Call drop 30 secs?

Asterisk Asterisk Support
1

i have installed the asterisk 1.8.6 . i can able to call also
but after 30sec , call is disconnected automatically and getting following warning message .

Using SIP RTP CoS mark 5
– Executing [2004@phones:1] Macro(“SIP/IMSI234261003917943-0000000b”, “dialGSM,IMSI234101493065396”) in new stack
– Executing [s@macro-dialGSM:1] Dial(“SIP/IMSI234261003917943-0000000b”, “SIP/IMSI234101493065396”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/IMSI234101493065396
– SIP/IMSI234101493065396-0000000c is ringing
– SIP/IMSI234101493065396-0000000c is ringing
– SIP/IMSI234101493065396-0000000c is ringing
– SIP/IMSI234101493065396-0000000c answered SIP/IMSI234261003917943-0000000b
– Locally bridging SIP/IMSI234261003917943-0000000b and SIP/IMSI234101493065396-0000000c
[Sep 14 01:06:04] WARNING[1224]: chan_sip.c:3620 retrans_pkt: Retransmission timeout reached on transmission 1726974616@127.0.0.1 for seqno 772 (Critical Response) – See wiki.asterisk.org/wiki/display/ … nsmissions
Packet timed out after 32000ms with no response
[Sep 14 01:06:04] WARNING[1224]: chan_sip.c:3649 retrans_pkt: Hanging up call 1726974616@127.0.0.1 - no reply to our critical packet (see wiki.asterisk.org/wiki/display/ … nsmissions).
== Spawn extension (macro-dialGSM, s, 1) exited non-zero on ‘SIP/IMSI234261003917943-0000000b’ in macro ‘dialGSM’
== Spawn extension (phones, 2004, 1) exited non-zero on ‘SIP/IMSI234261003917943-0000000b’

can anyone know the solution??

2

You need to provide SIP debug output to show which request is timing out. Is an X-Lite involved? At least some versions have broken re-invite handling which will cause this result, although I’m not sure whether you should have seen a “native bridging” message, or whether that only happens after the re-invite succeeds.

You could try disabling re-invites (directmedia=no).

3

I have tried with the directmedia = no and NAT = NO
NO SUCESSS call terimates after 30 sec…
here is sip debug

-- Called SIP/IMSI234261003917943

Retransmitting #1 (no NAT) to 127.0.0.1:5062:
INVITE sip:IMSI234261003917943@127.0.0.1:5062 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK25d7d1e5
Max-Forwards: 70
From: “IMSI234159103965685” sip:2001@127.0.0.1;tag=as0ec0e192
To: sip:IMSI234261003917943@127.0.0.1:5062
Contact: sip:2001@127.0.0.1:5060
Call-ID: 3315c0876d9e23ce78e5981b346dd11b@127.0.0.1:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.7.0-rc1
Date: Fri, 16 Sep 2011 22:38:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 252

v=0
o=root 1279317412 1279317412 IN IP4 127.0.0.1
s=Asterisk PBX 1.8.7.0-rc1
c=IN IP4 127.0.0.1
t=0 0
m=audio 17162 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<— SIP read from UDP:127.0.0.1:5062 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK25d7d1e5
From: “IMSI234159103965685” sip:2001@127.0.0.1;tag=as0ec0e192
To: sip:IMSI234261003917943@127.0.0.1:5062
Call-ID: 3315c0876d9e23ce78e5981b346dd11b@127.0.0.1:5060
CSeq: 102 INVITE
Contact: sip:IMSI234261003917943@127.0.0.1
Content-Length: 0

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

<— SIP read from UDP:127.0.0.1:5062 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK25d7d1e5
From: “IMSI234159103965685” sip:2001@127.0.0.1;tag=as0ec0e192
To: sip:IMSI234261003917943@127.0.0.1:5062;tag=xscjx
Call-ID: 3315c0876d9e23ce78e5981b346dd11b@127.0.0.1:5060
CSeq: 102 INVITE
Contact: sip:IMSI234261003917943@127.0.0.1
Content-Length: 0

<------------->
— (8 headers 0 lines) —
– SIP/IMSI234261003917943-0000000b is ringing

