Call hangsup as after 1st ring

Asterisk Asterisk SIP
1

Hi,

Let me first explain my network, i have two interfaces on server
1 for local LAN, ip 192.168.1.221
2 for SIP PRI, ip 10.24.33.126

SIP trunk ip: 43.242.100.1 which is registered via 2nd interface.

now when i make an inbound call, it lands fine, plays IVR, when i press any extension (3 digit) or press 0 (as defined in IVR to ring operator) it takes DTMF fine. and start ringing device (D-Link IP phone which are connect over local LAN IP)
when it sends ring to device, it gives below error:

Retransmission timeout reached on transmission 2b2271971aade8d6739cb5467610da1b@10.24.33.126:5060 for seqno 104 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

here is sip trace of extension:

Audio is at 14626
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.1.212:5060:
INVITE sip:129@192.168.1.212:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.221:5060;branch=z9hG4bK591e4e9b;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.1.221;tag=as4ec9fb03
To: sip:129@192.168.1.212:5060
Contact: sip:asterisk@192.168.1.221:5060
Call-ID: 564c12f80f72fabe042093591b566c70@192.168.1.221:5060
Seq: 102 INVITE
User-Agent: Asterisk PBX 1.8.24.0
Date: Thu, 13 Sep 2018 17:19:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 1340143399 1340143399 IN IP4 192.168.1.221
s=Asterisk PBX 1.8.24.0
c=IN IP4 192.168.1.221
t=0 0
m=audio 14626 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

-- Called SIP/129

<— SIP read from UDP:192.168.1.212:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.221:5060;branch=z9hG4bK591e4e9b;rport=5060
From: “asterisk” sip:asterisk@192.168.1.221;tag=as4ec9fb03
To: sip:129@192.168.1.212:5060
Call-ID: 564c12f80f72fabe042093591b566c70@192.168.1.221:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Length: 0

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:192.168.1.212:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.221:5060;branch=z9hG4bK591e4e9b;rport=5060
From: “asterisk” sip:asterisk@192.168.1.221;tag=as4ec9fb03
To: sip:129@192.168.1.212:5060;tag=2739831761
Call-ID: 564c12f80f72fabe042093591b566c70@192.168.1.221:5060
CSeq: 102 INVITE
Contact: sip:129@192.168.1.212:5060
User-Agent: DLINK DPH-150SE FRU2.2.1328.545
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Length: 0

<------------->
— (10 headers 0 lines) —
list_route: hop: sip:129@192.168.1.212:5060
– SIP/129-00000005 is ringing
Reliably Transmitting (NAT) to 192.168.1.212:5060:
CANCEL sip:129@192.168.1.212:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.221:5060;branch=z9hG4bK591e4e9b;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.1.221;tag=as4ec9fb03
To: sip:129@192.168.1.212:5060
Call-ID: 564c12f80f72fabe042093591b566c70@192.168.1.221:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.8.24.0
Content-Length: 0

<— SIP read from UDP:192.168.1.212:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.221:5060;branch=z9hG4bK591e4e9b;rport=5060
From: “asterisk” sip:asterisk@192.168.1.221;tag=as4ec9fb03
To: sip:129@192.168.1.212:5060;tag=2739831761
Call-ID: 564c12f80f72fabe042093591b566c70@192.168.1.221:5060
CSeq: 102 CANCEL
User-Agent: DLINK DPH-150SE FRU2.2.1328.545
Content-Length: 0

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:192.168.1.212:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.221:5060;branch=z9hG4bK591e4e9b;rport=5060
From: “asterisk” sip:asterisk@192.168.1.221;tag=as4ec9fb03
To: sip:129@192.168.1.212:5060;tag=2739831761
Call-ID: 564c12f80f72fabe042093591b566c70@192.168.1.221:5060
CSeq: 102 INVITE
User-Agent: DLINK DPH-150SE FRU2.2.1328.545
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Transmitting (NAT) to 192.168.1.212:5060:
ACK sip:129@192.168.1.212:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.221:5060;branch=z9hG4bK591e4e9b;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.1.221;tag=as4ec9fb03
To: sip:129@192.168.1.212:5060;tag=2739831761
Contact: sip:asterisk@192.168.1.221:5060
Call-ID: 564c12f80f72fabe042093591b566c70@192.168.1.221:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.24.0
Content-Length: 0

