I think the "488 Not acceptable here" is occurring because the channel
connecting through is not T.38 capable, that will be the IAX channel
from iaxmomdem.
I've not used PJSIP so cannot offer any advice regarding it however you
may try to make iaxmodem connect through another context using either
SIP or IAX (experiment with both bu most probably IAX) in an attempt to
prevent the rejection of the T.38 establishment forcing the call to
terminate. What I seem to recall when experimenting with SIP as the
trunk, have UDPTL disabled i.e. t38pt_udptl=no, this would also induce
"488 Not acceptable here".
Looking at a legacy configuration where I tested iaxmodem
(context=faxgateway-iax) going through Asterisk 1.2 which then forwarded
the request to Asterisk 11 (context=FAX-T30) where it then went out
through the trunk with Fax Gateway enabled.
In short;
Asterisk 1.2
IAX Modem in context faxgateway-iax, could change to faxgateway-sip.
[faxgateway-iax]
; Incoming calls from iaxmodem to Asterisk are directed to
; other Asterisk server.
exten => _XX.,1,Dial(IAX2/faxgw-iax at faxgw-iax/${EXTEN},55,t)
exten => _XX.,n,Wait(1)
exten => _XX.,n,Hangup
;
[faxgateway-sip]
; Incoming calls from iaxmodem to Asterisk are directed to
; other Asterisk server.
exten => _XX.,1,Dial(SIP/${EXTEN}@faxgw-sip,55,t)
exten => _XX.,n,Wait(1)
exten => _XX.,n,Hangup
;
Asterisk 11
IAX user faxgw-iax is in context FAX-T30
extensions.ael on Asterisk 11 contains
context FAX-T30 {
<snip>
_XXXXXXXX => {
// Set(FAXOPT(t38gateway)=yes);
Dial(SIP/${EXTEN}@itsp-fax,55);
Hangup();
};
<snip>
};
One other note, enable alaw & ulaw in iaxmomdem and your iax peer
configuration in Asterisk, just to be sure!
I know this isn't specific to your case but maybe you can make something
from this that helps.
Please note, I don't have the old set up to test so I can't be certain
of the above configurations.
Cheers,
Larry.
On 27/07/2015 11:15 AM, Jean-Denis Girard wrote:> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi list,
>
> 2 weks ago I asked questions about PJSIP and T.38 but got no replies. I
> upgraded Asterisk to git as of yesterday (309dd2a), and I'm still
having
> the same issues.
>
> In the trace below, I'm sending a fax from Hylafax server through
> iaxmodem on Asterisk-13 (tiare) to a second Asterisk (11.18.0 t0gw)
> connected to the PSTN via ISDN; the call is to my test fax machine,
> connected to the PSTN. chan_pjsip is used on Asterisk-13, and chan_sip
> is used on Asterisk-11.
>
> This how endpoint t0gw (Asterisk-11) is configured on tiare (Asterisk-13
> ):
> tiare*CLI> pjsip show endpoint t0gw
> ...
> t38_udptl : true
> t38_udptl_ec : fec
> t38_udptl_ipv6 : false
> t38_udptl_maxdatagram : 400
> t38_udptl_nat : false
> ...
>
> Could someone explain why I'm getting "Not acceptable" below?
