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
*sorry pake bhs inggris, copas dari grup sebelah, baru ngeh kalau ada forum ID nya,
mohon bantuan nya para suhu asterisk, hehe
Regards,
Utomo