PROBLEM T.38 and FAX PJSIP - 488 NOT ACCEPT HERE

learbia · March 13, 2018, 9:43pm

Hello, we have a problem with negotiation T.38 in PJSIP.

I make a call
When carrier know the call is fax, he send UPDATE in SIP with T.38
But PJSIP Asterisk 15. response 488 (NOT ACCPETED HERE)

U 2018/03/13 16:28:32.001928 172.16.0.34:5060 -> 192.168.25.1:5060

INVITE sip:30276761@192.168.25.1 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.34:5060;rport;branch=z9hG4bKPjab0d4b1c-1c24-47b9-aa21-079bc9aa8144
From: “4001” sip:1130276850@10.213.18.170;tag=c03285cf-5f47-45d7-8367-c5b1c550e58f
To: sip:30276761@192.168.25.1
Contact: sip:handphone@172.16.0.34:5060
Call-ID: 50b1851e-c278-4fed-ab6f-e215f52e89b4
CSeq: 31518 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: “4001” sip:1130276850@10.213.18.170
Remote-Party-ID: “4001” sip:1130276850@10.213.18.170;privacy=off;screen=no
Max-Forwards: 70
User-Agent: handphone
Content-Type: application/sdp
Content-Length: 266

v=0
o=- 943676200 943676200 IN IP4 172.16.0.34
s=HANDPHONE_XHAND
c=IN IP4 10.213.18.170
t=0 0
m=audio 11402 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=ptime:20
a=maxptime:150
a=sendrecv

U 2018/03/13 16:28:32.015600 192.168.25.1:5060 -> 172.16.0.34:5060

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.0.34:5060;received=172.16.0.34;branch=z9hG4bKPjab0d4b1c-1c24-47b9-aa21-079bc9aa8144;rport=5060
From: “4001” sip:1130276850@10.213.18.170;tag=c03285cf-5f47-45d7-8367-c5b1c550e58f
To: sip:30276761@192.168.25.1
Call-ID: 50b1851e-c278-4fed-ab6f-e215f52e89b4
CSeq: 31518 INVITE

U 2018/03/13 16:28:33.118217 192.168.25.1:5060 -> 172.16.0.34:5060

SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.34:5060;received=172.16.0.34;branch=z9hG4bKPjab0d4b1c-1c24-47b9-aa21-079bc9aa8144;rport=5060
From: “4001” sip:1130276850@10.213.18.170;tag=c03285cf-5f47-45d7-8367-c5b1c550e58f
To: sip:30276761@192.168.25.1;tag=0331vrg0-CC-14
Call-ID: 50b1851e-c278-4fed-ab6f-e215f52e89b4
CSeq: 31518 INVITE
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,UPDATE,INFO,REFER,NOTIFY
Require: timer
Session-Expires: 1800;refresher=uac
Contact: sip:30276761@192.168.25.1:5060;Dpt=7bb8-400;transport=udp
Content-Length: 225
Content-Type: application/sdp

v=0
o=HuaweiATS9900 177209542 177209542 IN IP4 192.168.25.1
s=Sip Call
c=IN IP4 192.168.25.1
t=0 0
m=audio 10096 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

U 2018/03/13 16:28:33.119290 172.16.0.34:5060 -> 192.168.25.1:5060

ACK sip:30276761@192.168.25.1:5060;transport=udp;Dpt=7bb8-400 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.34:5060;rport;branch=z9hG4bKPj2ec1c3fc-5740-4662-a472-c246cc881eb3
From: “4001” sip:1130276850@10.213.18.170;tag=c03285cf-5f47-45d7-8367-c5b1c550e58f
To: sip:30276761@192.168.25.1;tag=0331vrg0-CC-14
Call-ID: 50b1851e-c278-4fed-ab6f-e215f52e89b4
CSeq: 31518 ACK
Max-Forwards: 70
User-Agent: handphone
Content-Length: 0