>
> -- Accepting AUTHENTICATED call from 127.0.0.1:4570:
> -- > requested format = slin,
> -- > requested prefs = (),
> -- > actual format = slin,
> -- > host prefs = (slin),
> -- > priority = mine
> -- Executing [40ZZZZZZ at fax-sortant:1]
NoOp("IAX2/iaxmodem0-7838", "
> calls 40ZZZZZZ (local)") in new stack
> -- Executing [40ZZZZZZ at fax-sortant:2]
Set("IAX2/iaxmodem0-7838",
> "FAXOPT(gateway)=yes") in new stack
> -- Executing [40ZZZZZZ at fax-sortant:3]
Dial("IAX2/iaxmodem0-7838",
> "PJSIP/40ZZZZZZ at t0gw") in new stack
> -- Called PJSIP/40ZZZZZZ at t0gw
> <--- Transmitting SIP request (936 bytes) to UDP:192.168.0.10:5060
--->
> INVITE sip:40ZZZZZZ at gw.sysnux.pf SIP/2.0
> Via: SIP/2.0/UDP
> 192.168.0.200:5060;rport;branch=z9hG4bKPjba38816d-b5fe-4d5f-8bda-d0968e3
> 8e5f1
> From: "SysNux"
> <sip:+68940XXXXXX at
192.168.0.200>;tag=22d8369f-061d-4232-9c4d-5068e81bc5c
> 5
> To: <sip:40ZZZZZZ at gw.sysnux.pf>
> Contact: <sip:63035284-ad7d-484f-8e54-f5ea54f39104 at
192.168.0.200:5060>
> Call-ID: 57283616-94e0-4052-bdff-491b31fdd229
> CSeq: 31693 INVITE
> Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL,
> UPDATE, PRACK, MESSAGE, REGISTER, REFER
> Supported: 100rel, timer, replaces, norefersub
> Session-Expires: 1800
> Min-SE: 90
> Max-Forwards: 70
> User-Agent: Asterisk GPL PBX
> Content-Type: application/sdp
> Content-Length: 238
>
> v=0
> o=- 1710591484 1710591484 IN IP4 192.168.0.200
> s=Asterisk
> c=IN IP4 192.168.0.200
> t=0 0
> m=audio 8834 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv
>
> <--- Received SIP response (585 bytes) from UDP:192.168.0.10:5060
--->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP
> 192.168.0.200:5060;branch=z9hG4bKPjba38816d-b5fe-4d5f-8bda-d0968e38e5f1;
> received=192.168.0.200;rport=5060
> From: "SysNux"
> <sip:+68940XXXXXX at
192.168.0.200>;tag=22d8369f-061d-4232-9c4d-5068e81bc5c
> 5
> To: <sip:40ZZZZZZ at gw.sysnux.pf>
> Call-ID: 57283616-94e0-4052-bdff-491b31fdd229
> CSeq: 31693 INVITE
> Server: Asterisk PBX 11.18.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Session-Expires: 1800;refresher=uas
> Contact: <sip:40ZZZZZZ at 192.168.0.10:5060>
> Content-Length: 0
>
>
> <--- Received SIP response (895 bytes) from UDP:192.168.0.10:5060
--->
> SIP/2.0 183 Session Progress
> Via: SIP/2.0/UDP
> 192.168.0.200:5060;branch=z9hG4bKPjba38816d-b5fe-4d5f-8bda-d0968e38e5f1;
> received=192.168.0.200;rport=5060
> From: "SysNux"
> <sip:+68940XXXXXX at
192.168.0.200>;tag=22d8369f-061d-4232-9c4d-5068e81bc5c
> 5
> To: <sip:40ZZZZZZ at gw.sysnux.pf>;tag=as7bba6b0d
> Call-ID: 57283616-94e0-4052-bdff-491b31fdd229
> CSeq: 31693 INVITE
> Server: Asterisk PBX 11.18.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Session-Expires: 1800;refresher=uas
> Contact: <sip:40ZZZZZZ at 192.168.0.10:5060>
> Content-Type: application/sdp
> Require: timer
> Content-Length: 236
>
> v=0
> o=root 2087714374 2087714374 IN IP4 192.168.0.10
> s=Asterisk PBX 11.18.0
> c=IN IP4 192.168.0.10
> t=0 0
> m=audio 16834 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
>
> -- PJSIP/t0gw-0000001a is making progress passing it to
> IAX2/iaxmodem0-7838
> <--- Received SIP response (601 bytes) from UDP:192.168.0.10:5060
--->
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP
> 192.168.0.200:5060;branch=z9hG4bKPjba38816d-b5fe-4d5f-8bda-d0968e38e5f1;
> received=192.168.0.200;rport=5060
> From: "SysNux"
> <sip:+68940XXXXXX at
192.168.0.200>;tag=22d8369f-061d-4232-9c4d-5068e81bc5c
> 5
> To: <sip:40ZZZZZZ at gw.sysnux.pf>;tag=as7bba6b0d
> Call-ID: 57283616-94e0-4052-bdff-491b31fdd229
> CSeq: 31693 INVITE
