disconnect cause (cc) 57 disconnect cause (sip) 403

[mastergreg]

Hi Everyone,

I have an issue with disconnect cause (cc) 57 disconnect cause (sip) 403 from my lab. I have CUCM, SWITC 3560 POE, router 2821  and ITSP and internet connection. Inbound and outbound calls are not working. I think I missed something in my configuration. I need your help!!

Attached is my configuration and debug ccsip and voice ccapi.

[Deepak Mehta]

For inbound i donot see any codec being sent in the initial invite message from SP.

So you may want to check on that ..

==========================

INVITE sip:15144477598@192.168.2.2:5060 SIP/2.0
Record-Route: <sip:208.43.27.75;ftag=e535847ae4aeab949cf7064556b26109;lr>
Via: SIP/2.0/UDP 208.43.27.75;branch=z9hG4bK9576.d8dadde192eebaafc1e9813c9eff9d1
e.0
Via: SIP/2.0/UDP 208.43.27.75:5070;branch=z9hG4bK77fa2be60dbafe82f2ff48d11df8e84
6;rport=5070
Max-Forwards: 69
From: "GREGORY SOUILLE" <sip:4387648753@208.43.27.75>;tag=e535847ae4aeab949cf706
4556b26109
To: <sip:15144477598@208.43.27.75>
Call-ID: F67BF699@208.72.120.66-b2b_1
CSeq: 200 INVITE
Contact: Anonymous <sip:4387648753@208.43.27.75:5060>
Expires: 300
User-Agent: Sippy Softswitch v4.5-PRODUCTION.174
cisco-GUID: 3332204332-901724443-2712176348-4199365119
h323-conf-id: 3332204332-901724443-2712176348-4199365119
Content-Length: 184
Content-Type: application/sdp

v=0
o=- 3546920932 3546920932 IN IP4 208.43.27.75
s=-
c=IN IP4 208.43.27.75
t=0 0
m=audio 25952 RTP/AVP 18 0 101
a=rtpmap:101 telephone-event/8000>>>>>>>>RTP-NTE DTMF type
a=ptime:20
a=nortpproxy:yes

for outbound it looks like call is failing after we recieve authentication failure message back from SIp server

======================

SIP/2.0 403 Auth Failed (1)
Via: SIP/2.0/UDP 192.168.2.2:5060;received=184.144.244.168;branch=z9hG4bK25111CE
;rport=53503
Record-Route: <sip:208.43.27.75;ftag=103C494-10D6;lr>
From: <sip:5144477598@sbc1.ixica.com>;tag=103C494-10D6
To: <sip:4387648753@208.43.27.75>;tag=1bf22324f578190f5cf38ebfc6319709
Call-ID: B73F08A0-E1E611E6-805AE3A5-8E41901B@192.168.2.2
CSeq: 102 INVITE
Server: Sippy Softswitch v4.5-PRODUCTION.174
Content-Length: 0

[Ayodeji Okanlawon]

Deepak,

There are codecs in the inbound invite

"m=audio 25952 RTP/AVP 18 0 101"

Greg,

Your inbound call is failing because your authentication credetials are wrong. Check with your provider on what it should be

Jan 25 03:29:28.074: //298/C69D672CA1A8/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:7777@172.16.88.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.88.1:5060;branch=z9hG4bK2391F4
From: "GREGORY SOUILLE" <sip:4387648753@sbc1.ixica.com>;tag=FA8F30-1C15
To: <sip:7777@192.168.2.2:54419>

---

Authorization: Digest username="egvoxx",realm="sippysoft.com",uri="sip:7777@172.
16.88.10:5060",response="f707095655e5858b8f52d89f90379c14",nonce="30802a1848c5ec
9734a7ea7422ef2485104c41b",algorithm=md5

---
*Jan 25 03:29:28.138: //298/C69D672CA1A8/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 403 Auth Failed (1)
Via: SIP/2.0/UDP 172.16.88.1:5060;received=184.144.244.168;branch=z9hG4bK2391F4;
rport=54419
Record-Route: <sip:208.43.27.75;ftag=FA8F30-1C15;lr>
From: "GREGORY SOUILLE" <sip:4387648753@sbc1.ixica.com>;tag=FA8F30-1C15
To: <sip:7777@192.168.2.2:54419>;tag=245268e6c7ed9d73dcc938927f8d67b0
Call-ID: 4F6CC775-E1E511E6-804EE3A5-8E41901B@172.16.88.1
CSeq: 102 INVITE
Server: Sippy Softswitch v4.5-PRODUCTION.174
Content-Length: 0

The same issue happens on outbound leg too. So check with your provider.

[Deepak Mehta]

Thank you for correction, i didn't see any specific attributes for codecs like we normally do.  However Media attribute does list the codecs as you showed.

[mastergreg]

Thank You so much Deepak and Ayodeji. I will check with my provider. If I need any help, I will let you know. Thank you again.