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6771
Views
32
Helpful
16
Replies

CUBE One-Way Audio

Go to solution
3MAD
Level 1
Level 1

‎04-30-2020 01:19 AM

 

Dears, hope you all doing well,

We have this environment (call flow):
ITSP (SIP Trunk) --> CUBE --> CUCM (5001 is AA) --> CUC (Customer Service 0 is a hunt group on CUCM extension number 1888)

 

We have one way-audio and the call will disconnect after 10-15 seconds when calling to this hunt-group, the weird thing is when I do a "debug ccsip messages" the call will work fine

 

This is the running config on my CUBE

 

!
voice-card 0
 dspfarm
 dsp services dspfarm
!
!
voice rtp send-recv
!
voice service voip
 ip address trusted list
  ipv4 172.25.227.29 255.255.255.255
  ipv4 10.141.40.233 255.255.255.255
  ipv4 10.170.16.69 255.255.255.255
 mode border-element
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 signaling forward unconditional
 fax protocol pass-through g711alaw
 sip
  early-offer forced
  no call service stop
!
voice class codec 1
 codec preference 1 g711alaw
 codec preference 2 g711ulaw
 codec preference 3 g729r8
 codec preference 4 g729br8
!
!
voice class dualtone-detect-params 1
 freq-max-power 0
 freq-min-power 13
 freq-power-twist 4
 cadence-variation 8
!
!
voice class e164-pattern-map 1
 description ##ALL INCOMING NUMBERS FROM STC##
  e164 ^01142087..
  e164 ^42087..
 !
!
voice class e164-pattern-map 2
 description ##INCOMING NUMBERS FROM CUCM##
  e164 ^01142087..
  e164 ^42087..
 !
!
voice class server-group 1
 ipv4 172.25.227.29 preference 1
 description ##CUCM##
!
!
!
!
!
voice translation-rule 2
 rule 1 /^.*/ /5001/
!
!
voice translation-rule 4
 rule 1 /^9/ //
!
!
voice translation-profile IN-FROM-STC
translate called 2
!
voice translation-profile OUT-TO-STC
translate called 4
!
!
!
!
!
!
!
interface GigabitEthernet0/0
 description ## LAN ##
 ip address 172.25.227.126 255.255.255.0
 duplex auto
 speed auto
!
!
interface GigabitEthernet0/2
 description ## STC_SIP_CIRCUIT ##
 ip address 10.170.16.70 255.255.255.252
 duplex auto
 speed auto
!
!
no ip http server
no ip http secure-server
!
ip route 10.141.40.233 255.255.255.255 10.170.16.69
ip ssh version 2
!
!
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
mgcp profile default
!
!
!
!
dspfarm profile 1 transcode universal
 codec g729abr8
 codec g729ar8
 codec g711alaw
 codec g711ulaw
 codec g729r8
 codec g722-64
 maximum sessions 10
 associate application CUBE
!
dial-peer voice 1 voip
 description ##ALL INCOMING NUMBERS FROM ITSP##
 translation-profile incoming IN-FROM-STC
 session protocol sipv2
 session target ipv4:172.25.227.29
 session transport udp
 incoming called e164-pattern-map 1
 voice-class codec 1
 voice-class sip bind control source-interface GigabitEthernet0/2
 voice-class sip bind media source-interface GigabitEthernet0/2
 dtmf-relay rtp-nte sip-kpml
 no fax-relay sg3-to-g3
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 no vad
!
dial-peer voice 2 voip
 description ##TO CUCM##
 destination-pattern 5001
 session protocol sipv2
 session transport udp
 session server-group 1
 voice-class codec 1
 voice-class sip bind control source-interface GigabitEthernet0/0
 voice-class sip bind media source-interface GigabitEthernet0/0
 dtmf-relay rtp-nte sip-kpml
 fax-relay ecm disable
 no fax-relay sg3-to-g3
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 no vad
!
dial-peer voice 3 voip
 description ##ALL INCOMING NUMBERS FROM CUCM##
 session protocol sipv2
 session transport udp
 incoming calling e164-pattern-map 2
 voice-class codec 1
 voice-class sip bind control source-interface GigabitEthernet0/0
 voice-class sip bind media source-interface GigabitEthernet0/0
 dtmf-relay rtp-nte
 fax-relay ecm disable
 no fax-relay sg3-to-g3
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 no vad
!
dial-peer voice 4 voip
 description ## OUTGOING CALL TO ITSP##
 translation-profile outgoing OUT-TO-STC
 destination-pattern 9T
 session protocol sipv2
 session target ipv4:10.141.40.233
 session transport udp
 voice-class codec 1 offer-all
 voice-class sip early-offer forced
 voice-class sip bind control source-interface GigabitEthernet0/2
 voice-class sip bind media source-interface GigabitEthernet0/2
 dtmf-relay rtp-nte cisco-rtp h245-alphanumeric h245-signal sip-kpml sip-notify
 no vad
!

Thanks in advance

0 Helpful
1 Accepted Solution

Accepted Solutions

Go to solution
Roger Kallberg
VIP
VIP
In response to 3MAD

‎05-01-2020 03:07 AM - edited ‎05-01-2020 03:21 AM

In general your dial peer configuration is a little bit messy. One thing that sticks out is that you have the IP for the CUCM as the destination on your inbound dial peer from the ITSP. This must be a mistake I assume. I would make these changes to your configuration.

ip route 0.0.0.0 0.0.0.0 172.25.227.xxx !set this to the gateway for your internal network.

voice class uri 1 sip
host ipv4:10.141.40.233
!
voice class uri 2 sip
host ipv4:172.25.227.29

 

dial-peer voice 1 voip
description ##ALL INCOMING NUMBERS FROM ITSP##
translation-profile incoming IN-FROM-STC
session protocol sipv2
session transport udp
incoming uri via 1
incoming called e164-pattern-map 1
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/2
voice-class sip bind media source-interface GigabitEthernet0/2
dtmf-relay rtp-nte cisco-rtp sip-kpml sip-notify
no fax-relay sg3-to-g3
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
no vad
!
!
dial-peer voice 10 voip
description ## OUTGOING CALL TO ITSP##
translation-profile outgoing OUT-TO-STC
destination-pattern 9T
session protocol sipv2
session target ipv4:10.141.40.233
session transport udp
voice-class codec 1 offer-all
voice-class sip early-offer forced
voice-class sip bind control source-interface GigabitEthernet0/2
voice-class sip bind media source-interface GigabitEthernet0/2
dtmf-relay rtp-nte cisco-rtp sip-kpml sip-notify
no fax-relay sg3-to-g3
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
no vad
!
dial-peer voice 2 voip
description ##ALL INCOMING NUMBERS FROM CUCM##
session protocol sipv2
session transport udp
incoming uri via 2
incoming calling e164-pattern-map 2
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0
dtmf-relay rtp-nte sip-kpml
no fax-relay sg3-to-g3
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
no vad
!

dial-peer voice 20 voip
description ##TO CUCM##
destination-pattern 5001
session protocol sipv2
session transport udp
session server-group 1
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0
dtmf-relay rtp-nte sip-kpml
no fax-relay sg3-to-g3
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
no vad
!

Keeping uneven number starting dial peers as used for ITSP and even numbered used by CUCM will help you quickly see the call flow moving forward. In my example single number means it's an inbound dial peer and double numbers means it's an outbound dial peer. Purely cosmetics, but does help to identify things more easily. 

You should also verify what port range your ITSP uses for RTP. As you have not defined it in configuration it will use the default and that might not work for your ITSP. What kind of router do you use? If it's a ISR4K the default RTP port range is 8000-48198. You can see this by running this command "sh voip rtp connections" when you have an ongoing call. The output would be similar to this.

VoIP RTP Port Usage Information:

Max Ports Available: 19999, Ports Reserved: 101, Ports in Use: 2

Port range not configured

Min Max Ports Ports Ports

Media-Address Range Port Port Available Reserved In-use

------------------------------------------------------------------------------

Global Media Pool 8000 48198 19999 101 2

------------------------------------------------------------------------------

VoIP RTP active connections :

No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP MPSS VRF

1 2908314 2908315 40098 61704 10.119.217.246 84.241.227.55 NO NA

2 2908315 2908314 40102 24578 10.82.62.10 10.64.122.49 NO

// Notice the min and max port range in the above output. You will see it is 8000-48198.



