Re-invite to non-existing call leg on other UA

danilo7 · October 10, 2012, 9:01am

Hi everyone,
I’m trying to set up two Asterisk boxes with a trunk, but the calls (without timeouts) on the trunk keep getting dropped after around 1 minute ringing.
Here are more details:

on both boxes there’s Asterisk Version: 1.6.2.13
the sip.conf on the first box

[code]
[general]
language=it
bindaddr=10.1.30.100
register => astBox01:trunkpw@10.2.31.240

useragent = Asterisk-PBX-01

match_auth_username=yes

ignoresdpversion=yes
session-timers=refuse
session-expires=120

[astBox02]
username=astBox02
authname=astBox02
type=peer
secret=trunkpw
qualify=yes
nat=no
language=it
insecure=port,invite
host=dynamic
disallow=all
allow=alaw
context=from-ast02
canreinvite=no[/code]

the sip.conf on the second box

[code]
[general]
language=it
bindaddr=10.2.31.240
register => astBox02:trunkpw@10.1.30.100
useragent=Asterisk-PBX-02

tos_sip=ef ; Sets TOS for SIP packets.
tos_audio=ef ; Sets TOS for RTP audio packets.

;match_auth_username=yes

ignoresdpversion=yes
session-timers=refuse
session-expires=120

[astBox01]
username=astBox01
authname=astBox01
type=peer
secret=trunkpw
qualify=yes
nat=no
language=it
insecure=port,invite
host=dynamic
disallow=all
allow=alaw
context=from-OCC
canreinvite=no[/code]

I try to place a call from a phone registered on Asterisk PBX 01 (extension 01000101) to a phone registered on Asterisk PBX 02 (extension 00100600)

But after around 1 minute ringing (I need it to keep ringing without timeout) the call drops and I get this

[Oct 9 18:39:36] WARNING[7314]: chan_sip.c:17946 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '26ececda0b44da2e3ff2f22f5d220438@10.1.30.100'. Giving up. -- SIP/00100600-00000009 is circuit-busy

I attach the trace during the call using the debug mode

[code]SIP Debugging enabled

<— SIP read from UDP:10.2.31.240:5060 —>
INVITE sip:00100600@10.1.30.100 SIP/2.0
Via: SIP/2.0/UDP 10.2.31.240:5060;branch=z9hG4bK3e7afad9;rport
Max-Forwards: 70
From: “HP 002 DS” sip:01000203@10.2.31.240;tag=as1300bb6a
To: sip:00100600@10.1.30.100
Contact: sip:01000203@10.2.31.240
Call-ID: 6d3ce4fa5149289826bde34648731554@10.2.31.240
CSeq: 102 INVITE
User-Agent: Asterisk-PBX-02
Date: Tue, 09 Oct 2012 16:37:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 1635627417 1635627417 IN IP4 10.2.31.240
s=Asterisk PBX 1.6.2.13
c=IN IP4 10.2.31.240
t=0 0
m=audio 14986 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------->
— (14 headers 11 lines) —
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Sending to 10.2.31.240 : 5060 (no NAT)
Using INVITE request as basis request - 6d3ce4fa5149289826bde34648731554@10.2.31.240
Found peer ‘01000203’ for ‘01000203’ from 10.2.31.240:5060

<— Reliably Transmitting (no NAT) to 10.2.31.240:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.2.31.240:5060;branch=z9hG4bK3e7afad9;received=10.2.31.240;rport=5060
From: “HP 002 DS” sip:01000203@10.2.31.240;tag=as1300bb6a
To: sip:00100600@10.1.30.100;tag=as365e18f3
Call-ID: 6d3ce4fa5149289826bde34648731554@10.2.31.240
CSeq: 102 INVITE
Server: Asterisk-PBX-01
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="281487f6"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘6d3ce4fa5149289826bde34648731554@10.2.31.240’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:10.2.31.240:5060 —>
ACK sip:00100600@10.1.30.100 SIP/2.0
Via: SIP/2.0/UDP 10.2.31.240:5060;branch=z9hG4bK3e7afad9;rport
Max-Forwards: 70
From: “HP 002 DS” sip:01000203@10.2.31.240;tag=as1300bb6a
To: sip:00100600@10.1.30.100;tag=as365e18f3
Contact: sip:01000203@10.2.31.240
Call-ID: 6d3ce4fa5149289826bde34648731554@10.2.31.240
CSeq: 102 ACK
User-Agent: Asterisk-PBX-02
Content-Length: 0