<— Transmitting (no NAT) to 127.0.0.1:5062 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK12167;received=127.0.0.1
From: IMSI234159103965685 sip:IMSI234159103965685@127.0.0.1;tag=ulnus
To: sip:2001@127.0.0.1;tag=as1de260e2
Call-ID: 1615669690@127.0.0.1
CSeq: 33 INVITE
Server: Asterisk PBX 1.8.7.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:2001@127.0.0.1:5060
Content-Length: 0

<------------>

<— SIP read from UDP:127.0.0.1:5062 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK25d7d1e5
From: “IMSI234159103965685” sip:2001@127.0.0.1;tag=as0ec0e192
To: sip:IMSI234261003917943@127.0.0.1:5062;tag=xscjx
Call-ID: 3315c0876d9e23ce78e5981b346dd11b@127.0.0.1:5060
CSeq: 102 INVITE
Contact: sip:IMSI234261003917943@127.0.0.1:5062;expires=3600
Content-Type: application/sdp
Content-Length: 135

v=0
o=IMSI234261003917943 0 0 IN IP4 127.0.0.1
s=Talk Time
t=0 0
m=audio 16544 RTP/AVP 3
c=IN IP4 127.0.0.1
a=rtpmap:3 GSM/8000
<------------->
— (9 headers 7 lines) —
Found RTP audio format 3
Found audio description format GSM for ID 3
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x2 (gsm)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x2 (gsm)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 127.0.0.1:16544
list_route: hop: sip:IMSI234261003917943@127.0.0.1:5062
set_destination: Parsing sip:IMSI234261003917943@127.0.0.1:5062 for address/port to send to
set_destination: set destination to 127.0.0.1:5062
Transmitting (no NAT) to 127.0.0.1:5062:
ACK sip:IMSI234261003917943@127.0.0.1:5062 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK27401ad6
Max-Forwards: 70
From: “IMSI234159103965685” sip:2001@127.0.0.1;tag=as0ec0e192
To: sip:IMSI234261003917943@127.0.0.1:5062;tag=xscjx
Contact: sip:2001@127.0.0.1:5060
Call-ID: 3315c0876d9e23ce78e5981b346dd11b@127.0.0.1:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.7.0-rc1
Content-Length: 0

-- SIP/IMSI234261003917943-0000000b answered SIP/IMSI234261003917943-0000000a

Audio is at 5060
Adding codec 0x2 (gsm) to SDP

<— Reliably Transmitting (no NAT) to 127.0.0.1:5062 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK12167;received=127.0.0.1
From: IMSI234159103965685 sip:IMSI234159103965685@127.0.0.1;tag=ulnus
To: sip:2001@127.0.0.1;tag=as1de260e2
Call-ID: 1615669690@127.0.0.1
CSeq: 33 INVITE
Server: Asterisk PBX 1.8.7.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:2001@127.0.0.1:5060
Content-Type: application/sdp
Content-Length: 204

v=0
o=root 1671736310 1671736310 IN IP4 127.0.0.1
s=Asterisk PBX 1.8.7.0-rc1
c=IN IP4 127.0.0.1
t=0 0
m=audio 17130 RTP/AVP 3
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
– Locally bridging SIP/IMSI234261003917943-0000000a and SIP/IMSI234261003917943-0000000b

<— SIP read from UDP:127.0.0.1:5062 —>
ACK sip:2001@127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK61943
From: IMSI234159103965685 sip:IMSI234159103965685@127.0.0.1;tag=as1de260e2
To: sip:2001@127.0.0.1;tag=ulnus
Call-ID: 1615669690@127.0.0.1
CSeq: 33 INVITE
Max-Forwards: 70
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Retransmitting #1 (no NAT) to 127.0.0.1:5062:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK12167;received=127.0.0.1
From: IMSI234159103965685 sip:IMSI234159103965685@127.0.0.1;tag=ulnus
To: sip:2001@127.0.0.1;tag=as1de260e2
Call-ID: 1615669690@127.0.0.1
CSeq: 33 INVITE
Server: Asterisk PBX 1.8.7.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:2001@127.0.0.1:5060
Content-Type: application/sdp
Content-Length: 204