Response Signature


View solution in original post

16 Replies 16

Go to solution
3MAD
Level 1
Level 1

‎04-30-2020 04:20 AM

I was able to regenerate the issue and did a debug

 

Spoiler

SIP Call messages tracing is enabled
VGW01#
*Apr 30 10:29:24.115: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:560032888@172.25.227.126:5060 SIP/2.0

Via: SIP/2.0/UDP 172.25.227.29:5060;branch=z9hG4bK3fecbacaff7

From: <sip:5001@172.25.227.29>;tag=254760~1bfdf346-5239-45d3-85f0-26020bcb99c6-26328730

To: <sip:560032888@172.25.227.126>;tag=3419A6C8-1CB7

Date: Thu, 30 Apr 2020 10:42:51 GMT

Call-ID: 42FD8F2E-8A0411EA-B074B4DA-CB90D725@172.25.227.126

Supported: timer,resource-priority,replaces

Cisco-Guid: 1123833413-2315522538-2960045274-3415267109

User-Agent: Cisco-CUCM11.5

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

CSeq: 101 INVITE

Max-Forwards: 70

Expires: 180

Allow-Events: presence

Session-ID: 01f970f9d4f467343ab7030eab254761;remote=d41e0b8235b74f8a2d1449a2ab254760

Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=MIXED

Supported: X-cisco-srtp-fallback

Supported: Geolocation

Session-Expires: 1800;refresher=uac

Min-SE: 1800

P-Preferred-Identity: <sip:5000@172.25.227.29>

Remote-Party-ID: <sip:5000@172.25.227.29>;party=calling;screen=no;privacy=off

Contact: <sip:5001@172.25.227.29:5060>;automata

Content-Type: application/sdp

Content-Length: 245

 

v=0

o=CiscoSystemsCCM-SIP 254760 2 IN IP4 172.25.227.29

s=SIP Call

c=IN IP4 0.0.0.0

b=TIAS:64000

b=AS:64

t=0 0

m=audio 21298 RTP/AVP 0 101

a=ptime:20

a=rtpmap:0 PCMU/8000

a=inactive

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15


*Apr 30 10:29:24.123: //58974/42FC5645B06E/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 172.25.227.29:5060;branch=z9hG4bK3fecbacaff7

From: <sip:5001@172.25.227.29>;tag=254760~1bfdf346-5239-45d3-85f0-26020bcb99c6-26328730

To: <sip:560032888@172.25.227.126>;tag=3419A6C8-1CB7

Date: Thu, 30 Apr 2020 10:29:24 GMT

Call-ID: 42FD8F2E-8A0411EA-B074B4DA-CB90D725@172.25.227.126

CSeq: 101 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-15.5.3.M

Content-Length: 0

 


*Apr 30 10:29:24.123: //58973/42FC5645B06E/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:560032888@10.141.40.233:5060;user=phone;transport=udp SIP/2.0

Via: SIP/2.0/UDP 10.170.16.70:5060;branch=z9hG4bKD372626

Remote-Party-ID: <sip:5000@10.170.16.70>;party=calling;screen=no;privacy=off

From: <sip:4208730@10.170.16.70;user=phone>;tag=3419A6E4-945

To: <sip:560032888@10.141.40.233;user=phone>;tag=fsf2fb2e-CC-35

Date: Thu, 30 Apr 2020 10:29:24 GMT

Call-ID: kbttbpus2csedashoupddud4hbbs2obk@SoftX3000

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

Min-SE: 1800

Cisco-Guid: 1123833413-2315522538-2960045274-3415267109

User-Agent: Cisco-SIPGateway/IOS-15.5.3.M

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

CSeq: 101 INVITE

Max-Forwards: 70

Timestamp: 1588242564

Contact: <sip:4208730@10.170.16.70:5060>

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Length: 259

 

v=0

o=CiscoSystemsSIP-GW-UserAgent 5869 9015 IN IP4 10.170.16.70

s=SIP Call

c=IN IP4 10.170.16.70

t=0 0

m=audio 18544 RTP/AVP 0 116

c=IN IP4 10.170.16.70

a=rtpmap:0 PCMU/8000

a=rtpmap:116 telephone-event/8000

a=fmtp:116 0-15

a=ptime:20

a=inactive


*Apr 30 10:29:24.271: //58973/42FC5645B06E/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.170.16.70:5060;branch=z9hG4bKD372626

From: <sip:4208730@10.170.16.70;user=phone>;tag=3419A6E4-945

To: <sip:560032888@10.141.40.233;user=phone>;tag=fsf2fb2e-CC-35

Call-ID: kbttbpus2csedashoupddud4hbbs2obk@SoftX3000

CSeq: 101 INVITE

Timestamp: 1588242564

Contact: <sip:560032888@10.141.40.233:5060;user=phone;transport=udp>

Content-Length: 217

Content-Type: application/sdp

 

v=0

o=HuaweiSoftX3000 276661 276662 IN IP4 10.141.40.233

s=Sip Call

c=IN IP4 10.141.40.233

t=0 0

m=audio 57106 RTP/AVP 0 116

a=rtpmap:0 PCMU/8000

a=rtpmap:116 telephone-event/8000

a=fmtp:116 0-15

a=inactive


*Apr 30 10:29:24.275: //58973/42FC5645B06E/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:560032888@10.141.40.233:5060;user=phone;transport=udp SIP/2.0

Via: SIP/2.0/UDP 10.170.16.70:5060;branch=z9hG4bKD38740

From: <sip:4208730@10.170.16.70;user=phone>;tag=3419A6E4-945

To: <sip:560032888@10.141.40.233;user=phone>;tag=fsf2fb2e-CC-35

Date: Thu, 30 Apr 2020 10:29:24 GMT

Call-ID: kbttbpus2csedashoupddud4hbbs2obk@SoftX3000

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

 


*Apr 30 10:29:24.279: //58974/42FC5645B06E/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 172.25.227.29:5060;branch=z9hG4bK3fecbacaff7

From: <sip:5001@172.25.227.29>;tag=254760~1bfdf346-5239-45d3-85f0-26020bcb99c6-26328730

To: <sip:560032888@172.25.227.126>;tag=3419A6C8-1CB7

Date: Thu, 30 Apr 2020 10:29:24 GMT

Call-ID: 42FD8F2E-8A0411EA-B074B4DA-CB90D725@172.25.227.126

CSeq: 101 INVITE

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

Allow-Events: telephone-event

Remote-Party-ID: <sip:560032888@172.25.227.126>;party=called;screen=no;privacy=off

Contact: <sip:560032888@172.25.227.126:5060>

Supported: replaces

Supported: sdp-anat

Server: Cisco-SIPGateway/IOS-15.5.3.M

Require: timer

Session-Expires: 1800;refresher=uac

Supported: timer

Content-Type: application/sdp

Content-Length: 265

 

v=0

o=CiscoSystemsSIP-GW-UserAgent 4894 1461 IN IP4 172.25.227.126

s=SIP Call

c=IN IP4 172.25.227.126

t=0 0

m=audio 18546 RTP/AVP 0 101

c=IN IP4 172.25.227.126

a=inactive

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20


*Apr 30 10:29:24.279: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:560032888@172.25.227.126:5060 SIP/2.0

Via: SIP/2.0/UDP 172.25.227.29:5060;branch=z9hG4bK3fed64ef8f9d

From: <sip:5001@172.25.227.29>;tag=254760~1bfdf346-5239-45d3-85f0-26020bcb99c6-26328730

To: <sip:560032888@172.25.227.126>;tag=3419A6C8-1CB7

Date: Thu, 30 Apr 2020 10:42:51 GMT

Call-ID: 42FD8F2E-8A0411EA-B074B4DA-CB90D725@172.25.227.126

User-Agent: Cisco-CUCM11.5

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: presence

Content-Length: 0

 


*Apr 30 10:29:24.339: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
UPDATE sip:560032888@172.25.227.126:5060 SIP/2.0

Via: SIP/2.0/UDP 172.25.227.29:5060;branch=z9hG4bK3fef40fb7eaf

From: <sip:5001@172.25.227.29>;tag=254760~1bfdf346-5239-45d3-85f0-26020bcb99c6-26328730

To: <sip:560032888@172.25.227.126>;tag=3419A6C8-1CB7

Date: Thu, 30 Apr 2020 10:42:51 GMT

Call-ID: 42FD8F2E-8A0411EA-B074B4DA-CB90D725@172.25.227.126

User-Agent: Cisco-CUCM11.5

Max-Forwards: 70

Supported: timer,resource-priority,replaces

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

CSeq: 102 UPDATE

Session-ID: 00000000000000000000000000000000;remote=d41e0b8235b74f8a2d1449a2ab254760

Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=MIXED

Supported: X-cisco-srtp-fallback

Supported: Geolocation

Session-Expires: 1800;refresher=uac

Min-SE: 1800

P-Asserted-Identity: "Customer Support" <sip:1888@172.25.227.29>

Remote-Party-ID: "Customer Support" <sip:1888@172.25.227.29>;party=calling;screen=yes;privacy=off

Contact: <sip:5001@172.25.227.29:5060>;automata

Content-Length: 0

 


*Apr 30 10:29:24.339: //58974/42FC5645B06E/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 172.25.227.29:5060;branch=z9hG4bK3fef40fb7eaf

From: <sip:5001@172.25.227.29>;tag=254760~1bfdf346-5239-45d3-85f0-26020bcb99c6-26328730

To: <sip:560032888@172.25.227.126>;tag=3419A6C8-1CB7

Date: Thu, 30 Apr 2020 10:29:24 GMT

Call-ID: 42FD8F2E-8A0411EA-B074B4DA-CB90D725@172.25.227.126

Server: Cisco-SIPGateway/IOS-15.5.3.M

CSeq: 102 UPDATE

Allow-Events: telephone-event

Contact: <sip:560032888@172.25.227.126:5060>

Require: timer

Session-Expires: 1800;refresher=uac

Supported: timer

Content-Length: 0

Remote-Party-ID: <sip:560032888@172.25.227.126>;party=called;screen=no;privacy=off

 


*Apr 30 10:29:24.343: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:4208730@10.170.16.70:5060 SIP/2.0

