Solved: SIP to SCCP call not OK

Karl Rogers Muller · ‎06-27-2011

Hey all,

I got an issue kind of interesting here. It's a CME and there are two stations: 1 SCCP and 1 SIP.

The point is:

Call from SCCP to SIP: the call is established

Call from SIP to SCCP: reorder tone.

I've got a trace from the SIP to SCCP call:

it7-rot-df-01#

*Jun 27 18:46:02.922: //1913/8ACC373B88F6/SIP/Call/sipSPICallInfo:

The Call Setup Information is:

Call Control Block (CCB) : 0x31711200

State of The Call : STATE_DEAD

TCP Sockets Used : NO

Calling Number : 7063

Called Number : 7060

Source IP Address (Sig ): 172.23.0.1

Destn SIP Req Addr:Port : 172.23.0.21:0

Destn SIP Resp Addr:Port : 172.23.0.21:5060

Destination Name : 172.23.0.21

it7-rot-df-01#

*Jun 27 18:46:02.922: //1913/8ACC373B88F6/SIP/Call/sipSPIMediaCallInfo:

Number of Media Streams: 1

Media Stream : 1

Negotiated Codec : g729r8

Negotiated Codec Bytes : 0

Nego. Codec payload : 18 (tx), 18 (rx)

Negotiated Dtmf-relay : 6

Dtmf-relay Payload : 101 (tx), 101 (rx)

Source IP Address (Media): 172.23.0.1

Source IP Port (Media): 0

Destn IP Address (Media): 172.23.0.21

Destn IP Port (Media): 16454

Orig Destn IP Address:Port (Media): [ - ]:0

*Jun 27 18:46:02.922: //1913/8ACC373B88F6/SIP/Call/sipSPICallInfo:

Disconnect Cause (CC) : 65

Disconnect Cause (SIP) : 488

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

The config is the following:

voice call send-alert

voice rtp send-recv

!

voice service voip

clid network-provided

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

supplementary-service h450.12

no supplementary-service sip moved-temporarily

no supplementary-service sip refer

fax protocol pass-through g711alaw

modem passthrough nse codec g711alaw

sip

registrar server expires max 600 min 60

!

voice class codec 1

codec preference 1 g729br8

codec preference 2 g711alaw

codec preference 3 g711ulaw

!

voice class h323 1

h225 timeout tcp establish 5

!

voice register global

mode cme

source-address 172.23.0.1 port 5060

max-dn 20

max-pool 10

authenticate register

authenticate realm cme

file text

create profile sync 0402163529809355

!

voice register dn 3

number 7063

allow watch

name wip310

no-reg

label wip310

!

voice register dn 4

number 7064

allow watch

name WIP310

no-reg

label WIP310

!

voice register dn 5

number 7066

allow watch

name WIP310

no-reg

label WIP310

!

voice register dn 6

number 7067

allow watch

name WIP310

no-reg

label WIP310

!

voice register dn 7

number 7068

allow watch

name WIP310

no-reg

label WIP310

!

voice register pool 3

id mac 0026.CB0E.7775

number 3 dn 3

presence call-list

dtmf-relay rtp-nte

username 7063 password 1234

!

voice register pool 4

id mac 0026.CB0E.6890

number 4 dn 4

presence call-list

dtmf-relay rtp-nte

username 7064 password 1234

!

voice register pool 5

id mac 0026.CB0E.807F

number 5 dn 5

presence call-list

dtmf-relay rtp-nte

username 7066 password 1234

!

voice register pool 6

id mac 0026.CB0E.7872

number 6 dn 6

presence call-list

dtmf-relay rtp-nte

username 7067 password 1234

!

voice register pool 7

id mac 0026.CB0E.7AD7

number 7 dn 7

presence call-list

dtmf-relay rtp-nte

username 7068 password 1234

!

