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]

Hi,

I have looked at your debugs and they are not complete. I didnt see the end of the call in your debugs.

Is this your call flow

SIP Provider------->CCME---->CUE (AA) ?

If thats correct can you send the sh run of your CCME router.

Can you also let us know what happens exactly? Do you get an engaged tone? when you call AA. When you forward to VM what happens exactly..

Do a atest again and send

debug ccsip messages

debug voip ccapi inout

[mohd abdul malik]

Yes thats my call flow.

the call gets transferred from ccme to cue and as u can see from cue trace the call gets transferred to cue the bell rings but the AA does not answer and play the prompt the bell rings for about 90 secs and gets disconnected.This is a really busy site and its really difficult for me to get the whole logs. I will get the configuration by tomorrow and will try to get the logs again by tomorrow as the client seems to have left the office already.

[mohd abdul malik]

   Pls find the attached logs u asked for.

I have also added the cue traces for ur convenience.

[syed baseeruddin]

Pls find the attached file of sip trunks for auto attendant configuration

[Ayodeji Okanlawon]

Mohd,

Can you tell me what you get when you call into the AA..and what you want to happen..

THE cue logs showed that the AA plays

1. Silence.wav prompt

5085 05/20 10:15:14.330 ACCN LMED 0 Adding File: /usr/wfavvid/Prompts/system/default/silence/silence_500.wav

5085 05/20 10:15:14.330 ACCN LMED 0 In play(): Now going to wait() for prompt to finish

2. Then another prompt is played..(main-menu-eng.wav)

5085 05/20 10:15:14.876 ACCN ENGN 0 Execute step of Task 60000008460 : Play Prompt (contact: --Triggering Contact--, prompt: "main-menu-eng.wav")

5085 05/20 10:15:14.877 ACCN LMED 0 In play(): Now going to wait() for prompt to finish

3. Next a dial by extension step is executed.

085 05/20 10:15:25.396 ACCN ENGN 0 Execute step of Task 60000008460 : Dial-by-Extension Menu (contact: --Triggering Contact--, prompt: "lang-option.wav")

4. Then cue executes digit collection step

5085 05/20 10:15:25.396 ACCN STMD 0 Task:60000008460 Executing DigitCollectMenuStep

085 05/20 10:15:25.396 ACCN ENGN 0 Execute step of Task 60000008460 : Dial-by-Extension Menu (contact: --Triggering Contact--, prompt: "lang-option.wav")
5085 05/20 10:15:25.396 ACCN STMD 0 Task:60000008460 Executing DigitCollectMenuStep

5. This is where I see the call failing..

5156 05/20 10:15:33.831 DSSP LWRE 0 Received UDP packet on 10.0.0.11:5060 ,source 10.0.0.1:63449
BYE sip:2293100@10.0.0.11:5060 SIP/2.0

Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK4CE41C26

From: <531808130>;tag=E73C4DC-A9

To: <2293100>;tag=cueb91350

Date: Sun, 20 May 2012 07:15:14 GMT

Call-ID: 60FEBB18-A18211E1-BA90C25C-D5D81B27@10.0.0.1

User-Agent: Cisco-SIPGateway/IOS-12.x

Max-Forwards: 70

Timestamp: 1337498133

CSeq: 102 BYE

++++Can you confirm this is where you are having p roblems. Entering digit to dial an extension++++

If that is so them you have a dtmf issue..we can work on that...

[mohd abdul malik]

Thanks a lot for u spending so much time checking debugs.

The actual problem is that when anybody calls the AA the bell keeps ringing but nothing is played nor the call is answered.

The bell keeps ringing for almost 1 min and gets disconnected it does not play any any prompt though the logs show the prompt is played but caller does not hear anything but rings. So there is no question of entering digits.

[Ayodeji Okanlawon]

Ok,

What did you use to record the prompts? The prompts needs to recorded in the following format:

G.711 U-law, 8 kHz, 8 bit, Mono.

Are your pormpts recorded this way?

[mohd abdul malik]

I already mentioned that I can still use these prompts for Analog lines on FXO ports, but its not working for SIP the bell rings and no answer and one more thing is happening the voicemail is also not responding, the CFNA to VM is also not working since this SIP configuration was made.