Via: SIP/2.0/UDP 10.141.40.233:5060;branch=z9hG4bK1c7de910agpsrin30er0ch0000020.1

Call-ID: kbttbpus2csedashoupddud4hbbs2obk@SoftX3000

From: <sip:560032888@10.141.40.233;user=phone>;tag=fsf2fb2e-CC-35

To: <sip:4208730@10.170.16.70;user=phone>;tag=3419A6E4-945

CSeq: 4 OPTIONS

Accept: application/sdp

Max-Forwards: 69

Content-Length: 0

 


*Apr 30 10:29:24.343: //58973/42FC5645B06E/SIP/Msg/ccsipDisplayMsg:
Sent:
UPDATE sip:560032888@10.141.40.233:5060;user=phone;transport=udp SIP/2.0

Via: SIP/2.0/UDP 10.170.16.70:5060;branch=z9hG4bKD3965F

From: <sip:4208730@10.170.16.70;user=phone>;tag=3419A6E4-945

To: <sip:560032888@10.141.40.233;user=phone>;tag=fsf2fb2e-CC-35

Date: Thu, 30 Apr 2020 10:29:24 GMT

Call-ID: kbttbpus2csedashoupddud4hbbs2obk@SoftX3000

User-Agent: Cisco-SIPGateway/IOS-15.5.3.M

Max-Forwards: 70

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

Timestamp: 1588242564

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

CSeq: 102 UPDATE

Contact: <sip:4208730@10.170.16.70:5060>

Min-SE: 1800

Remote-Party-ID: "Customer Support" <sip:1888@10.170.16.70>;party=called;screen=yes;privacy=off

Content-Length: 0

 


*Apr 30 10:29:24.347: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:560032888@172.25.227.126:5060 SIP/2.0

Via: SIP/2.0/UDP 172.25.227.29:5060;branch=z9hG4bK3ff738094728

From: <sip:5001@172.25.227.29>;tag=254760~1bfdf346-5239-45d3-85f0-26020bcb99c6-26328730

To: <sip:560032888@172.25.227.126>;tag=3419A6C8-1CB7

Date: Thu, 30 Apr 2020 10:42:51 GMT

Call-ID: 42FD8F2E-8A0411EA-B074B4DA-CB90D725@172.25.227.126

Supported: timer,resource-priority,replaces

Cisco-Guid: 1123833413-2315522538-2960045274-3415267109

User-Agent: Cisco-CUCM11.5

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

CSeq: 103 INVITE

Max-Forwards: 70

Expires: 180

Allow-Events: presence

Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=MIXED

Supported: X-cisco-srtp-fallback

Supported: Geolocation

Session-Expires: 1800;refresher=uac

Min-SE: 1800

Session-ID: 00000000000000000000000000000000;remote=d41e0b8235b74f8a2d1449a2ab254760

P-Asserted-Identity: "Customer Support" <sip:1888@172.25.227.29>

Remote-Party-ID: "Customer Support" <sip:1888@172.25.227.29>;party=calling;screen=yes;privacy=off

Contact: <sip:5001@172.25.227.29:5060>;automata

Content-Length: 0

 


*Apr 30 10:29:24.347: //58974/42FC5645B06E/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 172.25.227.29:5060;branch=z9hG4bK3ff738094728

From: <sip:5001@172.25.227.29>;tag=254760~1bfdf346-5239-45d3-85f0-26020bcb99c6-26328730

To: <sip:560032888@172.25.227.126>;tag=3419A6C8-1CB7

Date: Thu, 30 Apr 2020 10:29:24 GMT

Call-ID: 42FD8F2E-8A0411EA-B074B4DA-CB90D725@172.25.227.126

CSeq: 103 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-15.5.3.M

Content-Length: 0

 


*Apr 30 10:29:24.395: //58973/42FC5645B06E/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.170.16.70:5060;branch=z9hG4bKD3965F

From: <sip:4208730@10.170.16.70;user=phone>;tag=3419A6E4-945

To: <sip:560032888@10.141.40.233;user=phone>;tag=fsf2fb2e-CC-35

Call-ID: kbttbpus2csedashoupddud4hbbs2obk@SoftX3000

Timestamp: 1588242564

CSeq: 102 UPDATE

Contact: <sip:560032888@10.141.40.233:5060;user=phone;transport=udp>

Content-Length: 0

 


*Apr 30 10:29:24.395: //58973/42FC5645B06E/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.141.40.233:5060;branch=z9hG4bK1c7de910agpsrin30er0ch0000020.1

From: <sip:560032888@10.141.40.233;user=phone>;tag=fsf2fb2e-CC-35

To: <sip:4208730@10.170.16.70;user=phone>;tag=3419A6E4-945

Date: Thu, 30 Apr 2020 10:29:24 GMT

Call-ID: kbttbpus2csedashoupddud4hbbs2obk@SoftX3000

Server: Cisco-SIPGateway/IOS-15.5.3.M

CSeq: 4 OPTIONS

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

Allow-Events: telephone-event

Accept: application/sdp

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

Content-Type: application/sdp

Content-Length: 468

 

v=0

o=CiscoSystemsSIP-GW-UserAgent 5869 9015 IN IP4 10.170.16.70

s=SIP Call

c=IN IP4 10.170.16.70

t=0 0

m=audio 18544 RTP/AVP 0 116

c=IN IP4 10.170.16.70

a=rtpmap:0 PCMU/8000

a=rtpmap:116 telephone-event/8000

a=fmtp:116 0-15

a=ptime:20

a=inactive

m=image 18544 udptl t38

c=IN IP4 10.170.16.70

a=T38FaxVersion:0

a=T38MaxBitRate:14400

a=T38FaxRateManagement:transferredTCF

a=T38FaxMaxBuffer:200

a=T38FaxMaxDatagram:320

a=T38FaxUdpEC:t38UDPRedundancy


*Apr 30 10:29:24.395: //58973/42FC5645B06E/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:560032888@10.141.40.233:5060;user=phone;transport=udp SIP/2.0

Via: SIP/2.0/UDP 10.170.16.70:5060;branch=z9hG4bKD3A17F

Remote-Party-ID: "Customer Support" <sip:1888@10.170.16.70>;party=calling;screen=yes;privacy=off

From: <sip:4208730@10.170.16.70;user=phone>;tag=3419A6E4-945

To: <sip:560032888@10.141.40.233;user=phone>;tag=fsf2fb2e-CC-35

Date: Thu, 30 Apr 2020 10:29:24 GMT

Call-ID: kbttbpus2csedashoupddud4hbbs2obk@SoftX3000

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

Min-S
VGW01#E: 1800

Cisco-Guid: 1123833413-2315522538-2960045274-3415267109

User-Agent: Cisco-SIPGateway/IOS-15.5.3.M

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

CSeq: 103 INVITE

Max-Forwards: 70

Timestamp: 1588242564

Contact: <sip:4208730@10.170.16.70:5060>

Expires: 180

Allow-Events: telephone-event

Content-Length: 0

 


*Apr 30 10:29:24.475: //58973/42FC5645B06E/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.170.16.70:5060;branch=z9hG4bKD3A17F

From: <sip:4208730@10.170.16.70;user=phone>;tag=3419A6E4-945

To: <sip:560032888@10.141.40.233;user=phone>;tag=fsf2fb2e-CC-35

Call-ID: kbttbpus2csedashoupddud4hbbs2obk@SoftX3000

CSeq: 103 INVITE

Timestamp: 1588242564

Contact: <sip:560032888@10.141.40.233:5060;user=phone;transport=udp>

Content-Length: 205

Content-Type: application/sdp

 

v=0

o=HuaweiSoftX3000 276661 276663 IN IP4 10.141.40.233

s=Sip Call

c=IN IP4 10.141.40.233

t=0 0

m=audio 57106 RTP/AVP 0 116

a=rtpmap:0 PCMU/8000

a=rtpmap:116 telephone-event/8000

a=fmtp:116 0-15


*Apr 30 10:29:24.479: //58974/42FC5645B06E/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 172.25.227.29:5060;branch=z9hG4bK3ff738094728

From: <sip:5001@172.25.227.29>;tag=254760~1bfdf346-5239-45d3-85f0-26020bcb99c6-26328730

To: <sip:560032888@172.25.227.126>;tag=3419A6C8-1CB7

Date: Thu, 30 Apr 2020 10:29:24 GMT

Call-ID: 42FD8F2E-8A0411EA-B074B4DA-CB90D725@172.25.227.126

CSeq: 103 INVITE

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

Allow-Events: telephone-event

Remote-Party-ID: <sip:560032888@172.25.227.126>;party=called;screen=no;privacy=off

Contact: <sip:560032888@172.25.227.126:5060>

Supported: replaces

Supported: sdp-anat

Server: Cisco-SIPGateway/IOS-15.5.3.M

Require: timer

Session-Expires: 1800;refresher=uac

Supported: timer

Content-Type: application/sdp

Content-Length: 325

 

v=0

o=CiscoSystemsSIP-GW-UserAgent 4894 1462 IN IP4 172.25.227.126

s=SIP Call

c=IN IP4 172.25.227.126

t=0 0

m=audio 18546 RTP/AVP 0 8 18 101

c=IN IP4 172.25.227.126

a=rtpmap:0 PCMU/8000

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

a=ptime:20


*Apr 30 10:29:24.495: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:560032888@172.25.227.126:5060 SIP/2.0

