Asterisk sends invite to transferor after refer

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.

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.)

1 Like

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

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

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.