CUCM hold issue

kalashnikovsg · ‎10-24-2012

Hi,

We are using CUCM 7.1 with Cisco 2901 15.2.1T as a CUBE. There is SIP trunk between CUCM and 2901, and another SIP trunk between 2901 and SIP provider. When we want to hold/transfer external call, which is coming from SIP provider, it is terminated. Internal calls have no issues.

In captured SIP traffic I see SIP INVITE to 0.0.0.0, 200 OK, ACK and then BYE from SIP provider. I tried configure "mid-call signaling block", but there is no any changes in behavior of CUBE. How can I fix this issue?

Thanks,

Sergey

Suresh Subramanian · ‎10-25-2012

Did you try 'midcall-signaling passthru' ? what is the disconnect cause code you see in the bye message? can you post the CUBE config and the debug messages pls?

//Suresh Please rate all the useful posts.

kalashnikovsg · ‎10-25-2012

Yes, I tried "midcall-signaling passthru".

Config of CUBE:

version 15.2

voice-card 0

dsp services dspfarm

!

voice service pots

fax rate disable

!

voice service voip

ip address trusted list

ipv4 10.50.0.3 255.255.255.255

ipv4 10.50.0.0 255.255.254.0

ipv4 81.200.208.154 255.255.255.255

dtmf-interworking rtp-nte

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

no supplementary-service h450.3

supplementary-service h450.12

no supplementary-service sip moved-temporarily

no supplementary-service sip refer

redirect ip2ip

fax protocol t38 nse force version 0 ls-redundancy 2 hs-redundancy 2 fallback none

h323

emptycapability

sip

midcall-signaling passthru media-change

!

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g711alaw

codec preference 3 g729r8

codec preference 4 g729br8

!

voice class custom-cptone class1

dualtone disconnect

frequency 425

cadence 500 1000

!

voice translation-rule 7

rule 1 /2189/ /9791190/

!

voice translation-rule 9

rule 1 /^9/ //

!

voice translation-rule 10

rule 1 /.*/ /9791190/

!

voice translation-profile GOROD

translate calling 10

translate called 9

!

application

service aatest flash:aatest.vxml

!

interface Loopback0

description -==Interface for SCCP application==-

ip address 172.16.0.1 255.255.255.0

!

interface GigabitEthernet0/0

description -== WAN ==-

ip address 192.168.223.2 255.255.255.252

duplex auto

speed auto

!

interface GigabitEthernet0/1

description -== LAN ==-

ip address 10.63.0.1 255.255.255.0

ip virtual-reassembly in

duplex auto

speed auto

!

ip route 0.0.0.0 0.0.0.0 192.168.223.1

!

tftp-server flash:aatest.vxml

tftp-server flash:vxml_allagentsbusy.au

tftp-server flash:vxml_disconnect.au

tftp-server flash:vxml_invalidoption.au

tftp-server flash:vxml_secretary_branch.au

tftp-server flash:vxml_welcome_sam.au

!

control-plane

!

voice-port 0/0/0

supervisory disconnect dualtone mid-call

supervisory custom-cptone class1

cptone RU

timeouts call-disconnect 1

timeouts ringing 20

timeouts wait-release 1

timing hookflash-out 500

connection plar opx 9791190

description -==connection to PSTN1==-

caller-id enable

!

voice-port 0/0/1

supervisory disconnect dualtone mid-call

supervisory custom-cptone class1

cptone RU

timeouts call-disconnect 1

timeouts ringing 20

timeouts wait-release 1

timing hookflash-out 500

connection plar opx 9791189

description -==connection to PSTN2==-

caller-id enable

!

voice-port 0/0/2

shutdown

!

voice-port 0/0/3

shutdown

!

voice-port 0/1/0

cptone RU

description -==FAX==-

station-id name Fax

station-id number 2590

caller-id enable

!

voice-port 0/1/1

cptone RU

description -==analog phone==-

station-id name Sklad

station-id number 2550

caller-id enable

!

mgcp profile default

!

sccp local GigabitEthernet0/1

sccp ccm 10.63.0.1 identifier 2 version 7.0

sccp ccm 10.50.0.3 identifier 1 version 7.0

sccp

!

sccp ccm group 2

bind interface GigabitEthernet0/1

associate ccm 2 priority 1

associate profile 1 register mtp28940fb887d1

switchover method immediate

switchback method immediate

!

