[asterisk-users] T.38 Fax With Flowroute SIP Provider

Thu May 6 18:17:39 CDT 2010

I wasn't sure how the lines were counted. Here is the debug output
from Asterisk where it is building the invite packet. I looked at the
a=T38 lines and nothing is standing out to me.

Ryan

[May  6 13:29:05] DEBUG[32389] chan_sip.c:  Header  0 [ 47]: INVITE
sip:+number at x.x.x.x:5060 SIP/2.0
[May  6 13:29:05] DEBUG[32389] chan_sip.c:  Header  1 [ 63]: Via:
SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK2837f4cf;rport
[May  6 13:29:05] DEBUG[32389] chan_sip.c:  Header  2 [ 54]: Route:
<sip:x.x.x.x;lr>,<sip:x.x.x.x;lr>
[May  6 13:29:05] DEBUG[32389] chan_sip.c:  Header  3 [ 16]: Max-Forwards: 70
[May  6 13:29:05] DEBUG[32389] chan_sip.c:  Header  4 [ 59]: From:
<sip:+number at x.x.x.x:5060>;tag=as7d21d6f3
[May  6 13:29:05] DEBUG[32389] chan_sip.c:  Header  5 [ 53]: To:
<sip:+number at x.x.x.x:5060>;tag=gK0d4c48f7
[May  6 13:29:05] DEBUG[32389] chan_sip.c:  Header  6 [ 39]: Contact:
<sip:number at x.x.x.x>
[May  6 13:29:05] DEBUG[32389] chan_sip.c:  Header  7 [ 39]: Call-ID:
302861516_123483666 at x.x.x.x
[May  6 13:29:05] DEBUG[32389] chan_sip.c:  Header  8 [ 16]: CSeq: 102 INVITE
[May  6 13:29:05] DEBUG[32389] chan_sip.c:  Header  9 [ 36]:
User-Agent: Asterisk PBX 1.6.2.7-rc3
[May  6 13:29:05] DEBUG[32389] chan_sip.c:  Header 10 [ 72]: Allow:
INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
[May  6 13:29:05] DEBUG[32389] chan_sip.c:  Header 11 [ 26]:
Supported: replaces, timer
[May  6 13:29:05] DEBUG[32389] chan_sip.c:  Header 12 [ 52]:
X-asterisk-Info: SIP re-invite (External RTP bridge)
[May  6 13:29:05] DEBUG[32389] chan_sip.c:  Header 13 [ 29]:
Content-Type: application/sdp
[May  6 13:29:05] DEBUG[32389] chan_sip.c:  Header 14 [ 19]: Content-Length: 293
[May  6 13:29:05] DEBUG[32389] chan_sip.c:  Header 15 [  0]:
[May  6 13:29:05] DEBUG[32389] chan_sip.c:    Body  0 [  3]: v=0
[May  6 13:29:05] DEBUG[32389] chan_sip.c:    Body  1 [ 48]: o=root
2048302926 2048302927 IN IP4 x.x.x.x
[May  6 13:29:05] DEBUG[32389] chan_sip.c:    Body  2 [ 26]:
s=Asterisk PBX 1.6.2.7-rc3
[May  6 13:29:05] DEBUG[32389] chan_sip.c:    Body  3 [ 21]: c=IN IP4 x.x.x.x
[May  6 13:29:05] DEBUG[32389] chan_sip.c:    Body  4 [  5]: t=0 0
[May  6 13:29:05] DEBUG[32389] chan_sip.c:    Body  5 [ 22]: m=image
4575 udptl t38
[May  6 13:29:05] DEBUG[32389] chan_sip.c:    Body  6 [ 17]: a=T38FaxVersion:0
[May  6 13:29:05] DEBUG[32389] chan_sip.c:    Body  7 [ 21]:
a=T38MaxBitRate:14400
[May  6 13:29:05] DEBUG[32389] chan_sip.c:    Body  8 [ 22]:
a=T38FaxFillBitRemoval
[May  6 13:29:05] DEBUG[32389] chan_sip.c:    Body  9 [ 37]:
a=T38FaxRateManagement:transferredTCF
[May  6 13:29:05] DEBUG[32389] chan_sip.c:    Body 10 [ 24]:
a=T38FaxMaxDatagram:1400
[May  6 13:29:05] DEBUG[32389] chan_sip.c:    Body 11 [ 23]:
a=T38FaxUdpEC:t38UDPFEC

On Thu, May 6, 2010 at 6:54 PM, Kevin P. Fleming <kpfleming at digium.com> wrote:
> On 05/06/2010 05:46 PM, Ryan Wagoner wrote:
>> Does anybody have T.38 faxing working with Flowroute? I am running
>> Asterisk 1.6.2.7-rc3 with spandsp 0.0.6-pre17. I can successfully
>> receive faxes over ulaw. I enabled T.38 with t38pt_udptl=yes in
>> sip.conf. When I receive a fax it tries to negotiate T.38 and
>> Flowroute sends back a Bad Request response saying I have a SIP syntax
>> error.
>>
>> Flowroute support is recommending that I try again after removing
>> externip and localnet from sip.conf. They state that their service
>> will recognize the private IP and rewrite the SIP packets. However
>> this is going to cause issues for my remote SIP phones.
>>
>> Thanks,
>> Ryan
>>
>> DEBUG[32389] app_fax.c: Negotiating T.38 for receive on SIP/flowroute-00000000
>>
>> INVITE sip:+number at xx.xx.xx.xx:5060 SIP/2.0
>> ...
>> CSeq: 102 INVITE
>> User-Agent: Asterisk PBX 1.6.2.7-rc3
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
>> Supported: replaces, timer
>> X-asterisk-Info: SIP re-invite (External RTP bridge)
>> Content-Type: application/sdp
>> Content-Length: 293
>>
>> v=0
>> o=root 2048302926 2048302927 IN IP4 xx.xx.xx.xx
>> s=Asterisk PBX 1.6.2.7-rc3
>> c=IN IP4 xx.xx.xx.xx
>> t=0 0
>> m=image 4575 udptl t38
>> a=T38FaxVersion:0
>> a=T38MaxBitRate:14400
>> a=T38FaxFillBitRemoval
>> a=T38FaxRateManagement:transferredTCF
>> a=T38FaxMaxDatagram:1400
>> a=T38FaxUdpEC:t38UDPFEC
>>
>> SIP/2.0 400 Bad Request
>> ...
>> CSeq: 102 INVITE
>> Error-Info: <sip:+number at xx.xx.xx.xx>;cause="[line 023] SIP syntax error"
>> Content-Length: 0
>
> Which line is 'line 23' of the T.38 re-INVITE?
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> skype: kpfleming | jabber: kfleming at digium.com
> Check us out at www.digium.com & www.asterisk.org
>
> --
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