Via: SIP/2.0/UDP 172.25.227.29:5060;branch=z9hG4bK3ff85942f57a

From: <sip:5001@172.25.227.29>;tag=254760~1bfdf346-5239-45d3-85f0-26020bcb99c6-26328730

To: <sip:560032888@172.25.227.126>;tag=3419A6C8-1CB7

Date: Thu, 30 Apr 2020 10:42:51 GMT

Call-ID: 42FD8F2E-8A0411EA-B074B4DA-CB90D725@172.25.227.126

User-Agent: Cisco-CUCM11.5

Max-Forwards: 70

CSeq: 103 ACK

Allow-Events: presence

Session-ID: 00000000000000000000000000000000;remote=d41e0b8235b74f8a2d1449a2ab254760

Content-Type: application/sdp

Content-Length: 182

 

v=0

o=CiscoSystemsCCM-SIP 254760 4 IN IP4 172.25.227.29

s=SIP Call

c=IN IP4 172.25.227.29

t=0 0

m=audio 25006 RTP/AVP 0

a=X-cisco-media:umoh

a=ptime:20

a=rtpmap:0 PCMU/8000


*Apr 30 10:29:24.503: //58973/42FC5645B06E/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:560032888@10.141.40.233:5060;user=phone;transport=udp SIP/2.0

Via: SIP/2.0/UDP 10.170.16.70:5060;branch=z9hG4bKD3BB27

From: <sip:4208730@10.170.16.70;user=phone>;tag=3419A6E4-945

To: <sip:560032888@10.141.40.233;user=phone>;tag=fsf2fb2e-CC-35

Date: Thu, 30 Apr 2020 10:29:24 GMT

Call-ID: kbttbpus2csedashoupddud4hbbs2obk@SoftX3000

Max-Forwards: 70

CSeq: 103 ACK

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Length: 247

 

v=0

o=CiscoSystemsSIP-GW-UserAgent 5869 9016 IN IP4 10.170.16.70

s=SIP Call

c=IN IP4 10.170.16.70

t=0 0

m=audio 18544 RTP/AVP 0 116

c=IN IP4 10.170.16.70

a=rtpmap:0 PCMU/8000

a=rtpmap:116 telephone-event/8000

a=fmtp:116 0-16

a=ptime:20


*Apr 30 10:29:24.559: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:4208730@10.170.16.70:5060 SIP/2.0

Via: SIP/2.0/UDP 10.141.40.233:5060;branch=z9hG4bK1c7de910agpsrin30er0ch0000g20.1

Call-ID: kbttbpus2csedashoupddud4hbbs2obk@SoftX3000

From: <sip:560032888@10.141.40.233;user=phone>;tag=fsf2fb2e-CC-35

To: <sip:4208730@10.170.16.70;user=phone>;tag=3419A6E4-945

CSeq: 5 OPTIONS

Accept: application/sdp

Max-Forwards: 69

Content-Length: 0

 


*Apr 30 10:29:24.559: //58973/42FC5645B06E/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.141.40.233:5060;branch=z9hG4bK1c7de910agpsrin30er0ch0000g20.1

From: <sip:560032888@10.141.40.233;user=phone>;tag=fsf2fb2e-CC-35

To: <sip:4208730@10.170.16.70;user=phone>;tag=3419A6E4-945

Date: Thu, 30 Apr 2020 10:29:24 GMT

Call-ID: kbttbpus2csedashoupddud4hbbs2obk@SoftX3000

Server: Cisco-SIPGateway/IOS-15.5.3.M

CSeq: 5 OPTIONS

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

Allow-Events: telephone-event

Accept: ap
VGW01#plication/sdp

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

Content-Type: application/sdp

Content-Length: 456

 

v=0

o=CiscoSystemsSIP-GW-UserAgent 5869 9016 IN IP4 10.170.16.70

s=SIP Call

c=IN IP4 10.170.16.70

t=0 0

m=audio 18544 RTP/AVP 0 116

c=IN IP4 10.170.16.70

a=rtpmap:0 PCMU/8000

a=rtpmap:116 telephone-event/8000

a=fmtp:116 0-16

a=ptime:20

m=image 18544 udptl t38

c=IN IP4 10.170.16.70

a=T38FaxVersion:0

a=T38MaxBitRate:14400

a=T38FaxRateManagement:transferredTCF

a=T38FaxMaxBuffer:200

a=T38FaxMaxDatagram:320

a=T38FaxUdpEC:t38UDPRedundancy


VGW01#
*Apr 30 10:29:27.227: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:560032888@172.25.227.126:5060 SIP/2.0

Via: SIP/2.0/UDP 172.25.227.29:5060;branch=z9hG4bK3ff970620117

From: <sip:5001@172.25.227.29>;tag=254760~1bfdf346-5239-45d3-85f0-26020bcb99c6-26328730

To: <sip:560032888@172.25.227.126>;tag=3419A6C8-1CB7

Date: Thu, 30 Apr 2020 10:42:54 GMT

Call-ID: 42FD8F2E-8A0411EA-B074B4DA-CB90D725@172.25.227.126

Supported: timer,resource-priority,replaces

Cisco-Guid: 1123833413-2315522538-2960045274-3415267109

User-Agent: Cisco-CUCM11.5

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

CSeq: 104 INVITE

Max-Forwards: 70

Expires: 180

Allow-Events: presence

Session-ID: 00000000000000000000000000000000;remote=d41e0b8235b74f8a2d1449a2ab254760

Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=VIDEO_UNSPECIFIED

Supported: X-cisco-srtp-fallback

Supported: Geolocation

Session-Expires: 1800;refresher=uac

Min-SE: 1800

P-Asserted-Identity: "Customer Support" <sip:1888@172.25.227.29>

Remote-Party-ID: "Customer Support" <sip:1888@172.25.227.29>;party=calling;screen=yes;privacy=off

Contact: <sip:5001@172.25.227.29:5060>;automata

Content-Type: application/sdp

Content-Length: 188

 

v=0

o=CiscoSystemsCCM-SIP 254760 5 IN IP4 172.25.227.29

s=SIP Call

c=IN IP4 0.0.0.0

t=0 0

m=audio 25006 RTP/AVP 0

a=X-cisco-media:umoh

a=ptime:20

a=rtpmap:0 PCMU/8000

a=inactive


*Apr 30 10:29:27.231: //58974/42FC5645B06E/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 172.25.227.29:5060;branch=z9hG4bK3ff970620117

From: <sip:5001@172.25.227.29>;tag=254760~1bfdf346-5239-45d3-85f0-26020bcb99c6-26328730

To: <sip:560032888@172.25.227.126>;tag=3419A6C8-1CB7

Date: Thu, 30 Apr 2020 10:29:27 GMT

Call-ID: 42FD8F2E-8A0411EA-B074B4DA-CB90D725@172.25.227.126

CSeq: 104 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-15.5.3.M

Content-Length: 0

 


*Apr 30 10:29:27.231: //58973/42FC5645B06E/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:560032888@10.141.40.233:5060;user=phone;transport=udp SIP/2.0

Via: SIP/2.0/UDP 10.170.16.70:5060;branch=z9hG4bKD3C26AE

Remote-Party-ID: "Customer Support" <sip:1888@10.170.16.70>;party=calling;screen=yes;privacy=off

From: <sip:4208730@10.170.16.70;user=phone>;tag=3419A6E4-945

To: <sip:560032888@10.141.40.233;user=phone>;tag=fsf2fb2e-CC-35

Date: Thu, 30 Apr 2020 10:29:27 GMT

Call-ID: kbttbpus2csedashoupddud4hbbs2obk@SoftX3000

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

Min-SE: 1800

Cisco-Guid: 1123833413-2315522538-2960045274-3415267109

User-Agent: Cisco-SIPGateway/IOS-15.5.3.M

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

CSeq: 104 INVITE

Max-Forwards: 70

Timestamp: 1588242567

Contact: <sip:4208730@10.170.16.70:5060>

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Length: 331

 

v=0

o=CiscoSystemsSIP-GW-UserAgent 5869 9017 IN IP4 10.170.16.70

s=SIP Call

c=IN IP4 10.170.16.70

t=0 0

m=audio 18544 RTP/AVP 0 8 18 116

c=IN IP4 10.170.16.70

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=yes

a=rtpmap:116 telephone-event/8000

a=fmtp:116 0-16

a=ptime:20

a=inactive


*Apr 30 10:29:27.391: //58973/42FC5645B06E/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.170.16.70:5060;branch=z9hG4bKD3C26AE

From: <sip:4208730@10.170.16.70;user=phone>;tag=3419A6E4-945

To: <sip:560032888@10.141.40.233;user=phone>;tag=fsf2fb2e-CC-35

Call-ID: kbttbpus2csedashoupddud4hbbs2obk@SoftX3000

CSeq: 104 INVITE

Timestamp: 1588242567

Contact: <sip:560032888@10.141.40.233:5060;user=phone;transport=udp>

Content-Length: 217

Content-Type: application/sdp

 

v=0

o=HuaweiSoftX3000 276661 276664 IN IP4 10.141.40.233

s=Sip Call

c=IN IP4 10.141.40.233

t=0 0

m=audio 57106 RTP/AVP 0 116

a=rtpmap:0 PCMU/8000

a=rtpmap:116 telephone-event/8000

a=fmtp:116 0-15

a=inactive


*Apr 30 10:29:27.399: //58973/42FC5645B06E/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:560032888@10.141.40.233:5060;user=phone;transport=udp SIP/2.0

