Sip Fax lot of error

Asterisk Asterisk Support
1

Hi

I use Asterisk for sending and receiving faxes.
The solution works, but I meet a lot of error:

  • (Incoming and outgoing) The call dropped prematurely
  • (Incoming and outgoing) Unexpected message received
  • (Incoming) Disconnected after permitted retries
  • (Outgoing) Invalid response after sending a page

Asterisk 1.6.2.17.2

A part of my config
sip.conf

[general]
# sipdebug=yes
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=no
language=fr

tos_sip=cs3
tos_audio=ef
tos_video=af41

;videosupport=yes
t38pt_udptl=yes,redundancy,maxdatagram=400
t38pt_rtp=no
t38pt_tcp=no

notifyringing=yes
notifyhold=yes
limitonpeers=yes

rtcachefriends=yes
rtsavesysname=yes
rtupdate=yes
rtautoclear=no
ignoreregexpire=yes

[authentication]

#include <sip_trunks.conf>

sip_trunks.conf

[basic-opts](!)
type=friend
insecure=port,invite
canreinvite=no
disallow=all
allow=alaw
dtmfmode=rfc2833
nat=no

[sbc1](basic-opts)
context=inbound
disallow=all
allow=ulaw
host=10.8.254.20
dtmfmode=rfc2833
2

Are you using T.38 on the SIP Trunk on which you are receiving faxes?

3

Yes I do
Yes, why ask me that?

4

I on kernel 2.6.18-194.el5 x86_64

I find this in my log :

[Sep 15 16:10:57] VERBOSE[31928] chan_sip.c:
<--- SIP read from UDP:10.8.254.20:5060 --->
BYE sip:**********@10.249.10.66 SIP/2.0
Via: SIP/2.0/UDP 10.8.254.20:5060;branch=z9hG4bKtojgh596ra9seaprjnl8lrvvb4-1sa
Max-Forwards: 69
From: <sip:**********@10.8.254.20>;tag=1c497186327
To: "**********" <sip:**********@10.249.10.66>;tag=as412577bc
Call-ID: 6ab846bf44b946a91e6aa4ee446357df@10.249.10.66
CSeq: 2 BYE
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.035.004
Reason: Q.850 ;cause=16 ;text="local"
Content-Length: 0


<------------->
5

[quote=“g.tour”]Yes I do
Yes, why ask me that?[/quote]

Because T.38 is the only way you get a guaranteed fax delivery on VoIP system.

Can you please post the output of the “sip set debug on” command? (you type it in before you send/receive a call)

6

[code] – Attempting call on SIP/@sbc1 for @outbound:1 (Retry 1)
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
Audio is at 10.249.10.162 port 11768
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.8.254.20:5060:
INVITE sip:@10.8.254.20 SIP/2.0
Via: SIP/2.0/UDP 10.249.10.162:5060;branch=z9hG4bK27075cf4;rport
Max-Forwards: 70
From: "" sip:**********@10.249.10.162;tag=as3dee4028
To: sip:**********@10.8.254.20
Contact: sip:**********@10.249.10.162
Call-ID: 6feafcc012e52a7f47c3ade8730ce5d9@10.249.10.162
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.17.2
Date: Thu, 29 Sep 2011 15:16:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 266

v=0
o=root 838049699 838049699 IN IP4 10.249.10.162
s=Asterisk PBX 1.6.2.17.2
c=IN IP4 10.249.10.162
t=0 0
m=audio 11768 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<— SIP read from UDP:10.8.254.20:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.249.10.162:5060;received=10.249.10.162;branch=z9hG4bK27075cf4;rport=5060
From: “**********” sip:**********@10.249.10.162;tag=as3dee4028
To: sip:**********@10.8.254.20
Call-ID: 6feafcc012e52a7f47c3ade8730ce5d9@10.249.10.162
CSeq: 102 INVITE
Content-Length: 0

