could you provide an example config?

[ivanbalashov239]

Couldn't figure out configuration of tg2sip on first try.

asterisk and tg2sip are on the same server
I'm using asterisk with this configs:

• sip.conf:

[telegram]
host=127.0.0.1
type=peer
context=from-telegram
port=5061
disallow=all
allow=opus

• codecs.conf

[opus]
type=opus
fec=yes
dtx=yes
cbr=yes
bitrate=48000
samprate=48000
max_playback_rate=48000

• extensions.conf (MYREPLACEDNUMBER is known by used tg2sip account and different of its own)

exten => 3,1,Dial(SIP/MYREPLACEDNUMBER@telegram,30,r)

tg2sip settings.ini:

[logging]
core=1
tgvoip=1
pjsip=1
sip_messages=true

[sip]
port=5061
id_uri=sip:127.0.0.1:5060

tg2sip.log:

[14:47:10.761][t:25041][p:25041][pjsip][debug] Module "mod-pjsua-log" registered
[14:47:10.762][t:25041][p:25041][pjsip][debug] Module "mod-tsx-layer" registered
[14:47:10.762][t:25041][p:25041][pjsip][debug] Module "mod-stateful-util" registered
[14:47:10.762][t:25041][p:25041][pjsip][debug] Module "mod-ua" registered
[14:47:10.762][t:25041][p:25041][pjsip][debug] Module "mod-100rel" registered
[14:47:10.762][t:25041][p:25041][pjsip][debug] Module "mod-pjsua" registered
[14:47:10.768][t:25041][p:25041][pjsip][debug] Module "mod-invite" registered
[14:47:10.768][t:25041][p:25041][pjsip][debug] select() I/O Queue created (0x2f46d18)
[14:47:10.814][t:25041][p:25041][pjsip][debug] Module "mod-evsub" registered
[14:47:10.814][t:25041][p:25041][pjsip][debug] Module "mod-presence" registered
[14:47:10.814][t:25041][p:25041][pjsip][debug] Module "mod-mwi" registered
[14:47:10.814][t:25041][p:25041][pjsip][debug] Module "mod-refer" registered
[14:47:10.814][t:25041][p:25041][pjsip][debug] Module "mod-pjsua-pres" registered
[14:47:10.815][t:25041][p:25041][pjsip][debug] Module "mod-pjsua-im" registered
[14:47:10.815][t:25041][p:25041][pjsip][debug] Module "mod-pjsua-options" registered
[14:47:10.815][t:25041][p:25041][pjsip][debug] 1 SIP worker threads created
[14:47:10.815][t:25041][p:25041][pjsip][info] pjsua version 2.8-svn for Linux-4.19.42.1/x86_64/glibc-2.12 initialized
[14:47:10.815][t:25041][p:25041][pjsip][debug] PJSUA state changed: CREATED --> INIT
[14:47:10.815][t:25041][p:25041][pjsip][debug] Setting null sound device..
[14:47:10.816][t:25041][p:25041][pjsip][debug] Opening null sound device..
[14:47:10.817][t:25041][p:25041][pjsip][debug] SIP UDP socket reachable at 192.168.1.170:5061
[14:47:10.817][t:25041][p:25041][pjsip][debug] SIP UDP transport started, published address is 192.168.1.170:5061
[14:47:10.817][t:25041][p:25041][pjsip][debug] PJSUA state changed: INIT --> STARTING
[14:47:10.817][t:25041][p:25041][pjsip][debug] Module "mod-unsolicited-mwi" registered
[14:47:10.817][t:25041][p:25041][pjsip][debug] PJSUA state changed: STARTING --> RUNNING
[14:47:10.817][t:25041][p:25041][pjsip][debug] Adding account: id=sip:127.0.0.1:5060
[14:47:10.818][t:25041][p:25041][pjsip][debug] Account sip:127.0.0.1:5060 added with id 0
[14:47:11.250][t:25055][p:25041][core][info] TG client authorization ready
[14:47:11.250][t:25041][p:25041][core][info] Loading contacts cache
[14:47:11.524][t:25041][p:25041][core][info] Loaded 27 usernames and 121 phones into contacts cache
[14:47:11.818][t:25049][p:25041][pjsip][debug] Closing sound device after idle for 1 second(s)
[14:47:11.818][t:25049][p:25041][pjsip][debug] Closing null sound device..
[14:47:32.530][t:25049][p:25041][pjsip][debug] RX 813 bytes Request msg INVITE/cseq=102 (rdata0x2f77768) from UDP 127.0.0.1:5060:
INVITE sip:MYREPLACEDNUMBER@127.0.0.1:5061 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK182b58a0
Max-Forwards: 70
From: <sip:MYSIPNUMBER@127.0.0.1>;tag=as54c242fd
To: <sip:MYREPLACEDNUMBER@127.0.0.1:5061>
Contact: <sip:MYSIPNUMBER@127.0.0.1:5060>
Call-ID: 05a43aa100fd83c52aa1423334b95a1b@127.0.0.1:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.4.0
Date: Sun, 23 Jun 2019 11:47:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 237

