ATA 187 Fax is failing with CUCM 8.5 & 3925 SIP Gateway.

[ashbandhujha]

Dear All,

Here is the call flow for failed fax :

ATA 187---------- CUCM 8.5 ----------3925 SIP Gateway------------ Huawei softx3000 SIP provider

FAX mode is ATA is Fax pass through.

Fax is failing with strange RTP payload Type Behaviour.

RTP pt vary from 97 ,101 ,127 etc.

attached debug file.          

[Ayodeji Okanlawon]

Hi,

first of all what is the calling and called number?

Secondly, rtp pt 97 and 101 are dtmf-relay type..Not fax. Huwaei always uses pt 97 for rtp-nte while cisco and most others user pt 101. So those telephone-events are not fax related.

Thirdly,

From your logs I see the ff: SDP attributes from Huwaei

v=0

o=HuaweiSoftX3000 1139083 1139083 IN IP4 10.209.4.58

s=Sip Call

c=IN IP4 10.209.4.58

t=0 0

m=audio 28154 RTP/AVP 8 97

a=rtpmap:8 PCMA/8000

a=rtpmap:97 telephone-event/8000

a=fmtp:97 0-15

As you can see RTP/AVP 8= G711alaw codec and RTP/AVP 97= dtmf pt=97 and the supported dtmf digits are 0-15

Just a few things to think about..You can send me calling and called number...

send a new log with debug ccsip all

[ashbandhujha]

Thank you.

SIP Gateway is in remote site.

As soon as i will get debug , i will post it .

[ashbandhujha]

Hi

Here is the debug file attached with show config also.

fax calling number: 2279179

fax called number:022871188

CUCM:                    10.116.44.10

SIP GW:                 10.196.101.198

SIP Provider server:  10.209.4.58

[Ayodeji Okanlawon · Accepted solution]

Ashish,

Sorry for the delay or late response. Been busy..

I have looked at the traces and here is what I found..

It looks as though your SIP provider wants to use G711ulaw for fax-passthrough and you have configured G711alaw.

Can you try and configure g711ulaw for fax pasthrouhg and also use g711ulaw on your voip dial-peers and test again...

May 31 12:21:27.697: //81138/75AD58800000/SIP/Info/sip_iwf_sip_copy_sdp_to_channelInfo: Selecting codec g711ulaw
May 31 12:21:27.697: //81138/75AD58800000/SIP/Info/codec_found:
Codec to be matched: 5
May 31 12:21:27.697: //81138/75AD58800000/SIP/Info/codec_found: No match for the codecs found..
May 31 12:21:27.697: //81138/75AD58800000/SIP/Info/sip_iwf_sip_copy_sdp_to_channelInfo: ADD AUDIO CODEC 5

May 31 12:21:27.697: //81138/75AD58800000/SIP/Info/sipSPI_ipip_update_codec_params_in_channelInfo:
May 31 12:21:27.697: //81138/75AD58800000/SIP/Info/sipSPI_ipip_update_codec_params_in_channelInfo: Parsing from stream media address
May 31 12:21:27.697: //81138/75AD58800000/SIP/Info/sipSPI_ipip_GetCodecBytePtimeFromSdp:
May 31 12:21:27.697: //-1/xxxxxxxxxxxx/SIP/Info/convert_ptime_to_codec_bytes: Values :Codec: g711ulaw ptime :20, codecbytes: 160
May 31 12:21:27.697: //81138/75AD58800000/SIP/Info/sipSPI_ipip_update_codec_params_in_channelInfo:
Adding negotiated codec 5 ptype 0 time 20,   bytes 160 as channel 1 mline 1 ss 0 10.209.4.58:24388
May 31 12:21:27.697: //81138/75AD58800000/SIP/Info/sipSPIUpdateCallEntry:
Call 81138 set InfoType to FAX

[Ayodeji Okanlawon]

Once you have made the change can and if it doesnt work can  you try sending the ff: debug

debug voip vtsp all

debug fax relay t30 all-level-1

debug ccsip all

[Ayodeji Okanlawon]

Ashish,

How are you progressing with this. Di you see my last post? Let me know if you still need help

[ashbandhujha]

Thank you Aokanlawon

i will do the necessary and see it through.

soon update you.

[ashbandhujha]

            Hi

Thank you , sorry for delay.

Here is the debug file you told.

we configured g711ulaw for the dial-peer as well as in fax it is using NSE fax-pass through g711ulaw.

failed fax calling number: 2279107

              called number: 2833373     

[Ayodeji Okanlawon · Accepted solution]

Where are the other two debugs I asked for?

debug voip vtsp all

debug fax relay t30 all-level-1

Please rate useful posts

[Ayodeji Okanlawon · Accepted solution]

The sip debug is incomplete. Pls send the full sip debug...

Please rate useful posts

[ashbandhujha]

Dear  Aokanlawon,

i vl resend all the debug soon.its live network so the hassle.

Thank you.

[ashbandhujha]

                   Here is the debug . there is delay due to remote.

                   but in debug negotiated codec is still g711alaw. 

[Ayodeji Okanlawon · Accepted solution]

Asish,

I am going to be honest with you. I am not happy with the way you are responding on this issue. I am trying to help but you are not helping me.

You sent me a log with totally different numbers..

your vtsp log has the ff: calling and called number

Calling Number=2279281,

   Called Number=238301

Your sip logs has the ff: 

calling number: 2279107

called number: 2833373     

I am not sure why you dont seem to want to resolve this problem...Maybe its not important to you but I am speding lots of time trying to help you...Pls le tme know if this is not important and then I can spend my sundays looking at other things...

Please if you are interested in resolving this issue then please provide accurate information....

Now lets start again assuming you are interetsed in solving it...

1. Where is your fax machine connected to? Is it to the ATA?

2. Can you connect an analogue phone to the ATA and make a phone call?

3. Can you send me an update configuration of your router. The sip log you sent to me does not show that the call is going to cucm. Is this fax machine now on your CUBE gateway?

4. IS your ATA configured for Fax passthrough?

5. Is your ATA running the latest firmware (9.2.3 firmware)

+++Please do the following ++++++

1. Configure your ATA in CUCM for fax-passthrough by setting  the fax mode under the ATA device level to "Fax Pass-Through".

2. configure

"voice service voip

fax protocol none"

3. On your dial-peers

remove "fax protocol pass-through g711alaw"

4. Configure region between your ATA and CUBE gateway to Use G711 codec.  You can do this by assigning the same device pool to the gateway and ATA or use different device pool and put them in same region

5.finally ensure your ATA is using  latest firmware 9.2.1 0r 9.2.3

6. try and use codec g711ulaw on your dial-peers..

7. The ATA 187 does not support NSE -based modem passthrough. So please remove this configuration if you have it

Test again...

send the ff:

debug voip vtsp all

debug fax relay t30 all-level-1

debug ccsip all

debug voip ccapi inout

Please rate useful posts

[ashbandhujha]

Dear Aokanlawon,

Grateful to you for your valauble time.

This issue is prime importance to us.

The SIP Gateways and ATA fax all are in remote site.

  During vtsp debug of failed fax, no debug message coming in term monitor.

  but after 5 minute some debug message  came,but it was for another call.sory for confusion.

  so i repeat, during fax no vtsp & fax relay t30 debug message is coming in my terminal monitor.

  may be no fax event is happening in gateawy.its only telephony event.

  i will do the the steps , you said. & update you.

Thank you

Ashish