Auto attendant CUE SIP Trunking

[kotushaMx]

Hi, 

I'm having problems to redirect incoming call from sip trunk to  CUE AA,

- I have a Cisco 3945 and  a NME-CUE , VID: V04

- the sip trunk provider only gives the last four digits.

- sip trunk number 8800 redirects to extension 100. Works fine.

- sip trunk number 5624 redirects to extension 100. Works fine.

- sip trunk number 8006 redirects to extension 600, which is the AA. Does not work.

- When I dial 600 from any extension AA work fine.

Here is a briefing of the configuration, if you need any more info please let me know. Thank you in advance.

[Roger Kallberg]

Please do not create duplicate posts. https://community.cisco.com/t5/ip-telephony-and-phones/auto-attendant-cme-and-cue-with-sip-trunking/m-p/5203264

[kotushaMx]

sorry, the other post was taken down, I did not know why and I created this shorter and it came back overnight. I muted the post I did not find how to erase it.

Thank you

[Nithin Eluvathingal]

You can delete the duplicate post before someone replies to it.

Could you clarify what you mean by ‘not working’? Are you getting a fast busy signal, or are you accessing the AA menu but the DTMF is not working? What exactly is not functioning?

Could you share the debug CCSIP messages and debug ccapi in/out in a different attachment?

[kotushaMx]

it does not transfer the call to AA in the dial peer 600, it just silence fot 45 seconds and then it hang up the call, I will share the ccsip tomorrow.

thank u

[kotushaMx]

i cannot find the option to delete post.

[Roger Kallberg]

I’m not sure if the option to delete your own response is available to you. @Nithin Eluvathingal I think that option is available for us in the VIP community, for “regular” members I think that they would need to use the option to notify the moderator team to have it removed. This said I did notify the moderator team about the duplicate post and moved the post to a holding space the VIP community has access to, so the duplicate post is not accessible any more.

[kotushaMx]

Hi, Im attaching the files you ask 

-   CCSIP messages two files when a dial sip trunk ending number 8800 that works and 8806 that does not work to compare.

-    ccapi in/out  same two files

-   as well as debug dialpeer when dial some two number 8800.

Thank you very much, let me know if you need anything-else.

[kotushaMx]

it did not let me attach file with txt extension, Im attaching it in word format 

[Nithin Eluvathingal]

I see a ‘Service Unavailable’ warning and a warning message indicating that the transcoder is not configured.

CUE supports only the G.711 codec, so you need to specify this codec on the dial-peer pointing to the CUE.

[kotushaMx]

thank you, I will try it and let u know.

[kotushaMx]

I will not be able to test it until Monday night, because the sip trunk is operational on another phone system, but I think everything is on g711ulaw, but I will add g711alaw on monday and test it. Thank you very much.

voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
!
dial-peer voice 9000 voip
description ** INBOUND CALLS from telecoms **
translation-profile incoming PSTN_INCOMING
session protocol sipv2
incoming called-number .%
incoming uri via PSTN
voice-class codec 1
voice-class sip early-offer forced
dtmf-relay rtp-nte
no vad
!
dial-peer voice 100 voip
description **** OUTBOUND CALLS to telecoms *****
translation-profile outgoing SIP
destination-pattern 9T
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip early-offer forced
dtmf-relay rtp-nte
no vad
!
dial-peer voice 4500 voip
description **** VoiceMail *****
destination-pattern 45..
session protocol sipv2
session target ipv4:172.16.1.2
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 600 voip
description **** Auto Attendant*****
destination-pattern 60.
session protocol sipv2
session target ipv4:172.16.1.2
no voice-class sip early-offer forced
dtmf-relay sip-notify
codec g711ulaw
no vad
!

[kotushaMx]

I change all dials peers to g711alaw instead of g711ulaw, and it did not work, any ideas what else could the problem be?  it is giving me the same error.

thank you

[Nithin Eluvathingal]

I continue to see the 503 Service Unavailable error with the message ‘Transcorder Not Configured’ in the CCSIP logs.

 The codec on the dial-peer must be G.711alaw, as the ISP is sending ‘m=audio 60228 RTP/AVP 8 18 98’ in the SDP.

Your problem will be solved if you configure the transcoder.

Why did you configure ‘no voice-class sip early-offer forced’ on dial-peer 600 towards Unity Express?

dial-peer voice 600 voip
description **** Auto Attendant*****
destination-pattern 60.
session protocol sipv2
session target ipv4:172.16.1.2
no voice-class sip early-offer forced
dtmf-relay sip-notify
codec g711ulaw
no vad

[kotushaMx]

I cancel command "no voice-class sip early-offer forced" because it did not fix the problem, do you have an article or example in how correctly configure a transcoder? thank you