Dial - Bridge Event sequentially resend

Asterisk Asterisk APIs
1

Hi,

When I made a call with Dial() command, and caller answer the call, system send sequentially (every 15 seconds) an Unlink and a Link Bridge event. How could I disable this to resend events or why Asterisk rebridge the call?

Event_1_15sec = {
‘Event’ => ‘Bridge’,
‘CallerID2’ => ‘xxxx’,
‘Bridgestate’ => ‘Unlink’,
‘Bridgetype’ => ‘core’,
‘Channel1’ => ‘SIP/xxxx-00000055’,
‘Uniqueid1’ => ‘1476272283.85’,
‘Privilege’ => ‘call,all’,
‘CallerID1’ => ‘xxxx’,
‘Uniqueid2’ => ‘1476272283.86’,
‘Channel2’ => ‘SIP/xxxx-00000056’
};
Event_1_15sec = {
‘Uniqueid1’ => ‘1476272283.85’,
‘Privilege’ => ‘call,all’,
‘CallerID1’ => ‘xxxx’,
‘Uniqueid2’ => ‘1476272283.86’,
‘Channel2’ => ‘SIP/xxxx-00000056’,
‘Event’ => ‘Bridge’,
‘Bridgetype’ => ‘core’,
‘CallerID2’ => ‘xxxx’,
‘Bridgestate’ => ‘Link’,
‘Channel1’ => ‘SIP/xxxx-00000055’
};

Event_2_30sec = {
‘Privilege’ => ‘call,all’,
‘Uniqueid1’ => ‘1476272283.85’,
‘Channel2’ => ‘SIP/xxxx-00000056’,
‘Uniqueid2’ => ‘1476272283.86’,
‘CallerID1’ => ‘xxxx’,
‘Event’ => ‘Bridge’,
‘Channel1’ => ‘SIP/xxxx-00000055’,
‘CallerID2’ => ‘xxxx’,
‘Bridgestate’ => ‘Unlink’,
‘Bridgetype’ => ‘core’
};
$Event_2_30sec = {
‘CallerID2’ => ‘xxxx’,
‘Bridgetype’ => ‘core’,
‘Bridgestate’ => ‘Link’,
‘Channel1’ => ‘SIP/xxxx-00000055’,
‘Event’ => ‘Bridge’,
‘CallerID1’ => ‘xxxx’,
‘Uniqueid2’ => ‘1476272283.86’,
‘Channel2’ => ‘SIP/xxxx-00000056’,
‘Uniqueid1’ => ‘1476272283.85’,
‘Privilege’ => ‘call,all’
};

…

Thanks

2

What Dial string are you using and what is the complete console output?

3

Dial:
exten => _X.,n,Dial(SIP/${DIALED}@${PROVIDER},120,${OUTGOINGTRANSFER}RL(${HANGUPTIME})M(outgoing-mixmonitor,${REC_OUT}))

Console output:
Executing [xxxx@outgoing:38] Dial(“SIP/xxxx01-00000059”, “SIP/xxxx@provider,120,TRL(7200000)M(outgoing-mixmonitor,1)”) in new stack

And here is the sip show peer result of the caller:

  • Name : xxxx
    Description :
    Realtime peer: Yes, cached
    Secret :
    MD5Secret :
    Remote Secret:
    Context : xxxx
    Record On feature : automon
    Record Off feature : automon
    Subscr.Cont. : xxxx-hints
    Language :
    Tonezone :
    Accountcode : SIP/xxxx
    AMA flags : Unknown
    Transfer mode: open
    CallingPres : Presentation Allowed, Not Screened
    Callgroup :
    Pickupgroup :
    Named Callgr :
    Nam. Pickupgr:
    MOH Suggest :
    Mailbox :
    VM Extension : asterisk
    LastMsgsSent : 0/0
    Call limit : 1
    Max forwards : 0
    Dynamic : Yes
    Callerid : “” <201>
    MaxCallBR : 384 kbps
    Expire : 97
    Insecure : no
    Force rport : Yes
    Symmetric RTP: Yes
    ACL : No
    DirectMedACL : No
    T.38 support : No
    T.38 EC mode : Unknown
    T.38 MaxDtgrm: 4294967295
    DirectMedia : No
    PromiscRedir : No
    User=Phone : No
    Video Support: No
    Text Support : No
    Ign SDP ver : No
    Trust RPID : No
    Send RPID : No
    TrustIDOutbnd: Legacy
    Subscriptions: Yes
    Overlap dial : No
    DTMFmode : auto
    Timer T1 : 500
    Timer B : 32000
    ToHost :
    Addr->IP : XXXX
    Defaddr->IP : (null)
    Prim.Transp. : TCP
    Allowed.Trsp : UDP,TCP
    Def. Username: xxxx
    SIP Options : (none)
    Codecs : (gsm|ulaw|alaw|opus|vp8)
    Codec Order : (gsm:20,alaw:20,ulaw:20,opus:20,vp8:0)
    Auto-Framing : No
    Status : OK (46 ms)
    Useragent : XXXX
    Reg. Contact : sip:xxxx@XXXX;transport=TCP
    Qualify Freq : 120000 ms
    Keepalive : 0 ms
    Sess-Timers : Accept
    Sess-Refresh : uas
    Sess-Expires : 1800 secs
    Min-Sess : 90 secs
    RTP Engine : asterisk
    Parkinglot :
    Use Reason : No
    Encryption : No
4

It is likely something to do with features in use, and there’s no way to explicitly disable those repeated ones.

5

Hi,

I made a sip debug and every 15 secs it’s look like we get a re-invite, but in sip configuration everywhere we disabled it: canreinvite = no

Could you please check this log:

<— SIP read from UDP:X.X.X.X:1405 —>
INVITE sip: CALLER_NUMBER@Y.Y.Y.Y:5060 SIP/2.0
Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK-d8754z-804b4e4c564f7903-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:CALLED_NUMBER@X.X.X.X:5060
To: <sip CALLER_NUMBER@Y.Y.Y.Y> tag=as7d31b97f
From: <sip: CALLED_NUMBER@SIP_PROVIDER:5060>;tag=ce996f1501de205cb372da46f7dffd25
Call-ID: 2f1f634a3bc94e080f9d7e591e70ca8a@Y.Y.Y.Y:5060
CSeq: 1476369700 INVITE
Allow: INVITE, CANCEL, ACK, BYE, REGISTER, OPTIONS, REFER, INFO, PRACK
Content-Type: application/sdp
User-Agent: Deverto Tequet SoftSwitch 6.2.2-8
Content-Length: 237

v=0
o=- 1688838050 35212037 IN IP4 X.X.X.X
s=-
c=IN IP4 185.51.66.70
t=0 0
m=audio 15292 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
— (12 headers 12 lines) —
Sending to X.X.X.X:1405 (NAT)

<— Transmitting (NAT) to X.X.X.X:1405 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK-d8754z-804b4e4c564f7903-1—d8754z-;received=X.X.X.X;rport=1405
From: sip:CALLED_NUMBER@SIP_PROVIDER:5060;tag=ce996f1501de205cb372da46f7dffd25
To: sip:CALLER_NUMBER@Y.Y.Y.Y tag=as7d31b97f
Call-ID: 2f1f634a3bc94e080f9d7e591e70ca8a@Y.Y.Y.Y:5060
CSeq: 1476369700 INVITE
Server: MinervaTel
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:CALLER_NUMBER@Y.Y.Y.Y:5060
Content-Length: 0

6

That’s a reinvite from the remote side. You can’t control if/when they will do that.