Need Help, Call to SIP Trunk Provider Randomly Disconnected after 10 seconds (call connected)

Asterisk Asterisk SIP
1

Hi Everyone,

I have problem with my Asterisk (new implementation),
IP Phone (Yealink T19) able to do outbond call to PSTN via SIP Trunk,
able to talk two ways audio with called party,
but suddenly call disconnected after (around) 10 seconds,
this outbond call issue happen randomly,
somehow it happen but somehow call are normal.
Already talk with our SIP Trunk provider but they only give us codec preferences (alaw & ulaw), 5060 port, pitime 20, and SIP Trunk IP address.
And they said error code in their system indicate for error code 102 (call setup timeup failure).
There is no NAT and firewall in my environment.
Inbound call are normal.

Bellow link download for verbose 5 and sip debug logs from normal and disconnected call, also peer detail for SIP trunk provider :
https://drive.google.com/drive/folders/1rbeVghRVeyeU1cLJ34jGv0CA7tQTkrfN?usp=sharing

Calling number = 541024
Called number = 79021803XX20 (prepend 79, send 021803XX20 to SIP trunk Provider)
Please help solve this issue.

Bellow my environtment :
FreePBX 2.11.0.42
Asterisk 11.21.0
Yealink IP Phone T19

#sip show peer sip_telkom (SIP Trunk to Provider)
vt100*CLI> sip show peer sip_telkom

  • Name : sip_telkom
    Description :
    Secret :
    MD5Secret :
    Remote Secret:
    Context : from-trunk-sip-sip_telkom
    Record On feature : automon
    Record Off feature : automon
    Subscr.Cont. :
    Language :
    Tonezone :
    AMA flags : Unknown
    Transfer mode: open
    CallingPres : Presentation Allowed, Not Screened
    Callgroup :
    Pickupgroup :
    Named Callgr :
    Nam. Pickupgr:
    MOH Suggest :
    Mailbox :
    VM Extension : *97
    LastMsgsSent : 0/0
    Call limit : 0
    Max forwards : 0
    Dynamic : No
    Callerid : “” <>
    MaxCallBR : 384 kbps
    Expire : -1
    Insecure : no
    Force rport : Auto (No)
    Symmetric RTP: No
    ACL : No
    DirectMedACL : No
    T.38 support : No
    T.38 EC mode : Unknown
    T.38 MaxDtgrm: 4294967295
    DirectMedia : Yes
    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 : Yes
    DTMFmode : rfc2833
    Timer T1 : 500
    Timer B : 32000
    ToHost : 10.36.0.137
    Addr->IP : 10.36.0.137:5060
    Defaddr->IP : (null)
    Prim.Transp. : UDP
    Allowed.Trsp : UDP
    Def. Username:
    SIP Options : 100rel precondition replaces replace timer
    Codecs : (gsm|ulaw|alaw)
    Codec Order : (gsm:20,ulaw:20,alaw:20)
    Auto-Framing : No
    Status : OK (4 ms)
    Useragent :
    Reg. Contact :
    Qualify Freq : 60000 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

Regards,

Ovindo

2

There is no such error code in SIP. It is something internal for their exchange.

I do not see any error messages or faults. Looks like a normal dialog termination message (BYE) is coming from the provider’s side:

<--- SIP read from UDP:10.36.0.137:5060 --->
BYE sip:0541...@10.36.0.138:5060 SIP/2.0
Via: SIP/2.0/UDP 10.36.0.137:5060;branch=z9hG4bK00Bd3cce7885388b09d
From: <sip:02180...@10.36.0.137>;tag=gK00d1e24c
To: <sip:0541...@10.36.0.138>;tag=as060eeab2
Call-ID: 4ba6285e612c25562fb7d6c32476d54c@10.36.0.138:5060
CSeq: 292053 BYE
Max-Forwards: 70
Content-Length: 0
3

Sadly, Our SIP trunk provider said that there is no problem with their side,
the problem come from our (Asterisk) side,