Hi

####################################################
This is the inbound part to freeswitch when they are calling me:

2019/02/12 12:44:00.258755 217.0.15.67:5060 -> 10.0.200.2:49881
INVITE sip:auto_to_user@87.157.X.X:5080;transport=tcp;gw=sip-trunk.telekom.de SIP/2.0
Via: SIP/2.0/TCP 217.0.15.67:5060;branch=XYXYXYXYXYXYXYX
Record-Route: <sip:reg.sip-trunk.telekom.de;transport=tcp;lr>
Max-Forwards: 57
To: +49XXXXXX18 <sip:+49XXXXXX18@TELEKOM.DE;user=phone>
From: <sip:+49XXXXXX11@sip-trunk.telekom.de;transport=udp;user=phone>;tag=799d34ab
Call-ID: a6a01cb4d9f46c96@62.156.111.70
Contact: <sip:SxmWXkZ2+Ob+Zli0Jp3jABUfr2GdT4MBP0kzwJgGCuGtkw38S668+tgCxOE6QHU1@th1>
Supported: 100rel,histinfo,timer
Session-Expires: 1900;refresher=uac
CSeq: 12119032 INVITE
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER, UPDATE
Min-SE: 900
P-Asserted-Identity: <sip:+49XXXXXX11@sip-trunk.telekom.de;user=phone>
P-Called-Party-ID: <sip:+49XXXXXX18@sip-trunk.telekom.de;user=phone>
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 499

v=0
o=- 40086765 40086765 IN IP4 217.0.15.67
s=on transit
c=IN IP4 217.0.132.153
t=0 0
m=audio 18570 RTP/AVP 8 118 126 101
a=rtpmap:8 PCMA/8000
a=rtpmap:118 PCMA/8000
a=gpmd:118 vbd=yes
a=rtpmap:126 CLEARMODE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
m=image 14956 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:3000
a=T38FaxMaxDatagram:500
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
a=sendrecv

2019/02/12 12:44:00.258755 10.0.200.2:49881 -> 217.0.15.67:5060
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 217.0.15.67:5060;branch=XYXYXYXYXYXYXYX;rport=5060
Record-Route: <sip:reg.sip-trunk.telekom.de;transport=tcp;lr>
From: <sip:+49XXXXXX11@sip-trunk.telekom.de;transport=udp;user=phone>;tag=799d34ab
To: +49XXXXXX18 <sip:+49XXXXXX18@TELEKOM.DE;user=phone>
Call-ID: a6a01cb4d9f46c96@62.156.111.70
CSeq: 12119032 INVITE
Content-Length: 0

2019/02/12 12:44:00.406766 10.0.200.2:49881 -> 217.0.15.67:5060
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 217.0.15.67:5060;branch=XYXYXYXYXYXYXYX;rport=5060
Record-Route: <sip:reg.sip-trunk.telekom.de;transport=tcp;lr>
From: <sip:+49XXXXXX11@sip-trunk.telekom.de;transport=udp;user=phone>;tag=799d34ab
To: +49XXXXXX18 <sip:+49XXXXXX18@TELEKOM.DE;user=phone>;tag=7K7y11N0937Fj
Call-ID: a6a01cb4d9f46c96@62.156.111.70
CSeq: 12119032 INVITE
Contact: <sip:+49XXXXXX18@87.157.X.X:5080;transport=tcp>
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Length: 0
P-Asserted-Identity: "+49XXXXXX18" <sip:+49XXXXXX18@TELEKOM.DE>


2019/02/12 12:44:02.234905 217.0.15.67:5060 -> 10.0.200.2:49881
BYE sip:+49XXXXXX18@87.157.X.X:5080;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 217.0.15.67:5060;branch=ABABABABAB
Record-Route: <sip:reg.sip-trunk.telekom.de;transport=tcp;lr>
Max-Forwards: 61
To: +49XXXXXX18 <sip:+49XXXXXX18@TELEKOM.DE;user=phone>;tag=7K7y11N0937Fj
From: <sip:+49XXXXXX11@sip-trunk.telekom.de;transport=udp;user=phone>;tag=799d34ab
Call-ID: a6a01cb4d9f46c96@62.156.111.70
Contact: <sip:SxmWXkZ2+Ob+Zli0Jp3jABUfr2GdT4MBP0kzwJgGCuGtkw38S668+tgCxOE6QHU1@th1>
CSeq: 12119033 BYE
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, OPTIONS, PRACK, REGISTER, UPDATE
Warning: 399 MSX_SIPCOM "0 RELEASE FROM CC! ExternalRel:1, PeerType:278, Call State:9, Reason:25"
Content-Length: 0


2019/02/12 12:44:02.246906 10.0.200.2:49881 -> 217.0.15.67:5060
SIP/2.0 200 OK
Via: SIP/2.0/TCP 217.0.15.67:5060;branch=ABABABABAB;rport=5060
From: <sip:+49XXXXXX11@sip-trunk.telekom.de;transport=udp;user=phone>;tag=799d34ab
To: +49XXXXXX18 <sip:+49XXXXXX18@TELEKOM.DE;user=phone>;tag=7K7y11N0937Fj
Call-ID: a6a01cb4d9f46c96@62.156.111.70
CSeq: 12119033 BYE
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Content-Length: 0


2019/02/12 12:44:02.246906 10.0.200.2:49881 -> 217.0.15.67:5060
SIP/2.0 487 Early Session Terminated
Via: SIP/2.0/TCP 217.0.15.67:5060;branch=XYXYXYXYXYXYXYX;rport=5060
From: <sip:+49XXXXXX11@sip-trunk.telekom.de;transport=udp;user=phone>;tag=799d34ab
To: +49XXXXXX18 <sip:+49XXXXXX18@TELEKOM.DE;user=phone>;tag=7K7y11N0937Fj
Call-ID: a6a01cb4d9f46c96@62.156.111.70
CSeq: 12119032 INVITE
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Length: 0

