01-07-2014 08:11 AM - edited 03-16-2019 09:08 PM
We have just received a Xerox Workcentre 5330 with SIP capabilities. We have SIP up and working and the printer is authenticated back to the CUCM. We are also able to receive faxes just fine but we're unable to send. I see the call hitting the router and the remote fax machine actually rings, but after it rings and starts to process the call, the call just ends and the fax is aborted. Any suggestions on what would be causing this?
Thanks
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01-15-2014 06:02 AM
Joseph,
I don't believe that I've ever deployed something similar, but it still sounds like an MTP issue to me. Not necessarily an MTP issue per se, but an issue that can be solved with an MTP. H323 has similar constructs to SIP in terms of fast start vs early offer and slow start vs delayed offer --- at least from the depths of my memory. Do you have MTP required checked on your h323 gateway configs? Again, that would be a really quick test to confirm the theory. If that does work, then there might be a more optimal ios-based solution.
hope that helps,
will
edit: I'd still run some h323 debugs at your gateway. I don't find them as easy to read as SIP messaging, but if you do it at an off-peak time and there aren't any other calls it's not too bad.
01-07-2014 09:20 AM
Codec issues. Which Codec are you using?
Sent from Cisco Technical Support Android App
01-08-2014 12:17 PM
The location that the device is setup in is using G711ulaw. Also, on the gateway the dialpeer is set to G711ulaw. I made another test call out to my cell phone and when I pick up I hear the fax machine beeping. Just intermittent beeps, not the full modem sound that you normally get. I heard this for about 60 minutes straight before the call dropped. Should I be hearing even a tone if it were a codec issue?
01-09-2014 05:26 AM
Hi,
Just to query your config, does it look something like this?
voice service voip
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
h323
sip
!
dial-peer voice 100 voip
description ## for right fax ###
destination-pattern 1448
session target ipv4:xxx.xxx.xxx.xxx
dtmf-relay h245-alphanumeric
codec g711(a/u)law
no vad
ial-peer voice 2470 voip
description # Xerox fax printer #####
destination-pattern 2470
session target ipv4:xxx.xxx.xxx.xxx
session transport udp
dtmf-relay h245-alphanumeric
codec g711(a/u)law
fax rate 14400
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
01-09-2014 06:06 AM
Here is the config in my router---noticed I'm missing a few things from what you've listed.
!
voice service voip
signaling forward unconditional<----------does this need to go
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
fax-relay ans-disable<---------------does this need to go
h323
(sip)<---------does this need to be added
!
Excuse my ignorance also, what do you mean by "right fax"
Below is the dial-peer config for the Fax machine
dial-peer voice 6801 voip
description Xerox
destination-pattern 6801
session target ipv4:10.101.0.4
codec g711ulaw
no vad
I can add the options you've listed and give it a try.
01-09-2014 11:17 PM
sorry right fax is a 3rd party sip faxing server.
01-09-2014 11:23 PM
Why don't you point this dial-peer straight to the SIP fax machine and cut out the H323 connection to the cucm?
Also maybe have a look at the following thread:
https://supportforums.cisco.com/thread/2143147
Below is the dial-peer config for the Fax machine
dial-peer voice 6801 voip
description Xerox
destination-pattern 6801
session target ipv4:10.101.0.4
codec g711ulaw
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip pass-thru content sdp
dtmf-relay rtp-nte
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
no vad
no vad
01-09-2014 06:23 AM
Before trying the options you suggested I did a show voice call status and show voip rtp connections and got the following.
Andy_Fiber_2800#sho voice call status
CallID CID ccVdb Port Slot/DSP:Ch Called # Codec MLPP Dia
l-peers
0x5A9B 2B3D 0x4889BDE8 0/0/0:23.23 0/2:1 *2564929109 g711ulaw 67
50/9
1 active call found
--
Andy_Fiber_2800#sho voip rtp connections
VoIP RTP active connections :
No. CallId dstCallId LocalRTP RmtRTP LocalIP
RemoteIP
1 23194 23195 16842 5004 10.101.255.254
10.1.15.123
Found 1 active RTP connections
===========================================
I added your suggestions and still had the same issues. Fax is being aborted.
01-09-2014 09:32 AM
I don't think that there's a default session protocol, so always define it under each voip peer as follows:
session protocol sipv2
But that's probably not the issue. If you can receive faxes but not send, it sounds like an SDP/delayed offer issue. Are you using an MTP on the UCM SIP trunk to your gateway?
As for right fax - it's an ip fax product.
edit: You can confirm by enabling sip debugging (just messages is fine). Check the invite from UCM for an SDP section. My guess is you're doing delayed offer from UCM to router and also on gateway to PSTN. If so, you can check MTP on your UCM trunk or enable early offer at your CUBE.
01-09-2014 09:51 AM
We do not have a SIP trunk back to the gateway from the CUCM. It connects H.323. On the H.323 connection we have a media resource group list specified but there is nothing associated with it and no DSP resources available on the router to setup MTP or Transcoding on the router itself.
01-09-2014 10:00 AM
So the fax machine is a 3rd party sip device on CUCM? H.323 between CUCM and CUBE and ISDN to PSTN?
01-09-2014 10:02 AM
Yes that is correct:
Simple Diagram:
PSTN<-PRI->Cisco 2821<-H.323->CUCM<-SIP->3rd Party SIP fax machine
01-15-2014 04:03 AM
Does anyone have any other suggestions?
01-15-2014 06:02 AM
Joseph,
I don't believe that I've ever deployed something similar, but it still sounds like an MTP issue to me. Not necessarily an MTP issue per se, but an issue that can be solved with an MTP. H323 has similar constructs to SIP in terms of fast start vs early offer and slow start vs delayed offer --- at least from the depths of my memory. Do you have MTP required checked on your h323 gateway configs? Again, that would be a really quick test to confirm the theory. If that does work, then there might be a more optimal ios-based solution.
hope that helps,
will
edit: I'd still run some h323 debugs at your gateway. I don't find them as easy to read as SIP messaging, but if you do it at an off-peak time and there aren't any other calls it's not too bad.
01-18-2014 10:51 AM
I'm going to be onsite this week and will look into this and let you know what I find out. Thanks for all of your help.
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