Yes, I just did, but same result

After the message "WARNING[1094003008]: app_t38gateway.c:396
t38gateway_exec: Bridging frames" the audio becomes silence.


    -- Executing SipT38SwitchOver("IAX2/216.155.127.26:4569-5", "") in new stack
Feb  5 04:48:24 WARNING[1094003008]: chan_sip.c:16651
sip_t38switchover: This function can only be used on SIP channels.
    -- Executing T38Gateway("IAX2/216.155.127.26:4569-5",
"SIP/00908F03206D-L1|90|r") in new stack
We're at 216.155.127.30 port 16340
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x1 (g723) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726) to SDP
Adding codec 0x20 (dvi) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x400 (ilbc) to SDP
13 headers, 20 lines
Reliably Transmitting (NAT) to 64.238.173.171:5060:
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 216.155.127.30:5060;branch=z9hG4bK744d87f6;rport
From: "openpbx" <sip:[EMAIL PROTECTED]>;tag=as4b0909e3
To: <sip:[EMAIL PROTECTED]:5060>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: OpenPBX
Max-Forwards: 70
Date: Mon, 05 Feb 2007 09:48:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 467

v=0
o=root 10100 10100 IN IP4 216.155.127.30
s=session
c=IN IP4 216.155.127.30
t=0 0
m=audio 16340 RTP/AVP 0 122 4 3 8 111 5 10 7 18 110 97
a=rtpmap:0 PCMU/8000
a=rtpmap:122 t38/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=silenceSupp:off - - - -

---
Retransmitting #1 (NAT) to 64.238.173.171:5060:
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 216.155.127.30:5060;branch=z9hG4bK744d87f6;rport
From: "openpbx" <sip:[EMAIL PROTECTED]>;tag=as4b0909e3
To: <sip:[EMAIL PROTECTED]:5060>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: OpenPBX
Max-Forwards: 70
Date: Mon, 05 Feb 2007 09:48:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 467

v=0
o=root 10100 10100 IN IP4 216.155.127.30
s=session
c=IN IP4 216.155.127.30
t=0 0
m=audio 16340 RTP/AVP 0 122 4 3 8 111 5 10 7 18 110 97
a=rtpmap:0 PCMU/8000
a=rtpmap:122 t38/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=silenceSupp:off - - - -

---
web04*CLI>
<-- SIP read from 64.238.173.171:5060:
SIP/2.0 180 Ringing
From: "openpbx"<sip:[EMAIL PROTECTED]>;tag=as4b0909e3
To: <sip:[EMAIL 
PROTECTED]:5060>;tag=100c3250-24fba8c0-13c4-45c6fc10-5fa5a19a-45c6fc10
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 216.155.127.30:5060;rport=5060;branch=z9hG4bK744d87f6
Supported: replaces
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL
Contact: <sip:[EMAIL PROTECTED]:5060>
Content-Length: 0
web04*CLI>

--- (10 headers 0 lines) ---
web04*CLI>
<-- SIP read from 64.238.173.171:5060:
SIP/2.0 180 Ringing
From: "openpbx"<sip:[EMAIL PROTECTED]>;tag=as4b0909e3
To: <sip:[EMAIL 
PROTECTED]:5060>;tag=100c3250-24fba8c0-13c4-45c6fc10-5fa5a19a-45c6fc10
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 216.155.127.30:5060;rport=5060;branch=z9hG4bK744d87f6
Supported: replaces
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL
Contact: <sip:[EMAIL PROTECTED]:5060>
Content-Length: 0
web04*CLI>

--- (10 headers 0 lines) ---
web04*CLI>
<-- SIP read from 64.238.173.171:5060:
SIP/2.0 200 OK
From: "openpbx"<sip:[EMAIL PROTECTED]>;tag=as4b0909e3
To: <sip:[EMAIL 
PROTECTED]:5060>;tag=100c3250-24fba8c0-13c4-45c6fc10-5fa5a19a-45c6fc10
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 216.155.127.30:5060;rport=5060;branch=z9hG4bK744d87f6
Supported: replaces
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL
Contact: <sip:[EMAIL PROTECTED]:5060>
Content-Type: application/sdp
Content-Length: 259

v=0
o=rtp 1170669305 1170669305 IN IP4 192.168.251.36
s=-
c=IN IP4 64.238.173.171
t=0 0
m=audio 5004 RTP/AVP 0 0 0 0 0 0
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000

