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calls from UK PSTN to CUE fails

Chris Drew
Level 1
Level 1

I am getting disconnected when i forward calls from the pstn to CUE. Works fine internally but from the PSTN the call fails. I think this may be a trasncoding issue as the CUE can only talk g711ulaw but i am not sure where or how to verify this My CUE is in the same device as the PSTN GW a E1 PRI. TIA

24 Replies 24

What dial peers are you using on your router? Do any of them use G711alaw?

-nick

my voip dial peers to the ccm have a codec class with 711alaw yes. thanks

Should be G.711u, even if you are in UK, that the "internal standard.

However the CUE isse is likely due to something else at DP level.

sorry. what is the DP level. thanks

dial-peer, somehow the call comes in and is not sent to the CME correctly.

Hello everybody,

any results on this problem?

I am having the same issue, calls from PSTN do not make it to the CUE when busy/NoAnsw, the call just ends (they are redirected):

ephone-dn  8  dual-line
 number xxx
 label xxx-xxx
 name 3xxx
 call-forward busy 3199 (Voice mailbox pilot number)
 call-forward noan 3199 timeout 20

!

Instead, if I call from an IP phone registered to the CME to another colleague's phone (also registered) and he/she is busy/NoAnsw the call goes to CUE directly without problem and I can leave a message.

After doing several debugs I also think it is a transcode problem, but I do not know how to resolve it.

I thought of using something like this but I read it should only be used as last-resort method:

dspfarm profile 1 transcode universal
 codec g729abr8
 codec g729ar8
 codec g711alaw
 codec g711ulaw
 codec g729r8
 codec g729br8
 maximum sessions 5
 associate application SCCP

!

Here my dial-peer for voice mailbox:

dial-peer voice 91 voip
 description Unity Express Voice Mailbox pilot number
 destination-pattern 91
 session protocol sipv2
 session target ipv4:A.B.C.D !!--> IP@ ISM0/0 CUE
 dtmf-relay sip-notify
 codec g711ulaw
 no vad

 

When debugging ccapi I see how the call is redirected to 91.

To compare the calls making it to CUE from those ones that do not, I run a debug and, until the point I am showing to you, both debugs are more or less the same but in this point the debug (calling from PSTN) displays this ccSetMediaclass, ccGet... messages that I do not know what they mean whereas the debug from calls within the network do not show this kind of messages.

*Dec  5 14:04:25.064:  feature call forward featname is 3
*Dec  5 14:04:25.064: //557/86D8E9B78B8B/CCAPI/ccGetMediaClassTag:
   media class tag 0
*Dec  5 14:04:25.064: //557/86D8E9B78B8B/CCAPI/ccSetMediaclassIp2ipTags:
   media class tags set: NR 0, ASP 0
*Dec  5 14:04:25.064: //555/86D8E9B78B8B/CCAPI/ccGetMediaClassTag:
   media class tag 0
*Dec  5 14:04:25.064: //555/86D8E9B78B8B/CCAPI/ccSetMediaclassIp2ipTags:
   media class tags set: NR 0, ASP 0
*Dec  5 14:04:25.064: //557/86D8E9B78B8B/CCAPI/ccGet_xc_nr_asp_info:
   media class tags: NR 0, ASP 0
*Dec  5 14:04:25.064: //555/86D8E9B78B8B/CCAPI/ccGet_xc_nr_asp_info:
   media class tags: NR 0, ASP 0
*Dec  5 14:04:25.064: //557/86D8E9B78B8B/CCAPI/ccGetMediaClassTag:
   media class tag 0
*Dec  5 14:04:25.064: //557/86D8E9B78B8B/CCAPI/ccSetMediaclassIp2ipTags:
   media class tags set: NR 0, ASP 0
*Dec  5 14:04:25.064: //555/86D8E9B78B8B/CCAPI/ccGetMediaClassTag:
   media class tag 0
*Dec  5 14:04:25.064: //555/86D8E9B78B8B/CCAPI/ccSetMediaclassIp2ipTags:
   media class tags set: NR 0, ASP 0
*Dec  5 14:04:25.064: //557/86D8E9B78B8B/CCAPI/ccGet_xc_nr_asp_info:
   media class tags: NR 0, ASP 0
*Dec  5 14:04:25.064: //555/86D8E9B78B8B/CCAPI/ccGet_xc_nr_asp_info:
   media class tags: NR 0, ASP 0
*Dec  5 14:04:25.064: //557/86D8E9B78B8B/CCAPI/cc_api_call_disconnected:
   Cause Value=47, Interface=0x3D82B49C, Call Id=557
*Dec  5 14:04:25.068: //557/86D8E9B78B8B/CCAPI/cc_api_call_disconnected:
   Call Entry(Responsed=TRUE, Cause Value=47, Retry Count=0)

