Asterisk sends invite to transferor after refer

Asterisk Asterisk SIP
1

Hello
I am a beginner at asterisk.
Scenario

I have 3 Terminal connected to an Asterisk PBX.

  1. I called Terminal-B from Terminal-A using PJSUA
  2. After the call established I initiated transfer from Terminal-B to transfer Terminal-A to Terminal-C
    3.Terminal-B sends REFER to Asterisk.
  3. Asterisk sends invite to Terminal-A and Terminal-C.
    5.Terminal-A sends an invite to Terminal-B via Asterisk.
    extensions.conf
    [internal]
    exten => 7001,1,Answer()
    exten => 7001,2,Dial(PJSIP/7001,90)
    exten => 7001,3,Playback(vm-nobodyavail)
    exten => 7001,4,VoiceMail(7001@main)
    exten => 7001,5,Hangup()

exten => 7002,1,Answer()
exten => 7002,2,Dial(PJSIP/7002,90)
exten => 7002,3.Playback(vm-nobodyavail)
exten => 7002,4,VoiceMail(7002@main)
exten => 7002,5,Hangup()

exten => 7003,1,Answer()
exten => 7003,2,Dial(PJSIP/7003,90)
exten => 7003,3,Playback(vm-nobodyavail)
exten => 7003,4.Voicemail(7003@main)
exten => 7003,5,Hangup()

exten => 7004,1,Answer()
exten => 7004,2,Dial(PJSIP/7004,90)
exten => 7004,3,Playback(vm-nobodyavail)
exten => 7004,4.Voicemail(7004@main)
exten => 7004,5,Hangup()

exten => 8001,1,VoicemailMain(7001@main)
exten => 8002,2,Hangup()

exten => 8002,1,VoicemailMain(7002@main)
exten => 8002,2,Hangup()

exten => 8003,1,VoicemailMain(7003@main)
exten => 8003,2,Hangup()

exten => 8004,1,VoicemailMain(7004@main)
exten => 8004,2,Hangup()

pjsip.conf
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0

[7001]
type=endpoint
context=internal
disallow=all
allow=ulaw
auth=7001
aors=7001

[7001]
type=auth
auth_type=userpass
password=7001
username=7001

[7001]
type=aor
max_contacts=1

My problem is why is Terminal-A is sending INVITE to Terminal-B after REFER.
I have attached the flow sequence marking the problem.

Capture
Capture853Ă—499 50.8 KB

2

Please use text for logs, preferably that created by “pjsip set logger on”, and taken from the log file, not a screen scrape. Even enlarging the images, I can’t read your traces.

Given that Asterisk doesn’t de-trombone on transfers, I assume it is sending connected line update information to both the remaining parties. (Some people use hairpin rather than trombone.)

3

I am uploading a partial part of flow graph which i got using
tcpdump on asterisk.

Capture
Capture1228Ă—644 83.8 KB

4

There’s still not enough information. You need to provide the complete SIP traffic.