Hello everyone,
I’m having a hard time trying to transfer a call to another user through Asterisk.
Here’s my current setup:
- Two SIP users: 601 and 602, both with usernames and passwords.
- If I register both users locally (on my own machine) using different SIP clients, calls between 601 and 602 work perfectly — audio is fine both ways.
- When I call 100 (the bot/orchestrator) from 601, the bot answers correctly. However, when it tries to connect the call to 602, nothing happens — 602 never receives the call.
Background:
JAT is a call center that manages multiple SIP extensions (like 601, 602, etc.) and handles registration, media, and routing.
pjsip.conf
[JAT]
type=endpoint
transport=transport-udp-nat
context=public
disallow=all
allow=g722
allow=ulaw
allow=alaw
direct_media=no
rtp_symmetric=yes
aors=JAT
set_var=PROJECT_ID=Dev
set_var=ORIGIN=phone-sip
[JAT]
type=identify
endpoint=JAT
match=83.240.xxx.xxx
[JAT]
type=aor
contact=sip:83.240.xxx.xxx:5060
extensions.conf
same => n,Dial(PJSIP/602@JAT,60)
SIP INVITE from user 601 when dialing 100 (incoming to Asterisk):
Invite from 601 when dials 100:
INVITE sip:100@135.236.xxx.xxxSIP/2.0
Via: SIP/2.0/UDP 83.240.xxx.xxx;rport;branch=z9hG4bKy1Nae853t8mQg
Max-Forwards: 69
From: “601” sip:FreeSWITCH@135.236.xxx.xxx;tag=U4KcaFvS57rZr
To: sip:100@135.236.xxx.xxx
Call-ID: 4cb4ed43-3b1c-123f-6bba-0050569094df
CSeq: 106960625 INVITE
Contact: sip:gw+07143571-a044-4cb3-9650-d52267267b09@83.240.xxx.xxx:5060;transport=udp;gw=07143571-a044-4cb3-9650-d52267267b09
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 299
X-FS-Support: update_display,send_info
Remote-Party-ID: “601” sip:601@135.236.xxx.xxx;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1763008824 1763008825 IN IP4 83.240.xxx.xxx
s=FreeSWITCH
c=IN IP4 83.240.xxx.xxx
t=0 0
m=audio 20010 RTP/AVP 9 0 8 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=ptime:20
Asterisk INVITE attempting to call 602 (outgoing)
INVITE sip:602@83.240.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/UDP 135.236.xxx.xxx:5060;rport;branch=z9hG4bKPj66009da7-e711-4f65-912c-5ef261f6be25
From: “601” sip:FreeSWITCH@172.18.0.4;tag=3c57ef40-1119-48b2-b248-cb62382fbf51
To: sip:602@83.240.200.237
Contact: sip:asterisk@135.236.xxx.xxx:5060
Call-ID: 151f0bc0-e5ff-45b1-9ce6-b8237570b93f
CSeq: 31045 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, INFO
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.24.3
Content-Type: application/sdp
Content-Length: 289
v=0
o=- 265572700 265572700 IN IP4 135.236.xxx.xxx
s=Asterisk
c=IN IP4 135.236.xxx.xxx
t=0 0
m=audio 10004 RTP/AVP 9 0 8 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrec
If anyone has dealt with similar routing issues or knows how to make Asterisk correctly send an INVITE to a remote user (602) on the same call center SIP server, I’d really appreciate the help.
Thank you.