[Ayodeji Okanlawon]

Ok. Can you do the following:

remove the ff : commands from dial-peer 2293100 voip

voice-class sip dtmf-relay force rtp-nte

voice-class sip outbound-proxy  ipv4:10.0.0.11

For the AA:

Then test first from a ccme phone...record the results

then test from an outside phone..records the results

please let me know the outcome

can you also try the ff for CFNA to vm

1. test CFNA to VM from the CCME

2. add the ff: command and test from outside

voice service voip

no supplementary−service sip moved−temporarily

[mohd abdul malik]

Hi,

I tried all the above said options but still AA does not work. The AA works like this

mainmenu prompt : 2293100

English prompt : 7802
Arabic prompt : 7804

Internally 7802,7804 are working but 2293100 is not working i tried making a new script but still its not working.

Although VM is working internally but externally its stll the same.

[Ayodeji Okanlawon]

Mohd,

Lets work on one thing at a time. Lets start with External voicemail.

Can you send me debug voip ccapi inout and debug ccsip messages for external VM. If you can send cue logs too that will be good.

[Ayodeji Okanlawon]

So after looking at your trace in detail..I see where we could be having problems from outside call to VM

1. ++++ We receive call from service provide++++

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=z9hG4bKc78b7vh7yv7cjoucoccenveb7

Call-ID: SBCe1f5kgm3p3k5sefhcm1fotieo1i51k1h@SoftX3000

From: <531808130>;tag=pemkm33s-CC-24

To: <2293100>

2. ++++We send a Trying to Service provider+++++++

Sent:
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 10.208.9.69:5060;branch=z9hG4bKc78b7vh7yv7cjoucoccenveb7

From: <531808130>;tag=pemkm33s-CC-24

To: <2293100>

3. ++++We send an Invite to CUE++++++

Sent:
INVITE sip:2293100@10.0.0.11:5060 SIP/2.0

Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK4D021790

Remote-Party-ID: <531808130>;party=calling;screen=no;privacy=off

From: <531808130>;tag=E780260-B43

4. +++After a couple of trying from CUE and Session prgress, we get a 200 ok

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK4D021790

To: <2293100>;tag=cue38d240d7

From: <531808130>

c=IN IP4 10.0.0.11

5. +++We send a 200 Ok to service provider++++

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.208.9.69:5060;branch=z9hG4bKiivxhklheuoylxe7oeouekvvi

From: <531808130>;tag=pemkm33s-CC-24

To: <2293100>;tag=E780260-226C

Date: Sun, 20 May 2012 07:19:52 GMT

c=IN IP4 10.66.4.246

Now, lets look at how the RTP connection looks like..This is where the problem is I think..

10.208.9.69---10.66.4.246 (call leg 1)

10.0.0.1----10.0.0.11(call leg 2)

On the first call leg, RTP stream is between the CCME router and Service provider (this is expected)

on the 2nd call leg, RTP stream is setup between the hidden ip of your CCME router and CUE(Hence there is no way the service provider can listen on the 2nd call leg)

Ideally for the call to work, your RTP connection should look like this..

10.208.9.69---10.66.4.246 (call leg 1)

10.66.4.246----10.0.0.11(call leg 2)

This way your service provider can listen to the RTP stream to the CUE for voicemail and AA.

So we ned to find a way to fix this in your configuration..

Can you configure this and apply it to the dial-peer as show and test again..

voice class sip-profiles 10
request invite sip-header From modify "10.0.0.1" "10.66.4.246"

dial-peer voice 9000 voip
voice-class sip profiles 10

I am not sure if this will resolve the problem though..Because for this issue to be resolved, we need CUE to send its RTP conection to the address that yoru service provider is listening on which is "10.66.4.246"

[mohd abdul malik]

very well said,

when i checked now, i could not reach SIP from my CUE.

I will check this configuration tomorrow and get back to you.

Again Thanks a lot.....I would like to know one more thing that how to identify things in the logs and particular call and how to identify call id.

[Ayodeji Okanlawon]

It is not hard to understand traces. But first of all you need to understand the underlying component in any call  flow. E.g to understand sip traces, you need to understand how SIP work. I suggest you read about sip from a telephony perspective. How calls are routed over sip trunks etc.  The link sysed attached for configuring cme-sip trunk is quite good.

So start by understanding how SIP work, then you will understand the traces and you will be able to identify when things go wrong.

And dont forget to rate useful posts!