telephony-service

no auto-reg-ephone

em logout 0:0 0:0 0:0

fxo hook-flash

max-ephones 25

max-dn 25

ip source-address 172.23.0.1 port 2000

timeouts ringing 30

system message IT7-DF

cnf-file location flash:

cnf-file perphone

user-locale U1 load CME-locale-pt_BR-Portuguese-7.0.1.1.tar

load 7942 SCCP42.9-1-1SR1S.loads

date-format dd-mm-yy

max-conferences 8 gain -6

web admin system name WebAdmin secret 5 $1$9S1/$Uipxic1IdEotBLNx/FusR/

transfer-system full-consult

secondary-dialtone 0

fac custom callfwd all *1

fac custom callfwd cancel *2

fac custom pickup local *3

fac custom pickup group *4

fac custom pickup direct *6

create cnf-files version-stamp Jan 01 2002 00:00:00

!

ephone-dn 1

number 7060

pickup-group 1

description Brasilia 1

corlist incoming DF

no huntstop

!

ephone-dn 2

number 7061

pickup-group 1

description Brasilia 2

name Brasilia 2

corlist incoming DF

no huntstop

!

ephone 1

no multicast-moh

device-security-mode none

mac-address 0817.3515.970B

codec g729r8

type 7942

button 1:1

!

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

do you guys have any tips?

thanks in advance,

Karl

paolo bevilacqua · ‎06-28-2011

Under sip phone, configure codec g711ulaw.

If strill troubles, take "debug ccsip message". Do not take any other debugs.

View solution in original post

paolo bevilacqua · ‎06-28-2011

Under sip phone, configure codec g711ulaw.

If strill troubles, take "debug ccsip message". Do not take any other debugs.

Karl Rogers Muller · ‎06-28-2011

Thanks buddy. It worked just perfect. Anyway, I need to change to the G729 codec. This is actually the reason because it was set to G729. I'm trying to set the codec to G729, but it's not accepting the call. It seems as the wireless sip phone do not accept this codec. I tried to force G729 through its web interface, but I got no success. When the invite is generated, it receives the G729 as the primary codec:

v=0

o=- 36243 36243 IN IP4 172.23.0.21

s=-

c=IN IP4 172.23.0.21

t=0 0

m=audio 16454 RTP/AVP 18 0 2 8 101

a=rtpmap:18 G729a/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:2 G726-32/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

a=sendrecv

I don't why it doesn't answer when the codec is set to G729. As I've just said, I forced the G729 on it already. Any tips?

The whole trace is below:

it7-rot-df-01#

*Jun 28 14:06:32.361: //2139/00F4F13E6800/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:7063@172.23.0.21:5060 SIP/2.0

Via: SIP/2.0/UDP 172.23.0.1:5060;branch=z9hG4bK982510

Remote-Party-ID: <7034>;party=calling;screen=yes;privacy=off

From: <7034>;tag=32B8ECDC-925

To: <7063>

Date: Tue, 28 Jun 2011 14:06:32 GMT

Call-ID: A92FEA17-A0C611E0-8A2189E5-29A3983@172.23.0.1

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE: 1800

Cisco-Guid: 0016052542-4057895392-1744851201-2886992562

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: 1309269992

Contact: <7034>

Expires: 180

Allow-Events: telephone-event

Content-Length: 0

*Jun 28 14:06:32.861: //2139/00F4F13E6800/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:7063@172.23.0.21:5060 SIP/2.0

Via: SIP/2.0/UDP 172.23.0.1:5060;branch=z9hG4bK982510

Remote-Party-ID: <7034>;party=calling;screen=yes;privacy=off

From: <7034>;tag=32B8ECDC-925

To: <7063>

Date: Tue, 28 Jun 2011 14:06:32 GMT

Call-ID: A92FEA17-A0C611E0-8A2189E5-29A3983@172.23.0.1

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE: 1800

Cisco-Guid: 0016052542-4057895392-1744851201-2886992562

User-Agent: Cisco-SIPGateway/IOS-12.x

it7-rot-df-01#

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Max-Forwards: 70

Timestamp: 1309269992

Contact: <7034>

Expires: 180

Allow-Events: telephone-event

Content-Length: 0

*Jun 28 14:06:33.861: //2139/00F4F13E6800/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:7063@172.23.0.21:5060 SIP/2.0