--- (11 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 0
Found RTP audio format 0
Found RTP audio format 0
Found RTP audio format 0
Found RTP audio format 0
Peer audio RTP is at port 64.238.173.171:5004
Found description format PCMU
Found description format PCMU
Found description format PCMU
Found description format PCMU
Found description format PCMU
Found description format PCMU
Capabilities: us - 0x3f07ff
(g723|gsm|ulaw|alaw|g726|dvi|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263|h263p|h264),
peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0
(nothing), combined - 0x0 (nothing)
list_route: hop: <sip:[EMAIL PROTECTED]:5060>
set_destination: Parsing <sip:[EMAIL PROTECTED]:5060> for
address/port to send to
set_destination: set destination to 64.238.173.171, port 5060
Transmitting (NAT) to 64.238.173.171:5060:
ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 216.155.127.30:5060;branch=z9hG4bK4b0cea50;rport
From: "openpbx" <sip:[EMAIL PROTECTED]>;tag=as4b0909e3
To: <sip:[EMAIL 
PROTECTED]:5060>;tag=100c3250-24fba8c0-13c4-45c6fc10-5fa5a19a-45c6fc10
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: OpenPBX
Max-Forwards: 70
Content-Length: 0


---
Feb  5 04:48:26 WARNING[1094003008]: app_t38gateway.c:396
t38gateway_exec: Bridging frames
Scheduling destruction of call
'[EMAIL PROTECTED]' in 32000 ms
set_destination: Parsing <sip:[EMAIL PROTECTED]:5060> for
address/port to send to
set_destination: set destination to 64.238.173.171, port 5060
Reliably Transmitting (NAT) to 64.238.173.171:5060:
BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 216.155.127.30:5060;branch=z9hG4bK35072bde;rport
From: "openpbx" <sip:[EMAIL PROTECTED]>;tag=as4b0909e3
To: <sip:[EMAIL 
PROTECTED]:5060>;tag=100c3250-24fba8c0-13c4-45c6fc10-5fa5a19a-45c6fc10
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 103 BYE
User-Agent: OpenPBX
Max-Forwards: 70
Content-Length: 0


---
    -- Hungup 'IAX2/216.155.127.26:4569-5'


On 2/5/07, Ivan C Myrvold <[EMAIL PROTECTED]> wrote:
> Have you tried with 9600 instead of 14400?
>
> Ivan
>
> Den 5. feb. 2007 kl. 09.50 skrev Andrew Joakimsen:
>
> > Do you use T38Gateway? I've tried SIP (T.38) -> OpenPBX -> IAX2 ->
> > Asterisk -> PRI
> >
> > PRI -> Asterisk -> IAX -> OpenPBX -> SIP T.38
> >
> > And it never works.
> >
> > I see with "sip debug" that the ATA is sending a T38 request, but
> > nothing back from OpenPBX the call just proceeds as Ulaw, the
> > T38SwitchOver doesnt help either.  Is there any speed limiation? The
> > ATA is requesting 14,400kbps.
> >
> > Also I am wondering if OpenPBX is 100% compatible with Asterisk
> > configuration files, and if there is any plan to change that?
> >
> > On 2/5/07, Checkov, Andrew <[EMAIL PROTECTED]> wrote:
> >
> >> I have stable T.38 operation between any pair of devices. All of them
> >> (except) Cisco works thru the NAT with disabled Reinvite.
> > _______________________________________________
> > Openpbx-users mailing list
> > [email protected]
> > http://lists.openpbx.org/mailman/listinfo/openpbx-users
> >
>
> _______________________________________________
> Openpbx-users mailing list
> [email protected]
> http://lists.openpbx.org/mailman/listinfo/openpbx-users
>
_______________________________________________
Openpbx-users mailing list
[email protected]
http://lists.openpbx.org/mailman/listinfo/openpbx-users

Reply via email to