2911 to ISR4431 call

[dufour001]

Hello,

I have a new ISR4431 for voice.

I have configured an ephone/ephone-dn on it.

On another site, I have an already working CCME 2911

I try to make a call from one end to the other.

On 2911 :

dial-peer voice 4001 voip
 destination-pattern 265
 session protocol sipv2
 session target ipv4:x.x.x.x(ISR4431)
 dtmf-relay rtp-nte
 no vad
!

On 4431, no dial-peer

=> With this config, I can call from 2911 to 4431

But (normal), not from 4431 to 2911 ...

So I add this dial-peer on 4431

dial-peer voice 4002 voip
 destination-pattern 504
 session protocol sipv2
 session target ipv4:x.x.x.x (2911)
 dtmf-relay rtp-nte
 no vad
!

Now, I can call from 4431 to 2911.

But a call from 2911 to 4431 is now failed ...

I tried to add this dial-peer to 4431 (I think I must do that on my old config to work and accept call)

dial-peer voice 5012 voip
 description Appels VOIP vers 2xx (Tournai)
 incoming called-number 265

But that don't help me

What can I do ?

Thx

[Rajan]

Can you provide the below debugs for the failed call along with the calling and called number to check;

debug ccsip messages

debug voip ccapi inout

Thanks

Rajan

[dufour001]

*Jul 18 14:04:29.041: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:265@192.168.101.12:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.167.2:5060;branch=z9hG4bK96BB79
Remote-Party-ID: "Cedric Casbas" <sip:504@192.168.167.2>;party=calling;screen=no;privacy=off
From: "Cedric Casbas" <sip:504@192.168.167.2>;tag=5CAE156C-1A96
To: <sip:265@192.168.101.12>
Date: Tue, 18 Jul 2017 14:04:21 GMT
Call-ID: D5034B26-6AF811E7-9275D248-FEC5D72@192.168.167.2
Supported: 100rel,timer,resource-priority,replaces
Min-SE:  1800
Cisco-Guid: 3567242441-1794642407-2456867400-267148658
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1500386661
Contact: <sip:504@192.168.167.2:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 250

v=0
o=CiscoSystemsSIP-GW-UserAgent 1290 2340 IN IP4 192.168.167.2
s=SIP Call
c=IN IP4 192.168.167.2
t=0 0
m=audio 17178 RTP/AVP 0 101
c=IN IP4 192.168.167.2
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

*Jul 18 14:04:29.044: //74/D49FCCC99270/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/UDP 192.168.167.2:5060;branch=z9hG4bK96BB79
From: "Cedric Casbas" <sip:504@192.168.167.2>;tag=5CAE156C-1A96
To: <sip:265@192.168.101.12>;tag=D950F2-6F5
Date: Tue, 18 Jul 2017 14:04:29 GMT
Call-ID: D5034B26-6AF811E7-9275D248-FEC5D72@192.168.167.2
Timestamp: 1500386661
CSeq: 101 INVITE
Allow-Events: telephone-event
Reason: Q.850;cause=65
Server: Cisco-SIPGateway/IOS-15.5.3.S4b
Content-Length: 0


*Jul 18 14:04:29.097: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:265@192.168.101.12:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.167.2:5060;branch=z9hG4bK96BB79
From: "Cedric Casbas" <sip:504@192.168.167.2>;tag=5CAE156C-1A96
To: <sip:265@192.168.101.12>;tag=D950F2-6F5
Date: Tue, 18 Jul 2017 14:04:21 GMT
Call-ID: D5034B26-6AF811E7-9275D248-FEC5D72@192.168.167.2
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

[Rajan]

Call fails with cause code 65 which is media negotiation failure. In the invite we see only g711ulaw being advertised. Are you limiting the calls to use only this codec anywhere in the configuration ?

[dufour001]

No,

I don't find any reference to this codec ont eh 2 configurations ...

[Rajan]

can you provide debug voip ccapi inout for this call. Also check in the cme sip/sccp phone configuration whether you are locking any specific codec at phone level or globally.