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

<— SIP read from UDP:10.8.254.20:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.249.10.162:5060;received=10.249.10.162;branch=z9hG4bK27075cf4;rport=5060
From: “**********” sip:**********@10.249.10.162;tag=as3dee4028
To: sip:**********@10.8.254.20;tag=1c870885245
Call-ID: 6feafcc012e52a7f47c3ade8730ce5d9@10.249.10.162
CSeq: 102 INVITE
Contact: sip:10.8.254.20:5060;transport=udp
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-/v.5.80A.035.004
Content-Type: application/sdp
Content-Length: 258

v=0
o=AudiocodesGW 870962719 870962453 IN IP4 10.8.254.20
s=Phone-Call
c=IN IP4 10.8.254.20
t=0 0
m=audio 6436 RTP/AVP 0 105 101
a=rtpmap:0 PCMU/8000
a=rtpmap:105 X-NSE/8000
a=ptime:20
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

<------------->
— (12 headers 12 lines) —
Found RTP audio format 0
Found RTP audio format 105
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format X-NSE for ID 105
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x4000004 (ulaw|red)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.8.254.20:6436

<— SIP read from UDP:10.8.254.20:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.249.10.162:5060;received=10.249.10.162;branch=z9hG4bK27075cf4;rport=5060
From: “**********” sip:**********@10.249.10.162;tag=as3dee4028
To: sip:**********@10.8.254.20;tag=1c870885245
Call-ID: 6feafcc012e52a7f47c3ade8730ce5d9@10.249.10.162
CSeq: 102 INVITE
Contact: sip:10.8.254.20:5060;transport=udp
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-/v.5.80A.035.004
Content-Type: application/sdp
Content-Length: 258

v=0
o=AudiocodesGW 870962719 870962453 IN IP4 10.8.254.20
s=Phone-Call
c=IN IP4 10.8.254.20
t=0 0
m=audio 6436 RTP/AVP 0 105 101
a=rtpmap:0 PCMU/8000
a=rtpmap:105 X-NSE/8000
a=ptime:20
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

<------------->
— (12 headers 12 lines) —
list_route: hop: sip:10.8.254.20:5060;transport=udp
set_destination: Parsing sip:10.8.254.20:5060;transport=udp for address/port to send to
set_destination: set destination to 10.8.254.20, port 5060
Transmitting (no NAT) to 10.8.254.20:5060:
ACK sip:10.8.254.20:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.10.162:5060;branch=z9hG4bK505aa9d5;rport
Max-Forwards: 70
From: “**********” sip:**********@10.249.10.162;tag=as3dee4028
To: sip:**********@10.8.254.20;tag=1c870885245
Contact: sip:**********@10.249.10.162
Call-ID: 6feafcc012e52a7f47c3ade8730ce5d9@10.249.10.162
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.17.2
Content-Length: 0

   > Channel SIP/sbc1-0000108e was answered.
-- Executing [**********@outbound:1] Set("SIP/sbc1-0000108e", "GROUP()=SBC1_OUT") in new stack
-- Executing [**********@outbound:2] Verbose("SIP/sbc1-0000108e", ""**** SENDING FAX : /tmp/scim.tmp/ASWS-d71b2ea9-ad08-47c6-a1ed-adc2a8d449b9/out-ASWS-d71b2ea9-ad08-47c6-a1ed-adc2a8d449b9.tif ****"") in new stack

**** SENDING FAX : /tmp/scim.tmp/ASWS-d71b2ea9-ad08-47c6-a1ed-adc2a8d449b9/out-ASWS-d71b2ea9-ad08-47c6-a1ed-adc2a8d449b9.tif ****
– Executing [@outbound:3] Set(“SIP/sbc1-0000108e”, “LOCALHEADERINFO=”) in new stack
– Executing [@outbound:4] Set(“SIP/sbc1-0000108e”, "LOCALSTATIONID=") in new stack
– Executing [@outbound:5] Set(“SIP/sbc1-0000108e”, "MYSRCFROM=") in new stack
– Executing [@outbound:6] SendFAX(“SIP/sbc1-0000108e”, “/tmp/scim.tmp/ASWS-d71b2ea9-ad08-47c6-a1ed-adc2a8d449b9/out-ASWS-d71b2ea9-ad08-47c6-a1ed-adc2a8d449b9.tif”) in new stack