sccp ccm group 1

bind interface GigabitEthernet0/1

associate ccm 1 priority 1

associate profile 2 register XCODEsamara

!

dspfarm profile 2 transcode

codec g729abr8

codec g729ar8

codec g711alaw

codec g711ulaw

codec g729r8

codec g729br8

maximum sessions 2

associate application SCCP

!

dspfarm profile 1 transcode

codec g729br8

codec g729r8

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

maximum sessions 2

associate application SCCP

!

dial-peer voice 201 pots

destination-pattern 2590

fax rate disable

port 0/1/0

no sip-register

!

dial-peer voice 102 voip

description -==Incoming from SIP ==-

service aatest

session protocol sipv2

session target sip-server

incoming called-number 2189

voice-class sip dtmf-relay force rtp-nte

dtmf-relay sip-notify rtp-nte sip-kpml

codec g711ulaw

no vad

!

dial-peer voice 12 voip

description -==Outgoing to SIP ==-

translation-profile outgoing GOROD

preference 2

destination-pattern 9T

session protocol sipv2

session target sip-server

voice-class sip dtmf-relay force rtp-nte

dtmf-relay sip-notify rtp-nte sip-kpml

codec g711ulaw

no vad

!

dial-peer voice 202 pots

description -==Sklad Samara==-

destination-pattern 2550

port 0/1/1

no register e164

no sip-register

!

dial-peer voice 30 voip

description -==to internal numbers==-

destination-pattern [1245]...

session protocol sipv2

session target ipv4:10.50.0.3

voice-class codec 1

dtmf-relay sip-notify rtp-nte sip-kpml

no vad

!

sip-ua

credentials username 2189 password 7 133246025A595D7232121E realm 81.200.208.154:5060

authentication username 2189 password 7 1079580954424B5315321C

no remote-party-id

retry invite 3

retry register 10

timers connect 1000

registrar ipv4:81.200.208.154:5060 expires 1800

sip-server ipv4:81.200.208.154:5060

!

gatekeeper

shutdown

!

telephony-service

sdspfarm units 1

sdspfarm transcode sessions 2

sdspfarm tag 1 mtp28940fb887d1

srst mode auto-provision none

srst ephone description Cisco SRST : Jul 02 2012 12:30:08

srst dn line-mode dual-octo

fxo hook-flash

max-ephones 25

max-dn 20

ip source-address 10.63.0.1 port 2000

timeouts interdigit 5

system message AKVATORIYA TEPLA

cnf-file location flash:

user-locale RU load CME-locale-ru_RU-Russian-8.1.2.1.tar

network-locale RU

network-locale 1 RU

network-locale 2 RU

network-locale 3 RU

network-locale 4 RU

time-zone 32

time-format 24

date-format dd-mm-yy

max-conferences 4 gain -6

call-forward pattern .T

moh "music-on-hold.au"

web admin system name zva password WebAdm

dn-webedit

time-webedit

transfer-system full-consult

transfer-pattern .T

transfer-pattern 2490 blind

kalashnikovsg · ‎10-25-2012

Here is "debug ccsip mesages" and pcap dump from gateway with traffic from CUBE to SIP provider.

Suresh Subramanian · ‎10-25-2012

Hi Sergey,

I dont think the debug is complete.

If I understand correctly extn: 2189 called 4952874964 and tranffered the call to 2501. I dont see any Refer-to message in the debugs attached.

by the way, why there is "no supplementary-service sip refer"? can you try removing this and let me know how it works?

dont the service provider support Refer message?

in the debug I see CUBE received an INVITE message from CUCM with c=0.0.0.0 from the calling party 2501 and called party 4952874964. This indicates the Call Hold.