U 2018/03/13 16:28:36.262653 192.168.25.1:5060 -> 172.16.0.34:5060

UPDATE sip:handphone@172.16.0.34:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.25.1:5060;branch=z9hG4bKsvdfir008ov2ctsk4100sm0000g00.1
Call-ID: 50b1851e-c278-4fed-ab6f-e215f52e89b4
From: sip:30276761@192.168.25.1;tag=0331vrg0-CC-14
To: “4001” sip:1130276850@10.213.18.170;tag=c03285cf-5f47-45d7-8367-c5b1c550e58f
CSeq: 1 UPDATE
Max-Forwards: 65
Supported: timer
Session-Expires: 1800
Contact: sip:30276761@192.168.25.1:5060;Dpt=7bb8-400;transport=udp
Min-SE: 600
Content-Length: 280
Content-Type: application/sdp

v=0
o=HuaweiATS9900 177209542 177209543 IN IP4 192.168.25.1
s=Sip Call
c=IN IP4 192.168.25.1
t=0 0
m=image 10096 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxDatagram:1400
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
a=sendrecv

U 2018/03/13 16:28:36.262872 172.16.0.34:5060 -> 192.168.25.1:5060

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 192.168.25.1:5060;rport=5060;received=192.168.25.1;branch=z9hG4bKsvdfir008ov2ctsk4100sm0000g00.1
Call-ID: 50b1851e-c278-4fed-ab6f-e215f52e89b4
From: sip:30276761@192.168.25.1;tag=0331vrg0-CC-14
To: “4001” sip:1130276850@10.213.18.170;tag=c03285cf-5f47-45d7-8367-c5b1c550e58f
CSeq: 1 UPDATE
Server: handphone
Content-Length: 0

U 2018/03/13 16:28:41.162899 172.16.0.34:5060 -> 192.168.25.1:5060

BYE sip:30276761@192.168.25.1:5060;transport=udp;Dpt=7bb8-400 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.34:5060;rport;branch=z9hG4bKPj4bbbfa0e-8149-45ae-9ddd-97c2e31f5b41
From: “4001” sip:1130276850@10.213.18.170;tag=c03285cf-5f47-45d7-8367-c5b1c550e58f
To: sip:30276761@192.168.25.1;tag=0331vrg0-CC-14
Call-ID: 50b1851e-c278-4fed-ab6f-e215f52e89b4
CSeq: 31519 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: handphone
Content-Length: 0

U 2018/03/13 16:28:41.250322 192.168.25.1:5060 -> 172.16.0.34:5060

SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.34:5060;received=172.16.0.34;branch=z9hG4bKPj4bbbfa0e-8149-45ae-9ddd-97c2e31f5b41;rport=5060
From: “4001” sip:1130276850@10.213.18.170;tag=c03285cf-5f47-45d7-8367-c5b1c550e58f
To: sip:30276761@192.168.25.1;tag=0331vrg0-CC-14
Call-ID: 50b1851e-c278-4fed-ab6f-e215f52e89b4
CSeq: 31519 BYE
Content-Length: 0

jcolp · March 13, 2018, 9:59pm

You have not provided the configuration you are using. As well I don’t think UPDATE has been tested for this, just a re-invite has.

learbia · March 13, 2018, 10:12pm

My configuration

is: pjsip.conf

[transport-udp]
allow_reload=yes
type=transport
protocol=udp
[NOVATELECOM]
type=aor
contact=sip:187.33.43.180
max_contacts=1

[NOVATELECOM]
type=endpoint
context=tronco
outbound_auth=auth_NOVATELECOM
aors=NOVATELECOM
from_user=
allow=g729
allow=alaw
allow=ulaw
inband_progress=yes
language=pt_BR
direct_media=no
sdp_session=HANDPHONE_XHAND
subscribe_context=hint-handphone
t38_udptl=yes
t38_udptl_ec=redundancy
allow_subscribe=yes

[NOVATELECOM]
type=identify
endpoint=NOVATELECOM
match=187.33.43.180

learbia · March 13, 2018, 10:13pm

The principals carries in Brazil now use UPDATE in SIP

jcolp · March 14, 2018, 10:19am

And what is the complete console output?