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

<— SIP read from UDP:10.2.31.240:5060 —>
INVITE sip:00100600@10.1.30.100 SIP/2.0
Via: SIP/2.0/UDP 10.2.31.240:5060;branch=z9hG4bK0716f5c0;rport
Max-Forwards: 70
From: “HP 002 DS” sip:01000203@10.2.31.240;tag=as1300bb6a
To: sip:00100600@10.1.30.100
Contact: sip:01000203@10.2.31.240
Call-ID: 6d3ce4fa5149289826bde34648731554@10.2.31.240
CSeq: 103 INVITE
User-Agent: Asterisk-PBX-02
Authorization: Digest username=“astBox01”, realm=“asterisk”, algorithm=MD5, uri="sip:00100600@10.1.30.100", nonce=“281487f6”, response="05074046141fe4f89999d410dcf609e7"
Date: Tue, 09 Oct 2012 16:37:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 1635627417 1635627418 IN IP4 10.2.31.240
s=Asterisk PBX 1.6.2.13
c=IN IP4 10.2.31.240
t=0 0
m=audio 14986 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------->
— (15 headers 11 lines) —
Sending to 10.2.31.240 : 5060 (no NAT)
Using INVITE request as basis request - 6d3ce4fa5149289826bde34648731554@10.2.31.240
Found peer ‘astBox02’ for ‘astBox01’ from 10.2.31.240:5060
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.2.31.240:14986
Looking for 00100600 in from-ast02 (domain 10.1.30.100)
list_route: hop: sip:01000203@10.2.31.240

<— Transmitting (no NAT) to 10.2.31.240:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.2.31.240:5060;branch=z9hG4bK0716f5c0;received=10.2.31.240;rport=5060
From: “HP 002 DS” sip:01000203@10.2.31.240;tag=as1300bb6a
To: sip:00100600@10.1.30.100
Call-ID: 6d3ce4fa5149289826bde34648731554@10.2.31.240
CSeq: 103 INVITE
Server: Asterisk-PBX-01
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:00100600@10.1.30.100
Content-Length: 0

<------------>
– Executing [00100600@from-ast02:1] NoOp(“SIP/astBox02-00000008”, “Incoming from Asterisk PBX 02”) in new stack
– Executing [00100600@from-ast02:2] Answer(“SIP/astBox02-00000008”, “”) in new stack
Audio is at 10.1.30.100 port 12916
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 10.2.31.240:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.31.240:5060;branch=z9hG4bK0716f5c0;received=10.2.31.240;rport=5060
From: “HP 002 DS” sip:01000203@10.2.31.240;tag=as1300bb6a
To: sip:00100600@10.1.30.100;tag=as777125ad
Call-ID: 6d3ce4fa5149289826bde34648731554@10.2.31.240
CSeq: 103 INVITE
Server: Asterisk-PBX-01
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:00100600@10.1.30.100
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 1356065173 1356065173 IN IP4 10.1.30.100
s=Asterisk PBX 1.6.2.13
c=IN IP4 10.1.30.100
t=0 0
m=audio 12916 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

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

<— SIP read from UDP:10.2.31.240:5060 —>
ACK sip:00100600@10.1.30.100 SIP/2.0
Via: SIP/2.0/UDP 10.2.31.240:5060;branch=z9hG4bK3662a37e;rport
Max-Forwards: 70
From: “HP 002 DS” sip:01000203@10.2.31.240;tag=as1300bb6a
To: sip:00100600@10.1.30.100;tag=as777125ad
Contact: sip:01000203@10.2.31.240
Call-ID: 6d3ce4fa5149289826bde34648731554@10.2.31.240
CSeq: 103 ACK
User-Agent: Asterisk-PBX-02
Content-Length: 0