Via: SIP/2.0/UDP 172.23.0.1:5060;branch=z9hG4bK982510

Remote-Party-ID: <7034>;party=calling;screen=yes;privacy=off

From: <7034>;tag=32B8ECDC-925

To: <7063>

Date: Tue, 28 Jun 2011 14:06:33 GMT

Call-ID: A92FEA17-A0C611E0-8A2189E5-29A3983@172.23.0.1

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE: 1800

Cisco-Guid: 0016052542-4057895392-1744851201-2886992562

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: 1309269993

Contact: <7034>

Expires: 180

Allow-Events: telephone-event

Content-Length: 0

it7-rot-df-01#

*Jun 28 14:06:34.477: //2139/00F4F13E6800/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Trying

To: <7063>

From: <7034>;tag=32B8ECDC-925

Call-ID: A92FEA17-A0C611E0-8A2189E5-29A3983@172.23.0.1

CSeq: 101 INVITE

Via: SIP/2.0/UDP 172.23.0.1:5060;branch=z9hG4bK982510

Timestamp: 1309269992

Server: Cisco/WIP310-5.0.12

Content-Length: 0

*Jun 28 14:06:34.573: //2139/00F4F13E6800/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Trying

To: <7063>

From: <7034>;tag=32B8ECDC-925

Call-ID: A92FEA17-A0C611E0-8A2189E5-29A3983@172.23.0.1

CSeq: 101 INVITE

Via: SIP/2.0/UDP 172.23.0.1:5060;branch=z9hG4bK982510

Timestamp: 1309269992

Server: Cisco/WIP310-5.0.12

Content-Length: 0

*Jun 28 14:06:34.577: //2139/00F4F13E6800/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Trying

To: <7063>

From: <7034>;tag=32B8ECDC-925

Call-ID: A92FEA17-A0C611E0-8A2189E5-29A3983@172.23.0.1

CSeq: 101 INVITE

Via: SIP/2.0/UDP 172.23.0.1:5060;branch=z9hG4bK982510

Timestamp: 1309269993

Server: Cisco/WIP310-5.0.12

Content-Length: 0

it7-rot-df-01#

*Jun 28 14:06:34.577: //2139/00F4F13E6800/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 180 Ringing

To: <7063>;tag=f2885f29467bc67ei0

From: <7034>;tag=32B8ECDC-925

Call-ID: A92FEA17-A0C611E0-8A2189E5-29A3983@172.23.0.1

CSeq: 101 INVITE

Via: SIP/2.0/UDP 172.23.0.1:5060;branch=z9hG4bK982510

Timestamp: 1309269992

Server: Cisco/WIP310-5.0.12

Content-Length: 0

Allow-Events: dialog

it7-rot-df-01#

*Jun 28 14:06:39.257: //2139/00F4F13E6800/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 200 OK

To: <7063>;tag=f2885f29467bc67ei0

From: <7034>;tag=32B8ECDC-925

Call-ID: A92FEA17-A0C611E0-8A2189E5-29A3983@172.23.0.1

CSeq: 101 INVITE

Via: SIP/2.0/UDP 172.23.0.1:5060;branch=z9hG4bK982510

Timestamp: 1309269992

Contact: "7063" <7063>

Server: Cisco/WIP310-5.0.12

Content-Length: 278

Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE

Allow-Events: dialog

Supported: replaces

Content-Type: application/sdp

v=0

o=- 4372 4372 IN IP4 172.23.0.21

s=-

c=IN IP4 172.23.0.21

t=0 0

m=audio 16448 RTP/AVP 18 0 2 8 101

a=rtpmap:18 G729a/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:2 G726-32/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

a=sendrecv

*Jun 28 14:06:39.261: //2139/00F4F13E6800/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:7063@172.23.0.21:5060 SIP/2.0