<— SIP read from UDP:10.8.254.20:5060 —>
INVITE sip:@10.249.10.162 SIP/2.0
Via: SIP/2.0/UDP 10.8.254.20:5060;branch=z9hG4bKh0a4n293oh7vv7d3drjuo1lfs0-g0ka
Max-Forwards: 69
From: sip:**********@10.8.254.20;tag=1c870885245
To: "" sip:**********@10.249.10.162;tag=as3dee4028
Call-ID: 6feafcc012e52a7f47c3ade8730ce5d9@10.249.10.162
CSeq: 1 INVITE
Contact: sip:10.8.254.20:5060;transport=udp
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.035.004
Content-Type: application/sdp
Content-Length: 289

v=0
o=AudiocodesGW 870962719 870962454 IN IP4 10.8.254.20
s=Phone-Call
c=IN IP4 10.8.254.20
t=0 0
m=image 6436 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:122
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy

<------------->
— (13 headers 12 lines) —
Sending to 10.8.254.20 : 5060 (no NAT)
Got T.38 offer in SDP in dialog 6feafcc012e52a7f47c3ade8730ce5d9@10.249.10.162
Capabilities: us - 0x4 (ulaw), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.

<— Transmitting (no NAT) to 10.8.254.20:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.8.254.20:5060;branch=z9hG4bKh0a4n293oh7vv7d3drjuo1lfs0-g0ka;received=10.8.254.20
From: sip:**********@10.8.254.20;tag=1c870885245
To: “**********” sip:**********@10.249.10.162;tag=as3dee4028
Call-ID: 6feafcc012e52a7f47c3ade8730ce5d9@10.249.10.162
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.17.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:**********@10.249.10.162
Content-Length: 0

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

<— Reliably Transmitting (no NAT) to 10.8.254.20:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.8.254.20:5060;branch=z9hG4bKh0a4n293oh7vv7d3drjuo1lfs0-g0ka;received=10.8.254.20
From: sip:**********@10.8.254.20;tag=1c870885245
To: “**********” sip:**********@10.249.10.162;tag=as3dee4028
Call-ID: 6feafcc012e52a7f47c3ade8730ce5d9@10.249.10.162
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.17.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:**********@10.249.10.162
Content-Type: application/sdp
Content-Length: 275

v=0
o=root 838049699 838049700 IN IP4 10.249.10.162
s=Asterisk PBX 1.6.2.17.2
c=IN IP4 10.249.10.162
t=0 0
m=image 4556 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:1400
a=T38FaxUdpEC:t38UDPRedundancy

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

<— SIP read from UDP:10.8.254.20:5060 —>
ACK sip:@10.249.10.162 SIP/2.0
Via: SIP/2.0/UDP 10.8.254.20:5060;branch=z9hG4bKh0a4n293oh7vv7d3drjuo1lfs0-g0oa
Max-Forwards: 69
From: sip:**********@10.8.254.20;tag=1c870885245
To: "" sip:**********@10.249.10.162;tag=as3dee4028
Call-ID: 6feafcc012e52a7f47c3ade8730ce5d9@10.249.10.162
CSeq: 1 ACK
Contact: sip:10.8.254.20:5060
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.035.004
Content-Length: 0

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

<— SIP read from UDP:10.8.254.20:5060 —>
BYE sip:@10.249.10.162 SIP/2.0
Via: SIP/2.0/UDP 10.8.254.20:5060;branch=z9hG4bKh0a4n293oh7vv7d3drjuo1lfs0-1sa
Max-Forwards: 69
From: sip:**********@10.8.254.20;tag=1c870885245
To: "" sip:**********@10.249.10.162;tag=as3dee4028
Call-ID: 6feafcc012e52a7f47c3ade8730ce5d9@10.249.10.162
CSeq: 2 BYE
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.035.004
Reason: Q.850 ;cause=16 ;text="local"
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 10.8.254.20 : 5060 (no NAT)