<------------->
— (10 headers 0 lines) —
– Executing [00100600@from-ast02:3] Goto(“SIP/astBox02-00000008”, “from-internal,00100600,1”) in new stack
– Goto (from-internal,00100600,1)
– Executing [00100600@from-internal:1] NoOp(“SIP/astBox02-00000008”, “Direct dial”) in new stack
– Executing [00100600@from-internal:2] NoOp(“SIP/astBox02-00000008”, “Exten: 00100600”) in new stack
– Executing [00100600@from-internal:3] Dial(“SIP/astBox02-00000008”, “SIP/00100600,tTkKr”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 10.1.30.100 port 11228
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Reliably Transmitting (no NAT) to 10.2.30.221:5060:
INVITE sip:00100600@10.2.30.221:5060;transport=udp;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.1.30.100:5060;branch=z9hG4bK19c7b23b
Max-Forwards: 70
From: “HP 002 DS” sip:01000203@10.1.30.100;tag=as261eb43e
To: sip:00100600@10.2.30.221:5060;transport=udp;user=phone
Contact: sip:01000203@10.1.30.100
Call-ID: 26ececda0b44da2e3ff2f22f5d220438@10.1.30.100
CSeq: 102 INVITE
User-Agent: Asterisk-PBX-01
Date: Tue, 09 Oct 2012 16:38:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 226

v=0
o=root 1531566680 1531566680 IN IP4 10.1.30.100
s=Asterisk PBX 1.6.2.13
c=IN IP4 10.1.30.100
t=0 0
m=audio 11228 RTP/AVP 8 0 3
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=ptime:20
a=sendrecv

-- Called 00100600

<— SIP read from UDP:10.2.30.221:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.30.100:5060;branch=z9hG4bK19c7b23b
From: “HP 002 DS” sip:01000203@10.1.30.100;tag=as261eb43e
To: sip:00100600@10.2.30.221:5060;transport=udp;user=phone
Call-ID: 26ececda0b44da2e3ff2f22f5d220438@10.1.30.100
CSeq: 102 INVITE
User-Agent: Grandstream GXP2020 1.2.5.3
Content-Length: 0

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

<— SIP read from UDP:10.2.30.221:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.30.100:5060;branch=z9hG4bK19c7b23b
From: “HP 002 DS” sip:01000203@10.1.30.100;tag=as261eb43e
To: sip:00100600@10.2.30.221:5060;transport=udp;user=phone;tag=e5c6cad00ce002ad
Call-ID: 26ececda0b44da2e3ff2f22f5d220438@10.1.30.100
CSeq: 102 INVITE
User-Agent: Grandstream GXP2020 1.2.5.3
Contact: sip:00100600@10.2.30.221:5060;transport=udp;user=phone
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0

<------------->
— (10 headers 0 lines) —
– SIP/00100600-00000009 is ringing
Really destroying SIP dialog ‘0b106e801b68b02f4ab65d2e778d9601@10.2.31.240’ Method: OPTIONS
Really destroying SIP dialog ‘2a416fe8214bbc6b723e4de47eb9b505@10.1.30.100’ Method: REGISTER
Really destroying SIP dialog ‘58045b9e73cad1273de2e99419efa96b@10.2.31.240’ Method: OPTIONS
Really destroying SIP dialog ‘61c33d3e54ea77337da3a5511311f7b1@10.1.30.100’ Method: REGISTER

<— SIP read from UDP:10.2.30.221:5060 —>

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

<— SIP read from UDP:10.2.30.221:5060 —>

<------------->
Really destroying SIP dialog ‘091189203466666135bb040146696927@10.2.31.240’ Method: REGISTER
Reliably Transmitting (no NAT) to 10.2.30.221:5060:
OPTIONS sip:00100600@10.2.30.221:5060;transport=udp;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.1.30.100:5060;branch=z9hG4bK3f012eb8
Max-Forwards: 70
From: “Unknown” sip:Unknown@10.1.30.100;tag=as5797cd8c
To: sip:00100600@10.2.30.221:5060;transport=udp;user=phone
Contact: sip:Unknown@10.1.30.100
Call-ID: 65869d1560be16b678528fee6e5ff656@10.1.30.100
CSeq: 102 OPTIONS
User-Agent: Asterisk-PBX-01
Date: Tue, 09 Oct 2012 16:39:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:10.2.30.221:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.30.100:5060;branch=z9hG4bK3f012eb8
From: “Unknown” sip:Unknown@10.1.30.100;tag=as5797cd8c
To: sip:00100600@10.2.30.221:5060;transport=udp;user=phone;tag=e5c6cad00ce002ad
Call-ID: 65869d1560be16b678528fee6e5ff656@10.1.30.100
CSeq: 102 OPTIONS
User-Agent: Grandstream GXP2020 1.2.5.3
Contact: sip:00100600@10.2.30.221:5060;transport=udp;user=phone
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog ‘65869d1560be16b678528fee6e5ff656@10.1.30.100’ Method: OPTIONS