Oct 25 10:30:20.780: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:4952874964@CCM-SBC IP-10.63.0.1:5060 SIP/2.0

Date: Thu, 25 Oct 2012 10:29:30 GMT

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

From: <2501>>;tag=2baadd03-a7b4-4f20-964a-6fbc9aec0881-21804043

Allow-Events: presence, kpml

P-Asserted-Identity: "Sinjakov Vladimir" <2501>

Supported: timer,resource-priority,replaces

Min-SE: 1800

Cisco-Guid: 3075301440-0499454434-2419700941-1776400203

Remote-Party-ID: "Sinjakov Vladimir" <2501>;party=calling;screen=yes;privacy=off

Content-Length: 238

User-Agent: Cisco-CUCM7.1

To: "4952874964" <4952874964>;tag=3622FFC-164D

Contact: <2501>

Expires: 180

Content-Type: application/sdp

Call-ID: BCA49098-1DC511E2-903FB4CD-69E1B74B@CCM-SBC IP-10.63.0.1

Via: SIP/2.0/UDP CUCM=10.50.0.3:5060;branch=z9hG4bK10f1f75bbf853

CSeq: 101 INVITE

Session-Expires: 1800;refresher=uac

Max-Forwards: 70

v=0

o=CiscoSystemsCCM-SIP 2000 2 IN IP4 CUCM=10.50.0.3

s=SIP Call

c=IN IP4 0.0.0.0

t=0 0

m=audio 23380 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=ptime:20

a=inactive

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

m=video 0 RTP/AVP 34

This call leg is disconnected with cause code 86 by CUCM as below

Oct 25 10:30:52.820: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

BYE sip:4952874964@CCM-SBC IP-10.63.0.1:5060 SIP/2.0

Reason: Q.850;cause=86

Date: Thu, 25 Oct 2012 10:29:30 GMT

From: <2501>;tag=2baadd03-a7b4-4f20-964a-6fbc9aec0881-21804043

Content-Length: 0

User-Agent: Cisco-CUCM7.1

To: "4952874964" <4952874964>;tag=3622FFC-164D

Call-ID: BCA49098-1DC511E2-903FB4CD-69E1B74B@CCM-SBC IP-10.63.0.1

Via: SIP/2.0/UDP CUCM=10.50.0.3:5060;branch=z9hG4bK10f234f7afee9

CSeq: 102 BYE

Max-Forwards: 70

Oct 25 10:30:52.824: //119165/B74D60409039/SIP/Msg/ccsipDisplayMsg:

Sent:

BYE sip:4952874964@PSTN=81.200.208.154:5060 SIP/2.0

Via: SIP/2.0/UDP CUBE=192.168.223.2:5060;branch=z9hG4bK48842583

From: <2189>;tag=3620D08-10AA

To: "4952874964" <4952874964>;tag=as0de60ebd

Date: Thu, 25 Oct 2012 10:29:51 GMT

Call-ID: 00b2276a7d11631d35dc6aff56af3e4e@PSTN=81.200.208.154

User-Agent: Cisco-SIPGateway/IOS-15.2(1)T1,

Max-Forwards: 70

Timestamp: 1351161052

CSeq: 102 BYE

Reason: Q.850;cause=16

P-RTP-Stat: PS=2011,OS=321760,PR=3645,OR=579144,PL=0,JI=0,LA=0,DU=73

Content-Length: 0

//Suresh Please rate all the useful posts.

kalashnikovsg · ‎10-29-2012

Hi sureshsub2,

This is debug of failed hold. When call is placed on Hold, there is no any refer-to messages. May be I am wrong. I don't understand why after receiving INVITE with c=0.0.0.0, GW receives BYE message from CUCM.

Concerning incomplete debug, before making call, I run "debug ccsip messages" on GW. All messages were captured by router's buffer. I don't think that I missed any messages.

I can try to remove "no supplementary-service sip refer". But if I am not mistaken, this command take a part only during call transfer/forward.

May be it is better to capture debug of failed transfer?

Suresh Subramanian · ‎10-30-2012

Hi Sergey,

The Refer-To message is only for Call transfer not for Call Hold. The phone which put the call on hold will send the INVITE message with c=0.0.0.0 or a=sendonly.

From the given debugs, I could see two phones involved. one is 2189 and another one is 2501.

The phone with extn 2501 put the call of 4952874964 on Hold. CUCM sent the INVITE with c=0.0.0.0 to CUBE and then CUCM sends BYE message with cause code: 86. It indicates that the network has received a call identity information element indicating a suspended call that has in the meantime been cleared wile suspended.

http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/voice_troubleshooting/old/vts_appa.html#wp1007443

