06-27-2011 12:41 PM - edited 03-16-2019 05:39 AM
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
Solved! Go to Solution.
06-28-2011 03:27 AM
Under sip phone, configure codec g711ulaw.
If strill troubles, take "debug ccsip message". Do not take any other debugs.
06-28-2011 03:27 AM
Under sip phone, configure codec g711ulaw.
If strill troubles, take "debug ccsip message". Do not take any other debugs.
06-28-2011 07:03 AM
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#
it7-rot-df-01#
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=off7034>
From: <7034>;tag=32B8ECDC-9257034>
To: <7063>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>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=off7034>
From: <7034>;tag=32B8ECDC-9257034>
To: <7063>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>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=off7034>
From: <7034>;tag=32B8ECDC-9257034>
To: <7063>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>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>7063>
From: <7034>;tag=32B8ECDC-9257034>
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>7063>
From: <7034>;tag=32B8ECDC-9257034>
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>7063>
From: <7034>;tag=32B8ECDC-9257034>
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=f2885f29467bc67ei07063>
From: <7034>;tag=32B8ECDC-9257034>
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=f2885f29467bc67ei07063>
From: <7034>;tag=32B8ECDC-9257034>
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>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-9257034>
To: <7063>;tag=f2885f29467bc67ei07063>
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-9257034>
To: <7063>;tag=f2885f29467bc67ei07063>
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=f2885f29467bc67ei07063>
To: <7034>;tag=32B8ECDC-9257034>
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=f2885f29467bc67ei07063>
To: <7034>;tag=32B8ECDC-9257034>
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-9257034>
To: <7063>;tag=f2885f29467bc67ei07063>
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=67d4ffee3a3841bo07063>
To: "7063" <7063>7063>
Call-ID: be67b269-e1527bcb@172.23.0.21
CSeq: 29138 REGISTER
Max-Forwards: 70
Contact: "7063" <7063>;expires=36007063>
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=67d4ffee3a3841bo07063>
To: "7063" <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=67d4ffee3a3841bo07063>
To: "7063" <7063>;tag=32B91C88-FD67063>
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=67d4ffee3a3841bo07063>
To: "7063" <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=36007063>
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=67d4ffee3a3841bo07063>
To: "7063" <7063>;tag=32B91C88-FD67063>
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=67d4ffee3a3841bo07063>
To: "7063" <7063>;tag=32B91C88-FD67063>
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=6007063>
Expires: 600
Content-Length: 0
06-28-2011 10:26 AM
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!
06-28-2011 10:36 AM
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!!!
06-28-2011 10:41 AM
Try small business forum, linksys phones do belong there.
06-28-2011 10:55 AM
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!
06-29-2011 11:29 AM
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=239698f741af419bo07063>
To: <7034>7034>
Call-ID: 1bf3c127-5ce0b72b@172.23.0.21
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "7063" <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=239698f741af419bo07063>
To: <7034>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=239698f741af419bo07063>
To: <7034>;tag=38C3D524-1B0A7034>
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=off7034>
Contact: <7034>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=239698f741af419bo07063>
To: <7034>;tag=38C3D524-1B0A7034>
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=239698f741af419bo07063>
To: <7034>;tag=38C3D524-1B0A7034>
Call-ID: 1bf3c127-5ce0b72b@172.23.0.21
CSeq: 101 ACK
Max-Forwards: 70
Contact: "7063" <7063>7063>
User-Agent: Cisco/WIP310-5.0.13
Content-Length: 0
Allow-Events: dialog
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