learbia · March 14, 2018, 11:01am

Mar 14 08:00:51] WARNING[13798][C-0000015f]: res_fax.c:1974 receivefax_t38_init: channel ‘PJSIP/TELEFONICA-000001a0’ refused to negotiate T.38
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 000.146469 ], channel sent 7 frames (140 ms) of silence.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 000.166640 ], channel sent 1 frames (20 ms) of energy.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 000.218978 ], stack sent 10 frames (200 ms) of silence.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 000.266635 ], channel sent 5 frames (100 ms) of silence.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 000.286613 ], channel sent 1 frames (20 ms) of energy.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 000.326587 ], channel sent 2 frames (40 ms) of silence.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 000.386644 ], channel sent 3 frames (60 ms) of energy.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 002.819107 ], stack sent 130 frames (2600 ms) of energy.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 002.879018 ], stack sent 3 frames (60 ms) of silence.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 002.986736 ], channel sent 130 frames (2600 ms) of silence.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 003.066756 ], channel sent 4 frames (80 ms) of energy.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 003.086741 ], channel sent 1 frames (20 ms) of silence.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 003.126660 ], channel sent 2 frames (40 ms) of energy.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 004.959006 ], stack sent 104 frames (2080 ms) of energy.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 005.706482 ], channel sent 129 frames (2580 ms) of silence.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 005.746746 ], channel sent 2 frames (40 ms) of energy.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 005.806807 ], channel sent 3 frames (60 ms) of silence.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 005.826782 ], channel sent 1 frames (20 ms) of energy.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 006.126755 ], channel sent 15 frames (300 ms) of silence.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 006.146748 ], channel sent 1 frames (20 ms) of energy.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 006.286686 ], channel sent 7 frames (140 ms) of silence.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 006.306689 ], channel sent 1 frames (20 ms) of energy.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 006.406722 ], channel sent 5 frames (100 ms) of silence.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 006.426819 ], channel sent 1 frames (20 ms) of energy.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 006.546787 ], channel sent 6 frames (120 ms) of silence.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 006.566860 ], channel sent 1 frames (20 ms) of energy.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 006.686674 ], channel sent 6 frames (120 ms) of silence.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 006.706581 ], channel sent 1 frames (20 ms) of energy.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 006.986869 ], channel sent 14 frames (280 ms) of silence.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 007.006843 ], channel sent 1 frames (20 ms) of energy.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 007.406803 ], channel sent 20 frames (400 ms) of silence.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 007.426844 ], channel sent 1 frames (20 ms) of energy.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 007.906844 ], channel sent 24 frames (480 ms) of silence.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 007.926805 ], channel sent 1 frames (20 ms) of energy.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 008.006864 ], channel sent 4 frames (80 ms) of silence.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 008.246693 ], channel sent 12 frames (240 ms) of energy.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 008.458987 ], stack sent 175 frames (3500 ms) of silence.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 008.566865 ], channel sent 16 frames (320 ms) of silence.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 008.626601 ], channel sent 3 frames (60 ms) of energy.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 008.686942 ], channel sent 3 frames (60 ms) of silence.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 008.706811 ], channel sent 1 frames (20 ms) of energy.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 008.806860 ], channel sent 5 frames (100 ms) of silence.
Channel ‘PJSIP/TELEFONICA-000001a0’ fax session ‘80’, [ 008.826708 ], channel sent 1 frames (20 ms) of energy.
– Channel PJSIP/4001-0000019e left ‘simple_bridge’ basic-bridge
– Channel PJSIP/TELEFONICA2-0000019f left ‘simple_bridge’ basic-bridge
== Spawn extension (geral, A30276761, 55) exited non-zero on ‘PJSIP/4001-0000019e’

jcolp · March 14, 2018, 11:02am

Okay, so the channel is going to a ReceiveFax?

learbia · March 14, 2018, 11:52am

For Receive and send it’s a same problem

jcolp · March 14, 2018, 11:54am

Then I’d suggest filing an issue[1] with the complete trace, configuration, console output, and the SIP trace. I have no timeframe on when it would get looked into though.

