Cannot send out via asterisk asterisk-10.6.1 t.38 gateway

hi all,

I am trying to send out fax via asterisk-10.6.1 as below:

hylafax -> iaxmodem -> asterisk 10.6.1 as t38 gateway -> sip t38 gateway provider -> ptsn

the destination rings, but fax is not sending out. here below is the asterisk log, there is an error " [Jul 27 17:27:44] ERROR[24792] astobj2.c: user_data is NULL".

your advice is much appreciated on this issue.

thanks very much.

[Jul 30 10:37:07] VERBOSE[11394] chan_iax2.c: – Accepting AUTHENTICATED call from 127.0.0.1:
> requested format = alaw,
> requested prefs = (),
> actual format = alaw,
> host prefs = (alaw|ulaw|slin),
> priority = mine
[Jul 30 10:37:07] VERBOSE[11505] pbx.c: – Executing [1500773085230196079@zonefax:1] NoOp(“IAX2/zonefax119-6897”, ““IAXmodem” <>”) in new stack
[Jul 30 10:37:07] VERBOSE[11505] pbx.c: – Executing [1500773085230196079@zonefax:2] Set(“IAX2/zonefax119-6897”, “CALLERID(num)=15007730”) in new stack
[Jul 30 10:37:07] VERBOSE[11505] pbx.c: – Executing [1500773085230196079@zonefax:3] Set(“IAX2/zonefax119-6897”, “FAXOPT(gateway)=yes”) in new stack
[Jul 30 10:37:07] VERBOSE[11505] pbx.c: – Executing [1500773085230196079@zonefax:4] Dial(“IAX2/zonefax119-6897”, “SIP/vsc/30196079”) in new stack
[Jul 30 10:37:07] VERBOSE[11505] netsock2.c: == Using SIP RTP CoS mark 5
[Jul 30 10:37:07] VERBOSE[11505] chan_sip.c: Audio is at 12384
[Jul 30 10:37:07] VERBOSE[11505] chan_sip.c: Adding codec 100004 (alaw) to SDP
[Jul 30 10:37:07] VERBOSE[11505] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[Jul 30 10:37:07] VERBOSE[11505] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Jul 30 10:37:07] VERBOSE[11505] chan_sip.c: Reliably Transmitting (NAT) to 202.130.146.100:5060:
INVITE sip:30196079@202.130.146.100 SIP/2.0
Via: SIP/2.0/UDP 202.130.146.125:1812;branch=z9hG4bK29deb679;rport
Max-Forwards: 70
From: “IAXmodem” sip:15007730@202.130.146.125:1812;tag=as0bce6b68
To: sip:30196079@202.130.146.100
Contact: sip:15007730@202.130.146.125:1812
Call-ID: 6d0a846a5ab302d925f383586af1cb30@202.130.146.125:1812
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.6.1
Date: Mon, 30 Jul 2012 02:37:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 292

v=0
o=root 1012076007 1012076007 IN IP4 202.130.146.125
s=Asterisk PBX 10.6.1
c=IN IP4 202.130.146.125
t=0 0
m=audio 12384 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


[Jul 30 10:37:07] VERBOSE[11505] app_dial.c: – Called SIP/vsc/30196079
[Jul 30 10:37:07] VERBOSE[11401] chan_sip.c:
<— SIP read from UDP:202.130.146.100:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 202.130.146.125:1812;rport=1812;received=202.130.146.125;branch=z9hG4bK29deb679
Call-ID: 6d0a846a5ab302d925f383586af1cb30@202.130.146.125:1812
From: “IAXmodem” sip:15007730@202.130.146.125;tag=as0bce6b68
To: sip:30196079@202.130.146.100
CSeq: 102 INVITE
Content-Length: 0

<------------->
[Jul 30 10:37:07] VERBOSE[11401] chan_sip.c: — (7 headers 0 lines) —
[Jul 30 10:37:08] VERBOSE[11401] chan_sip.c:
<— SIP read from UDP:202.130.146.100:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 202.130.146.125:1812;rport=1812;received=202.130.146.125;branch=z9hG4bK29deb679
Call-ID: 6d0a846a5ab302d925f383586af1cb30@202.130.146.125:1812
From: “IAXmodem” sip:15007730@202.130.146.125;tag=as0bce6b68
To: sip:30196079@202.130.146.100;tag=5df3c9ab-b857-4fbd-8bb4-03d7197c9e42
CSeq: 102 INVITE
Contact: sip:202.130.146.100:5060
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS
Content-Type: application/sdp
Content-Length: 221