Via: SIP/2.0/UDP 172.23.0.1:5060;branch=z9hG4bK991084

From: <7034>;tag=32B8ECDC-925

To: <7063>;tag=f2885f29467bc67ei0

Date: Tue, 28 Jun 2011 14:06:33 GMT

Call-ID: A92FEA17-A0C611E0-8A2189E5-29A3983@172.23.0.1

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

*Jun 28 14:06:39.261: //2139/00F4F13E6800/SIP/Msg/ccsipDisplayMsg:

Sent:

BYE sip:7063@172.23.0.21:5060 SIP/2.0

Via: SIP/2.0/UDP 172.23.0.1:5060;branch=z9hG4bK9A1AA7

From: <7034>;tag=32B8ECDC-925

To: <7063>;tag=f2885f29467bc67ei0

Date: Tue, 28 Jun 2011 14:06:33 GMT

Call-ID: A92FEA17-A0C611E0-8A2189E5-29A3983@172.23.0.1

User-Agent: Cisco-SIPGateway/IOS-12.x

Max-Forwards: 70

Timestamp: 1309269999

CSeq: 102 BYE

Reason: Q.850;cause=65

P-RTP-Stat: PS=0,OS=0,PR=0,OR=0,PL=0,JI=0,LA=0,DU=850986

Content-Length: 0

*Jun 28 14:06:39.521: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

BYE sip:7034@172.23.0.1:5060 SIP/2.0

Via: SIP/2.0/UDP 172.23.0.21:5060;branch=z9hG4bK-74e3796

From: <7063>;tag=f2885f29467bc67ei0

To: <7034>;tag=32B8ECDC-925

Call-ID: A92FEA17-A0C611E0-8A2189E5-29A3983@172.23.0.1

CSeq: 101 BYE

Max-Forwards: 70

User-Agent: Cisco/WIP310-5.0.12

Content-Length: 0

*Jun 28 14:06:39.521: //2139/00F4F13E6800/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 172.23.0.21:5060;branch=z9hG4bK-74e3796

From: <7063>;tag=f2885f29467bc67ei0

To: <7034>;tag=32B8ECDC-925

Date: Tue, 28 Jun 2011 14:06:39 GMT

Call-ID: A92FEA17-A0C611E0-8A2189E5-29A3983@172.23.0.1

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 101 BYE

Reason: Q.850;cause=65

P-RTP-Stat: PS=0,OS=0,PR=0,OR=0,PL=0,JI=0,LA=0,DU=850987

Content-Length: 0

*Jun 28 14:06:39.757: //2139/00F4F13E6800/SIP/Msg/ccsipDisplayMsg:

Sent:

it7-rot-df-01#BYE sip:7063@172.23.0.21:5060 SIP/2.0

Via: SIP/2.0/UDP 172.23.0.1:5060;branch=z9hG4bK9A1AA7

From: <7034>;tag=32B8ECDC-925

To: <7063>;tag=f2885f29467bc67ei0

Date: Tue, 28 Jun 2011 14:06:39 GMT

Call-ID: A92FEA17-A0C611E0-8A2189E5-29A3983@172.23.0.1

User-Agent: Cisco-SIPGateway/IOS-12.x

Max-Forwards: 70

Timestamp: 1309269999

CSeq: 102 BYE

Reason: Q.850;cause=65

P-RTP-Stat: PS=0,OS=0,PR=0,OR=0,PL=0,JI=0,LA=0,DU=850986

Content-Length: 0

it7-rot-df-01#

*Jun 28 14:06:44.557: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

REGISTER sip:172.23.0.1 SIP/2.0

Via: SIP/2.0/UDP 172.23.0.21:5060;branch=z9hG4bK-850f424b

From: "7063" <7063>;tag=67d4ffee3a3841bo0

To: "7063" <7063>

Call-ID: be67b269-e1527bcb@172.23.0.21

CSeq: 29138 REGISTER

Max-Forwards: 70

Contact: "7063" <7063>;expires=3600

User-Agent: Cisco/WIP310-5.0.12

P-Station-Name: ;mac=0026cb0e7775

Content-Length: 0

Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE

Allow-Events: dialog

Supported: replaces

*Jun 28 14:06:44.561: //2140/B0757C118A22/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 172.23.0.21:5060;branch=z9hG4bK-850f424b

