Re: calls from UK PSTN to CUE fails

Chris Drew · ‎10-01-2009

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

Nicholas Matthews · ‎10-01-2009

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

-nick

Chris Drew · ‎10-01-2009

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

paolo bevilacqua · ‎10-01-2009

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.

Chris Drew · ‎10-01-2009

sorry. what is the DP level. thanks

paolo bevilacqua · ‎10-01-2009

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

CSCO12205445 · ‎12-09-2014

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.

yahsiel2004 · ‎12-09-2014

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

Regards,

Yosh

HTH Regards, Yosh

CSCO12205445 · ‎12-09-2014

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.

afsal abdul gafoor · ‎12-09-2014

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

CSCO12205445 · ‎12-09-2014

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.

afsal abdul gafoor · ‎12-09-2014

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

CSCO12205445 · ‎12-09-2014

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".

CSCO12205445 · ‎12-12-2014

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.

yahsiel2004 · ‎12-09-2014

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