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@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@fax-sortant:1] NoOp("IAX2/iaxmodem0-7838", "
calls 40ZZZZZZ (local)") in new stack
-- Executing [40ZZZZZZ@fax-sortant:2] Set("IAX2/iaxmodem0-7838",
"FAXOPT(gateway)=yes") in new stack
-- Executing [40ZZZZZZ@fax-sortant:3] Dial("IAX2/iaxmodem0-7838",
"PJSIP/40ZZZZZZ@t0gw") in new stack
-- Called PJSIP/40ZZZZZZ@t0gw
<--- Transmitting SIP request (936 bytes) to UDP:192.168.0.10:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP
192.168.0.200:5060;rport;branch=z9hG4bKPjba38816d-b5fe-4d5f-8bda-d0968e3
8e5f1
From: "SysNux"
<sip:[email protected]>;tag=22d8369f-061d-4232-9c4d-5068e81bc5c
5
To: <sip:[email protected]>
Contact: <sip:[email protected]: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:[email protected]>;tag=22d8369f-061d-4232-9c4d-5068e81bc5c
5
To: <sip:[email protected]>
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:[email protected]: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:[email protected]>;tag=22d8369f-061d-4232-9c4d-5068e81bc5c
5
To: <sip:[email protected]>;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:[email protected]: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:[email protected]>;tag=22d8369f-061d-4232-9c4d-5068e81bc5c
5
To: <sip:[email protected]>;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:[email protected]: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:[email protected]>;tag=22d8369f-061d-4232-9c4d-5068e81bc5c
5
To: <sip:[email protected]>;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:[email protected]: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:[email protected]: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:[email protected]>;tag=22d8369f-061d-4232-9c4d-5068e81bc5c
5
To: <sip:[email protected]>;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:[email protected]:5060 SIP/2
.0
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK4fd84f17;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as7bba6b0d
To: "SysNux"
<sip:[email protected]>;tag=22d8369f-061d-4232-9c4d-5068e81bc5c
5
Contact: <sip:[email protected]: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:[email protected]>;tag=as7bba6b0d
To: "SysNux"
<sip:[email protected]>;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 Systèmes Linux en Polynésie française
http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27
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