Scheduling destruction of SIP dialog ‘564c12f80f72fabe042093591b566c70@192.168.1.221:5060’ in 6400 ms (Method: INVITE)
– <SIP/cyber-00000004>AGI Script agi://127.0.0.1:4577/call_log–HVcauses–PRI-----NODEBUG-----18-----CANCEL---------- completed, returning 0
[Sep 13 22:19:26] NOTICE[53476]: pbx_spool.c:385 attempt_thread: Call completed to SIP/cyber/03318676062

<— SIP read from UDP:192.168.1.212:5060 —>
OPTIONS sip:192.168.1.221:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.212:5060;branch=z9hG4bK26884218801408323915
From: “Qundeel Ahmed” sip:129@192.168.1.221:5060;tag=274027953
To: sip:192.168.1.221:5060
Call-ID: 98011376722495-110801078332341@192.168.1.212
CSeq: 1 OPTIONS
Max-Forwards: 70
User-Agent: DLINK DPH-150SE FRU2.2.1328.545
Accept: application/sdp
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Looking for s in trunkinbound (domain 192.168.1.221)

<— Transmitting (NAT) to 192.168.1.212:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.212:5060;branch=z9hG4bK26884218801408323915;received=192.168.1.212;rport=5060
From: “Qundeel Ahmed” sip:129@192.168.1.221:5060;tag=274027953
To: sip:192.168.1.221:5060;tag=as78d7f0f6
Call-ID: 98011376722495-110801078332341@192.168.1.212
CSeq: 1 OPTIONS
Server: Asterisk PBX 1.8.24.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘98011376722495-110801078332341@192.168.1.212’ in 32000 ms (Method: OPTIONS)
Reliably Transmitting (NAT) to 192.168.1.212:5060:
OPTIONS sip:129@192.168.1.212:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.221:5060;branch=z9hG4bK6b9d0bce;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.1.221;tag=as3628c805
To: sip:129@192.168.1.212:5060
Contact: sip:asterisk@192.168.1.221:5060
Call-ID: 7758a1266e6e7d471eab921f16f85bae@192.168.1.221:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.24.0
Date: Thu, 13 Sep 2018 17:19:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.1.212:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.221:5060;branch=z9hG4bK6b9d0bce;rport=5060
From: “asterisk” sip:asterisk@192.168.1.221;tag=as3628c805
To: sip:129@192.168.1.212:5060;tag=10561730
Call-ID: 7758a1266e6e7d471eab921f16f85bae@192.168.1.221:5060
CSeq: 102 OPTIONS
Contact: sip:129@192.168.1.212:5060
Supported: 100rel, replaces, timer
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Accept: application/sdp, message/sipfrag, application/dtmf-relay
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog ‘7758a1266e6e7d471eab921f16f85bae@192.168.1.221:5060’ Method: OPTIONS
Really destroying SIP dialog ‘564c12f80f72fabe042093591b566c70@192.168.1.221:5060’ Method: INVITE

2

I have set nat=force_rport in sip trunk settings.

asterisk version is 11.22-vici

3

Sometimes , i’ve the same problem and i resolve that with rebooting the network and the server. I think that the problem is the trafic in your network and your will must create a vlan to isolate the sip trafic.

4

I have already restarted things, but didn’t work anything. it was working fine, suddenly it started making this issue.

5

do you have vlan for your voice trafic?

Le jeu. 13 sept. 2018 à 20:07, s.khan asterisk@discoursemail.com a écrit :

6

no, server has lan ip and all the phones are connected directly to the server

7

The log shows that your end aborted the call, but doesn’t have enough information to indicate why it should have decided to do that.

8

should i share any other logs?

9

Update:

I tested the case with my SIP provider, and as per him, my system send INVITE packet again when it rings any extension, which is creating issue.
how can i limit invite packet? Also i am using this on Hyper-V and my physical interface is connected with SIP interface.

10

Re-INVITEs are perfectly valid and should not cause a problem with a valid peer (retransmitted ones certainly shouldn’t).

However your log shows a call being cancelled, before the final response. Asterisk will never send INVITE whilst an inbound one is incomplete.

A Re-INVITE, after the final response to the initial INVITE could be the result of:

Direct media: drectmedia=yes (or the obsolete form, canreinvite=yes).

Connected line update (sendrpid anything other than no).

Session timer refreshes or initialisation (the peer tries to negotiate them and Asterisk is not set to refuse them).