Via: SIP/2.0/UDP 10.170.16.70:5060;branch=z9hG4bKD3D2154

From: <sip:4208730@10.170.16.70;user=phone>;tag=3419A6E4-945

To: <sip:560032888@10.141.40.233;user=phone>;tag=fsf2fb2e-CC-35

Date: Thu, 30 Apr 2020 10:29:27 GMT

Call-ID: kbttbpus2csedashoupddud4hbbs2obk@SoftX3000

Max-Forwards: 70

CSeq: 104 ACK

Allow-Events: telephone-event

Content-Length: 0

 


*Apr 30 10:29:27.399: //58974/42FC5645B06E/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 172.25.227.29:5060;branch=z9hG4bK3ff970620117

From: <sip:5001@172.25.227.29>;tag=254760~1bfdf346-5239-45d3-85f0-26020bcb99c6-26328730

To: <sip:560032888@172.25.227.126>;tag=3419A6C8-1CB7

Date: Thu, 30 Apr 2020 10:29:27 GMT

Call-ID: 42FD8F2E-8A0411EA-B074B4DA-CB90D725@172.25.227.126

CSeq: 104 INVITE

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

Allow-Events: telephone-event

Remote-Party-ID: <sip:560032888@172.25.227.126>;party=called;screen=no;privacy=off

Contact: <sip:560032888@172.25.227.126:5060>

Supported: replaces

Supported: sdp-anat

Server: Cisco-SIPGateway/IOS-15.5.3.M

Require: timer

Session-Expires: 1800;refresher=uac

Supported: timer

Content-Type: application/sdp

Content-Length: 209

 

v=0

o=CiscoSystemsSIP-GW-UserAgent 4894 1463 IN IP4 172.25.227.126

s=SIP Call

c=IN IP4 172.25.227.126

t=0 0

m=audio 18546 RTP/AVP 0

c=IN IP4 172.25.227.126

a=inactive

a=rtpmap:0 PCMU/8000

a=ptime:20


*Apr 30 10:29:27.399: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:560032888@172.25.227.126:5060 SIP/2.0

Via: SIP/2.0/UDP 172.25.227.29:5060;branch=z9hG4bK3ffbe7e94e

From: <sip:5001@172.25.227.29>;tag=254760~1bfdf346-5239-45d3-85f0-26020bcb99c6-26328730

To: <sip:560032888@172.25.227.126>;tag=3419A6C8-1CB7

Date: Thu, 30 Apr 2020 10:42:54 GMT

Call-ID: 42FD8F2E-8A0411EA-B074B4DA-CB90D725@172.25.227.126

User-Agent: Cisco-CUCM11.5

Max-Forwards: 70

CSeq: 104 ACK

Allow-Events: presence

Content-Length: 0

 


*Apr 30 10:29:27.403: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:560032888@172.25.227.126:5060 SIP/2.0

Via: SIP/2.0/UDP 172.25.227.29:5060;branch=z9hG4bK3ffc4d732955

From: <sip:5001@172.25.227.29>;tag=254760~1bfdf346-5239-45d3-85f0-26020bcb99c6-26328730

To: <sip:560032888@172.25.227.126>;tag=3419A6C8-1CB7

Date: Thu, 30 Apr 2020 10:42:54 GMT

Call-ID: 42FD8F2E-8A0411EA-B074B4DA-CB90D725@172.25.227.126

Supported: timer,resource-priority,replaces

Cisco-Guid: 1123833413-2315522538-2960045274-3415267109

User-Agent: Cisco-CUCM11.5

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

CSeq: 105 INVITE

Max-Forwards: 70

Expires: 180

Allow-Events: presence

Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=VIDEO_UNSPECIFIED

Supported: X-cisco-srtp-fallback

Supported: Geolocation

Session-Expires: 1800;refresher=uac

Min-SE: 1800

Session-ID: 00000000000000000000000000000000;remote=d41e0b8235b74f8a2d1449a2ab254760

P-Asserted-Identity: "Customer Support" <sip:1888@172.25.227.29>

Remote-Party-ID: "Customer Support" <sip:1888@172.25.227.29>;party=calling;screen=yes;privacy=off

Contact: <sip:5001@172.25.227.29:5060>;video;audio;automata

Content-Length: 0

 


*Apr 30 10:29:27.403: //58974/42FC5645B06E/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 172.25.227.29:5060;branch=z9hG4bK3ffc4d732955

From: <sip:5001@172.25.227.29>;tag=254760~1bfdf346-5239-45d3-85f0-26020bcb99c6-26328730

To: <sip:560032888@172.25.227.126>;tag=3419A6C8-1CB7

Date: Thu, 30 Apr 2020 10:29:27 GMT

Call-ID: 42FD8F2E-8A0411EA-B074B4DA-CB90D725@172.25.227.126

CSeq: 105 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-15.5.3.M

Content-Length: 0

 


*Apr 30 10:29:27.403: //58973/42FC5645B06E/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:560032888@10.141.40.233:5060;user=phone;transport=udp SIP/2.0

Via: SIP/2.0/UDP 10.170.16.70:5060;branch=z9hG4bKD3EA2F

Remote-Party-ID: "Customer Support" <sip:1888@10.170.16.70>;party=calling;screen=yes;privacy=off

From: <sip:4208730@10.170.16.70;user=phone>;tag=3419A6E4-945

To: <sip:560032888@10.141.40.233;user=phone>;tag=fsf2fb2e-CC-35

Date: Thu, 30 Apr 2020 10:29:27 GMT

Call-ID: kbttbpus2csedashoupddud4hbbs2obk@SoftX3000

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

Min-SE: 1800

Cisco-Guid: 1123833413-2315522538-2960045274-3415267109

User-Agent: Cisco-SIPGateway/IOS-15.5.3.M

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

CSeq: 105 INVITE

Max-Forwards: 70

Timestamp: 1588242567

Contact: <sip:4208730@10.170.16.70:5060>

Expires: 180

Allow-Events: telephone-event

Content-Length: 0

 


*Apr 30 10:29:27.459: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:

VGW01#OPTIONS sip:4208730@10.170.16.70:5060 SIP/2.0

Via: SIP/2.0/UDP 10.141.40.233:5060;branch=z9hG4bK1c7de910agpsrin30er0ch0000030.1

Call-ID: kbttbpus2csedashoupddud4hbbs2obk@SoftX3000

From: <sip:560032888@10.141.40.233;user=phone>;tag=fsf2fb2e-CC-35

To: <sip:4208730@10.170.16.70;user=phone>;tag=3419A6E4-945

CSeq: 6 OPTIONS

Accept: application/sdp

Max-Forwards: 69

Content-Length: 0

 


*Apr 30 10:29:27.467: //58973/42FC5645B06E/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.170.16.70:5060;branch=z9hG4bKD3EA2F

From: <sip:4208730@10.170.16.70;user=phone>;tag=3419A6E4-945

To: <sip:560032888@10.141.40.233;user=phone>;tag=fsf2fb2e-CC-35

Call-ID: kbttbpus2csedashoupddud4hbbs2obk@SoftX3000

CSeq: 105 INVITE

Timestamp: 1588242567

Contact: <sip:560032888@10.141.40.233:5060;user=phone;transport=udp>

Content-Length: 205

Content-Type: application/sdp

 

v=0

o=HuaweiSoftX3000 276661 276665 IN IP4 10.141.40.233

s=Sip Call

c=IN IP4 10.141.40.233

t=0 0

m=audio 57106 RTP/AVP 0 116

a=rtpmap:0 PCMU/8000

a=rtpmap:116 telephone-event/8000

a=fmtp:116 0-15


*Apr 30 10:29:27.959: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:4208730@10.170.16.70:5060 SIP/2.0

Via: SIP/2.0/UDP 10.141.40.233:5060;branch=z9hG4bK1c7de910agpsrin30er0ch0000030.1

Call-ID: kbttbpus2csedashoupddud4hbbs2obk@SoftX3000

From: <sip:560032888@10.141.40.233;user=phone>;tag=fsf2fb2e-CC-35

To: <sip:4208730@10.170.16.70;user=phone>;tag=3419A6E4-945

CSeq: 6 OPTIONS

Accept: application/sdp

Max-Forwards: 69

Content-Length: 0

 


*Apr 30 10:29:27.959: %SIP-3-BADPAIR: Unexpected event 38 (SIPSPI_EV_CC_OPTIONS_RESP) in state 25 (SIP_STATE_MIDCALL_RECD_SUCCESS) substate 0 (SUBSTATE_NONE)
*Apr 30 10:29:28.059: //58973/42FC5645B06E/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.170.16.70:5060;branch=z9hG4bKD3EA2F

