Putting Call on Hold breaks Audio Stream

[fieryhail]

Interesting situation I'm running into.  I have a SIP trunk from bandwidth.com coming into a CUBE and then from there to CUCM 8 via h.323.  I also have FXO on the same h.323 gateway and those calls are all working perfectly.  When a call comes into the SIP DID, it goes to CCM just fine and the call quality is decent.  However, when I put the call on hold from the IP Phone (7970, SCCP), the PSTN caller hears MoH and that's fine.  When I take the call off hold, the call still shows as being "connected"  on the IP Phone display but there is no audio.  After a few seconds the call disconnects.  Any ideas why this would happen?  I'm stymied.  Thanks in advance.  Also, any ccsip debugging or diag info that would assist, please let me know what would help and I'll supply it as attachments.  Thanks again.

[srgudava]

Hi

Can you please give the CUBE configuration you have under Voice service Voip ?

Sri Gudavalli

[fieryhail]

Here is my voice-service config.  I am also running CUBE 12.4.15T(8).  The lastest my router can support 3745 router.

voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
h323
  no h225 timeout keepalive
  no call service stop
sip

Also, I'm running as I stated earlier, H.323 to CUCM.  Is there a way to transport the calls via SIP from the CUBE to the CUCM?  And would there be any advantages in this if it is possible?  Thanks again.

An edit to my original post, the caller on the PSTN side does NOT hear MoH.  They hear a "static/clicking sound".  Maybe a Media Resource issue on CUCM also?

Here are the ccsip debug ourput also, the first section is when the call is established and active prior to the call being placed on hold or transferred.:

The Call Setup Information is:
Call Control Block (CCB) : 0x67B7AE98
State of The Call        : STATE_ACTIVE
TCP Sockets Used         : NO
Calling Number           : 5555551825
Called Number            : 5552819314
Source IP Address (Sig  ): 96.xx.xx.xx
Destn SIP Req Addr:Port  : 216.82.224.202:5060
Destn SIP Resp Addr:Port : 216.82.224.202:5060
Destination Name         : 216.82.224.202

RCS-3745(config)#
005251: *Jan 31 19:58:08.340 GMT: //1408/AB9E2E8F80D9/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g711ulaw
Negotiated Codec Bytes   : 160
Nego. Codec payload      : 0 (tx), 0 (rx)
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media): 96.xx.xx.xx
Source IP Port    (Media): 17282
Destn  IP Address (Media): 63.215.27.125
Destn  IP Port    (Media): 60232
Orig Destn IP Address:Port (Media): 0.0.0.0:0

This is the output once the call is put on hold/transferred and ultimately disconnected.

RCS-3745(config)#
005252: *Jan 31 19:58:40.752 GMT: //1408/AB9E2E8F80D9/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x67B7AE98
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 5555551825
Called Number            : 5552819314
Source IP Address (Sig  ): 96.xx.xx.xx
Destn SIP Req Addr:Port  : 216.82.224.202:5060
Destn SIP Resp Addr:Port : 216.82.224.202:5060
Destination Name         : 216.82.224.202

RCS-3745(config)#
005253: *Jan 31 19:58:40.752 GMT: //1408/AB9E2E8F80D9/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g711ulaw
Negotiated Codec Bytes   : 160
Nego. Codec payload      : 0 (tx), 0 (rx)
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media): 96.xx.xx.xx
Source IP Port    (Media): 17282
Destn  IP Address (Media): 63.215.27.125
Destn  IP Port    (Media): 60232
Orig Destn IP Address:Port (Media): 0.0.0.0:0

005254: *Jan 31 19:58:40.752 GMT: //1408/AB9E2E8F80D9/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 47
Disconnect Cause (SIP)   : 200

[srgudava · Accepted solution]

Hi

Can you please add this line to the Configuration on the Cube

"midcall-signaling passthru"

====

voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
h323
  no h225 timeout keepalive
  no call service stop
sip

  midcall-signaling passthru

====

Please test the call and let me know.

HTH

Sri Gudavalli

[fieryhail]

Thank you for the response, I had found a Cisco document that had refernced this, however when I tried to insert the command, I received an "invalid input detected", the command must not be present in my IOS/CUBE version.  Any other thoughts?

Additional research I have been doing, seems to indicate that feature is only available in CUBE 12.4(20)T and beyond, unfortunately, not useable on my router.  Also, I have read that if the midcall-signaling passthru is not enabled than an MTP has to be setup on the SIP trunk from CUCM to the CUBE.  This also infers that I can setup a SIP trunk from CUCM to CUBE, and if so, then that may well solve my issue, by just setting an MTP for said SIP trunk from CUCM to CUBE.  Am I correct in this, and if so, do you have any suggestions on how to accomplish this?  Thanks again Sir.

[Christopher Graham · Accepted solution]

midcall-signaling passthru was integrated in 12.4(20)T, so you will need to upgrade to use this command. 

Please upgrade and add the command and test.  If the issue persists, it would tell us more if you could grab 'debug ccsip mess' during the issue.

thanks

chris graham

[fieryhail]

Just wanted to reply with an update.  I was able to get the call hold and call transfers working without the use of that command.  What I did was:

voice service voip

no supplementary-service sip move-temporarily

no supplementary service sip refer

and with that, calls can be placed on hold as well as transferred without dropping out.  Not sure but thought this may possibly help someone else experiencing a similar issue with SIP trunks to CUCM via CUBE.  Thanks again for all the help guys!