<— Transmitting (no NAT) to 10.8.254.20:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.8.254.20:5060;branch=z9hG4bKh0a4n293oh7vv7d3drjuo1lfs0-1sa;received=10.8.254.20
From: sip:**********@10.8.254.20;tag=1c870885245
To: “**********” sip:**********@10.249.10.162;tag=as3dee4028
Call-ID: 6feafcc012e52a7f47c3ade8730ce5d9@10.249.10.162
CSeq: 2 BYE
Server: Asterisk PBX 1.6.2.17.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>
== Spawn extension (outbound, , 6) exited non-zero on ‘SIP/sbc1-0000108e’
– Executing [h@outbound:1] Verbose(“SIP/sbc1-0000108e”, ““FAX SENT WITH PAGES, FAXSTATUS: FAILED , FAXERROR: The call dropped prematurely, FROM ********** TO h””) in new stack
FAX SENT WITH PAGES, FAXSTATUS: FAILED , FAXERROR: The call dropped prematurely, FROM ********** TO h
– Executing [h@outbound:2] Set(“SIP/sbc1-0000108e”, “CDR(pages_number)=”) in new stack
– Executing [h@outbound:3] Set(“SIP/sbc1-0000108e”, “CDR(extid)=ASWS-d71b2ea9-ad08-47c6-a1ed-adc2a8d449b9-0”) in new stack
– Executing [h@outbound:4] Set(“SIP/sbc1-0000108e”, “CDR(fax_status)=FAILED”) in new stack
– Executing [h@outbound:5] Set(“SIP/sbc1-0000108e”, "CDR(src_from)=") in new stack
– Executing [h@outbound:6] GotoIf(“SIP/sbc1-0000108e”, “1?xchanger”) in new stack
– Goto (outbound,h,9)
– Executing [h@outbound:9] Set(“SIP/sbc1-0000108e”, “CDR(fax_error)=The call dropped prematurely”) in new stack
– Executing [h@outbound:10] NoOp(“SIP/sbc1-0000108e”, “”) in new stack
> [INSERT INTO cdr (uniqueid,calldate,clid,src_from,dst,dcontext,channel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,fax_status,fax_error,extid) VALUES (‘1317309395.4243’,{ ts ‘2011-09-29 17:16:35’ },’"" <>’,‘’,'’,‘outbound’,‘SIP/sbc1-0000108e’,‘SendFAX’,’/tmp/scim.tmp/ASWS-d71b2ea9-ad08-47c6-a1ed-adc2a8d449b9/out-ASWS-d71b2ea9-ad08-’,57,47,‘ANSWERED’,3,‘14’,‘FAILED’,‘The call dropped prematurely’,‘ASWS-d71b2ea9-ad08-47c6-a1ed-adc2a8d449b9-0’)]
Really destroying SIP dialog ‘6feafcc012e52a7f47c3ade8730ce5d9@10.249.10.162’ Method: BYE
[/code]

7

This is my extensions.conf

[general]
static=no
writeprotect=yes
clearglobalvars=yes
language=fr

[globals]

[default]

;[tdmin]
;switch =>Realtime/inbound@inbounddid
;exten => _X.,1,Goto(fax2mail,${EXTEN},1)

;exten =>i,1,Verbose("ERROR: CALL FOR ${INVALID_EXTEN} FROM ${CALLERID(all)} RECEIVED BUT NUMBER IS NOT ASSIGNED.")
;exten =>i,n,Congestion(3)

[inbound]
;exten =>_XX.,1,Set(GROUP()=SBC1_IN)
;exten =>_XX.,n,Verbose("PROCESSING INCOMING CALL FROM ${CALLERID(all)} TO ${EXTEN}")
;exten =>_XX.,n,Goto(tdmin,${EXTEN},1)
exten => _XXX!,1,Set(GROUP()=SBC1_IN)
exten => _XXX!,n,Set(MYSRCFROM=${EXTEN})
exten => _XXX!,n,Verbose("PROCESSING INCOMING CALL FROM ${CALLERID(all)} TO ${EXTEN}")
exten => _XXX!,n,NoOp(Processing incoming fax for ${EXTEN} from ${CALLERID(all)}.)
exten => _XXX!,n,Set(ARRAY(EMAILS,ACCTCODE)=${ODBC_GETFROMDID(${EXTEN})})
exten => _XXX!,n,Answer()
exten => _XXX!,n,GotoIf($["${ACCTCODE}" != ""]?exists)
exten => _XXX!,n,Set(ARRAY(FAXSTATUS,FAXERROR,FAXMODE)=FAILED,No fax entry for ${EXTEN},none)
exten => _XXX!,n,Goto(end)
exten => _XXX!,n(exists),Set(CDR(accountcode)=${ACCTCODE})
exten => _XXX!,n,Set(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tiff)
;exten => _XXX!,n,Wait(3)
exten => _XXX!,n,ReceiveFAX(${FAXFILE})
exten => _XXX!,n(end),Hangup()