v=0
o=root 1671736310 1671736310 IN IP4 127.0.0.1
s=Asterisk PBX 1.8.7.0-rc1
c=IN IP4 127.0.0.1
t=0 0
m=audio 17130 RTP/AVP 3
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Retransmitting #2 (no NAT) to 127.0.0.1:5062:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK12167;received=127.0.0.1
From: IMSI234159103965685 sip:IMSI234159103965685@127.0.0.1;tag=ulnus
To: sip:2001@127.0.0.1;tag=as1de260e2
Call-ID: 1615669690@127.0.0.1
CSeq: 33 INVITE
Server: Asterisk PBX 1.8.7.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:2001@127.0.0.1:5060
Content-Type: application/sdp
Content-Length: 204

v=0
o=root 1671736310 1671736310 IN IP4 127.0.0.1
s=Asterisk PBX 1.8.7.0-rc1
c=IN IP4 127.0.0.1
t=0 0
m=audio 17130 RTP/AVP 3
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Retransmitting #3 (no NAT) to 127.0.0.1:5062:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK12167;received=127.0.0.1
From: IMSI234159103965685 sip:IMSI234159103965685@127.0.0.1;tag=ulnus
To: sip:2001@127.0.0.1;tag=as1de260e2
Call-ID: 1615669690@127.0.0.1
CSeq: 33 INVITE
Server: Asterisk PBX 1.8.7.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:2001@127.0.0.1:5060
Content-Type: application/sdp
Content-Length: 204

v=0
o=root 1671736310 1671736310 IN IP4 127.0.0.1
s=Asterisk PBX 1.8.7.0-rc1
c=IN IP4 127.0.0.1
t=0 0
m=audio 17130 RTP/AVP 3
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Retransmitting #4 (no NAT) to 127.0.0.1:5062:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK12167;received=127.0.0.1
From: IMSI234159103965685 sip:IMSI234159103965685@127.0.0.1;tag=ulnus
To: sip:2001@127.0.0.1;tag=as1de260e2
Call-ID: 1615669690@127.0.0.1
CSeq: 33 INVITE
Server: Asterisk PBX 1.8.7.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:2001@127.0.0.1:5060
Content-Type: application/sdp
Content-Length: 204

v=0
o=root 1671736310 1671736310 IN IP4 127.0.0.1
s=Asterisk PBX 1.8.7.0-rc1
c=IN IP4 127.0.0.1
t=0 0
m=audio 17130 RTP/AVP 3
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Retransmitting #5 (no NAT) to 127.0.0.1:5062:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK12167;received=127.0.0.1
From: IMSI234159103965685 sip:IMSI234159103965685@127.0.0.1;tag=ulnus
To: sip:2001@127.0.0.1;tag=as1de260e2
Call-ID: 1615669690@127.0.0.1
CSeq: 33 INVITE
Server: Asterisk PBX 1.8.7.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:2001@127.0.0.1:5060
Content-Type: application/sdp
Content-Length: 204

v=0
o=root 1671736310 1671736310 IN IP4 127.0.0.1
s=Asterisk PBX 1.8.7.0-rc1
c=IN IP4 127.0.0.1
t=0 0
m=audio 17130 RTP/AVP 3
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<— SIP read from UDP:127.0.0.1:5062 —>
INFO sip:2001@127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5062;branch=
From: IMSI234261003917943 sip:IMSI234261003917943@127.0.0.1;tag=as0ec0e192
To: sip:2001@127.0.0.1
Call-ID: 3315c0876d9e23ce78e5981b346dd11b@127.0.0.1
CSeq: 934 INFO
Content-Type: application/dtmf-relay
Content-Length: 21

Signal=2
Duration=200
<------------->
— (8 headers 2 lines) —
Retransmitting #6 (no NAT) to 127.0.0.1:5062:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK12167;received=127.0.0.1
From: IMSI234159103965685 sip:IMSI234159103965685@127.0.0.1;tag=ulnus
To: sip:2001@127.0.0.1;tag=as1de260e2
Call-ID: 1615669690@127.0.0.1
CSeq: 33 INVITE
Server: Asterisk PBX 1.8.7.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:2001@127.0.0.1:5060
Content-Type: application/sdp
Content-Length: 204

