AA not working on SIP

[mohd abdul malik]

  Hi all,

we have recently terminated sip trunk on one of our customers site there seems to be problem with Unity as AA and voicemail is not functioning properly.

The AA has number 2293100, extensions are 4 digits in range of 2293100-229399. I have to put complete number 2293100 for AA to work but it does not answer i tried looking at the logs but i could not understand anything. The same is with the Voicemail the calls are not getting forwarded to voicemail once the cfna timer expires.

When i bring it back to analog lines everything seems to be working fine.

I have attached logs for cme and cue.

this is 3800 series router with 12.4 ios and 4.x cme.

[Ayodeji Okanlawon]

Mohd,

After more investigation, it look slike I have see what the problem is.

Cisco CUBE doing SIP-SIP does not support DTMF translation from SIP-Notify to RFC2833 (rtp-nte)

What this mean is that you cant call the CUE AA directly.

So I will suggest you do the following:

1. On your inbound dial-peer from your ITSP, make sure you have only rtp-nte as your dtmf-relay

2. Then on your outbound dial-peer to CUE make sure its sip-notify as you have now

3. Make a test call to a phone, ensure the call is forwarded to voicemail....Does that work? If that work then great

4. Create an extension in CME with CUE AA number. On the number do a CFDWALL to CUE (so replace the AA pilot number in CUE with a new number and forward from CCME AA pilot number to this number

5. Finally cponfigure this on your gateway..

voice service voip

no supplementary-service sip moved-temporarily

Let me know what the outcome is...

NB: if it works, then great. But you may not be able to enter any dtmf digits because we still need to adjust your config for this provider on entering dtmf digits...

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[mohd abdul malik]

Hi Aoklawon,

Sorry i could not respond to ur post.

It will take me another 2-3 days to check this configuration.

I will get back to u once i have the results.

Thanks again.

[mohd abdul malik]

Hi Aoklawon,

My sip provder at last responded back with a reply that it is a codec issue and we are not sending  PCMA/8000 G711ALAW request that is why they are not responding pls have a look at the debug logs from their side i could understand none of it.

Can u tell me is the transcoding configuration done properly as the cue supports only g711ulaw??

[Ayodeji Okanlawon]

Malik,

I dont know what your provider is talking about. I believe what they are talking about is not correct. Why did I say so..Because in the invite sent by your provider G711ulaw was advertised...Also I dont know any SIP provider that does not support G711ulaw.

Here is a proof. (i have underlined the codecs advertised by them) Take it back to them and tell them that they advertise G711ulaw

Received:

INVITE sip:2293100@10.66.4.246:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 10.208.9.69:5060;branch=z9hG4bKlcokinlekebocbux7njekuueh

Call-ID:

SBC5py15kpgs15m3s35kmf35e5tshhgth5h@SoftX3000

From: <26171155>;tag=ikmhcg11-CC-38

To: <2293100>CSeq: 1 INVITE

Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFERMax-Forwards: 70

Supported: 100relUser-Agent: Huawei SoftX3000 V300R010

Contact: <26171155>Content-Length: 224Content-Type: application/sdp

v=0

o=HuaweiSoftX3000 2624115 2624115 IN IP4 10.208.9.69

s=Sip Callc=IN IP4 10.208.9.69

t=0 0

m=audio 19012 RTP/AVP 8 0 97

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:97 telephone-event/8000

a=fmtp:97 0-15

CUE will only support G711ulaw. This is the codec it will advertise when it responds with a 200 ok SDP. I dont think we can change this by trannscoding..

In the meantime while you respond to them, please try and add this command and test again...

+++++++++++++++++++++++++++

voice class sip-profiles 1

response 200 sdp-header Attribute add "a=rtpmap:97 telephone-event/8000"

Then add it to this dial-peer.. ( i assume you still have it in your config)

dial-peer voice 2 voip

translation-profile incoming DID

voice-class codec 1

voice class sip-profiles 1

dtmf-relay rtp-nte

session protocol sipv2

incoming called-number 2293100

+++++++++++++++++++++++++++

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