exten => h,1,NoOp(Incoming fax result: ${FAXSTATUS}. Error (if any): ${FAXERROR}. Mode: ${FAXMODE}. Localstationid: ${LOCALSTATIONID} Remotestationid: ${REMOTESTATIONID})
exten => h,n,Set(CDR(pages_number)=${FAXPAGES})
exten => h,n,Set(CDR(extid)=${MYEXTID})
exten => h,n,Set(CDR(fax_status)=${FAXSTATUS})
exten => h,n,Set(CDR(fax_error)=${FAXERROR})
exten => h,n,Set(CDR(src_from)=${REMOTESTATIONID})
exten => h,n,GotoIf($["${FAXSTATUS}" != "SUCCESS"]?end)
exten => h,n,GotoIf($["${EMAILS}" = ""]?end)
exten => h,n,Set(PDFFILE=/tmp/${UNIQUEID}.pdf) ; fax2mail
exten => h,n,System(tiff2pdf -o ${PDFFILE} ${FAXFILE}) ; fax2mail
exten => h,n,System(/usr/bin/python ) ; fax2mail
exten => h,n(end),NoOp()




[outbound]
;SEB

exten =>_XX.,1,Set(GROUP()=SBC1_OUT)
exten =>_XX.,n,Verbose("**** SENDING FAX : ${MYFILE} ****")
exten =>_XX.,n,Set(LOCALHEADERINFO=${MYHEADER})
exten =>_XX.,n,Set(LOCALSTATIONID=${MYNUMBER})
exten =>_XX.,n,Set(MYSRCFROM=${MYNUMBER})
exten =>_XX.,n,SendFax(${MYFILE})
exten =>_XX.,n,Hangup()

exten => failed,1,Set(FAXSTATUS=FAILED)
exten => failed,n,Set(FAXPAGES=0)
exten => failed,n,Set(MYSRCFROM=${MYNUMBER})
exten => failed,n,Hangup()

exten => h,1,Verbose("FAX SENT WITH ${FAXPAGES} PAGES, FAXSTATUS: ${FAXSTATUS} , FAXERROR: ${FAXERROR}, FROM ${MYNUMBER} TO ${EXTEN}")
exten => h,n,Set(CDR(pages_number)=${FAXPAGES})
exten => h,n,Set(CDR(extid)=${MYEXTID})
exten => h,n,Set(CDR(fax_status)=${FAXSTATUS})
exten => h,n,Set(CDR(src_from)=${MYSRCFROM})
exten => h,n,GotoIf($["${FAXERROR}" != ""]?xchanger)
exten => h,n,Set(CDR(fax_error)=${CDR(disposition)})
exten => h,n,Goto(end)
exten => h,n(xchanger),Set(CDR(fax_error)=${FAXERROR})
exten => h,n(end),NoOp

Thanks in advance

8

I have another plateform with another trunk_sip
There is a small difference :
t38pt_udptl=yes,redundancy
without ,maxdatagram=400
and
allow=alaw,ulaw
and vise versa

I’m not sure about the configuration of the two trunk_sip because it’s not mine

9

Any help ? please

I found a lot of warning message :
WARNING[6746] app_fax.c: Transmission error

Can i have to check ecm mode for Asterisk and the trunk_sip ? INVITE/REINVITE ? FaxMode Pass-Through or other ?

Thanks in advance

10

t38pt_udptl=yes,redundancy,maxdatagram=400
Is anyone know the options available?

I have compiled Asterisk with spandsp for having the module app_fax, I do not have many options available or the configuration file. ECM is usually enabled by default, but I’m not sure, where can I find it?