From: <sip:4208730@10.170.16.70;user=phone>;tag=3419A6E4-945

To: <sip:560032888@10.141.40.233;user=phone>;tag=fsf2fb2e-CC-35

Call-ID: kbttbpus2csedashoupddud4hbbs2obk@SoftX3000

CSeq: 105 INVITE

Timestamp: 1588242567

Contact: <sip:560032888@10.141.40.233:5060;user=phone;transport=udp>

Content-Length: 205

Content-Type: application/sdp

 

v=0

o=HuaweiSoftX3000 276661 276665 IN IP4 10.141.40.233

s=Sip Call

c=IN IP4 10.141.40.233

VGW01#
t=0 0

m=audio 57106 RTP/AVP 0 116

a=rtpmap:0 PCMU/8000

a=rtpmap:116 telephone-event/8000

a=fmtp:116 0-15


*Apr 30 10:29:28.995: //58973/42FC5645B06E/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.170.16.70:5060;branch=z9hG4bKD3EA2F

From: <sip:4208730@10.170.16.70;user=phone>;tag=3419A6E4-945

To: <sip:560032888@10.141.40.233;user=phone>;tag=fsf2fb2e-CC-35

Call-ID: kbttbpus2csedashoupddud4hbbs2obk@SoftX3000

CSeq: 105 INVITE

Timestamp: 1588242567

Contact: <sip:560032888@10.141.40.233:5060;user=phone;transport=udp>

Content-Length: 205

Content-Type: application/sdp

 

v=0

o=HuaweiSoftX3000 276661 276665 IN IP4 10.141.40.233

s=Sip Call

c=IN IP4 10.141.40.233

t=0 0

m=audio 57106 RTP/AVP 0 116

a=rtpmap:0 PCMU/8000

a=rtpmap:116 telephone-event/8000

a=fmtp:116 0-15


*Apr 30 10:29:28.995: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:4208730@10.170.16.70:5060 SIP/2.0

Via: SIP/2.0/UDP 10.141.40.233:5060;branch=z9hG4bK1c7de910agpsrin30er0ch0000030.1

Call-ID: kbttbpus2csedashoupddud4hbbs2obk@SoftX3000

From: <sip:560032888@10.141.40.233;user=phone>;tag=fsf2fb2e-CC-35

To: <sip:4208730@10.170.16.70;user=phone>;tag=3419A6E4-945

CSeq: 6 OPTIONS

Accept: application/sdp

Max-Forwards: 69

Content-Length: 0

 


VGW01#
*Apr 30 10:29:28.995: %SIP-3-BADPAIR: Unexpected event 38 (SIPSPI_EV_CC_OPTIONS_RESP) in state 25 (SIP_STATE_MIDCALL_RECD_SUCCESS) substate 0 (SUBSTATE_NONE)
VGW01#
*Apr 30 10:29:30.955: //58973/42FC5645B06E/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.170.16.70:5060;branch=z9hG4bKD3EA2F

From: <sip:4208730@10.170.16.70;user=phone>;tag=3419A6E4-945

To: <sip:560032888@10.141.40.233;user=phone>;tag=fsf2fb2e-CC-35

Call-ID: kbttbpus2csedashoupddud4hbbs2obk@SoftX3000

CSeq: 105 INVITE

Timestamp: 1588242567

Contact: <sip:560032888@10.141.40.233:5060;user=phone;transport=udp>

Content-Length: 205

Content-Type: application/sdp

 

v=0

o=HuaweiSoftX3000 276661 276665 IN IP4 10.141.40.233

s=Sip Call

c=IN IP4 10.141.40.233

t=0 0

m=audio 57106 RTP/AVP 0 116

a=rtpmap:0 PCMU/8000

a=rtpmap:116 telephone-event/8000

a=fmtp:116 0-15


*Apr 30 10:29:31.279: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:4208730@10.170.16.70:5060 SIP/2.0

Via: SIP/2.0/UDP 10.141.40.233:5060;branch=z9hG4bK1c7de910agpsrin30er0ch0000030.1

Call-ID: kbttbpus2csedashoupddud4hbbs2obk@SoftX3000

From: <sip:560032888@10.141.40.233;user=phone>;tag=fsf2fb2e-CC-35

To: <sip:4208730@10.170.16.70;user=phone>;tag=3419A6E4-945

CSeq: 6 OPTIONS

Accept: application/sdp

Max-Forwards: 69

Content-Length: 0

 


VGW01#
*Apr 30 10:29:31.279: %SIP-3-BADPAIR: Unexpected event 38 (SIPSPI_EV_CC_OPTIONS_RESP) in state 25 (SIP_STATE_MIDCALL_RECD_SUCCESS) substate 0 (SUBSTATE_NONE)
VGW01#
*Apr 30 10:29:32.963: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:10.170.16.70:5060 SIP/2.0

Via: SIP/2.0/UDP 10.141.40.233:5060;branch=z9hG4bKug0vg100909ft5fp5580.1

Call-ID: tpstho2ufp7tueac7uhhu2ptkpaocfd2@SoftX3000

From: <sip:10.141.40.233:5060>;tag=atfe4he4

To: <sip:10.170.16.70>

CSeq: 1 OPTIONS

Max-Forwards: 69

Content-Length: 0

 


*Apr 30 10:29:32.967: //58976/50E17680B076/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.141.40.233:5060;branch=z9hG4bKug0vg100909ft5fp5580.1

From: <sip:10.141.40.233:5060>;tag=atfe4he4

To: <sip:10.170.16.70>;tag=341A01D0-153A

Date: Thu, 30 Apr 2020 10:29:32 GMT

Call-ID: tpstho2ufp7tueac7uhhu2ptkpaocfd2@SoftX3000

Server: Cisco-SIPGateway/IOS-15.5.3.M

CSeq: 1 OPTIONS

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

Allow-Events: telephone-event

Accept: application/sdp

Supported: 100rel,timer,resource-p
VGW01#riority,replaces,sdp-anat

Content-Type: application/sdp

Content-Length: 167

 

v=0

o=CiscoSystemsSIP-GW-UserAgent 8173 6454 IN IP4 10.170.16.70

s=SIP Call

c=IN IP4 10.170.16.70

t=0 0

m=audio 0 RTP/AVP 18 0 8 9 4 2 15

c=IN IP4 10.170.16.70


VGW01#
*Apr 30 10:29:34.951: //58973/42FC5645B06E/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.170.16.70:5060;branch=z9hG4bKD3EA2F

From: <sip:4208730@10.170.16.70;user=phone>;tag=3419A6E4-945

To: <sip:560032888@10.141.40.233;user=phone>;tag=fsf2fb2e-CC-35

Call-ID: kbttbpus2csedashoupddud4hbbs2obk@SoftX3000

CSeq: 105 INVITE

Timestamp: 1588242567

Contact: <sip:560032888@10.141.40.233:5060;user=phone;transport=udp>

Content-Length: 205

Content-Type: application/sdp

 

v=0

o=HuaweiSoftX3000 276661 276665 IN IP4 10.141.40.233

s=Sip Call

c=IN IP4 10.141.40.233

t=0 0

m=audio 57106 RTP/AVP 0 116

a=rtpmap:0 PCMU/8000

a=rtpmap:116 telephone-event/8000

a=fmtp:116 0-15


*Apr 30 10:29:35.459: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:4208730@10.170.16.70:5060 SIP/2.0

Via: SIP/2.0/UDP 10.141.40.233:5060;branch=z9hG4bK1c7de910agpsrin30er0ch0000030.1

Call-ID: kbttbpus2csedashoupddud4hbbs2obk@SoftX3000

From: <sip:560032888@10.141.40.233;user=phone>;tag=fsf2fb2e-CC-35

To: <sip:4208730@10.170.16.70;user=phone>;tag=3419A6E4-945

CSeq: 6 OPTIONS

Accept: application/sdp

Max-Forwards: 69

Content-Length: 0

 


VGW01#
*Apr 30 10:29:35.463: %SIP-3-BADPAIR: Unexpected event 38 (SIPSPI_EV_CC_OPTIONS_RESP) in state 25 (SIP_STATE_MIDCALL_RECD_SUCCESS) substate 0 (SUBSTATE_NONE)
VGW01#
*Apr 30 10:29:38.951: //58973/42FC5645B06E/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.170.16.70:5060;branch=z9hG4bKD3EA2F

From: <sip:4208730@10.170.16.70;user=phone>;tag=3419A6E4-945

To: <sip:560032888@10.141.40.233;user=phone>;tag=fsf2fb2e-CC-35

Call-ID: kbttbpus2csedashoupddud4hbbs2obk@SoftX3000

CSeq: 105 INVITE

Timestamp: 1588242567

Contact: <sip:560032888@10.141.40.233:5060;user=phone;transport=udp>

Content-Length: 205

Content-Type: application/sdp

 

v=0

o=HuaweiSoftX3000 276661 276665 IN IP4 10.141.40.233

s=Sip Call

c=IN IP4 10.141.40.233

t=0 0

m=audio 57106 RTP/AVP 0 116

a=rtpmap:0 PCMU/8000

a=rtpmap:116 telephone-event/8000

a=fmtp:116 0-15


*Apr 30 10:29:39.423: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:560032888@172.25.227.126:5060 SIP/2.0

Via: SIP/2.0/UDP 172.25.227.29:5060;branch=z9hG4bK3ffe5389f344

From: <sip:5001@172.25.227.29>;tag=254760~1bfdf346-5239-45d3-85f0-26020bcb99c6-26328730