v=0
o=GANG 506227872 506227873 IN IP4 202.130.146.100
s=-
c=IN IP4 202.130.146.100
t=0 0
m=audio 40166 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
[Jul 30 10:37:08] VERBOSE[11401] chan_sip.c: — (10 headers 11 lines) —
[Jul 30 10:37:08] VERBOSE[11401] chan_sip.c: list_route: hop: sip:202.130.146.100:5060
[Jul 30 10:37:08] VERBOSE[11401] chan_sip.c: Found RTP audio format 8
[Jul 30 10:37:08] VERBOSE[11401] chan_sip.c: Found RTP audio format 101
[Jul 30 10:37:08] VERBOSE[11401] chan_sip.c: Found audio description format PCMA for ID 8
[Jul 30 10:37:08] VERBOSE[11401] chan_sip.c: Found audio description format telephone-event for ID 101
[Jul 30 10:37:08] VERBOSE[11401] chan_sip.c: Capabilities: us - (ulaw|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
[Jul 30 10:37:08] VERBOSE[11401] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Jul 30 10:37:08] VERBOSE[11401] chan_sip.c: Peer audio RTP is at port 202.130.146.100:40166
[Jul 30 10:37:08] VERBOSE[11505] app_dial.c: – SIP/vsc-00000000 is ringing
[Jul 30 10:37:08] VERBOSE[11505] app_dial.c: – SIP/vsc-00000000 is making progress passing it to IAX2/zonefax119-6897
[Jul 30 10:37:10] VERBOSE[11401] chan_sip.c:
<— SIP read from UDP:202.130.146.100:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 202.130.146.125:1812;rport=1812;received=202.130.146.125;branch=z9hG4bK29deb679
Call-ID: 6d0a846a5ab302d925f383586af1cb30@202.130.146.125:1812
From: “IAXmodem” sip:15007730@202.130.146.125;tag=as0bce6b68
To: sip:30196079@202.130.146.100;tag=5df3c9ab-b857-4fbd-8bb4-03d7197c9e42
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS
Contact: sip:202.130.146.100:5060
Supported: 100rel
Content-Type: application/sdp
Content-Length: 221

v=0
o=GANG 506227872 506227873 IN IP4 202.130.146.100
s=-
c=IN IP4 202.130.146.100
t=0 0
m=audio 40166 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
[Jul 30 10:37:10] VERBOSE[11401] chan_sip.c: — (11 headers 11 lines) —
[Jul 30 10:37:10] VERBOSE[11401] chan_sip.c: list_route: hop: sip:202.130.146.100:5060
[Jul 30 10:37:10] VERBOSE[11401] chan_sip.c: set_destination: Parsing sip:202.130.146.100:5060 for address/port to send to
[Jul 30 10:37:10] VERBOSE[11401] chan_sip.c: set_destination: set destination to 202.130.146.100:5060
[Jul 30 10:37:10] VERBOSE[11401] chan_sip.c: Transmitting (NAT) to 202.130.146.100:5060:
ACK sip:202.130.146.100:5060 SIP/2.0
Via: SIP/2.0/UDP 202.130.146.125:1812;branch=z9hG4bK2493a59d;rport
Max-Forwards: 70
From: “IAXmodem” sip:15007730@202.130.146.125:1812;tag=as0bce6b68
To: sip:30196079@202.130.146.100;tag=5df3c9ab-b857-4fbd-8bb4-03d7197c9e42
Contact: sip:15007730@202.130.146.125:1812
Call-ID: 6d0a846a5ab302d925f383586af1cb30@202.130.146.125:1812
CSeq: 102 ACK
User-Agent: Asterisk PBX 10.6.1
Content-Length: 0


