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SIP Trunk Call hold fails (No Audio & No Resume) NO SDP from CUBE

Hi All,

I've got a strange issue where call holds fail with no audio and fail to resume both inbound and outbound.

I'm not seeing in the outbound SIP Message any SDP to the ITSP.

The Call flow is

SCCP Phone --> CUCM --> SIP TRUNK D/O --> CUBE --> SIP TRUNK E/O --> ITSP

CUBE Config:

voice service voip

address-hiding

allow-connections sip to sip

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none

sip

  early-offer forced

  midcall-signaling passthru

  pass-thru content sdp

dial-peer voice 20 voip

description *** Outbound dialpeer to SIP Trunk ***

translation-profile outgoing SIP-Outbound-Calls

destination-pattern 9T

session protocol sipv2

session target sip-server

voice-class codec 10

voice-class sip early-offer forced

dtmf-relay rtp-nte cisco-rtp

no vad

!

002240: *Jul 15 11:51:09.478 BST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:01234567890@XXX.XXX.XXX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bKae1b27a117
From: <sip:102929@XXX.XXX.XXX.XXX>;tag=132~4935096d-1f4c-478e-bcfc-1c6bfb163d5c-22740058
To: <sip:01234567890@XXX.XXX.XXX.XXX>;tag=23C5AF48-1BC6
Date: Mon, 15 Jul 2013 10:58:55 GMT
Call-ID: 449BB825-EC7311E2-8252EE50-1F4D0C0@XXX.XXX.XXX.XXX
Supported: 100rel,timer,resource-priority,replaces
Min-SE:  1800
Cisco-Guid: 3066954402-0110388267-3325763912-2396160765
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
P-Asserted-Identity: "TEST 1234" <sip:102929@XXX.XXX.XXX.XXX>
Remote-Party-ID: "TEST 1234" <sip:102929@XXX.XXX.XXX.XXX>;party=calling;screen=yes;privacy=off
Contact: <sip:102929@XXX.XXX.XXX.XXX:5060>
Content-Length: 0


002241: *Jul 15 11:51:09.482 BST: //232/B6CE02A2C63B/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bKae1b27a117
From: <sip:102929@XXX.XXX.XXX.XXX>;tag=132~4935096d-1f4c-478e-bcfc-1c6bfb163d5c-22740058
To: <sip:01234567890@XXX.XXX.XXX.XXX>;tag=23C5AF48-1BC6
Date: Mon, 15 Jul 2013 10:51:09 GMT
Call-ID: 449BB825-EC7311E2-8252EE50-1F4D0C0@XXX.XXX.XXX.XXX
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0


002242: *Jul 15 11:51:09.482 BST: //231/B6CE02A2C63B/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:XXX.XXX.XXX.XXX:5061 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK14D2B6
Remote-Party-ID: "TEST 1234" <sip:102929@XXX.XXX.XXX.XXX>;party=calling;screen=yes;privacy=off
From: <sip:448123456788@XXX.XXX.XXX.XXX>;tag=23C5AF64-2552
To: <sip:01234567890@XXX.XXX.XXX.XXX>;tag=5qy7ualijkt7a43t.o
Date: Mon, 15 Jul 2013 10:51:09 GMT
Call-ID: 336153-3582874724-629566@MSX9.gammatelecom.com
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 3066954402-0110388267-3325763912-2396160765
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: 1373885469
Contact: <sip:448123456788@XXX.XXX.XXX.XXX:5060>
Expires: 180
Allow-Events: telephone-event
Content-Length: 0


002243: *Jul 15 11:51:09.490 BST: //231/B6CE02A2C63B/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK14D2B6
From: <sip:448123456788@XXX.XXX.XXX.XXX>;tag=23C5AF64-2552
Server: Sippy
To: <sip:01234567890@XXX.XXX.XXX.XXX>;tag=5qy7ualijkt7a43t.o
CSeq: 101 INVITE
Call-ID: 336153-3582874724-629566@MSX9.gammatelecom.com


002244: *Jul 15 11:51:09.490 BST: //231/B6CE02A2C63B/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK14D2B6
From: <sip:448123456788@XXX.XXX.XXX.XXX>;tag=23C5AF64-2552
Server: Sippy
To: <sip:01234567890@XXX.XXX.XXX.XXX>;tag=5qy7ualijkt7a43t.o
CSeq: 101 INVITE
Call-ID: 336153-3582874724-629566@MSX9.gammatelecom.com


MKDCMBR-RTRPR01(config-dial-peer)#
002245: *Jul 15 11:51:09.990 BST: //231/B6CE02A2C63B/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK14D2B6
From: <sip:448123456788@XXX.XXX.XXX.XXX>;tag=23C5AF64-2552
Server: Sippy
To: <sip:01234567890@XXX.XXX.XXX.XXX>;tag=5qy7ualijkt7a43t.o
CSeq: 101 INVITE
Call-ID: 336153-3582874724-629566@MSX9.gammatelecom.com


MKDCMBR-RTRPR01(config-dial-peer)#
002246: *Jul 15 11:51:10.990 BST: //231/B6CE02A2C63B/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK14D2B6
From: <sip:448123456788@XXX.XXX.XXX.XXX>;tag=23C5AF64-2552
Server: Sippy
To: <sip:01234567890@XXX.XXX.XXX.XXX>;tag=5qy7ualijkt7a43t.o
CSeq: 101 INVITE
Call-ID: 336153-3582874724-629566@MSX9.gammatelecom.com

Any help would be great.

Thanks

Gus                   

1 REPLY
New Member

had the same issue alsoif i

had the same issue also

if i had mtp required checked on sip trunk it was working fine.

what i ended up doing is i unckecked the mtp required and on the cube i entered this command

 

voice service voip

  sip

    midcall-signaling block

    early-offer forced

 

originally i had it as 

voice service voip

  sip

    midcall-signaling pass-through

     early-offer forced

 

 

 

 

 

 

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