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.

[mohd abdul malik]

Sip Profiles are not working as this router is running on 12.4 ios and sip profiles were introduced in 15.0.

Can we not Nat those IP's i.e 10.0.0.1 and 10.0.0.11 on 10.66.4.246 will it work this way.

[Ayodeji Okanlawon]

You can try this with nat..

interface GigabitEthernet0/0.100
ip nat inside

interface GigabitEthernet0/1
ip nat outside

ip nat inside source list 1 interface GigabitEthernet0/1 overload

[Ayodeji Okanlawon]

Hi, you can try this with NAT:

interface GigabitEthernet0/0.100
ip nat inside

!

interface GigabitEthernet0/1
ip nat outside

!

ip nat inside source list 1 interface GigabitEthernet0/1 overload

[mohd abdul malik]

Hi,

Its not working even with Nat.

I have opened a case with TAC they say its a problem with the service provider that its not sending the 200 ok response.

So i have opened a case with the service provider to check with the issue.

can it happen this way that for all the voice calls the service provider has to send the 200 ok reply and not for the AA and Voicemail.

[Ayodeji Okanlawon]

Mohd,

It is true that the provider didnt send a 200 response back...

I think there is something they dint like in the 200 on your ccme router sent to them...

Can you do a test call for a normal succeful call and send me a

debug ccsip messages.

We can compare what we are sending in the 200 ok for that call and this one that is failing.

[mohd abdul malik]

                   Hi,

     Pls find the attached file for current logs to compare with previous logs.

     I am calling from 6191155 to 2293110.

[Ayodeji Okanlawon]

Mohd,

What happens with this call? Does the call fail too? Has your configuration change..it looks like the old config i have does not match what is on this trace..

Can you let me know what happens on this call. What I asked is a trace for a succesful call..did this call succeed? Can you also send me an updated config if it has change

[mohd abdul malik]

Hi Aokanlawon,

I will definetly get back to you with current configuration that was altered by TAC team for transcoding and sip interface binding.

Where as with the latest logs that i have attached is for a successful call that i placed to 2293110 from 6191155.

Man i am really beginning to respect and appreciate ur dedication to solve this problem, whereas i myself got fedup with this issue as Service provider does not respond with any reasonable reply and we cannot solve until now, hope i get better of myself with this problem.

[Ayodeji Okanlawon]

So the difference in what worked and what didnt work is that on the failed call you did not send any dtmf attributes to your sip provider, so  their switch ignores your call setup because they need dtmf...The call that worked had dtmf on it. Dont give up we can solve this problem..

So with Tac, is the problem resolved now?

[mohd abdul malik]

Nope,

TAC people wants a reply from service provider "Why they are not sending 200 Ack mesg".

It seems they wont do anything unless we get this reply.

So what can we do with the dtmf issue here.

[Ayodeji Okanlawon · Accepted solution]

Mohd,

Let us start again..I assume that you configure this number 2293110 on your CCME router for it to work.. Revert it back to what you want it to be

Send me your current 'sh run' and the ff: debug

debug ccsip messages

debug ccsip all

[mohd abdul malik]

Sorry it was a mistake i clicked on correct answer.

yes u r right, since u asked for a successful call debugs therefore i provided with one.

2293110 is an extension on which i tested successful call.

2293100 is our AA on which we have been working.

pls check the run-config and new logs u requested i am calling from 6191155 to 2293100

[Ayodeji Okanlawon]

Can you add this to your configuration and test again..send me a debug ccsip all

dial-peer voice 2 voip

translation-profile incoming DID

voice-class codec 1

dtmf-relay rtp-nte

session protocol sipv2

incoming called-number 2293100

[mohd abdul malik]

                   Hi,

sorry for being late in getting back to u with results.

Still not working.

have the ccsip msgs all.

[Ayodeji Okanlawon · Accepted solution]

+++Mohd the problem remains the same+++ I suggest you go back to cisco. This is a cisco problem not your service provider

1. when the provider sends an invite with SDP, they sent dtmf attributes in

a=rtpmap:97 telephone-event/8000

a=fmtp:97 0-15

INVITE sip:2293100@10.66.4.246:5060;user=phone SIP/2.0Via: SIP/2.0/UDP 10.208.9.69:5060;branch=z9hG4bKlcokinlekebocbux7njekuueh

Call-ID:

SBC5py15kpgs15m3s35kmf35e5tshhgth5h@SoftX3000

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

o: <2293100>

CSeq: 1 INVITEAllow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFERMax-Forwards: 70Supported: 100rel

User-Agent: Huawei SoftX3000 V300R010Contact: <26171155>Content-Length: 224Content-Type: application/sdp

v=0

o=HuaweiSoftX3000 2624115 2624115

IN IP4 10.208.9.69

s=Sip Call

c=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

Now the call goes to CUE and cue sends its 200 ok.

Received:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.66.4.246:5060;branch=z9hG4bKB6775DFTo: <2293100>;tag=cuef677b17c

From: <26171155>;tag=57312AA0-1DFDCall-ID:

C7491DE1-AC9B11E1-B824C25C-D5D81B27@10.66.4.246

CSeq: 101 INVITEContent-Length: 172Contact: <2293100>Content-Type: application/sdp

Call-Info: <10.0.0.11:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"Allow-Events: telephone-event

v=0

o=CiscoSystemsSIP-Workflow-App-UserAgent 3713 3713 Ieen=no;privacy=off

Contact: <2293100>Supported: replacesServer: Cisco-SIPGateway/IOS-12.x

Content-Type: application/sdpContent-Disposition: session;handling=requiredContent-Length: 188

v=0

o=CiscoSystemsSIP-GW-UserAgent 3642 9251 IN IP4 10.66.4.246s=SIP Callc=IN IP4 10.66.4.246

t=0 0

m=audio 19004 RTP/AVP 0

c=IN IP4 10.66.4.246

a=rtpmap:0 PCMU/8000

a=ptime:20

Finally CCME then sends a 200 ok to the provider with SDP but without DTMF attributes and the provider ingores it and doesnt send an ACK. Cisco need to investigate why even though dtmf relay-nte is configured on your inbound dial-peer

Sent:

SIP/2.0 200 OK

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

From: <26171155>;tag=ikmhcg11-CC-38To: <2293100>;tag=57312AB0-5E2Date: Sun, 03 Jun 2012 10:14:46 GMT

Call-ID:

SBC5py15kpgs15m3s35kmf35e5tshhgth5h@SoftX3000CSeq

: 1 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventRemote-Party-ID: <2293100>;party=called;screen=no;privacy=off

Contact: <2293100>Supported: replacesServer: Cisco-SIPGateway/IOS-12.xContent-Type: application/sdpContent-Disposition: session;handling=requiredContent-Length: 188

v=0

o=CiscoSystemsSIP-GW-UserAgent 3642 9251

IN IP4 10.66.4.246s=SIP Call

c=IN IP4 10.66.4.246

t=0 0

m=audio 19004 RTP/AVP 0c=IN IP4 10.66.4.246

a=rtpmap:0 PCMU/8000

a=ptime:20

******* (dtmf attributes missing)