The BYE message sent from CUCM to CUBE is

Oct 25 10:30:52.820: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

BYE sip:4952874964@CCM-SBC IP-10.63.0.1:5060 SIP/2.0

Reason: Q.850;cause=86

Date: Thu, 25 Oct 2012 10:29:30 GMT

From: <2501>;tag=2baadd03-a7b4-4f20-964a-6fbc9aec0881-21804043

Content-Length: 0

User-Agent: Cisco-CUCM7.1

To: "4952874964" <4952874964>;tag=3622FFC-164D

Call-ID: BCA49098-1DC511E2-903FB4CD-69E1B74B@CCM-SBC IP-10.63.0.1

Via: SIP/2.0/UDP CUCM=10.50.0.3:5060;branch=z9hG4bK10f234f7afee9

CSeq: 102 BYE

Max-Forwards: 70

But the CUBE sent another BYE message to PSTN with calling party number as 2901

Oct 25 10:30:52.824: //119165/B74D60409039/SIP/Msg/ccsipDisplayMsg:

Sent:

BYE sip:4952874964@PSTN=81.200.208.154:5060 SIP/2.0

Via: SIP/2.0/UDP CUBE=192.168.223.2:5060;branch=z9hG4bK48842583

From: <2189>>;tag=3620D08-10AA

To: "4952874964" <4952874964>>;tag=as0de60ebd

Date: Thu, 25 Oct 2012 10:29:51 GMT

Call-ID: 00b2276a7d11631d35dc6aff56af3e4e@PSTN=81.200.208.154

User-Agent: Cisco-SIPGateway/IOS-15.2(1)T1,

Max-Forwards: 70

Timestamp: 1351161052

CSeq: 102 BYE

Reason: Q.850;cause=16

P-RTP-Stat: PS=2011,OS=321760,PR=3645,OR=579144,PL=0,JI=0,LA=0,DU=73

Content-Length: 0

The debug was from failed CALL HOLD? or Did the phone try to TRANSFER the call?

Is it possible to collect CUCM traces as well as debug ccsip all for a failed call again?

may be, we will come to know what is happening between CUCM and CUBE.

Let me know if you have any questions please

//Suresh Please rate all the useful posts.

kalashnikovsg · ‎10-30-2012

This debugs is CALL HOLD

Call flow is follow: SIP provider - IVR (2189) on CUBE - dial by extention (2501) - call hold

If you analyze pcap dump, you will see that BYE message was sent by SIP provider. It is very strange.

Also if I use "midcall-signaling passthru", CUBE should supress INVITE messages with c=0.0.0.0 to SIP provider.

I think that it will be difficult to callect loges from CUCM. It is simpler to capture traffic from CUCM to CUBE.

Suresh Subramanian · ‎10-30-2012

so you have configured AA in CUBE and that AA will transfer the calls to IP phones registered in CUCM.

Lets say, instead of direct internal calls, if one IP phone-A calls AA number (2189) and if AA transfers the call to another IP phone-B, is that working fine?

//Suresh Please rate all the useful posts.

Suresh Subramanian · ‎10-30-2012

if the internal calls thru AA works fine, can you pls collect debug ccsip all and the packet capture for the external call issue?The previous pcap file was captured on 24th Oct and debug on 25th Oct.

To collect the detailed CCM SDI/SDL traces, please follow the link.

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a0080094e89.shtml#calm

also before collecting the debugs, can you pls try removing "no supplementary-service sip refer" and let me know how it works?

Message was edited by: Sureshsub2

//Suresh Please rate all the useful posts.

kalashnikovsg · ‎10-31-2012

I put command "supplementary-service sip refer", AA can't transfer any call.

I will try to collect. I have a little problem. If I turn "debug ccsip all" my router freeze. Of couse I collect debug only to buffer.

Ayodeji Okanlawon · ‎10-31-2012

Looking at the logs the CUCM is sending a bye to the CUBE. Hence CUCM is disconnecting the call. You need to look at CUCM logs to see whats going on..

If you pull CUCM traces and attach it here I can look at it for you

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

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kalashnikovsg · ‎10-31-2012

Therse are debuges and logs.

I called from 4952874964 to AA (2189) and then I was transfered by AA to 2599. After that I asked to transfer me. Call was terminated.

Ayodeji Okanlawon · ‎10-31-2012

The calls do not exist in the CUCM trace. Please send the correct traces. Ensure you have detailed tracing enabled on your cucm. Ensure you are taking the trace from the CUCM server that the IP phone is registered to

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

Please rate all useful posts

kalashnikovsg · ‎10-31-2012

Are you sure?

File "SDL001_100_000217.txt". Try to use ctrl-f "2599".