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

ashbandhujha
Beginner
Beginner

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.          

4 Accepted Solutions

Accepted Solutions

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

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Where are the other two debugs I asked for?

debug voip vtsp all

debug fax relay t30 all-level-1

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

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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

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