[Jul 30 10:37:10] VERBOSE[11505] app_dial.c: – SIP/vsc-00000000 answered IAX2/zonefax119-6897
[Jul 30 10:37:12] DTMF[11505] channel.c: DTMF end ‘2’ received on IAX2/zonefax119-6897, duration 0 ms
[Jul 30 10:37:12] DTMF[11505] channel.c: DTMF begin emulation of ‘2’ with duration 100 queued on IAX2/zonefax119-6897
[Jul 30 10:37:12] DTMF[11505] channel.c: DTMF end emulation of ‘2’ queued on IAX2/zonefax119-6897
[Jul 30 10:37:12] DTMF[11505] channel.c: DTMF end ‘3’ received on IAX2/zonefax119-6897, duration 0 ms
[Jul 30 10:37:12] DTMF[11505] channel.c: DTMF begin emulation of ‘3’ with duration 100 queued on IAX2/zonefax119-6897
[Jul 30 10:37:12] DTMF[11505] channel.c: DTMF end emulation of ‘3’ queued on IAX2/zonefax119-6897
[Jul 30 10:37:12] VERBOSE[11401] chan_sip.c:
<— SIP read from UDP:202.130.146.100:5060 —>
INVITE sip:15007730@202.130.146.125:1812 SIP/2.0
Via: SIP/2.0/UDP 202.130.146.100:5060;rport;branch=z9hG4bKPj003f61ad-0d8e-4b8e-8a7f-4a316fde16c3
Max-Forwards: 70
From: sip:30196079@202.130.146.100;tag=5df3c9ab-b857-4fbd-8bb4-03d7197c9e42
To: “IAXmodem” sip:15007730@202.130.146.125;tag=as0bce6b68
Contact: sip:202.130.146.100:5060
Call-ID: 6d0a846a5ab302d925f383586af1cb30@202.130.146.125:1812
CSeq: 11602 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS
Supported: 100rel
Content-Type: application/sdp
Content-Length: 358

v=0
o=GANG 506227872 506227874 IN IP4 202.130.146.100
s=-
c=IN IP4 202.130.146.100
t=0 0
m=image 40166 udptl t38
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:72
a=T38FaxVersion:0
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
[Jul 30 10:37:12] VERBOSE[11401] chan_sip.c: — (12 headers 15 lines) —
[Jul 30 10:37:12] VERBOSE[11401] chan_sip.c: Sending to 202.130.146.100:5060 (NAT)
[Jul 30 10:37:12] VERBOSE[11401] netsock.c: == Using UDPTL CoS mark 5
[Jul 30 10:37:12] VERBOSE[11401] chan_sip.c: Got T.38 offer in SDP in dialog 6d0a846a5ab302d925f383586af1cb30@202.130.146.125:1812
[Jul 30 10:37:12] VERBOSE[11401] chan_sip.c: Capabilities: us - (ulaw|alaw), peer - audio=(nothing)/video=(nothing)/text=(nothing), combined - (nothing)
[Jul 30 10:37:12] VERBOSE[11401] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
[Jul 30 10:37:12] VERBOSE[11401] chan_sip.c: Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
[Jul 30 10:37:12] VERBOSE[11401] chan_sip.c:
<— Transmitting (NAT) to 202.130.146.100:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 202.130.146.100:5060;branch=z9hG4bKPj003f61ad-0d8e-4b8e-8a7f-4a316fde16c3;received=202.130.146.100;rport=5060
From: sip:30196079@202.130.146.100;tag=5df3c9ab-b857-4fbd-8bb4-03d7197c9e42
To: “IAXmodem” sip:15007730@202.130.146.125;tag=as0bce6b68
Call-ID: 6d0a846a5ab302d925f383586af1cb30@202.130.146.125:1812
CSeq: 11602 INVITE
Server: Asterisk PBX 10.6.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:15007730@202.130.146.125:1812
Content-Length: 0

<------------>
[Jul 30 10:37:12] ERROR[11505] astobj2.c: user_data is NULL
[Jul 30 10:37:12] VERBOSE[11505] chan_sip.c:
<— Reliably Transmitting (NAT) to 202.130.146.100:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 202.130.146.100:5060;branch=z9hG4bKPj003f61ad-0d8e-4b8e-8a7f-4a316fde16c3;received=202.130.146.100;rport=5060
From: sip:30196079@202.130.146.100;tag=5df3c9ab-b857-4fbd-8bb4-03d7197c9e42
To: “IAXmodem” sip:15007730@202.130.146.125;tag=as0bce6b68
Call-ID: 6d0a846a5ab302d925f383586af1cb30@202.130.146.125:1812
CSeq: 11602 INVITE
Server: Asterisk PBX 10.6.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:15007730@202.130.146.125:1812
Content-Type: application/sdp
Content-Length: 277

v=0
o=root 1012076007 1012076008 IN IP4 202.130.146.125
s=Asterisk PBX 10.6.1
c=IN IP4 202.130.146.125
t=0 0
m=image 4690 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:1400
a=T38FaxUdpEC:t38UDPRedundancy