v=0
o=root 1671736310 1671736310 IN IP4 127.0.0.1
s=Asterisk PBX 1.8.7.0-rc1
c=IN IP4 127.0.0.1
t=0 0
m=audio 17130 RTP/AVP 3
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Retransmitting #7 (no NAT) to 127.0.0.1:5062:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK12167;received=127.0.0.1
From: IMSI234159103965685 sip:IMSI234159103965685@127.0.0.1;tag=ulnus
To: sip:2001@127.0.0.1;tag=as1de260e2
Call-ID: 1615669690@127.0.0.1
CSeq: 33 INVITE
Server: Asterisk PBX 1.8.7.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:2001@127.0.0.1:5060
Content-Type: application/sdp
Content-Length: 204

v=0
o=root 1671736310 1671736310 IN IP4 127.0.0.1
s=Asterisk PBX 1.8.7.0-rc1
c=IN IP4 127.0.0.1
t=0 0
m=audio 17130 RTP/AVP 3
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Retransmitting #8 (no NAT) to 127.0.0.1:5062:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK12167;received=127.0.0.1
From: IMSI234159103965685 sip:IMSI234159103965685@127.0.0.1;tag=ulnus
To: sip:2001@127.0.0.1;tag=as1de260e2
Call-ID: 1615669690@127.0.0.1
CSeq: 33 INVITE
Server: Asterisk PBX 1.8.7.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:2001@127.0.0.1:5060
Content-Type: application/sdp
Content-Length: 204

v=0
o=root 1671736310 1671736310 IN IP4 127.0.0.1
s=Asterisk PBX 1.8.7.0-rc1
c=IN IP4 127.0.0.1
t=0 0
m=audio 17130 RTP/AVP 3
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Retransmitting #9 (no NAT) to 127.0.0.1:5062:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK12167;received=127.0.0.1
From: IMSI234159103965685 sip:IMSI234159103965685@127.0.0.1;tag=ulnus
To: sip:2001@127.0.0.1;tag=as1de260e2
Call-ID: 1615669690@127.0.0.1
CSeq: 33 INVITE
Server: Asterisk PBX 1.8.7.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:2001@127.0.0.1:5060
Content-Type: application/sdp
Content-Length: 204

v=0
o=root 1671736310 1671736310 IN IP4 127.0.0.1
s=Asterisk PBX 1.8.7.0-rc1
c=IN IP4 127.0.0.1
t=0 0
m=audio 17130 RTP/AVP 3
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Retransmitting #10 (no NAT) to 127.0.0.1:5062:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK12167;received=127.0.0.1
From: IMSI234159103965685 sip:IMSI234159103965685@127.0.0.1;tag=ulnus
To: sip:2001@127.0.0.1;tag=as1de260e2
Call-ID: 1615669690@127.0.0.1
CSeq: 33 INVITE
Server: Asterisk PBX 1.8.7.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:2001@127.0.0.1:5060
Content-Type: application/sdp
ontent-Length: 204

v=0
o=root 1671736310 1671736310 IN IP4 127.0.0.1
s=Asterisk PBX 1.8.7.0-rc1
c=IN IP4 127.0.0.1
t=0 0
m=audio 17130 RTP/AVP 3
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

[Sep 16 15:39:20] WARNING[1016]: chan_sip.c:3622 retrans_pkt: Retransmission timeout reached on transmission 1615669690@127.0.0.1 for seqno 33 (Critical Response) – See wiki.asterisk.org/wiki/display/ … nsmissions
Packet timed out after 32001ms with no response
[Sep 16 15:39:20] WARNING[1016]: chan_sip.c:3651 retrans_pkt: Hanging up call 1615669690@127.0.0.1 - no reply to our critical packet (see wiki.asterisk.org/wiki/display/ … nsmissions).
[Sep 16 15:39:20] ERROR[23850]: cdr_csv.c:318 csv_log: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : Permission denied
Scheduling destruction of SIP dialog ‘3315c0876d9e23ce78e5981b346dd11b@127.0.0.1:5060’ in 32000 ms (Method: INVITE)
set_destination: Parsing sip:IMSI234261003917943@127.0.0.1:5062 for address/port to send to
set_destination: set destination to 127.0.0.1:5062
Reliably Transmitting (no NAT) to 127.0.0.1:5062:
BYE sip:IMSI234261003917943@127.0.0.1:5062 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK6141cc5d
Max-Forwards: 70
From: “IMSI234159103965685” sip:2001@127.0.0.1;tag=as0ec0e192
To: sip:IMSI234261003917943@127.0.0.1:5062;tag=xscjx
Call-ID: 3315c0876d9e23ce78e5981b346dd11b@127.0.0.1:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.8.7.0-rc1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

