11-13-2012 03:48 AM - edited 03-16-2019 02:09 PM
Hi
Setup:
ISP SIP--> VG SIP--> CUCM- Extensions
Sip Trunk in CUCM: Destination Port 5080 (Requested by ISP)
Below is the trace
Calling number:22551286
Called number:00302155203740
Extension:2803
CUCMs: 192.168.1.241 , 192.168.1.242
ISP network:10.224.71.68
In my interface connected with ISP modem:192.168.10.253
Problem::
Incoming calls never going to the extension number:
If i try to bind SIP media and call control then i lose the registration with ISP.
So i dont use sip binds
Nov 13 10:41:56.577: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:00302155203740@192.168.10.253:5080 SIP/2.0
Max-Forwards: 8
Via: SIP/2.0/UDP 10.224.71.68:5060;branch=z9hG4bKexvdikw6s2mon6c83x7hbdxw7
To: "ICT LOGICOM SOLUTIONS AE" <sip:00302155203740@fmc.cyta.com.gr>;cscf
From: <sip:22551286@anonymous.invalid;user=phone>;tag=h7g4Esbg_1305616646-1352803356905-
Call-ID: BW124236905131112-787426872@10.224.74.10
CSeq: 1018071157 INVITE
Contact: <sip:sgc_c@10.224.71.68;transport=udp>
Record-Route: <sip:10.224.71.68;transport=udp;lr>
Min-Se: 600
Privacy: none
Session-Expires: 1800
Supported: 100rel
Content-Type: application/sdp
Content-Length: 490
Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK, REFER, NOTIFY, INFO, UPDATE
Accept: application/sdp
Accept: application/media_control+xml
Accept: multipart/mixed
v=0
o=BroadWorks 820925148 1 IN IP4 10.224.71.68
s=-
c=IN IP4 10.224.71.5
t=0 0
m=audio 31494 RTP/AVP 8 18 100
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=sqn: 0
a=cdsc: 1 audio RTP/AVP 8 18 100
a=cpar: a=rtpmap:8 PCMA/8000
a=cpar: a=rtpmap:18 G729/8000
a=cpar: a=fmtp:18 annexb=yes
a=cpar: a=rtpmap:100 telephone-event/8000
a=cpar: a=fmtp:100 0-15
a=cdsc: 4 image udptl t38
a=sendrecv
a=ptime:20
Nov 13 10:41:56.593: //5019/94729857823C/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.224.71.68:5060;branch=z9hG4bKexvdikw6s2mon6c83x7hbdxw7
From: <sip:22551286@anonymous.invalid;user=phone>;tag=h7g4Esbg_1305616646-1352803356905-
To: "ICT LOGICOM SOLUTIONS AE" <sip:00302155203740@fmc.cyta.com.gr>;cscf
Date: Tue, 13 Nov 2012 10:41:56 GMT
Call-ID: BW124236905131112-787426872@10.224.74.10
CSeq: 1018071157 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M1
Content-Length: 0
Nov 13 10:41:56.597: //5020/94729857823C/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:2803@192.168.1.241:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.253:5080;branch=z9hG4bK12D32260
From: <sip:22551286@fmc.cyta.com.gr>;tag=E574B00-240E
To: <sip:2803@192.168.1.241>
Date: Tue, 13 Nov 2012 10:41:56 GMT
Call-ID: 94750990-2CB511E2-8242CD61-D98E522@fmc.cyta.com.gr
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2490538071-0750064098-2185022817-0228123938
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M1
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1352803316
Contact: <sip:22551286@192.168.10.253:5080>
Expires: 60
Allow-Events: telephone-event
Max-Forwards: 7
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 288
v=0
o=CiscoSystemsSIP-GW-UserAgent 8250 565 IN IP4 192.168.10.253
s=SIP Call
c=IN IP4 192.168.10.253
t=0 0
m=audio 16694 RTP/AVP 8 18 101
c=IN IP4 192.168.10.253
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Nov 13 10:41:56.709: //5020/94729857823C/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.10.253:5080;received=10.93.7.245;branch=z9hG4bK12D32260
To: <sip:2803@192.168.1.241>;tag=h7g4Esbg_16990-1352803357044
From: <sip:22551286@fmc.cyta.com.gr>;tag=E574B00-240E
Call-ID: 94750990-2CB511E2-8242CD61-D98E522@fmc.cyta.com.gr
CSeq: 101 INVITE
Contact: <sip:sgc_c@10.224.71.68;transport=udp>
Record-Route: <sip:10.224.71.68;transport=udp;lr>
Require: 100rel
RSeq: 1018071226
Content-Type: application/sdp
Content-Length: 224
Session: Media
Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE, INFO
v=0
o=BroadWorks 820925158 1 IN IP4 10.224.71.68
s=-
c=IN IP4 10.224.71.15
t=0 0
m=audio 11750 RTP/AVP 8 101
c=IN IP4 10.224.71.15
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Nov 13 10:41:56.713: //5020/94729857823C/SIP/Msg/ccsipDisplayMsg:
Sent:
PRACK sip:sgc_c@10.224.71.68;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.10.253:5080;branch=z9hG4bK12D4E9
From: <sip:22551286@fmc.cyta.com.gr>;tag=E574B00-240E
To: <sip:2803@192.168.1.241>;tag=h7g4Esbg_16990-1352803357044
Date: Tue, 13 Nov 2012 10:41:56 GMT
Call-ID: 94750990-2CB511E2-8242CD61-D98E522@fmc.cyta.com.gr
CSeq: 102 PRACK
RAck: 1018071226 101 INVITE
Route: <sip:10.224.71.68;transport=udp;lr>
Allow-Events: telephone-event
Max-Forwards: 70
Content-Length: 0
Nov 13 10:41:56.717: //5019/94729857823C/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.224.71.68:5060;branch=z9hG4bKexvdikw6s2mon6c83x7hbdxw7
From: <sip:22551286@anonymous.invalid;user=phone>;tag=h7g4Esbg_1305616646-1352803356905-
To: "ICT LOGICOM SOLUTIONS AE" <sip:00302155203740@fmc.cyta.com.gr>;cscf;tag=E574B78-1C5A
Date: Tue, 13 Nov 2012 10:41:56 GMT
Call-ID: BW124236905131112-787426872@10.224.74.10
CSeq: 1018071157 INVITE
Require: 100rel
RSeq: 8409
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:00302155203740@192.168.10.253:5080>
Record-Route: <sip:10.224.71.68;transport=udp;lr>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M1
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 252
v=0
o=CiscoSystemsSIP-GW-UserAgent 3041 881 IN IP4 192.168.10.253
s=SIP Call
c=IN IP4 192.168.10.253
t=0 0
m=audio 16692 RTP/AVP 8 100
c=IN IP4 192.168.10.253
a=rtpmap:8 PCMA/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=ptime:20
Nov 13 10:41:56.757: //5020/94729857823C/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.253:5080;received=10.93.7.245;branch=z9hG4bK12D4E9
To: <sip:2803@192.168.1.241>;tag=h7g4Esbg_16990-1352803357044
From: <sip:22551286@fmc.cyta.com.gr>;tag=E574B00-240E
Call-ID: 94750990-2CB511E2-8242CD61-D98E522@fmc.cyta.com.gr
CSeq: 102 PRACK
Content-Length: 0
Nov 13 10:41:56.833: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
PRACK sip:00302155203740@192.168.10.253:5080 SIP/2.0
Max-Forwards: 8
Via: SIP/2.0/UDP 10.224.71.68:5060;branch=z9hG4bK6x44d917ejxu6pkeo1u9l898f
To: "ICT LOGICOM SOLUTIONS AE" <sip:00302155203740@fmc.cyta.com.gr>;tag=E574B78-1C5A;cscf
From: <sip:22551286@anonymous.invalid;user=phone>;tag=h7g4Esbg_1305616646-1352803356905-
Call-ID: BW124236905131112-787426872@10.224.74.10
CSeq: 1018071158 PRACK
RAck: 8409 1018071157 INVITE
Content-Length: 0
Nov 13 10:41:56.833: //5019/94729857823C/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.224.71.68:5060;branch=z9hG4bK6x44d917ejxu6pkeo1u9l898f
From: <sip:22551286@anonymous.invalid;user=phone>;tag=h7g4Esbg_1305616646-1352803356905-
To: "ICT LOGICOM SOLUTIONS AE" <sip:00302155203740@fmc.cyta.com.gr>;cscf;tag=E574B78-1C5A
Date: Tue, 13 Nov 2012 10:41:56 GMT
Call-ID: BW124236905131112-787426872@10.224.74.10
Server: Cisco-SIPGateway/IOS-15.2.4.M1
CSeq: 1018071158 PRACK
Content-Length: 0
11-13-2012 03:55 AM
Trace shows that CM takes and ansewr the calls, so you should look what's happening there.
Nov 13 10:41:56.757: //5020/94729857823C/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.253:5080;received=10.93.7.245;branch=z9hG4bK12D4E9
To: <2803>;tag=h7g4Esbg_16990-13528033570442803>
From: <>22551286@fmc.cyta.com.gr>;tag=E574B00-240E>
Call-ID: 94750990-2CB511E2-8242CD61-D98E522@fmc.cyta.com.gr
CSeq: 102 PRACK
Content-Length: 0
11-13-2012 04:08 AM
Hi Paolo
I saw that , but the calls never coming to the extension.This is the strange
In the sip trunk config in CUCM i have 4 significant digits and the CSS include the PT for my extension
When i shut the sip dial peers in VG and no shut to the h323 dial peers then the phone ringing
(With H323 we have other problem and this is the reason that i tried to check the sip
With H323 when i answer the call its stay active ONLY for 10 sec.And then its going down
Any ideas FOR all ?
11-13-2012 04:16 AM
Are using NAT somewhere ? What is 10.93.7.245 ?
Nov 13 10:41:56.709: //5020/94729857823C/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.10.253:5080;received=10.93.7.245;branch=z9hG4bK12D32260
To: <2803>;tag=h7g4Esbg_16990-13528033570442803>
From: <>22551286@fmc.cyta.com.gr>;tag=E574B00-240E>
Call-ID: 94750990-2CB511E2-8242CD61-D98E522@fmc.cyta.com.gr
CSeq: 101 INVITE
Contact:
Record-Route: <10.224.71.68>10.224.71.68>
Require: 100rel
RSeq: 1018071226
Content-Type: application/sdp
Content-Length: 224
Session: Media
Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE, INFO
v=0
o=BroadWorks 820925158 1 IN IP4 10.224.71.68
s=-
c=IN IP4 10.224.71.15
t=0 0
m=audio 11750 RTP/AVP 8 101
c=IN IP4 10.224.71.15
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
11-13-2012 04:41 AM
Hi
10.93.7.245 is the voice ip in the ISP Modem
Nov 13 10:41:56.757: //5020/94729857823C/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.253:5080;received=10.93.7.245;branch=z9hG4bK12D4E9
To: <2803>;tag=h7g4Esbg_16990-13528033570442803>
11-13-2012 05:14 AM
10.93.7.245 is not in the same subnet as 192.168.10.253. I suspect there is NAT in there. Is that your router or their ?
Also, I don't understand why there is session progress coming in after invite.
11-13-2012 05:22 AM
192.168.10.0 subnet is in the modem site(ISP) for private ip
192.168.10.253 in in my interface .254 is in the modem side
The above is a private network that gives the ISP throught the modem
Now they have also configure in their modem 10.93.7.0.245 which is the voice ip.In this connection they will pass internet and also voice
So they have different subnets configured for that
ip route 10.93.7.0 255.255.255.0 192.168.10.254
ip route 10.199.0.0 255.255.255.0 10.100.0.4
ip route 10.224.50.0 255.255.255.0 192.168.10.254
ip route 10.224.64.0 255.255.255.0 192.168.10.254
ip route 10.224.71.0 255.255.255.0 192.168.10.254
ip route 10.224.75.0 255.255.255.0 192.168.10.254
ip route 192.168.10.0 255.255.255.0 192.168.10.254
I spoke with the ISP and they said that they see the invite for my extension
Is it normal?
i am not sure about that
11-13-2012 05:35 AM
In my expereince, one has to focus on the very first message that doesn't look normal, and that is the session progress above.
In CUCM, what is the address of SIP trunk, or H.323 GW ?
11-13-2012 05:41 AM
For sip trunk i used 10.110.0.3 :
Also 10.110.0.3 is binding in this router for the h323 communication (existing setup)
!
interface Loopback0
ip address 10.110.0.3 255.255.255.255
h323-gateway voip interface
h323-gateway voip bind srcaddr 10.110.0.3
!
11-13-2012 05:45 AM
Please replace that with VG local interface address. Also please configure VG for media flow-throgh, not flow-around.
11-13-2012 07:41 AM
Hi
Thank you for the replies
i used the voice subinterface but the same results..
!
interface GigabitEthernet0/2.20
description ##### VOICE Vlan #####
encapsulation dot1Q 20
ip address 10.17.20.1 255.255.255.0
ip flow ingress
ip flow egress
!
Do you have any ideas?
Its really very strange
11-13-2012 08:36 AM
You should have a router interface on subnet common to CM LAN interface.
Please use that and configure the corresponding address in CM.
11-13-2012 08:43 AM
Paolo
Its a remote site (SRST SITE) which the phones registered to cucm.The communcation is via tunnel
11-13-2012 08:49 AM
It seems to me there is some confusion, You would have to include a diagram with all the circuits, devices and addresses.
11-13-2012 09:20 AM
I also noticed in your trace that there is no "Trying" received from cucm before the "session progress".....
When CUBE sent an invite to cucm, the first response should be "trying" Thats how cucm says i am going to look for the number....Are you sending the full trace? If you are then something is wrong with the way CUCM is processing the sip invite
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