05-15-2009 03:55 AM - edited 03-15-2019 06:02 PM
I have problem with fax if it goes via SIP-to-SIP gateway.
The configuration is:
StoneFax <-H323-> CUCM6.1 <-SIP-> IOSGW <-SIP-> PSTN
The problem is: If PSTN gets SIP Invite message with âMedia Description, name and address (m): image 17856 udptl 101 t38â, then PSTN sends t38 messages that the Fax server cannot decode.
If I disable DTMF-relay in dial-peer configuration, then I can send and receive fax, but I do not use IVR function from outside. (âMedia Description, name and address (m): image 17948 udptl t38â)
Therefore I would like to remove only the media format 101 from the media type image in IOSGW.
How can I do this?
Thanks in advance,
Laszlo
05-15-2009 04:46 AM
Hi Laszlo,
This is precisely what SIP profiles are for. By default on the router it will not do any policy like conditionally advertising DTMF.
Here's the link for SIP profiles:
And this is what I'm guessing you could do:
voice class sip-profiles 100
request REINVITE sdp-header Audio-Media modify "udptl 101 t38" "udptl t38"
You may need to clean up the a= attribute line for it also, but that's trickier.
hth,
nick
05-15-2009 05:07 AM
Dear Nick,
I have tried it, and did not work.
This is the INVITE message that I want to change:
000535: May 15 14:50:38.835 CEST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:x.x.x.1:5060 SIP/2.0
Date: Fri, 15 May 2009 12:50:38 GMT
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
From: <3618148299>;tag=1FFE708-16B43618148299>
Allow-Events: telephone-event
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Remote-Party-ID: "StoneFax" <1999>;party=calling;screen=no;privacy=off1999>
Cisco-Guid: 3461229274-1082462686-2163723200-4041831498
Timestamp: 1242391838
Content-Length: 458
User-Agent: Cisco-SIPGateway/IOS-12.x
To: <>3613712722@pstn.hu;user=phone>;tag=19042-NZ-02ba4912-6190b9a62>
Contact: <3618148299>3618148299>
Expires: 180
Content-Disposition: session;handling=required
Content-Type: application/sdp
Call-ID: 19042-VP-02ba4911-6299685e1@pstn.hu
Via: SIP/2.0/UDP 81.160.133.217:5060;branch=z9hG4bK127207F
CSeq: 101 INVITE
Max-Forwards: 70
v=0
o=CiscoSystemsSIP-GW-UserAgent 5820 6480 IN IP4 x.x.x.217
s=SIP Call
c=IN IP4 x.x.x.217
t=0 0
m=image 19546 udptl 101 t38
c=IN IP4 x.x.x.217
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
I have some problem with SIP-profile:
This is an INVITE message instead of REINVITE. (I can set the change in profile to INVITE too but did not work)
The main problem is that the media is "image" instead of "audio". I cannot change parameter of image by SIP-profile. The SIP-profile is good for media of "audio" or "video".
Have you any idea?
Thanks,
Laszlo
05-15-2009 07:11 AM
Hi Laszlo,
These have to be applied to dial peers, I may have not made that as clear in my example.
I was also concerned with the lack of an option to change 'image', but I think it should match as audio.
Make sure the dial peer pointing towards CUCM that is matched has this applied:
voice-class sip profile 100
-nick
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