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Sue Fallon
Beginner

Dead air when putting a call on hold and transferring calls. 11.5.1

Anyone help with this problem please?

1. Call incoming (mobile etc) > Extension/Hunt Group. 

Put on hold. No MOH played just dead air. After 2 mins the call is disconnected.

 

2. Call incoming (mobile etc) > Extension/Hunt Group.

Trying to transfer the call to another internal extension. Dead air.

 

Cisco Call Manager v11.5.1.14900-11 > {SIP} > Cisco CUBE > {SIP} > ITSP

 

Recently we have moved to SIP (Gamma via Daisy) from ISDN but this seems to be new so not sure if that is related.

Help!

5 REPLIES 5
Adam Pawlowski
VIP Advocate

Are you maybe trying to invite the call to a MOH stream that won’t work because the CUBE is trying to pass it through ? That’s my only guess that you’d have to make sure the CUBE is doing address hiding or similar, otherwise the call may be failing out. 

 

Best guess off the top of my head. Or you’re running a codec that it can’t run and you’re not transcoding. 

Mohammed al Baqari
VIP Advisor

First check if the CUCM allocate MOH when you place the call onhold using
the CLI command admin:show perf query class "Cisco MOH Device" (apply it on
each node)

Check the counters and see if MOH is allocated (whether unitcast or
multicast)

Also, check the codec between MOH and CUBE dialpeer.

If both are fine, share the debugs in CUBE (debug ccsip message) while
placing the call on hold


This is the result to the CLI command:

 

admin:show perf query class "Cisco MOH Device"
==>query class :

- Perf class (Cisco MOH Device) has instances and values:
no values are returned

 

Debug of a call -

To: <sip:****Internal Extension Number*******@88.215.60.***:5060;user=phone>
From: <sip:*****Mobile Number******@88.215.60.***;user=phone>;tag=3748086009-785805
P-Asserted-Identity: <sip:*****Mobile Number******@88.215.60.***;user=phone>
Call-ID: 878258-3748086009-785799@MSX170.gammatelecom.com
CSeq: 1 INVITE
Allow: UPDATE,PRACK,INFO,NOTIFY,OPTIONS,BYE,INVITE,ACK,CANCEL
Via: SIP/2.0/UDP 88.215.60.***:5060;branch=z9hG4bK4739a0a83eafdf61246bec88f97a106c
Contact: <sip:*****Mobile Number******@88.215.60.***:5060>
Content-Type: application/sdp
Accept: application/sdp
Content-Length: 244

v=0
o=MSX170 8471867 8471867 IN IP4 88.215.60.***
s=sip call
c=IN IP4 88.215.60.***
t=0 0
a=sendrecv
m=audio 35634 RTP/AVP 8 101
b=RR:3000
b=RS:1000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

Can you please post the full debug of the call, from the Initial Invite (Call Establishment) to the Disconnect? As the debug you posted only has the invite, and does not contain enough info to diagnose the issue.  Thank you.

 

 

 

 

 

 

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