To: <sip:560032888@172.25.227.126>;tag=3419A6C8-1CB7

Date: Thu, 30 Apr 2020 10:42:54 GMT

Call-ID: 42FD8F2E-8A0411EA-B074B4DA-CB90D725@172.25.227.126

User-Agent: Cisco-CUCM11.5

Max-Forwards: 70

CSeq: 106 BYE

Reason: Q.850;cause=47

Session-ID: 1972727b00105000a00000af1fc0d69e;remote=d41e0b8235b74f8a2d1449a2ab254760

Content-Length: 0

 


*Apr 30 10:29:39.427: //58973/42FC5645B06E/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:560032888@10.141.40.233:5060;user=phone;transport=udp SIP/2.0

Via: SIP/2.0/UDP 10.170.16.70:5060;branch=z9hG4bKD3F1719

From: <sip:4208730@10.170.16.70;user=phone>;tag=3419A6E4-945

To: <sip:560032888@10.141.40.233;user=phone>;tag=fsf2fb2e-CC-35

Date: Thu, 30 Apr 2020 10:29:35 GMT

Call-ID: kbttbpus2csedashoupddud4hbbs2obk@SoftX3000

User-Agent: Cisco-SIPGateway/IOS-15.5.3.M

Max-Forwards: 70

Timestamp: 1588242579

CSeq: 106 BYE

Reason: Q.850;cause=16

P-RTP-Stat: PS=782,OS=124166,PR=830,OR=129524,PL=0,JI=0,LA=0,DU=29

Content-Length: 0

 


*Apr 30 10:29:39.427: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:

VGW01#SIP/2.0 200 OK

Via: SIP/2.0/UDP 172.25.227.29:5060;branch=z9hG4bK3ffe5389f344

From: <sip:5001@172.25.227.29>;tag=254760~1bfdf346-5239-45d3-85f0-26020bcb99c6-26328730

To: <sip:560032888@172.25.227.126>;tag=3419A6C8-1CB7

Date: Thu, 30 Apr 2020 10:29:39 GMT

Call-ID: 42FD8F2E-8A0411EA-B074B4DA-CB90D725@172.25.227.126

Server: Cisco-SIPGateway/IOS-15.5.3.M

CSeq: 106 BYE

Reason: Q.850;cause=16

P-RTP-Stat: PS=830,OS=129524,PR=782,OR=124166,PL=0,JI=0,LA=0,DU=29

Content-Length: 0

 


*Apr 30 10:29:39.479: //58973/42FC5645B06E/SIP/Msg/ccsipDisplayMsg:
Received:

VGW01#SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.170.16.70:5060;branch=z9hG4bKD3F1719

From: <sip:4208730@10.170.16.70;user=phone>;tag=3419A6E4-945

To: <sip:560032888@10.141.40.233;user=phone>;tag=fsf2fb2e-CC-35

Call-ID: kbttbpus2csedashoupddud4hbbs2obk@SoftX3000

Timestamp: 1588242579

CSeq: 106 BYE

Content-Length: 0

 


VGW01#
*Apr 30 10:29:42.951: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:4208730@10.170.16.70:5060 SIP/2.0

Via: SIP/2.0/UDP 10.141.40.233:5060;branch=z9hG4bK1c7de910agpsrin30er0ch0000030.1

Call-ID: kbttbpus2csedashoupddud4hbbs2obk@SoftX3000

From: <sip:560032888@10.141.40.233;user=phone>;tag=fsf2fb2e-CC-35

To: <sip:4208730@10.170.16.70;user=phone>;tag=3419A6E4-945

CSeq: 6 OPTIONS

Accept: application/sdp

Max-Forwards: 69

Content-Length: 0

 


*Apr 30 10:29:42.955: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:

VGW01#SIP/2.0 481 Call Leg/Transaction Does Not Exist

Via: SIP/2.0/UDP 10.141.40.233:5060;branch=z9hG4bK1c7de910agpsrin30er0ch0000030.1

From: <sip:560032888@10.141.40.233;user=phone>;tag=fsf2fb2e-CC-35

To: <sip:4208730@10.170.16.70;user=phone>;tag=3419A6E4-945

Date: Thu, 30 Apr 2020 10:29:42 GMT

Call-ID: kbttbpus2csedashoupddud4hbbs2obk@SoftX3000

Server: Cisco-SIPGateway/IOS-15.5.3.M

CSeq: 6 OPTIONS

Content-Length: 0

 


VGW01#
*Apr 30 10:29:53.027: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:10.170.16.70:5060 SIP/2.0

Via: SIP/2.0/UDP 10.141.40.233:5060;branch=z9hG4bKme71is30a04732bf5en0.1

Call-ID: escds72csdfa7psauhcpcp2cboufhhe2@SoftX3000

From: <sip:10.141.40.233:5060>;tag=ccck7fsk

To: <sip:10.170.16.70>

CSeq: 1 OPTIONS

Max-Forwards: 69

Content-Length: 0

 


*Apr 30 10:29:53.027: //58977/5CD662A7B078/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.141.40.233:5060;branch=z9hG4bKme71is30a04732bf5en0.1

From: <sip:10.141.40.233:5060>;tag=ccck7fsk

To: <sip:10.170.16.70>;tag=341A502C-C64

Date: Thu, 30 Apr 2020 10:29:53 GMT

Call-ID: escds72csdfa7psauhcpcp2cboufhhe2@SoftX3000

Server: Cisco-SIPGateway/IOS-15.5.3.M

CSeq: 1 OPTIONS

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

Allow-Events: telephone-event

Accept: application/sdp

Supported: 100rel,timer,resource-pr
VGW01#iority,replaces,sdp-anat

Content-Type: application/sdp

Content-Length: 167

 

v=0

o=CiscoSystemsSIP-GW-UserAgent 6982 2523 IN IP4 10.170.16.70

s=SIP Call

c=IN IP4 10.170.16.70

t=0 0

m=audio 0 RTP/AVP 18 0 8 9 4 2 15

c=IN IP4 10.170.16.70


VGW01#
*Apr 30 10:30:01.343: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:172.25.227.126:5060 SIP/2.0

Via: SIP/2.0/TCP 172.25.227.29:5060;branch=z9hG4bK40054a2054c0

From: <sip:172.25.227.29>;tag=523324653

To: <sip:172.25.227.126>

Date: Thu, 30 Apr 2020 10:43:28 GMT

Call-ID: 6bcb0d80-eaa1abd0-3954-1de319ac@172.25.227.29

User-Agent: Cisco-CUCM11.5

CSeq: 101 OPTIONS

Contact: <sip:172.25.227.29:5060;transport=tcp>

Max-Forwards: 0

Content-Length: 0

 


*Apr 30 10:30:01.343: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK

Via: SIP/2.0/TCP 172.25.227.29:5060;branch=z9hG4bK40054a2054c0

From: <sip:172.25.227.29>;tag=523324653

To: <sip:172.25.227.126>;tag=341A70A8-DC9

Date: Thu, 30 Apr 2020 10:30:01 GMT

Call-ID: 6bcb0d80-eaa1abd0-3954-1de319ac@172.25.227.29

Server: Cisco-SIPGateway/IOS-15.5.3.M

CSeq: 101 OPTIONS

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

Allow-Events: telephone-event

Accept: application/sdp

Supported: 100rel,timer,resource-priority,
VGW01#replaces,sdp-anat

Content-Type: application/sdp

Content-Length: 173

 

v=0

o=CiscoSystemsSIP-GW-UserAgent 9194 5508 IN IP4 172.25.227.126

s=SIP Call

c=IN IP4 172.25.227.126

t=0 0

m=audio 0 RTP/AVP 18 0 8 9 4 2 15

c=IN IP4 172.25.227.126


VGW01#u all
All possible debugging has been turned off
VGW01#

0 Helpful

Go to solution
Amine Nouasri
Level 3
Level 3
In response to 3MAD

‎04-30-2020 09:51 AM

Hi 3MAD,

Sorry I couldn't help in your previous post.

In your debug there is a message that stands out right before the call is disconnected:

VGW01#
*Apr 30 10:29:35.463: %SIP-3-BADPAIR: Unexpected event 38 (SIPSPI_EV_CC_OPTIONS_RESP) in state 25 (SIP_STATE_MIDCALL_RECD_SUCCESS) substate 0 (SUBSTATE_NONE)
VGW01#
*Apr 30 10:29:38.951: //58973/42FC5645B06E/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK

 

To be honest I am not sure if this is normal with your ITSP but it is worth checking if other working calls show the same message. If it is not the case, I would check if you are hitting the following bug:

https://bst.cloudapps.cisco.com/bugsearch/bug/CSCvf97230/?rfs=iqvred

 

Good Luck

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3MAD
Level 1
Level 1
In response to Amine Nouasri

‎04-30-2020 11:21 AM

Hi Amine,
No issues, thanks for your reply :)

I tried creating "voice class sip-profiles 14" from the link you provided and there's no difference in the results, the weird thing is that when we do a "debug ccsip messages" the issue will be resolved until I stop the debug or log-off the router.