From: "7063" <7063>;tag=67d4ffee3a3841bo0

To: "7063" <7063>

Date: Tue, 28 Jun 2011 14:06:44 GMT

Call-ID: be67b269-e1527bcb@172.23.0.21

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 29138 REGISTER

Content-Length: 0

*Jun 28 14:06:44.561: //2140/B0757C118A22/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 172.23.0.21:5060;branch=z9hG4bK-850f424b

From: "7063" <7063>;tag=67d4ffee3a3841bo0

To: "7063" <7063>;tag=32B91C88-FD6

Date: Tue, 28 Jun 2011 14:06:44 GMT

Call-ID: be67b269-e1527bcb@172.23.0.21

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 29138 REGISTER

WWW-Authenticate: Digest realm="cme",nonce="01D68B48051282DA",algorithm=MD5,qop="auth"

Content-Length: 0

*Jun 28 14:06:44.609: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

REGISTER sip:172.23.0.1 SIP/2.0

Via: SIP/2.0/UDP 172.23.0.21:5060;branch=z9hG4bK-3e7ef2f9

From: "7063" <7063>;tag=67d4ffee3a3841bo0

To: "7063" <7063>

Call-ID: be67b269-e1527bcb@172.23.0.21

CSeq: 29139 REGISTER

Max-Forwards: 70

Authorization: Digest username="7063",realm="cme",nonce="01D68B48051282DA",uri="sip:172.23.0.1",algorithm=MD5,response="703ae84a6da986b85ef77d21630fa527",qop=auth,nc=00000001,cnonce="bbd8f140"

Contact: "7063" <7063>;expires=3600

User-Agent: Cisco/WIP310-5.0.12

P-Station-Name: ;mac=0026cb0e7775

Content-Length: 0

Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE

Allow-Events: dialog

Supported: replaces

*Jun 28 14:06:44.609: //2140/B0757C118A22/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 172.23.0.21:5060;branch=z9hG4bK-3e7ef2f9

From: "7063" <7063>;tag=67d4ffee3a3841bo0

To: "7063" <7063>;tag=32B91C88-FD6

Date: Tue, 28 Jun 2011 14:06:44 GMT

Call-ID: be67b269-e1527bcb@172.23.0.21

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 29139 REGISTER

Content-Length: 0

it7-rot-df-01#

*Jun 28 14:06:44.609: //2140/B0757C118A22/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 172.23.0.21:5060;branch=z9hG4bK-3e7ef2f9

From: "7063" <7063>;tag=67d4ffee3a3841bo0

To: "7063" <7063>;tag=32B91C88-FD6

Date: Tue, 28 Jun 2011 14:06:44 GMT

Call-ID: be67b269-e1527bcb@172.23.0.21

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 29139 REGISTER

Contact: <7063>;expires=600

Expires: 600

Content-Length: 0

paolo bevilacqua · ‎06-28-2011

Try codec g729r8 or a voice class codec with the same.

Some phones do not support g.729 because it requires paid licensing.

Thanks for the nice rating and good luck!

Karl Rogers Muller · ‎06-28-2011

I tried G.729 already, but it doesn't work. If I change back to G711, it works just perfectly. The sip phone I have is a WIP310 and there is an option for G729. However, changing the phone codec options has no effect on the problem. In other hand, changing the codec options on CME does affect the codec negotiation. If I set to G711, G711 is negotiated and the call is OK. If I change to G729, the INVITE show the G729 as the first option, but the phone doesn't accept it, neither try to negotiate G711.

I really don't know buddy! It seems as this phone doesn't support G729 coming from CME!!!

paolo bevilacqua · ‎06-28-2011

Try small business forum, linksys phones do belong there.

Karl Rogers Muller · ‎06-28-2011

Hey buddy, I think I got it now. According to the datasheet, the WIP310 supports only G729ab. CME supports only G729a. As far as they are not compatible, my only option is set the codec to G711.

thanks by all the help!