<— SIP read from UDP:10.2.31.240:5060 —>
OPTIONS sip:s@10.1.30.100 SIP/2.0
Via: SIP/2.0/UDP 10.2.31.240:5060;branch=z9hG4bK7ec67cc8;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@10.2.31.240;tag=as41c86575
To: sip:s@10.1.30.100
Contact: sip:Unknown@10.2.31.240
Call-ID: 61f2fee74766679b527855347b2b86ee@10.2.31.240
CSeq: 102 OPTIONS
User-Agent: Asterisk-PBX-02
Date: Tue, 09 Oct 2012 16:38:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Looking for s in from-sip-external (domain 10.1.30.100)

<— Transmitting (no NAT) to 10.2.31.240:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.31.240:5060;branch=z9hG4bK7ec67cc8;received=10.2.31.240;rport=5060
From: “Unknown” sip:Unknown@10.2.31.240;tag=as41c86575
To: sip:s@10.1.30.100;tag=as575c0ff6
Call-ID: 61f2fee74766679b527855347b2b86ee@10.2.31.240
CSeq: 102 OPTIONS
Server: Asterisk-PBX-01
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:10.1.30.100
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘61f2fee74766679b527855347b2b86ee@10.2.31.240’ in 32000 ms (Method: OPTIONS)

<— SIP read from UDP:10.2.30.221:5060 —>

<------------->
Reliably Transmitting (no NAT) to 10.2.31.240:5060:
OPTIONS sip:s@10.2.31.240 SIP/2.0
Via: SIP/2.0/UDP 10.1.30.100:5060;branch=z9hG4bK4b12ba03;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@10.1.30.100;tag=as3b754674
To: sip:s@10.2.31.240
Contact: sip:Unknown@10.1.30.100
Call-ID: 21b36dd55dc85e775e9ba0eb5d31635b@10.1.30.100
CSeq: 102 OPTIONS
User-Agent: Asterisk-PBX-01
Date: Tue, 09 Oct 2012 16:39:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:10.2.31.240:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.30.100:5060;branch=z9hG4bK4b12ba03;received=10.1.30.100;rport=5060
From: “Unknown” sip:Unknown@10.1.30.100;tag=as3b754674
To: sip:s@10.2.31.240;tag=as780b79dd
Call-ID: 21b36dd55dc85e775e9ba0eb5d31635b@10.1.30.100
CSeq: 102 OPTIONS
Server: Asterisk-PBX-02
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:10.2.31.240
Accept: application/sdp
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Really destroying SIP dialog ‘21b36dd55dc85e775e9ba0eb5d31635b@10.1.30.100’ Method: OPTIONS

<— SIP read from UDP:10.2.30.221:5060 —>
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP 10.1.30.100:5060;branch=z9hG4bK19c7b23b
From: “HP 002 DS” sip:01000203@10.1.30.100;tag=as261eb43e
To: sip:00100600@10.2.30.221:5060;transport=udp;user=phone;tag=e5c6cad00ce002ad
Call-ID: 26ececda0b44da2e3ff2f22f5d220438@10.1.30.100
CSeq: 102 INVITE
User-Agent: Grandstream GXP2020 1.2.5.3
Content-Length: 0