[1] https://issues.asterisk.org/jira

thiagocnet · March 22, 2018, 1:57pm

Hi, I’m having a similar problem, but it’s with a INVITE and the caller is a mobile (???). Here’s the SIP trace and pjsip.conf, should I open a bug report too?

SIP ->

     Request
     INVITE sip:4731215050@10.143.92.98:5060;transport=UDP;user=phone SIP/2.0
     From:<sip:11992567632@10.150.129.68:5060;user=phone>;tag=3c4-45026-7f4a56-9daf24c-7f4a56
     To:<sip:4731215050@10.143.92.98:5060;user=phone>
     Call-ID:eb76d2304481960a13c47f4a56f13a6220a1b439c2a972a640-0008-7943
     CSeq: 1 INVITE
     User-agent:CS2000_NGSS/9.0
     P-Asserted-Identity:<sip:11992567632@10.150.129.68;user=phone>
     Allow:ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
     Via:SIP/2.0/UDP SOO2CS2K:5060;maddr=10.150.129.68;branch=z9hG4bK-7f4a56-f13a6220-5c9ab8e5
     Max-Forwards:140
     Contact:<sip:10.150.129.68:5060;transport=UDP>
     Supported:100rel,resource-priority
     Content-Type: application/sdp
     Content-Length:420

 SDP ->

     Version = 0.
     Owner = PVG 1521481511010 1521481511010 IN IP4 10.152.205.107.
     Session Name = -.
     Phone Address = +1 6135555555.
     Connection = IN IP4 10.152.205.107.
     Time = 0 0.
     Media Name = audio 56534 RTP/AVP 18 8 101.
     Media Attribute = rtpmap:101 telephone-event/8000.
     Media Attribute = a=fmtp:101 0-15.
     Media Attribute = a=ptime:20.
     Media Attribute = a=fmtp:18 annexb=no.
     Media Attribute = m=image 64726 udptl t38.
     Media Attribute = a=T38FaxVersion:0.
     Media Attribute = a=T38FaxMaxBuffer:1100.
     Media Attribute = a=T38FaxMaxDatagram:612.
     Media Attribute = a=T38MaxBitRate:14400.
     Media Attribute = a=T38FaxRateManagement:transferredTCF.
     Media Attribute = a=T38FaxUdpEC:t38UDPRedundancy.

 SIP ->

     Request
     INVITE sip:4731215050@10.143.92.98:5060;transport=UDP;user=phone SIP/2.0
     From:<sip:11992567632@10.150.129.68:5060;user=phone>;tag=3c4-45026-7f4a56-9daf24c-7f4a56
     To:<sip:4731215050@10.143.92.98:5060;user=phone>
     Call-ID:eb76d2304481960a13c47f4a56f13a6220a1b439c2a972a640-0008-7943
     CSeq: 1 INVITE
     User-agent:CS2000_NGSS/9.0
     P-Asserted-Identity:<sip:11992567632@10.150.129.68;user=phone>
     Allow:ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
     Via:SIP/2.0/UDP SOO2CS2K:5060;maddr=10.150.129.68;branch=z9hG4bK-7f4a56-f13a6220-5c9ab8e5
     Max-Forwards:140
     Contact:<sip:10.150.129.68:5060;transport=UDP>
     Supported:100rel,resource-priority
     Content-Type: application/sdp
     Content-Length:420

 SDP ->

     Version = 0.
     Owner = PVG 1521481511010 1521481511010 IN IP4 10.152.205.107.
     Session Name = -.
     Phone Address = +1 6135555555.
     Connection = IN IP4 10.152.205.107.
     Time = 0 0.
     Media Name = audio 56534 RTP/AVP 18 8 101.
     Media Attribute = rtpmap:101 telephone-event/8000.
     Media Attribute = a=fmtp:101 0-15.
     Media Attribute = a=ptime:20.
     Media Attribute = a=fmtp:18 annexb=no.
     Media Attribute = m=image 64726 udptl t38.
     Media Attribute = a=T38FaxVersion:0.
     Media Attribute = a=T38FaxMaxBuffer:1100.
     Media Attribute = a=T38FaxMaxDatagram:612.
     Media Attribute = a=T38MaxBitRate:14400.
     Media Attribute = a=T38FaxRateManagement:transferredTCF.
     Media Attribute = a=T38FaxUdpEC:t38UDPRedundancy.