v=0
o=root 726066901 726066901 IN IP4 127.0.0.1
s=Asterisk PBX 16.4.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 11434 RTP/AVP 107 101
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:60
a=sendrecv

--end msg--
[14:47:32.531][t:25049][p:25041][pjsip][debug] Incoming Request msg INVITE/cseq=102 (rdata0x2f77768)
[14:47:32.533][t:25049][p:25041][pjsip][debug] Call 0: initializing media..
[14:47:32.537][t:25049][p:25041][pjsip][debug] RTP socket reachable at 192.168.1.170:4000
[14:47:32.538][t:25049][p:25041][pjsip][debug] RTCP socket reachable at 192.168.1.170:4001
[14:47:32.539][t:25049][p:25041][pjsip][debug] Media index 0 selected for audio call 0
[14:47:32.539][t:25049][p:25041][core][debug] incoming SIP call #0 from <sip:MYSIPNUMBER@127.0.0.1> to <sip:MYREPLACEDNUMBER@127.0.0.1> with call-id 05a43aa100fd83c52aa1423334b95a1b@127.0.0.1:5060
[14:47:32.540][t:25049][p:25041][pjsip][debug] Call 0: deinitializing media..
[14:47:32.540][t:25049][p:25041][pjsip][debug] Call 0: cleaning up provisional media, prov_med_cnt=1, med_cnt=0
[14:47:32.540][t:25049][p:25041][pjsip][debug] TX 323 bytes Response msg 406/INVITE/cseq=102 (tdta0x7f0f5c00cdf8) to UDP 127.0.0.1:5060:
SIP/2.0 406 Not Acceptable
Via: SIP/2.0/UDP 127.0.0.1:5060;received=127.0.0.1;branch=z9hG4bK182b58a0
Call-ID: 05a43aa100fd83c52aa1423334b95a1b@127.0.0.1:5060
From: <sip:MYSIPNUMBER@127.0.0.1>;tag=as54c242fd
To: <sip:MYREPLACEDNUMBER@127.0.0.1>;tag=zjHMpFiPkHnvDV9xu3igAXkdIhWnfbEt
CSeq: 102 INVITE
Content-Length:  0


--end msg--
[14:47:32.543][t:25041][p:25041][core][debug] [25041-1] associated with SIP#0
[14:47:32.544][t:25041][p:25041][core][debug] [25041-1] setting SIP #0 in ringing mode
[14:47:32.544][t:25041][p:25041][pjsip][debug] Answering call 0: code=180
[14:47:32.546][t:25041][p:25041][pjsip][info] Invalid call_id 0 in pjsua_call_answer()
[14:47:32.546][t:25041][p:25041][pjsip][error] pjsua_call_answer2(id, param.p_opt, prm.statusCode, param.p_reason, param.p_msg_data) error: INVITE session already terminated (PJSIP_ESESSIONTERMINATED) (status=171140) [../src/pjsua2/call.cpp:582]
[14:47:32.550][t:25049][p:25041][pjsip][debug] RX 417 bytes Request msg ACK/cseq=102 (rdata0x7f0f5c0100b8) from UDP 127.0.0.1:5060:
ACK sip:MYREPLACEDNUMBER@127.0.0.1:5061 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK182b58a0
Max-Forwards: 70
From: <sip:MYSIPNUMBER@127.0.0.1>;tag=as54c242fd
To: <sip:MYREPLACEDNUMBER@127.0.0.1:5061>;tag=zjHMpFiPkHnvDV9xu3igAXkdIhWnfbEt
Contact: <sip:MYSIPNUMBER@127.0.0.1:5060>
Call-ID: 05a43aa100fd83c52aa1423334b95a1b@127.0.0.1:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 16.4.0
Content-Length: 0