<------------>
[Jul 30 10:37:12] VERBOSE[11401] chan_sip.c:
<— SIP read from UDP:202.130.146.100:5060 —>
ACK sip:15007730@202.130.146.125:1812 SIP/2.0
Via: SIP/2.0/UDP 202.130.146.100:5060;rport;branch=z9hG4bKPj58ce5bdc-8332-458c-9d3c-f3e75a5d74e3
Max-Forwards: 70
From: sip:30196079@202.130.146.100;tag=5df3c9ab-b857-4fbd-8bb4-03d7197c9e42
To: “IAXmodem” sip:15007730@202.130.146.125;tag=as0bce6b68
Call-ID: 6d0a846a5ab302d925f383586af1cb30@202.130.146.125:1812
CSeq: 11602 ACK
Content-Length: 0

<------------->
[Jul 30 10:37:12] VERBOSE[11401] chan_sip.c: — (8 headers 0 lines) —
[Jul 30 10:37:12] DTMF[11505] channel.c: DTMF end ‘1’ received on IAX2/zonefax119-6897, duration 0 ms
[Jul 30 10:37:12] DTMF[11505] channel.c: DTMF begin emulation of ‘1’ with duration 100 queued on IAX2/zonefax119-6897
[Jul 30 10:37:12] DTMF[11505] channel.c: DTMF end emulation of ‘1’ queued on IAX2/zonefax119-6897
[Jul 30 10:37:12] WARNING[11505] channel.c: Codec mismatch on channel IAX2/zonefax119-6897 setting write format to slin from alaw native formats (alaw)
[Jul 30 10:37:12] DTMF[11505] channel.c: DTMF end ‘4’ received on IAX2/zonefax119-6897, duration 0 ms
[Jul 30 10:37:12] DTMF[11505] channel.c: DTMF begin emulation of ‘4’ with duration 100 queued on IAX2/zonefax119-6897
[Jul 30 10:37:12] DTMF[11505] channel.c: DTMF end emulation of ‘4’ queued on IAX2/zonefax119-6897
[Jul 30 10:37:12] DTMF[11505] channel.c: DTMF end ‘3’ received on IAX2/zonefax119-6897, duration 0 ms
[Jul 30 10:37:12] DTMF[11505] channel.c: DTMF begin emulation of ‘3’ with duration 100 queued on IAX2/zonefax119-6897
[Jul 30 10:37:13] DTMF[11505] channel.c: DTMF end emulation of ‘3’ queued on IAX2/zonefax119-6897
[Jul 30 10:37:13] DTMF[11505] channel.c: DTMF end ‘6’ received on IAX2/zonefax119-6897, duration 0 ms
[Jul 30 10:37:13] DTMF[11505] channel.c: DTMF begin emulation of ‘6’ with duration 100 queued on IAX2/zonefax119-6897
[Jul 30 10:37:13] DTMF[11505] channel.c: DTMF end emulation of ‘6’ queued on IAX2/zonefax119-6897
[Jul 30 10:37:13] DTMF[11505] channel.c: DTMF end ‘3’ received on IAX2/zonefax119-6897, duration 0 ms
[Jul 30 10:37:13] DTMF[11505] channel.c: DTMF begin emulation of ‘3’ with duration 100 queued on IAX2/zonefax119-6897
[Jul 30 10:37:13] DTMF[11505] channel.c: DTMF end emulation of ‘3’ queued on IAX2/zonefax119-6897
[Jul 30 10:37:13] WARNING[11505] res_rtp_asterisk.c: RTP Read too short
[Jul 30 10:37:13] WARNING[11505] res_rtp_asterisk.c: RTP Read too short
[Jul 30 10:37:15] WARNING[11505] res_rtp_asterisk.c: RTP Read too short
[Jul 30 10:37:15] WARNING[11505] res_rtp_asterisk.c: RTP Read too short
[Jul 30 10:37:18] WARNING[11505] res_rtp_asterisk.c: RTP Read too short
[Jul 30 10:37:18] WARNING[11505] res_rtp_asterisk.c: RTP Read too short
[Jul 30 10:37:20] WARNING[11505] res_rtp_asterisk.c: RTP Read too short
[Jul 30 10:37:20] WARNING[11505] res_rtp_asterisk.c: RTP Read too short
[Jul 30 10:37:24] WARNING[11505] res_rtp_asterisk.c: RTP Read too short
[Jul 30 10:37:24] WARNING[11505] res_rtp_asterisk.c: RTP Read too short
[Jul 30 10:37:26] WARNING[11505] res_rtp_asterisk.c: RTP Read too short
[Jul 30 10:37:26] WARNING[11505] res_rtp_asterisk.c: RTP Read too short
[Jul 30 10:37:30] WARNING[11505] res_rtp_asterisk.c: RTP Read too short
[Jul 30 10:37:30] WARNING[11505] res_rtp_asterisk.c: RTP Read too short
[Jul 30 10:37:31] WARNING[11505] res_rtp_asterisk.c: RTP Read too short
[Jul 30 10:37:31] WARNING[11505] res_rtp_asterisk.c: RTP Read too short
[Jul 30 10:37:35] WARNING[11505] res_rtp_asterisk.c: RTP Read too short
[Jul 30 10:37:35] WARNING[11505] res_rtp_asterisk.c: RTP Read too short
[Jul 30 10:37:37] WARNING[11505] res_rtp_asterisk.c: RTP Read too short
[Jul 30 10:37:37] WARNING[11505] res_rtp_asterisk.c: RTP Read too short
[Jul 30 10:37:41] WARNING[11505] res_rtp_asterisk.c: RTP Read too short
[Jul 30 10:37:41] WARNING[11505] res_rtp_asterisk.c: RTP Read too short
[Jul 30 10:37:43] WARNING[11505] res_rtp_asterisk.c: RTP Read too short
[Jul 30 10:37:43] WARNING[11505] res_rtp_asterisk.c: RTP Read too short
[Jul 30 10:37:45] WARNING[11505] res_rtp_asterisk.c: RTP Read too short
[Jul 30 10:37:45] WARNING[11505] res_rtp_asterisk.c: RTP Read too short
[Jul 30 10:37:46] WARNING[11505] res_rtp_asterisk.c: RTP Read too short
[Jul 30 10:37:46] WARNING[11505] res_rtp_asterisk.c: RTP Read too short
[Jul 30 10:37:46] VERBOSE[11401] chan_sip.c:
<— SIP read from UDP:202.130.146.100:5060 —>
BYE sip:15007730@202.130.146.125:1812 SIP/2.0
Via: SIP/2.0/UDP 202.130.146.100:5060;rport;branch=z9hG4bKPjfd3df6d2-ad1c-4e03-8316-2f4bd54f4b2c
Max-Forwards: 70
From: sip:30196079@202.130.146.100;tag=5df3c9ab-b857-4fbd-8bb4-03d7197c9e42
To: “IAXmodem” sip:15007730@202.130.146.125;tag=as0bce6b68
Call-ID: 6d0a846a5ab302d925f383586af1cb30@202.130.146.125:1812
CSeq: 11603 BYE
User-Agent: IPSIP/2.1.3.2010.11.16
Reason: Q.850;cause=16
Content-Length: 0