 SIP <-

     Response
     SIP/2.0 488 Not Acceptable Here
     Via:SIP/2.0/UDP SOO2CS2K:5060;rport=5060;maddr=10.150.129.68;received=10.150.129.68;branch=z9hG4bK-7f4a56-f13a6220-5c9ab8e5
     Call-ID:eb76d2304481960a13c47f4a56f13a6220a1b439c2a972a640-0008-7943
     From:<sip:11992567632@10.150.129.68;user=phone>;tag=3c4-45026-7f4a56-9daf24c-7f4a56
     To:<sip:4731215050@10.143.92.98;user=phone>;tag=9f804ff5-215a-4c91-a93e-52e4689f866c
     CSeq: 1 INVITE
     Server:Asterisk PBX certified/13.13-cert7
     Content-Length:0

SIP <-

     Response
     SIP/2.0 488 Not Acceptable Here
     Via:SIP/2.0/UDP SOO2CS2K:5060;rport=5060;maddr=10.150.129.68;received=10.150.129.68;branch=z9hG4bK-7f4a56-f13a6220-5c9ab8e5
     Call-ID:eb76d2304481960a13c47f4a56f13a6220a1b439c2a972a640-0008-7943
     From:<sip:11992567632@10.150.129.68;user=phone>;tag=3c4-45026-7f4a56-9daf24c-7f4a56
     To:<sip:4731215050@10.143.92.98;user=phone>;tag=9f804ff5-215a-4c91-a93e-52e4689f866c
     CSeq: 1 INVITE
     Server:Asterisk PBX certified/13.13-cert7
     Content-Length:0

 SIP ->

     Request
     ACK sip:4731215050@10.143.92.98:5060;transport=UDP;user=phone SIP/2.0
     From:<sip:11992567632@10.150.129.68:5060;user=phone>;tag=3c4-45026-7f4a56-9daf24c-7f4a56
     To:<sip:4731215050@10.143.92.98;user=phone>;tag=9f804ff5-215a-4c91-a93e-52e4689f866c
     Call-ID:eb76d2304481960a13c47f4a56f13a6220a1b439c2a972a640-0008-7943
     CSeq: 1 ACK
     User-agent:CS2000_NGSS/9.0
     Max-Forwards:70
     Allow:ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
     Via:SIP/2.0/UDP SOO2CS2K:5060;maddr=10.150.129.68;branch=z9hG4bK-7f4a56-f13a6220-5c9ab8e5
     Contact:<sip:10.150.129.68:5060;transport=UDP>
     Supported:100rel,resource-priority
     Content-Length:0

SIP <-

     Request
     ACK sip:4731215050@10.143.92.98:5060;transport=UDP;user=phone SIP/2.0
     From:<sip:11992567632@10.150.129.68:5060;user=phone>;tag=3c4-45026-7f4a56-9daf24c-7f4a56
     To:<sip:4731215050@10.143.92.98;user=phone>;tag=9f804ff5-215a-4c91-a93e-52e4689f866c
     Call-ID:eb76d2304481960a13c47f4a56f13a6220a1b439c2a972a640-0008-7943
     CSeq: 1 ACK
     User-agent:CS2000_NGSS/9.0
     Max-Forwards:70
     Allow:ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
     Via:SIP/2.0/UDP SOO2CS2K:5060;maddr=10.150.129.68;branch=z9hG4bK-7f4a56-f13a6220-5c9ab8e5
     Contact:<sip:10.150.129.68:5060;transport=UDP>
     Supported:100rel,resource-priority
     Content-Length:0

pjsip.conf:

