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Summary:ASTERISK-17829: Duplicate SIP 180 / Ringing Messages in SIP trunk ingress
Reporter:JoshE (n8ideas)Labels:
Date Opened:2011-05-10 13:38:41Date Closed:2011-07-26 14:18:46
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:1.6.2.16 Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:Have a SIP trunk and a single SIP peer.  When generating an inbound call, I am seeing duplicate 180/Ringing SIP messages.  Carrier is complaining about this.  SIP trunk setup is:


[NBS]
qualify=no
ignoresdpversion=yes
nat=no
host=205.x,x,x
reinvite=no
dtmfmode=auto
context=from-outside
type=friend
insecure=port,invite
canreinvite=no
disallow=all
allow=ulaw
allow=g729

Peer is:

[101]
qualify=yes
nat=yes
pickupgroup=57
callerid= Demo <101>
context=from-inside
canreinvite=no
vmexten=101
secret=xxxx
host=dynamic
username=101
subscribecontext=local-extensions
callgroup=57
dtmfmode=rfc2833
type=friend
mailbox=101@default
disallow=all
allow=ulaw



****** ADDITIONAL INFORMATION ******

SIP Message Flow is:

SIP Trunk INVITE -> Asterisk Returns 100 Trying ->
Asterisk Sends 180 / Ringing ->
Asterisk Sends 180 / Ringing

Relevant scrubbed SIP debug output is here:


   -- Called 101

<--- Transmitting (no NAT) to 205.x.x.x:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 205.x.x.x:5060;branch=z9hG4bK04B97f53cc76cf6237c;received=205.x.x.x
From: "USER" <sip:+1nnnnnnnnnn@205.x.x.x;pstn-params=808482808882;otg=GLAB0>;tag=gK04681c85
To: <sip:214nnnnnnn@174.x.x.x>;tag=as2ca3199e
Call-ID: 134499423_1660842@205.x.x.x
CSeq: 2816 INVITE
Server: Asterisk PBX 1.6.2.16.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:+1pppppppppp@174.x.x.x>
Content-Length: 0


<------------>
   -- SIP/101-0000360a is ringing

<--- Transmitting (no NAT) to 205.x.x.x:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 205.x.x.x:5060;branch=z9hG4bK04B97f53cc76cf6237c;received=205.x.x.x
From: "USER" <sip:+1nnnnnnnnnn@205.x.x.x;pstn-params=808482808882;otg=GLAB0>;tag=gK04681c85
To: <sip:214nnnnnnn@174.x.x.x>;tag=as2ca3199e
Call-ID: 134499423_1660842@205.x.x.x
CSeq: 2816 INVITE
Server: Asterisk PBX 1.6.2.16.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:+1pppppppppp@174.x.x.x>
Content-Length: 0
Comments:By: Leif Madsen (lmadsen) 2011-05-11 13:03:06

Please provide the dialplan that causes this issue. Thanks
By: Leif Madsen (lmadsen) 2011-05-11 13:03:24

(02:02:19 PM) jcolp: they might have that 'r' option set on Dial so it tells the SIP channel to send ringing, which it does
(02:02:32 PM) jcolp: and then later it gets told again to send ringing, which it does, again
(02:02:45 PM) jcolp: (although it shouldn't, since it already did)
By: Russell Bryant (russell) 2011-07-26 14:18:16.086-0500

Per the Asterisk maintenance timeline page at http://www.asterisk.org/asterisk-versions maintenance (bug) support for the 1.4 and 1.6.x branches has ended. For continued maintenance support please move to the 1.8 branch which is a long term support (LTS) branch. For more information about branch support, please see https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

If this is still an issue, please open a new issue so it can be re-triaged appropriately. Thanks!