<------------->
— (8 headers 0 lines) —
[Oct 9 18:39:36] WARNING[7314]: chan_sip.c:17946 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog ‘26ececda0b44da2e3ff2f22f5d220438@10.1.30.100’. Giving up.
Transmitting (no NAT) to 10.2.30.221:5060:
ACK sip:00100600@10.2.30.221:5060;transport=udp;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.1.30.100:5060;branch=z9hG4bK19c7b23b
Max-Forwards: 70
From: “HP 002 DS” sip:01000203@10.1.30.100;tag=as261eb43e
To: sip:00100600@10.2.30.221:5060;transport=udp;user=phone;tag=e5c6cad00ce002ad
Contact: sip:01000203@10.1.30.100
Call-ID: 26ececda0b44da2e3ff2f22f5d220438@10.1.30.100
CSeq: 102 ACK
User-Agent: Asterisk-PBX-01
Content-Length: 0

Scheduling destruction of SIP dialog ‘26ececda0b44da2e3ff2f22f5d220438@10.1.30.100’ in 6400 ms (Method: INVITE)
– SIP/00100600-00000009 is circuit-busy
Scheduling destruction of SIP dialog ‘26ececda0b44da2e3ff2f22f5d220438@10.1.30.100’ in 6400 ms (Method: INVITE)
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [00100600@from-internal:4] Hangup(“SIP/astBox02-00000008”, “”) in new stack
== Spawn extension (from-internal, 00100600, 4) exited non-zero on ‘SIP/astBox02-00000008’
– Executing [h@from-internal:1] Macro(“SIP/astBox02-00000008”, “hangupcall”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/astBox02-00000008”, “1?noautomon”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] NoOp(“SIP/astBox02-00000008”, “TOUCH_MONITOR_OUTPUT=”) in new stack
– Executing [s@macro-hangupcall:4] GotoIf(“SIP/astBox02-00000008”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,7)
– Executing [s@macro-hangupcall:7] GotoIf(“SIP/astBox02-00000008”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,10)
– Executing [s@macro-hangupcall:10] GotoIf(“SIP/astBox02-00000008”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,12)
– Executing [s@macro-hangupcall:12] Hangup(“SIP/astBox02-00000008”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 12) exited non-zero on ‘SIP/astBox02-00000008’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/astBox02-00000008’
Scheduling destruction of SIP dialog ‘6d3ce4fa5149289826bde34648731554@10.2.31.240’ in 6400 ms (Method: ACK)
set_destination: Parsing sip:01000203@10.2.31.240 for address/port to send to
set_destination: set destination to 10.2.31.240, port 5060
Reliably Transmitting (no NAT) to 10.2.31.240:5060:
BYE sip:01000203@10.2.31.240 SIP/2.0
Via: SIP/2.0/UDP 10.1.30.100:5060;branch=z9hG4bK5ff56937;rport
Max-Forwards: 70
From: sip:00100600@10.1.30.100;tag=as777125ad
To: “HP 002 DS” sip:01000203@10.2.31.240;tag=as1300bb6a
Call-ID: 6d3ce4fa5149289826bde34648731554@10.2.31.240
CSeq: 102 BYE
User-Agent: Asterisk-PBX-01
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0

<— SIP read from UDP:10.2.31.240:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.30.100:5060;branch=z9hG4bK5ff56937;received=10.1.30.100;rport=5060
From: sip:00100600@10.1.30.100;tag=as777125ad
To: “HP 002 DS” sip:01000203@10.2.31.240;tag=as1300bb6a
Call-ID: 6d3ce4fa5149289826bde34648731554@10.2.31.240
CSeq: 102 BYE
Server: Asterisk-PBX-02
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------->
— (10 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog ‘6d3ce4fa5149289826bde34648731554@10.2.31.240’ Method: ACK
Really destroying SIP dialog ‘61f2fee74766679b527855347b2b86ee@10.2.31.240’ Method: OPTIONS
Really destroying SIP dialog ‘26ececda0b44da2e3ff2f22f5d220438@10.1.30.100’ Method: INVITE
ET-Ipbx*CLI> sip set debug off
SIP Debugging Disabled[/code]

It looks like a re-invite causes everything but I can’t find a solution
I’ve tried setting
ignoresdpversion=yes
session-timers=refuse
session-expires=120
canreinvite=no
nat=no
but nothing works.

Anybody experienced the same problem and found a solution?
Thanks for the support
D.

david55 · October 10, 2012, 10:34am

I can’t tell the timing, because you scraped the screen, rather than using a log file, and I can’t tell the Asterisk version, because you changed the user agent string, but Asterisk appears not to have sent an ACK to OK from 221.