 

Today I had an email from our ITSP, kindly check it also

 

Dear Emad,
Please check our analysis regarding your call forwarding scenario as per our live test :

Calling # 0506251016
Called # 011-4208730

INVITE message send to you with Calling # 0506251016 & Called # 011-4208730 from our SX HuaweiSoftx3000 Figure “1”.
Re-Invite send with Calling # 011-4208730 & Called # 0506251016 from your Cisco device “as call forwarding” Figure “2”.
UPDATE message with Calling # 011-4208730 & Called # 0506251016 and “Remote-Party-ID: "Customer Support" <sip:1888@10.170.16.70” Figure “3”.

Conclusion:-
The call is forwarded to same calling number which means there is no call forwarding to employees mobile numbers.
Need to add employees mobile numbers which need to forward the call to it in TO in the Re-Invite message.
STOP sending UPDATE message and keep forwarding the call by RE-INVITE MESSAGE.
Traces Attached for your references.

 

I'm sure that I see the employee number in my debugs!

 

Attached are the screenshots and some files from my ITSP please take a look at it also.

 

Thanks and I really appreciate your response.

Preview file
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Ayodeji Okanlawon
VIP Alumni
VIP Alumni
In response to 3MAD

‎04-30-2020 11:57 AM

I have not looked at debugs yet but when it comes to transfers on CUBE, you get a lot of RE-INVITEs and UPDATES that some ITSP cant process correctly. You can disable all this using midcall consumption on the CUBE.. Add this command to your dial-peer towards your ITSP or configure it in global mode..

voice service voip

sip

midcall-signaling passthru media-change

or dial-peer level:

dial-peer voice xx voip

voice-class sip mid-call signaling passthru media-change

 

Please rate all useful posts

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3MAD
Level 1
Level 1
In response to Ayodeji Okanlawon

‎04-30-2020 12:15 PM

Ayodeji I'm very thankful for your reply,

I did the "voice-class sip mid-call signaling passthru media-change" on the dial-peer level, we're still facing the same issue.

please note that this issue is not on the transferred calls only, all calls coming to the CUBE --> CUC IVR --> Hunt-List is having this one-way audio issue.

Thanks a lot
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Ayodeji Okanlawon
VIP Alumni
VIP Alumni
In response to 3MAD

‎04-30-2020 01:19 PM

Okay please do another test call, and enable the following

 

1. debug ccsip message

2. Enable packet capture on the cube as follows.

config t

ip traffic-export profile TAC mode capture

bidirectional

exit

 

int GigaBitEthernet X/X

ip traffic-export apply TAC size 10000000

end

 

traffic-export interface GigaBitEthernet X/X stop

traffic-export interface GigaBitEthernet X/X clear

traffic-export interface GigaBitEthernet X/X start

 

<-MAKE CALL->

 

++ stop the capture after issue has been reproduced ++

traffic-export interface GigaBitEthernet X/X stop

 

++ collect capture and send to tftp server ++

traffic-export interface GigaBitEthernet X/X copy tftp://IPaddress/Capture.pcap

 

Attache debug and capture here along with call details

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Amine Nouasri
Level 3
Level 3
In response to Ayodeji Okanlawon

‎05-01-2020 01:40 AM

Plus what Ayodeji Okanlawon suggested to do, can you tell us how is the Cube reaching your voice vlan?

I see only one route in the config, and it is for reaching your ITSP. This is assuming the Cube is in flow-through mode.

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3MAD
Level 1
Level 1
In response to Amine Nouasri

‎05-01-2020 04:46 AM

Thanks for your reply :)
currently we don't have a voice VLAN (data and voice are on the same VLAN 172.25.227.0/24, I know it's wrong)
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3MAD
Level 1
Level 1
In response to Ayodeji Okanlawon

‎05-01-2020 04:43 AM

Thanks for the reply,

I'm trying to regenerate the issue, running a debug is preventing the issue to happen, I kept the debug running and I'll upload the file once I face the issue again.

Thanks again
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Roger Kallberg
VIP
VIP
In response to 3MAD

‎05-01-2020 03:07 AM - edited ‎05-01-2020 03:21 AM

In general your dial peer configuration is a little bit messy. One thing that sticks out is that you have the IP for the CUCM as the destination on your inbound dial peer from the ITSP. This must be a mistake I assume. I would make these changes to your configuration.

ip route 0.0.0.0 0.0.0.0 172.25.227.xxx !set this to the gateway for your internal network.

voice class uri 1 sip
host ipv4:10.141.40.233
!
voice class uri 2 sip
host ipv4:172.25.227.29

 

dial-peer voice 1 voip
description ##ALL INCOMING NUMBERS FROM ITSP##
translation-profile incoming IN-FROM-STC
session protocol sipv2
session transport udp
incoming uri via 1
incoming called e164-pattern-map 1
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/2
voice-class sip bind media source-interface GigabitEthernet0/2
dtmf-relay rtp-nte cisco-rtp sip-kpml sip-notify
no fax-relay sg3-to-g3
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
no vad
!
!
dial-peer voice 10 voip
description ## OUTGOING CALL TO ITSP##
translation-profile outgoing OUT-TO-STC
destination-pattern 9T
session protocol sipv2
session target ipv4:10.141.40.233
session transport udp
voice-class codec 1 offer-all
voice-class sip early-offer forced
voice-class sip bind control source-interface GigabitEthernet0/2
voice-class sip bind media source-interface GigabitEthernet0/2
dtmf-relay rtp-nte cisco-rtp sip-kpml sip-notify
no fax-relay sg3-to-g3
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
no vad
!
dial-peer voice 2 voip
description ##ALL INCOMING NUMBERS FROM CUCM##
session protocol sipv2
session transport udp
incoming uri via 2
incoming calling e164-pattern-map 2
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0
dtmf-relay rtp-nte sip-kpml
no fax-relay sg3-to-g3
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
no vad
!

dial-peer voice 20 voip
description ##TO CUCM##
destination-pattern 5001
session protocol sipv2
session transport udp
session server-group 1
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0
dtmf-relay rtp-nte sip-kpml
no fax-relay sg3-to-g3
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
no vad
!

Keeping uneven number starting dial peers as used for ITSP and even numbered used by CUCM will help you quickly see the call flow moving forward. In my example single number means it's an inbound dial peer and double numbers means it's an outbound dial peer. Purely cosmetics, but does help to identify things more easily. 

You should also verify what port range your ITSP uses for RTP. As you have not defined it in configuration it will use the default and that might not work for your ITSP. What kind of router do you use? If it's a ISR4K the default RTP port range is 8000-48198. You can see this by running this command "sh voip rtp connections" when you have an ongoing call. The output would be similar to this.

VoIP RTP Port Usage Information:

Max Ports Available: 19999, Ports Reserved: 101, Ports in Use: 2

Port range not configured

Min Max Ports Ports Ports

Media-Address Range Port Port Available Reserved In-use

------------------------------------------------------------------------------

Global Media Pool 8000 48198 19999 101 2

------------------------------------------------------------------------------

VoIP RTP active connections :

No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP MPSS VRF

1 2908314 2908315 40098 61704 10.119.217.246 84.241.227.55 NO NA

2 2908315 2908314 40102 24578 10.82.62.10 10.64.122.49 NO

// Notice the min and max port range in the above output. You will see it is 8000-48198.



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3MAD
Level 1
Level 1
In response to Roger Kallberg

‎05-01-2020 05:18 AM - edited ‎05-01-2020 05:31 AM

Thanks I really appreciate your reply, it was very helpful, I did change our dial-peers to match yours.

 

Regarding the "What kind of router do you use?" our router is C2900-UNIVERSALK9-M (IOS: 15.0(1r)M16)

 

And this is the "sh voip rtp connections", what should I do after getting the RTP Port Range from my ITSP? I already asked about it from ITSP.

 

VGW01#sh voip rtp connections
VoIP RTP Port Usage Information:
Max Ports Available: 8091, Ports Reserved: 101, Ports in Use: 2
Min Max Ports Ports Ports
Media-Address Range Port Port Available Reserved In-use
------------------------------------------------------------------------------
Global Media Pool 16384 32766 8091 101 2
------------------------------------------------------------------------------
VoIP RTP active connections :
No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP MPSS
1 65207 65208 18856 10686 10.170.16.70 10.141.40.233 NO
2 65208 65207 18858 16414 172.25.227.126 172.25.227.30 NO
Found 2 active RTP connections

 

Thanks a lot

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Roger Kallberg
VIP
VIP
In response to 3MAD

‎05-01-2020 05:41 AM - edited ‎05-01-2020 05:47 AM

Once you know what range of ports your ITSP uses for RTP you set it with this command.

voice service voip

 rtp-port range range


The default for ISR2, aka 29xx and 39xx, is 

16384 32766

 



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3MAD
Level 1
Level 1
In response to Roger Kallberg

‎05-01-2020 05:56 AM

Thanks, I'll check and reply to you.
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3MAD
Level 1
Level 1
In response to Roger Kallberg

‎05-03-2020 04:29 AM

@Roger Kallberg wrote:

In general your dial peer configuration is a little bit messy. One thing that sticks out is that you have the IP for the CUCM as the destination on your inbound dial peer from the ITSP. This must be a mistake I assume. I would make these changes to your configuration.

    Hey, after editing my dial-peers everything is working fine now, no one-way audio, we was keeping an eye on it for the last couple of days.

 

    I would like to thank you very much for your support. And I really appreciate your time :)

 

Thanks a ton

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