Any ideas?

Thanks,

Reg.

Can you post your full config and a debus ccsip messages?

Regards,

Yosh

HTH Regards, Yosh

Hello yahsiel2004,

 

find attached the following files:

- CME config

- ccapi debug calling from pstn to my ip phone (97) --> doesn't work

- ccapi debug calling from ip phone (99) to ip phone (97) --> works

- ccsip debug pstn to my ip phone (97)

 

Thanks,

Reg.

RIght now also you are facing the same issue or not ?

Yes of course, that is why I send this message to Cisco Community. 

In the CCSIP debug it is said clearly that "no codec" has been negotiated but I do not understand why the calls go perfectly well and when it turns to the CUE the calls coming from "outside network/PSTN" cannot make it to it.

Thanks,

Reg.

Normally when a  PSTN call hits to the gateway, what is the codec using. If it is g711ulaw ,then normally transcoder not needed for this .

But if it is different it should need a transcoder.

 

Apart from that you have mentioned that incoming calls dont have any problem.

That is the strange part. Anyway try registering a transcoder and check the result.

We can remove it later

In the CCSIP debug it says:

 

*Dec  5 15:10:19.584: //583/A939D54C851C/SIP/Info/info/1/codec_found: Codec to be matched: g729r8(16)

 

When I call to my IP phone from PSTN and I issue the "sh voice call summary" and "sh voice call status" see what I get:

pabx#show voice call summary
PORT           CODEC     VAD VTSP STATE            VPM STATE
============== ========= === ==================== ======================
0/2/0         -          -  -                     FXSLS_ONHOOK
0/2/1         -          -  -                     FXSLS_ONHOOK
0/2/2         -          -  -                     FXSLS_ONHOOK
0/2/3         -          -  -                     FXSLS_ONHOOK
50/0/1  .1       -          -  -                     EFXS_ONHOOK
50/0/1  .2       -          -  -                     EFXS_ONHOOK
50/0/2  .1       -          -  -                     EFXS_ONHOOK
50/0/2  .2       -          -  -                     EFXS_ONHOOK
50/0/3  .1       g711ulaw   n  S_SETUP_REQ_PROC      EFXS_WAIT_OFFHOOK
50/0/3  .2       -          -  -                     EFXS_ONHOOK

 

----------------------------------------------------------------------------------------

pabx#sh voice call status
CallID     CID  ccVdb      Port        Slot/DSP:Ch  Called #   Codec    MLPP Dial-peers
0x2F1      175D 0x3EE2DDF0 50/0/3.0                *97         None     2/20003
1 active call found

On one hand it is g711ulaw, on the other hand it says "none".

 

Hello guys,

in the end I got it working following this site:

http://duzceli1979.blogspot.ch/2013/01/enable-cue-access-from-g729-enabled-wan.html

quite easy to understand and direct. I will update you if anything else goes wrong.

I did not need to enter any of the lines for sip in the voice group section.

 

Thanks for all your support.

Reg.

You don't have a PSTN, you have an ITSP. Since you have VoIP dial-peers which use G729r8 by default, you will need a transcoder and or you will need to change the codec on the incoming dial-peer to G711. Also you might need to put "no supplementary-service sip moved-temporarily" under the voice service voip because CUE doesn't like VoIP to VoIP calls forwards with sip moved-temporarily on.

voice service voip
no supplementary-service sip moved-temporarily

HTH

Yosh

HTH Regards, Yosh
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