Call disconnect after 6 seconds

i have issabel , my sip trunk is register and all phones can calls but when some one calls to our sip , calls disconnected after 6 seconds ( only incomings )

== Using SIP RTP CoS mark 5
> 0x7f0084249390 – Strict RTP learning after remote address set to: 10.1XX.XXX.XXX:32276
– Executing [000@from-trunk:1] NoOp(“SIP/SIP2-TCT–0000000e”, “Catch-All DID Match - Found 000 - You probably want a DID for this.”) in new stack
– Executing [000@from-trunk:2] Set(“SIP/SIP2-TCT–0000000e”, “__FROM_DID=000”) in new stack
– Executing [000@from-trunk:3] Goto(“SIP/SIP2-TCT–0000000e”, “ext-did,s,1”) in new stack
– Goto (ext-did,s,1)
– Executing [s@ext-did:1] ExecIf(“SIP/SIP2-TCT–0000000e”, “0?Set(__FROM_DID=s)”) in new stack
– Executing [s@ext-did:2] Gosub(“SIP/SIP2-TCT–0000000e”, “app-blacklist-check,s,1()”) in new stack
– Executing [s@app-blacklist-check:1] GotoIf(“SIP/SIP2-TCT–0000000e”, “0?blacklisted”) in new stack
– Executing [s@app-blacklist-check:2] Set(“SIP/SIP2-TCT–0000000e”, “CALLED_BLACKLIST=1”) in new stack
– Executing [s@app-blacklist-check:3] Return(“SIP/SIP2-TCT–0000000e”, “”) in new stack
– Executing [s@ext-did:3] Set(“SIP/SIP2-TCT–0000000e”, “__REC_POLICY_MODE=always”) in new stack
– Executing [s@ext-did:4] Set(“SIP/SIP2-TCT–0000000e”, “CDR(did)=74221000”) in new stack
– Executing [s@ext-did:5] ExecIf(“SIP/SIP2-TCT–0000000e”, “1 ?Set(CALLERID(name)=9394167131)”) in new stack
– Executing [s@ext-did:6] Set(“SIP/SIP2-TCT–0000000e”, “CHANNEL(musicclass)=Music-Hold”) in new stack
– Executing [s@ext-did:7] Set(“SIP/SIP2-TCT–0000000e”, “__MOHCLASS=Music-Hold”) in new stack
– Executing [s@ext-did:8] Set(“SIP/SIP2-TCT–0000000e”, “__CALLINGPRES_SV=allowed_not_screened”) in new stack
– Executing [s@ext-did:9] Set(“SIP/SIP2-TCT–0000000e”, “CALLERPRES()=allowed_not_screened”) in new stack
– Executing [s@ext-did:10] Goto(“SIP/SIP2-TCT–0000000e”, “ivr-2,s,1”) in new stack
– Goto (ivr-2,s,1)
– Executing [s@ivr-2:1] Set(“SIP/SIP2-TCT–0000000e”, “TIMEOUT_LOOPCOUNT=0”) in new stack
– Executing [s@ivr-2:2] Set(“SIP/SIP2-TCT–0000000e”, “INVALID_LOOPCOUNT=0”) in new stack
– Executing [s@ivr-2:3] Set(“SIP/SIP2-TCT–0000000e”, “_IVR_CONTEXT_ivr-2=”) in new stack
– Executing [s@ivr-2:4] Set(“SIP/SIP2-TCT–0000000e”, “_IVR_CONTEXT=ivr-2”) in new stack
– Executing [s@ivr-2:5] Set(“SIP/SIP2-TCT–0000000e”, “__IVR_RETVM=”) in new stack
– Executing [s@ivr-2:6] GotoIf(“SIP/SIP2-TCT–0000000e”, “0?skip”) in new stack
– Executing [s@ivr-2:7] Answer(“SIP/SIP2-TCT–0000000e”, “”) in new stack
– Executing [s@ivr-2:8] Wait(“SIP/SIP2-TCT–0000000e”, “1”) in new stack
– Executing [s@ivr-2:9] Set(“SIP/SIP2-TCT–0000000e”, “IVR_MSG=custom/Datis-IVR”) in new stack
– Executing [s@ivr-2:10] Set(“SIP/SIP2-TCT–0000000e”, “TIMEOUT(digit)=3”) in new stack
– Digit timeout set to 3.000
– Executing [s@ivr-2:11] ExecIf(“SIP/SIP2-TCT–0000000e”, “1?Background(custom/Datis-IVR)”) in new stack
– <SIP/SIP2-TCT–0000000e> Playing ‘custom/Datis-IVR.slin’ (language ‘en’)
> 0x7f00842490 – Strict RTP switching to RTP remote address 10.123.101.176:32276 as source
> 0x7f00842490 – Strict RTP learning complete - Locking on source address 10.123.101.176:32276
[2021-02-27 19:35:33] WARNING[3219]: chan_sip.c:4038 retrans_pkt: Retransmission timeout reached on transmission isbc0n1qj6q2qev8t1m6vjn2n31sj8tee66z@SoftX3000 for seqno 1 (Critical Response) – See SIP Retransmissions - Asterisk Project - Asterisk Project Wiki
Packet timed out after 6401ms with no response
[2021-02-27 19:35:33] WARNING[3219]: chan_sip.c:4067 retrans_pkt: Hanging up call isbc0n1qj6q2qev8t1m6vjn2n31sj8tee66z@SoftX3000 - no reply to our critical packet (see SIP Retransmissions - Asterisk Project - Asterisk Project Wiki).
== Spawn extension (ivr-2, s, 11) exited non-zero on ‘SIP/SIP2-TCT–0000000e’
– Executing [h@ivr-2:1] Hangup(“SIP/SIP2-TCT-74221-0000000e”, “”) in new stack
== Spawn extension (ivr-2, h, 1) exited non-zero on ‘SIP/SIP2-TCT–0000000e’

Have you looked at the link provided in the message?

Six seconds suggests you have overridden the normal timeouts, as this one normally happens at about 30 seconds. Incidentally, using the log file, rather than a screen scrape, helps a lot for timing issues, as it contains more time stamps.

This message means that the client never sent an ACK to the final response, or it got lost in transit. That can sometimes happen if you are behind NAT, but fail to provide your public address. Make sure that the Contact address, in the final response will actually get you back to your Asterisk system.

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