I facing a problem with Asterisk 13.38.2, and hope someone can help me. When my Asterisk receives a sip invite and then dial to extension (like Dial(SIP/103)) it works as expected, generating only one invite to extension 103.
When the same DID is routed to a queue, then Asterisk creates two sip invites (one for the extension as expected and another to the caller contact address).
This only happens with a specific provider, so I think it might be caused by some INVITE parameter.
Any idea why the last invite is created?
asterisk -rx “queue show 600”:
600 has 0 calls (max unlimited) in 'linear' strategy (0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 60s
Members:
Local/103@test-queue (ringinuse enabled) (dynamic) (Not in use) has taken no calls yet
No Callers
This is my test dial plan:
# extension dial context
[test-queue]
exten => _X.,1,Dial(SIP/103)
# incoming context
[test-int]
exten => s,1,Answer()
same => n,Queue(600)
These are the invites:
# Invite received by Asterisk
INVITE sip:+2020202020@1.1.1.180:5060 SIP/2.0
Record-Route: <sip:1.1.1.18:5060;r2=on;lr;ftag=p65540t1617838888m773513c7005s1_2017909308-538403679>
Record-Route: <sip:1.1.1.18;transport=tcp;r2=on;lr;ftag=p65540t1617838888m773513c7005s1_2017909308-538403679>
Via: SIP/2.0/UDP 1.1.1.18:5060;branch=z9hG4bK4b8e.e54a976359c351d06657262d335cf2ea.0;i=1
Via: SIP/2.0/TCP 1.1.1.38:5060;branch=z9hG4bKkitt7g10305hb5b2mao0.1
To: <sip:+2020202020@1.1.1.180;user=phone>
From: sip:+90909090@1.1.1.18;user=phone;tag=p65540t1617838888m773513c7005s1_2017909308-538403679
Call-ID: p65540t1617838888m773513c7005s2
CSeq: 1 INVITE
Max-Forwards: 65
Content-Length: 1060
Contact: <sip:p65540t1617838888m773513c7005s1@1.1.1.38:5060;transport=tcp>
Content-Type: application/sdp
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE
Accept: application/sdp
Accept-Contact: xxxxxxxxxxxxxxxxx
Supported: timer, 100rel, histinfo
P-Asserted-Identity: <sip:+90909090@1.1.1.38;cpc=ordinary>
History-Info: <sip:+2020202020@ims2xxxxxxxx;user=phone>;index=1
History-Info: <sip:+2020202020@ims2xxxxxxxx;cause=302;user=phone>;index=1.1
Min-SE: 900
Session-Expires: 1800
P-Charging-Vector: icid-value=mOTin606e4328d92c500100001b5d;icid-generated-at=ssdfd.xxxx.xxx.xx;orig-ioi=xxxx.xxx.xx;orig-icid=13212312131231
P-Early-Media: supported
Session-ID: a3a86bb42ac1e8c58592a9663bbd5ed5
# Invite sent to extension (correct)
INVITE sip:aguc9dt7@dachr2hc3gfe.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 1.1.1.180:5060;branch=z9hG4bK544ff0c7;rport
Max-Forwards: 70
From: "+90909090" <sip:+90909090@1.1.1.180>;tag=as185ed728
To: <sip:aguc9dt7@dachr2hc3gfe.invalid;transport=ws>
Contact: <sip:+90909090@1.1.1.180:5060;transport=ws>
Call-ID: 5ece1d376d522a03461cb9671f6af1ec@1.1.1.180:5060
CSeq: 102 INVITE
Date: Wed, 07 Apr 2021 23:41:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 794
# No idea why Dial(SIP/103) created this invite!
INVITE sip:p65540t1617838888m773513c7005s1@1.1.1.38:5060;transport=tcp SIP/2.0
Via: SIP/2.0/UDP 1.1.1.180:5060;branch=z9hG4bK77194dbb
Route: <sip:1.1.1.18:5060;r2=on;lr;ftag=p65540t1617838888m773513c7005s1_2017909308-538403679>,<sip:1.1.1.18;transport=tcp;r2=on;lr;ftag=p65540t1617838888m773513c7005s1_2017909308-538403679>
Max-Forwards: 70
From: <sip:+2020202020@1.1.1.180;user=phone>;tag=as7a03d243
To: sip:+90909090@1.1.1.18;user=phone;tag=p65540t1617838888m773513c7005s1_2017909308-538403679
Contact: <sip:+2020202020@1.1.1.180:5060>
Call-ID: p65540t1617838888m773513c7005s2
CSeq: 102 INVITE
Session-Expires: 1800;refresher=uac
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: "Test" <sip:103@1.1.1.18>
Content-Type: application/sdp
Content-Length: 614