2019/02/12 12:44:02.266908 217.0.15.67:5060 -> 10.0.200.2:49881
ACK sip:auto_to_user@87.157.X.X:5080;transport=tcp;gw=sip-trunk.telekom.de SIP/2.0
Via: SIP/2.0/TCP 217.0.15.67:5060;branch=XYXYXYXYXYXYXYX
Max-Forwards: 57
To: +49XXXXXX18 <sip:+49XXXXXX18@TELEKOM.DE;user=phone>;tag=7K7y11N0937Fj
From: <sip:+49XXXXXX11@sip-trunk.telekom.de;transport=udp;user=phone>;tag=799d34ab
Call-ID: a6a01cb4d9f46c96@62.156.111.70
CSeq: 12119032 ACK
Content-Length: 0

########################################
This is the invite to my phone behind freeswitch

2019/02/12 12:44:00.298758 10.0.200.2:5060 -> 10.0.200.18:5060
INVITE sip:18@10.0.200.18:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.200.2;rport;branch=z9hG4bKZ4t3F3ZrKrHQc
Max-Forwards: 55
From: "+49XXXXXXXX11" <sip:+49XXXXXXXX11@10.0.200.2>;tag=a3mj5XKSHyjjD
To: <sip:18@10.0.200.18:5060>
Call-ID: 5466d642-a95e-1237-f598-000024d0e264
CSeq: 432128 INVITE
Contact: <sip:mod_sofia@10.0.200.2:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Privacy: none
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 239
P-Called-Party-ID: <sip:+49XXXXXXXX18@sip-trunk.telekom.de;user=phone>
X-FS-Support: update_display,send_info
P-Asserted-Identity: "+49XXXXXXXX11" <sip:+49XXXXXXXX11@10.0.200.2>

v=0
o=FreeSWITCH 1549964758 1549964759 IN IP4 10.0.200.2
s=FreeSWITCH
c=IN IP4 10.0.200.2
t=0 0
m=audio 7082 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

Am Do., 23. Aug. 2018 um 17:50 Uhr schrieb Brian West <brian@freeswitch.com>:
it actually is probably fax related since the initial invite has T.38 stuff in it, I bet its getting tripped up because T.38 is a NO GO REJECT and it probably says something to that in the debug logs.

/b


On Wed, Aug 22, 2018 at 3:15 PM, Paul Muaddib <paul.muaddib83@gmail.com> wrote:
Hi,

It is definitely not a Fax. The guys, who are calling us from this number, are actually complaining that we are not reachable. But when we call them back the voice call perfectly works. Other Inbound calls are no problem.

Best regards,
Paul

Am Mi., 22. Aug. 2018 um 21:53 Uhr schrieb Vallimamod Abdullah <vma@vallimamod.org>:
Hi,

When there is udptl media type in sdp, freeswitch ignores the audio type. That's why you are getting the codec error (if you don't have t38 enabled.)
The caller looks like a fax machine, which explains why you get a direct t38 request and not a reinvite.


Best Regards,
-- 
Vallimamod Abdullah
SIP Solutions

On 22 Aug 2018, at 15:55, Paul Muaddib <paul.muaddib83@gmail.com> wrote:

vars.xml

  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA,PCMU"/>
  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA,PCMU"/>

My phones, all Grandstream GXP2140, accept PCMA and PCMU

I don't have trouble with other people calling me

Am Mi., 22. Aug. 2018 um 15:21 Uhr schrieb Thorsten Göllner <tg-maillistings@level5.de>:
Only PCMA@8000 is offered. Check device if codec is activated.

Am 21.08.2018 um 15:06 schrieb Paul Muaddib:
Hi,

I have a rarely error where I get this failure message in the log file. I am bit wondering because this is suposed to be a normal voice call and not a fax.

Best regards,
Paul.

fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 2018-08-20 14:04:54.056776 [INFO] switch_ivr_originate.c:1215 Sending early media
fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 2018-08-20 14:04:54.056776 [ERR] mod_sofia.c:2343 CODEC NEGOTIATION ERROR.  SDP:
fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 v=0^M
fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 o=- 20012711 20012711 IN IP4 217.0.15.67^M
fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 s=on transit^M
fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 c=IN IP4 X.X.X.X^M
fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 t=0 0^M
fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 m=audio 13274 RTP/AVP 8 118 126 101^M
fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=rtpmap:8 PCMA/8000^M
fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=rtpmap:118 PCMA/8000^M
fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=rtpmap:126 CLEARMODE/8000^M
fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=rtpmap:101 telephone-event/8000^M
fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=fmtp:101 0-15^M
fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=gpmd:118 vbd=yes^M
fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=ptime:20^M
fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 m=image 11232 udptl t38^M
fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=T38FaxVersion:0^M
fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=T38MaxBitRate:14400^M
fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=T38FaxMaxBuffer:3000^M
fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=T38FaxMaxDatagram:500^M
fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=T38FaxRateManagement:transferredTCF^M
fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 a=T38FaxUdpEC:t38UDPRedundancy^M
fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28
fe3eed7c-e03a-40af-bc2e-ae0bfe54bb28 2018-08-20 14:04:54.056776 [NOTICE] switch_channel.c:3515 Hangup sofia/external/+49XXXXXXXXX@sip-trunk.telekom.de [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]

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_________________________________________________________________________
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_________________________________________________________________________
Professional FreeSWITCH Services
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https://freeswitch.com

Official FreeSWITCH Sites
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