[voxip]
type=registration
outbound_auth=voxip
server_uri=sip:10.150.129.68
client_uri=sip:4730863277@10.150.129.68
auth_rejection_permanent=no

[voxip]
type=auth
auth_type=userpass
username=4730863277
password=4730863277

[voxip]
type=aor
contact=sip:10.150.129.68
qualify_frequency=60

[voxip]
type=endpoint
context=from-pstn
allow=!all,g729,alaw
;auth=voxip
outbound_auth=voxip
aors=voxip
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
from_user=4730863277
from_domain=10.150.129.68
t38_udptl=yes
t38_udptl_ec=redundancy
fax_detect=no
t38_udptl_nat=yes

[voxip]
type=identify
endpoint=voxip
match=10.150.129.68

jcolp · March 22, 2018, 1:59pm

It wouldn’t surprise me if that doesn’t work, we expect a reinvite changing the stream to T.38. We don’t support both an audio stream and a fax stream. Your version is also old and behavior may have changed since then such that we just reject the individual stream instead.

thiagocnet · March 22, 2018, 2:11pm

Hi @jcolp, I take a look on the changelog and don’t see anything related to T.38, do you believe that I should update the Asterisk? If so, what version do you recommend, 13.18-cert3, 13.20.0 or 15.3.0?

jcolp · March 22, 2018, 2:13pm

SDP negotiation is used for many things, including T.38.

As for what version to update to - if you have a support agreement with Digium then running certified is required for submitting any issues. If you don’t have one then using the latest LTS version (in this case 13.20.0) works fine for people.

learbia · March 22, 2018, 2:13pm

Hello Thiago, if is possible, try Use with new asterisk version and disable 100rel in your trunk, after that I think your t38 works fine.

thiagocnet · March 22, 2018, 2:18pm

Hi @learbia, the problem is that I’m not receiving a FAX, it occurs when some numbers (in this case a mobile from TIM) calls my number, Asterisk isn’t accepting the call.

david551 · March 22, 2018, 2:32pm

You are being offered a FAX media stream.

learbia · March 22, 2018, 3:06pm

Now I Understand, you have a normal call not fax. But your Asterisk think is Fax right ?

try put this in your trunk:
100rel=no

Maybe Asterisk will resend invite without T38, I don’t know its version (asterisk 13). can to that it

thiagocnet · March 22, 2018, 3:41pm

Hi @david551, the problem is that chan_sip works fine when I use it. I’ll have to use it until I find the solution

<--- SIP read from UDP:10.150.129.68:5060 --->
INVITE sip:4731215050@10.143.92.98:5060;transport=UDP;user=phone SIP/2.0
From: <sip:11987291094@10.150.129.68:5060;user=phone>;tag=-13c4-45026-1c4bf-8b2a32c-1c4bf
To: <sip:4731215050@10.143.92.98:5060;user=phone>
Call-ID: 7f9af3004481960a13c41c4bf6e88c5b4097a7dd504d8f40-0058-4883
CSeq: 1 INVITE
User-agent: CS2000_NGSS/9.0
P-Asserted-Identity: <sip:11987291094@10.150.129.68;user=phone>
Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
Via: SIP/2.0/UDP SOO2CS2K:5060;maddr=10.150.129.68;branch=z9hG4bK-1c4bf-6e88c5b-43954845
Max-Forwards: 140
Contact: <sip:10.150.129.68:5060;transport=UDP>
Supported: 100rel,resource-priority
Content-Type: application/sdp
Content-Length: 418

