds than was
; requested by server
;peer_flood_time=86400 ; Seconds to wait on PEER_FLOOD