--end msg--
[14:47:32.584][t:25041][p:25041][pjsip][debug] Call 0 hanging up: code=500..
[14:47:32.585][t:25041][p:25041][pjsip][info] Invalid call_id 0 in pjsua_call_hangup()
[14:47:32.585][t:25041][p:25041][pjsip][error] pjsua_call_hangup(id, prm.statusCode, param.p_reason, param.p_msg_data) error: INVITE session already terminated (PJSIP_ESESSIONTERMINATED) (status=171140) [../src/pjsua2/call.cpp:591]
[14:47:32.585][t:25041][p:25041][core][error] INVITE session already terminated (PJSIP_ESESSIONTERMINATED)

asterisk sip set debug on:

Audio is at 11434
Adding codec opus to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 127.0.0.1:5061:
INVITE sip:MYREPLACEDNUMBER@127.0.0.1:5061 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK182b58a0
Max-Forwards: 70
From: <sip:MYSIPNUMBER@127.0.0.1>;tag=as54c242fd
To: <sip:MYREPLACEDNUMBER@127.0.0.1:5061>
Contact: <sip:MYSIPNUMBER@127.0.0.1:5060>
Call-ID: 05a43aa100fd83c52aa1423334b95a1b@127.0.0.1:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.4.0
Date: Sun, 23 Jun 2019 11:47:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 237

v=0
o=root 726066901 726066901 IN IP4 127.0.0.1
s=Asterisk PBX 16.4.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 11434 RTP/AVP 107 101
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:60
a=sendrecv

---

<--- SIP read from UDP:127.0.0.1:5061 --->
SIP/2.0 406 Not Acceptable
Via: SIP/2.0/UDP 127.0.0.1:5060;received=127.0.0.1;branch=z9hG4bK182b58a0
Call-ID: 05a43aa100fd83c52aa1423334b95a1b@127.0.0.1:5060
From: <sip:MYSIPNUMBER@127.0.0.1>;tag=as54c242fd
To: <sip:MYREPLACEDNUMBER@127.0.0.1>;tag=zjHMpFiPkHnvDV9xu3igAXkdIhWnfbEt
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Transmitting (no NAT) to 127.0.0.1:5061:
ACK sip:MYREPLACEDNUMBER@127.0.0.1:5061 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK182b58a0
Max-Forwards: 70
From: <sip:MYSIPNUMBER@127.0.0.1>;tag=as54c242fd
To: <sip:MYREPLACEDNUMBER@127.0.0.1:5061>;tag=zjHMpFiPkHnvDV9xu3igAXkdIhWnfbEt
Contact: <sip:MYSIPNUMBER@127.0.0.1:5060>
Call-ID: 05a43aa100fd83c52aa1423334b95a1b@127.0.0.1:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 16.4.0
Content-Length: 0


---
Really destroying SIP dialog '05a43aa100fd83c52aa1423334b95a1b@127.0.0.1:5060' Method: INVITE

i will probably figure it out if invest some time in learning how it actually works.
It would be really helpful if you could help me to fix it now.
Thanks.

[danielgt82]

I was able to call the telegram I activated and answer the sip terminal(1000).

I created the two extension sip 1000 and 1009 (no password).

codec.conf

[opus]
type=opus
signal=voice
max_playback_rate=48000 
bitrate=max

setting.ini

i```
d_uri=sip:1009@192.168.0.200
callback_uri=sip:1000@192.168.0.200
raw_pcm=false

[ivanbalashov239]

@danielgt82 it worked, your config is exactly what was needed, i just made a mistake forgetting to change raw_pcm
Thank you.

[idiogolima]

I found this tutorial:

https://voxlink.ru/kb/asterisk-configuration/ustanovka-i-nastrojka-sip-shljuza-dlja-telegram/

[Infactum]

https://voxlink.ru/kb/asterisk-configuration/ustanovka-i-nastrojka-sip-shljuza-dlja-telegram/

Thats a good guide. Even though I don't recommend to waste time and comile everything yourself, since there is binaries on release page, including universal ones.