== Spawn extension (macro-dialGSM, s, 1) exited non-zero on ‘SIP/IMSI234261003917943-0000000a’ in macro ‘dialGSM’
== Spawn extension (phones, 2001, 1) exited non-zero on 'SIP/IMSI234261003917943-0000000a’
Scheduling destruction of SIP dialog ‘1615669690@127.0.0.1’ in 32000 ms (Method: INVITE)
set_destination: Parsing sip:IMSI234159103965685@127.0.0.1:5062 for address/port to send to
set_destination: set destination to 127.0.0.1:5062
Reliably Transmitting (no NAT) to 127.0.0.1:5062:
BYE sip:IMSI234159103965685@127.0.0.1:5062 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK6a80b003
Max-Forwards: 70
From: sip:2001@127.0.0.1;tag=as1de260e2
To: XXXXXXXXXXXXXXXX sip:XXXXXXXXX@127.0.0.1;tag=ulnus
Call-ID: 1615669690@127.0.0.1
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.7.0-rc1
X-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
Content-Length: 0

<— SIP read from UDP:127.0.0.1:5062 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK6a80b003
From: sip:2001@127.0.0.1;tag=as1de260e2
To: XXXXXXXXXXXXsip:XXXXXXXXXXX@127.0.0.1;tag=ulnus
Call-ID: 1615669690@127.0.0.1
CSeq: 102 BYE
Content-Length: 0

<------------->
— (7 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog ‘1615669690@127.0.0.1’ Method: INVITE

<— SIP read from UDP:127.0.0.1:5062 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK6141cc5d
From: “XXXXX” sip:2001@127.0.0.1;tag=as0ec0e192
To: sip:XXXXX@127.0.0.1:5062;tag=xscjx
Call-ID: 3315c0876d9e23ce78e5981b346dd11b@127.0.0.1:5060
CSeq: 103 BYE
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘3315c0876d9e23ce78e5981b346dd11b@127.0.0.1:5060’ Method: INVITE

4

It would help to see the original incoming INVITE, but either the outgoing OK is malformed, or the incoming ACK is malformed. The tags have got mixed up between From and To headers. As a result, Asterisk is not recognizing the ACK and is resending its OK until it times out.

I strongly suspect that the software running on port 5062 is broken, as, if Asterisk is broken, it would make its SIP implementation so broken that everyone would have noticed it.

5

yes ur right, asterisk is running on the port 5060, and sip is running the 5062. i didn’t face any problem when i used the asterisk 1.6.20, but when i moved to the asterisk 1.8 , there is a call drop after 30 when i call…
can you me help please …

6

The call is being dropped for the reasons I stated: Asterisk is not recognizing the ACK which completes the call setup so believes the setup has failed to complete.

It is not recognizing it because the tag fields are on the wrong headers. It seems likely that the fault is on the other side of the connection, but the only way of being sure is if you provide a trace that contains the whole of the INVITE - 200 OK - ACK handshake. However, given that Asterisk doesn’t use the legacy format for tags, I think it almost certain that it is the other side that is broken.

Maybe 1.6 didn’t bother to check the tags?

7

Hi,

I have the same problem, the call is dropped after 6399ms/6400ms…

If I set qualify=false, the call is dropped randomly, sometimes after 15 seconds sometimes after 20 seconds, but the quality is poor.

The router is set to forward 10000 to 20000 udp, but nothings change…
This issue is only in my version of asterisk 1.8.7, my old asterisk 1.6 works perfectly…

Any suggestion?

Thanks

Andrea

8

It is unlikely you have the same problem. Go back to the start of the thread and try the fixes suggested and obtain the same debugging information.

9

I have reisntalled following the same process, and now it seems to works fine…
This is the world of computer science… :wink:
Thanks in advance

A.