Karl Rogers Muller · ‎06-29-2011

Paolo, I've got some improvement on this issue and situation is following:

calling from 7034 to 7063: the call is OK (G711 is being negotiated)

calling from 7063 to 7034: for some reason, the calling party is no answering the SDP packet sent from the destination. I attached some Wireshark stuff.

Do you understand the situation? In one way, the call is OK. In the opposite way, the RTP payload is not being negociated. I forced G711 as the primary codec on both sides.

Any tips, buddy?

Thanks in advance,
Karl

PS: Ip addresses: 7063-172.23.0.21 and 7034: 172.20.2.178. CallManager 172.20.1.102

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

Trace from the CME side (7063 station - sip phone):

*Jun 29 18:16:09.978: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:7034@172.23.0.1 SIP/2.0

Via: SIP/2.0/UDP 172.23.0.21:5060;branch=z9hG4bK-48f89efb

From: "7063" <7063>;tag=239698f741af419bo0

To: <7034>

Call-ID: 1bf3c127-5ce0b72b@172.23.0.21

CSeq: 101 INVITE

Max-Forwards: 70

Contact: "7063" <7063>

Expires: 240

User-Agent: Cisco/WIP310-5.0.13

Content-Length: 274

Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE

Allow-Events: dialog

Supported: replaces

Content-Type: application/sdp

v=0

o=- 11175 11175 IN IP4 172.23.0.21

s=-

c=IN IP4 172.23.0.21

t=0 0

m=audio 16456 RTP/AVP 8 0 18 101

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729a/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

a=sendrecv

*Jun 29 18:16:09.994: //2952/B2F541D48CF8/SIP/Msg/ccsipDisplayMsg:

Sent:

it7-rot-df-01#SIP/2.0 100 Trying

Via: SIP/2.0/UDP 172.23.0.21:5060;branch=z9hG4bK-48f89efb

From: "7063" <7063>;tag=239698f741af419bo0

To: <7034>

Date: Wed, 29 Jun 2011 18:16:09 GMT

Call-ID: 1bf3c127-5ce0b72b@172.23.0.21

CSeq: 101 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

*Jun 29 18:16:10.470: //2952/B2F541D48CF8/SIP/Msg/ccsipDisplayMsg:

Sent:

it7-rot-df-01#SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 172.23.0.21:5060;branch=z9hG4bK-48f89efb

From: "7063" <7063>;tag=239698f741af419bo0

To: <7034>;tag=38C3D524-1B0A

Date: Wed, 29 Jun 2011 18:16:09 GMT

Call-ID: 1bf3c127-5ce0b72b@172.23.0.21

CSeq: 101 INVITE

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

Allow-Events: telephone-event

Remote-Party-ID: <7034>;party=called;screen=no;privacy=off

Contact: <7034>

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

it7-rot-df-01#

*Jun 29 18:16:25.190: //2952/B2F541D48CF8/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP 172.23.0.21:5060;branch=z9hG4bK-48f89efb

From: "7063" <7063>;tag=239698f741af419bo0

To: <7034>;tag=38C3D524-1B0A

Date: Wed, 29 Jun 2011 18:16:09 GMT

Call-ID: 1bf3c127-5ce0b72b@172.23.0.21

CSeq: 101 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=47

Content-Length: 0

*Jun 29 18:16:25.230: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

it7-rot-df-01#ACK sip:7034@172.23.0.1 SIP/2.0

Via: SIP/2.0/UDP 172.23.0.21:5060;branch=z9hG4bK-48f89efb

From: "7063" <7063>;tag=239698f741af419bo0

To: <7034>;tag=38C3D524-1B0A

Call-ID: 1bf3c127-5ce0b72b@172.23.0.21

CSeq: 101 ACK

Max-Forwards: 70

Contact: "7063" <7063>

User-Agent: Cisco/WIP310-5.0.13

Content-Length: 0

Allow-Events: dialog