> Server: Asterisk PBX 11.18.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Session-Expires: 1800;refresher=uas
> Contact: <sip:40ZZZZZZ at 192.168.0.10:5060>
> Content-Length: 0
>
>
> -- PJSIP/t0gw-0000001a is ringing
> <--- Received SIP response (881 bytes) from UDP:192.168.0.10:5060
--->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 192.168.0.200:5060;branch=z9hG4bKPjba38816d-b5fe-4d5f-8bda-d0968e38e5f1;
> received=192.168.0.200;rport=5060
> From: "SysNux"
> <sip:+68940XXXXXX at
192.168.0.200>;tag=22d8369f-061d-4232-9c4d-5068e81bc5c
> 5
> To: <sip:40ZZZZZZ at gw.sysnux.pf>;tag=as7bba6b0d
> Call-ID: 57283616-94e0-4052-bdff-491b31fdd229
> CSeq: 31693 INVITE
> Server: Asterisk PBX 11.18.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Session-Expires: 1800;refresher=uas
> Contact: <sip:40ZZZZZZ at 192.168.0.10:5060>
> Content-Type: application/sdp
> Require: timer
> Content-Length: 236
>
> v=0
> o=root 2087714374 2087714374 IN IP4 192.168.0.10
> s=Asterisk PBX 11.18.0
> c=IN IP4 192.168.0.10
> t=0 0
> m=audio 16834 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
>
> <--- Transmitting SIP request (412 bytes) to UDP:192.168.0.10:5060
--->
> ACK sip:40ZZZZZZ at 192.168.0.10:5060 SIP/2.0
> Via: SIP/2.0/UDP
> 192.168.0.200:5060;rport;branch=z9hG4bKPj8504e505-1222-4747-955f-4788fef
> f58d1
> From: "SysNux"
> <sip:+68940XXXXXX at
192.168.0.200>;tag=22d8369f-061d-4232-9c4d-5068e81bc5c
> 5
> To: <sip:40ZZZZZZ at gw.sysnux.pf>;tag=as7bba6b0d
> Call-ID: 57283616-94e0-4052-bdff-491b31fdd229
> CSeq: 31693 ACK
> Max-Forwards: 70
> User-Agent: Asterisk GPL PBX
> Content-Length: 0
>
>
> -- PJSIP/t0gw-0000001a answered IAX2/iaxmodem0-7838
> -- Channel PJSIP/t0gw-0000001a joined 'simple_bridge'
basic-bridge
> <56a7726f-44a3-4df3-aee0-d21020aa5be1>
> -- Channel IAX2/iaxmodem0-7838 joined 'simple_bridge'
basic-bridge
> <56a7726f-44a3-4df3-aee0-d21020aa5be1>
>
> <--- Received SIP request (954 bytes) from UDP:192.168.0.10:5060 --->
> UPDATE sip:63035284-ad7d-484f-8e54-f5ea54f39104 at 192.168.0.200:5060 SIP/2
> .0
> Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK4fd84f17;rport
> Max-Forwards: 70
> From: <sip:40ZZZZZZ at gw.sysnux.pf>;tag=as7bba6b0d
> To: "SysNux"
> <sip:+68940XXXXXX at
192.168.0.200>;tag=22d8369f-061d-4232-9c4d-5068e81bc5c
> 5
> Contact: <sip:40ZZZZZZ at 192.168.0.10:5060>
> Call-ID: 57283616-94e0-4052-bdff-491b31fdd229
> CSeq: 102 UPDATE
> User-Agent: Asterisk PBX 11.18.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> X-asterisk-Info: SIP re-invite (External RTP bridge)
> Content-Type: application/sdp
> Content-Length: 287
>
> v=0
> o=root 2087714374 2087714375 IN IP4 192.168.0.10
> s=Asterisk PBX 11.18.0
> c=IN IP4 192.168.0.10
> t=0 0
> m=image 5720 udptl t38
> c=IN IP4 192.168.0.10
> a=T38FaxVersion:0
> a=T38MaxBitRate:14400
> a=T38FaxRateManagement:transferredTCF
> a=T38FaxMaxDatagram:849
> a=T38FaxUdpEC:t38UDPFEC
>
> <--- Transmitting SIP response (376 bytes) to UDP:192.168.0.10:5060
--->
> SIP/2.0 488 Not Acceptable Here
> Via: SIP/2.0/UDP
> 192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK4fd84f1
> 7
> Call-ID: 57283616-94e0-4052-bdff-491b31fdd229
> From: <sip:40ZZZZZZ at gw.sysnux.pf>;tag=as7bba6b0d
> To: "SysNux"
> <sip:+68940XXXXXX at
192.168.0.200>;tag=22d8369f-061d-4232-9c4d-5068e81bc5c
> 5
> CSeq: 102 UPDATE
> Server: Asterisk GPL PBX
> Content-Length: 0
>
>
>
> Is anyone successfully using chan_pjsip and iaxmodem?
>
>
> Thanks,
> - --
> Jean-Denis Girard
>
> SysNux Syst?mes Linux en Polyn?sie fran?aise
> http://www.sysnux.pf/ T?l: +689 40.50.10.40 / GSM: +689 87.79.75.27
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