v=0
o=PVG 1521732832740 1521732832740 IN IP4 10.152.204.43
s=-
p=+1 6135555555
c=IN IP4 10.152.204.43
t=0 0
m=audio 49330 RTP/AVP 18 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=fmtp:18 annexb=no
m=image 57522 udptl t38
a=T38FaxVersion:0
a=T38FaxMaxBuffer:1100
a=T38FaxMaxDatagram:612
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (14 headers 18 lines) ---
Sending to 10.150.129.68:5060 (NAT)
Sending to 10.150.129.68:5060 (NAT)
Using INVITE request as basis request - 7f9af3004481960a13c41c4bf6e88c5b4097a7dd504d8f40-0058-4883
Found peer 'VOXIP_GVT' for '11987291094' from 10.150.129.68:5060
  == Using SIP RTP CoS mark 5
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
  == Using UDPTL CoS mark 5                                                                                                                                                                  [107/1736]
Got T.38 offer in SDP in dialog 7f9af3004481960a13c41c4bf6e88c5b4097a7dd504d8f40-0058-4883
Capabilities: us - (alaw), peer - audio=(alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.152.204.43:49330
Looking for 4731215050 in from-pstn (domain 10.143.92.98)
sip_route_dump: route/path hop: <sip:10.150.129.68:5060;transport=UDP>

<--- Transmitting (no NAT) to 10.150.129.68:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP SOO2CS2K:5060;maddr=10.150.129.68;branch=z9hG4bK-1c4bf-6e88c5b-43954845;received=10.150.129.68
From: <sip:11987291094@10.150.129.68:5060;user=phone>;tag=-13c4-45026-1c4bf-8b2a32c-1c4bf
To: <sip:4731215050@10.143.92.98:5060;user=phone>
Call-ID: 7f9af3004481960a13c41c4bf6e88c5b4097a7dd504d8f40-0058-4883
CSeq: 1 INVITE
Server: Asterisk PBX 13.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:4731215050@10.143.92.98:5060>
Content-Length: 0


<------------>
Audio is at 14648
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 10.150.129.68:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP SOO2CS2K:5060;maddr=10.150.129.68;branch=z9hG4bK-1c4bf-6e88c5b-43954845;received=10.150.129.68
From: <sip:11987291094@10.150.129.68:5060;user=phone>;tag=-13c4-45026-1c4bf-8b2a32c-1c4bf
To: <sip:4731215050@10.143.92.98:5060;user=phone>;tag=as2030a2ce
Call-ID: 7f9af3004481960a13c41c4bf6e88c5b4097a7dd504d8f40-0058-4883
CSeq: 1 INVITE
Server: Asterisk PBX 13.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:4731215050@10.143.92.98:5060>
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 413021425 413021425 IN IP4 10.143.92.98                                                                                                                                                [61/1736]
s=Asterisk PBX 13.20.0
c=IN IP4 10.143.92.98
t=0 0
m=audio 14648 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=image 0 udptl t38

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

<--- SIP read from UDP:10.150.129.68:5060 --->
ACK sip:4731215050@10.143.92.98:5060 SIP/2.0
From: <sip:11987291094@10.150.129.68:5060;user=phone>;tag=-13c4-45026-1c4bf-8b2a32c-1c4bf
To: <sip:4731215050@10.143.92.98:5060;user=phone>;tag=as2030a2ce
Call-ID: 7f9af3004481960a13c41c4bf6e88c5b4097a7dd504d8f40-0058-4883
CSeq: 1 ACK
User-agent: CS2000_NGSS/9.0
Max-Forwards: 70
Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
Via: SIP/2.0/UDP SOO2CS2K:5060;maddr=10.150.129.68;branch=z9hG4bK-1c4bf-6e88c76-6ec0f83c
Contact: <sip:10.150.129.68:5060;transport=UDP>
Supported: 100rel,resource-priority
Content-Length: 0

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

thiagocnet · March 22, 2018, 4:01pm

@jcolp I’ve updated the asterisk version, but the problem continues. Any suggestion? I’m stuck here.

jcolp · March 22, 2018, 4:08pm

You’d need to file an issue[1] with the provided details and the SDP negotiation could probably be made smarter so it behaved like chan_sip. I have no timeframe on when this would be done, though. Noone else has experienced this (or at least has reported it).

[1] https://issues.asterisk.org/jira