<------------->
[Jul 30 10:37:46] VERBOSE[11401] chan_sip.c: — (10 headers 0 lines) —
[Jul 30 10:37:46] VERBOSE[11401] chan_sip.c: Sending to 202.130.146.100:5060 (NAT)
[Jul 30 10:37:46] VERBOSE[11401] chan_sip.c: Scheduling destruction of SIP dialog ‘6d0a846a5ab302d925f383586af1cb30@202.130.146.125:1812’ in 32000 ms (Method: BYE)
[Jul 30 10:37:46] VERBOSE[11401] chan_sip.c:
<— Transmitting (NAT) to 202.130.146.100:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 202.130.146.100:5060;branch=z9hG4bKPjfd3df6d2-ad1c-4e03-8316-2f4bd54f4b2c;received=202.130.146.100;rport=5060
From: sip:30196079@202.130.146.100;tag=5df3c9ab-b857-4fbd-8bb4-03d7197c9e42
To: “IAXmodem” sip:15007730@202.130.146.125;tag=as0bce6b68
Call-ID: 6d0a846a5ab302d925f383586af1cb30@202.130.146.125:1812
CSeq: 11603 BYE
Server: Asterisk PBX 10.6.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

Hi.
I have the same problem.
Have you solved this problem?

[quote=“dereklinrs”]here is an error " [Jul 27 17:27:44] ERROR[24792] astobj2.c: user_data is NULL".
[/quote]

This indicates that there is a bug. You should raise this on issues.asterisk.org/jira/

Note, this could be a memory corruption problem, so may be difficult to solve.

Thank you David.
I will open a bug.

Daniele

I’m getting the exactly same symptoms… Have you been able to resolve this on your end?