Moreover, the call-ids and tags coming back for the apparent connect from 221 don’t match the invite. I wonder if you have a broken SIP aware router in the way?

danilo7 · October 10, 2012, 3:40pm

First of all, thanks for answering

Do you need the asterisk log for a deeper analysis? I’ve checked the Asterisk Version (1.6.2.13) using the sip show settings in the CLI, and the 1 minute period is visible on the phone screen.

[quote=“david55”]…because you changed the user agent string, but Asterisk appears not to have sent an ACK to OK from 221.

Moreover, the call-ids and tags coming back for the apparent connect from 221 don’t match the invite. I wonder if you have a broken SIP aware router in the way?[/quote]

There’s a router between the 2 boxes, but it doesn’t have any ACL configured and the traffic is not blocked in any point…a proof of it is that if i pick up the call it doesn’t drop, it only drops if I leave it ringing for more than a minute.

I’ve used wireshark on a mirrored port of the of called extension 00100600 (10.2.30.221) and i get this

Really have no clue

david55 · October 10, 2012, 3:47pm

You need to use wireshark in VoIP analysis mode. You have multiple dialogues all mixed up in the above trace and what I am saying is that the detailed trace appears to show that this extends down to the invite level, whereas you should have everythign with the same call-id. The wireshark VoIP analysis should make it easier to see if there is more than one call going on.

1.6.2.13 is an obsolete sub-version of an obsolete version. I hope you haven’t just installed it.

danilo7 · October 10, 2012, 4:07pm

[quote=“david55”]You need to use wireshark in VoIP analysis mode. You have multiple dialogues all mixed up in the above trace and what I am saying is that the detailed trace appears to show that this extends down to the invite level, whereas you should have everythign with the same call-id. The wireshark VoIP analysis should make it easier to see if there is more than one call going on.

1.6.2.13 is an obsolete sub-version of an obsolete version. I hope you haven’t just installed it.[/quote]

I know that the asterisk version is obsolete, unfortunately I can’t upgrade it
I used it, but during the ringing till the call drop all i get is this

*edit cause the image sucks

Start time: 233,410163
Stop time: 293,485905
Initial Speaker: 10.1.30.100
From: “HP 005 DS” sip:01000503@10.1.30.100
To: <sip:00100600@10.2.30.221:5060;transport=udp;user
Protocol: SIP
Packets: 2
State: CALL SETUP

david55 · October 10, 2012, 4:11pm

Flow it.

danilo7 · October 10, 2012, 4:15pm

What do you mean??

david55 · October 10, 2012, 4:22pm

Look at the buttons at the bottom of the wireshark screen. (I don’t think it is called Flow in my version, but I assume it is the same.)

danilo7 · October 10, 2012, 4:28pm

Yea you are right, i didn’t notice the button name
I did it and this is what i’ve got

david55 · October 10, 2012, 4:30pm

You are only seeing one side.

danilo7 · October 10, 2012, 4:58pm

I’ve mirrored both sides on the port I’m listening to
here’s the result
Voip Calls

first side

second side

david55 · October 10, 2012, 5:03pm

The “second side” is incomplete; it has none of the messages from the Grandstream, and those are what appear to be wrong.

You can use tcpdump on the Asterisk box to create pcap files representing what Asterisk is actually seeing. Note, you will normally need a non-default snap length. I think you can also get wireshark for Linux.

danilo7 · October 10, 2012, 5:46pm

Thanks for the other precious advices, but for today I can’t make any more tests
I’ll get the other part of the second side tomorrow and I’ll keep you posted

Thank you very much!!

danilo7 · October 15, 2012, 11:55am

[quote=“david55”]The “second side” is incomplete; it has none of the messages from the Grandstream, and those are what appear to be wrong.

You can use tcpdump on the Asterisk box to create pcap files representing what Asterisk is actually seeing. Note, you will normally need a non-default snap length. I think you can also get wireshark for Linux.[/quote]

I was finally able to find out what was the problem on my calls
There was a “Ring timeout” set on the Grandstream telephone that caused the call drop (and the weird warning message). After i’ve increased the timeout everything went fine, so it was a telephone setting not a trunk setting that created the problem

Thanks for the support david55! :mrgreen: