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SIP INCOMING CALLS NOT WORKING

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

Please rate all useful posts Regards Chrysostomos ""The Most Successful People Are Those Who Are Good At Plan B""
17 REPLIES 17
Highlighted
Hall of Fame Master

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_16990-1352803357044

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

Highlighted

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 ?

Please rate all useful posts Regards Chrysostomos ""The Most Successful People Are Those Who Are Good At Plan B""
Highlighted

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_16990-1352803357044

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>

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

Highlighted

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_16990-1352803357044

Please rate all useful posts Regards Chrysostomos ""The Most Successful People Are Those Who Are Good At Plan B""
Highlighted

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.

Highlighted

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

Please rate all useful posts Regards Chrysostomos ""The Most Successful People Are Those Who Are Good At Plan B""
Highlighted

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 ?

Highlighted


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

!

Please rate all useful posts Regards Chrysostomos ""The Most Successful People Are Those Who Are Good At Plan B""
Highlighted

Please replace that with VG local interface address. Also  please configure VG for media flow-throgh, not flow-around.

Highlighted

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

Please rate all useful posts Regards Chrysostomos ""The Most Successful People Are Those Who Are Good At Plan B""
Highlighted

You should have a router interface on subnet common to CM LAN interface.

Please use that and configure the corresponding address in CM.

Highlighted

Paolo

Its a remote site (SRST SITE) which the phones registered to cucm.The communcation is via tunnel

Please rate all useful posts Regards Chrysostomos ""The Most Successful People Are Those Who Are Good At Plan B""
Highlighted

It seems to me there is some confusion, You would have to include a diagram with all the circuits, devices and addresses.

Highlighted

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

Please rate all useful posts

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