05-19-2012 05:54 AM - edited 03-16-2019 11:14 AM
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.
Solved! Go to Solution.
05-21-2012 08:14 PM
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.
05-22-2012 01:14 AM
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
05-22-2012 01:16 AM
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
05-25-2012 09:34 PM
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.
05-26-2012 01:32 AM
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.
05-27-2012 02:17 AM
05-27-2012 07:49 AM
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
05-29-2012 01:40 PM
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.
05-29-2012 01:49 PM
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?
05-29-2012 09:04 PM
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.
05-30-2012 12:51 AM
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
05-30-2012 08:20 AM
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
05-30-2012 08:52 AM
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
06-03-2012 07:10 AM
06-03-2012 03:40 PM
+++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-38T26171155>
o: <2293100>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/sdp26171155>
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=cuef677b17c2293100>
From: <26171155>;tag=57312AA0-1DFDCall-ID: 26171155>
C7491DE1-AC9B11E1-B824C25C-D5D81B27@10.66.4.246
CSeq: 101 INVITEContent-Length: 172Contact: <2293100>Content-Type: application/sdp2293100>
Call-Info: <10.0.0.11:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"Allow-Events: telephone-event10.0.0.11:5060>
v=0
o=CiscoSystemsSIP-Workflow-App-UserAgent 3713 3713 Ieen=no;privacy=off
Contact: <2293100>Supported: replacesServer: Cisco-SIPGateway/IOS-12.x2293100>
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 GMT2293100>26171155>
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=off2293100>
Contact: <2293100>Supported: replacesServer: Cisco-SIPGateway/IOS-12.xContent-Type: application/sdpContent-Disposition: session;handling=requiredContent-Length: 1882293100>
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)
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