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one way audio issue after transfer the PSTN call

We are facing one way audio issue after transfering the call.

CUCM version 14.0. Its centralised deployment cluster over WAN.

There are 3 sites. site A, Site B and Site C. Site A has 4331 VG router. its working fine. but site B and C are having 2921 routers and have one-way audio issue at both sites. 


SITE A router - 4331

SITE B and C router - 2921


I cannot understand whats happening.

CUCM  -to(SIP TRUNK)- VG -to- PRI line

SIP trunk is configured between CUCM and VG. 


From PSTN call made to reception. reception transfer the call to Phone A . Phone A user picks up the call. Phone A user starts to talk and PSTN side user can listen to his audio. but Phone A user cannot listen voice from PSTN user.



Nithin Eluvathingal
VIP Mentor

For one way audio. First  thing is to check is the routing 


Also  check the codec 

Response Signature



I think u didn't understand issue.


All pstn calls are working fine. Once they transfer the call to another phone then it has one way audio issue.

All ip phones are in same voice vlan.


If it's codec or routing issue then it should have problem in the first receiving call itself. Problem is facing only if they transfer call to another phone.


Cisco tac couldn't solve the issue yet. All they said phone side no rtp packets from voice gateway. That we also understand before they said. Why only phone call get  transfered this issue happening.




kindly note no firewall beween voice gateway and IP Phone network. IP phones are in same vlan subnet range. 


But there is firewall between sites.



The issues may be related to SIP mid call invite. Can you see your SIP re-invite configuration (Mid-call Signaling Passthrough  )on the gateway?

Aslo, can you tell about yout MTP configuration on your CUCM (SIP trunk to VG)? try with MTP required or best effort.






In SIP trunk its already was in best effort. it was one way audio after transfering call to other phone.


then i enabled MTP. result was same. 


now I changed it toe madatory option.


I didnt understand the part Mid-call Signaling Passthrough in gateway.


Remember one way audio is happening after transfering the call . The initial call made is PSTN call. there is no problem with incoming PSTN or outgoing call. the issue happening only after transfering call and transfered cal lcannot hear audio from PSTN side . ie from gateway to transfered phone no audio. but PSTN side they can hear the audio .


As I mentioned we are facing issue with 2921 voice gateway sites only. there is no issue with 4331 voice gateway.


Hi Jaheshkhan, 


Could you please share your gateway configuration here?  The gateway requires you to change the endpoint details when you transfer the call. So we may have to check the gateway's ability to change the information in for transferred calls. Also, can you share the IOS version of the voice gateway?



Version details are as follows:


Cisco IOS Software, C2900 Software (C2900-UNIVERSALK9-M), Version 15.4(3)M2, RELEASE SOFTWARE (fc2)
Technical Support:
Copyright (c) 1986-2015 by Cisco Systems, Inc.
Compiled Fri 06-Feb-15 17:29 by prod_rel_team


System image file is "flash0:c2900-universalk9-mz.SPA.154-3.M2.bin"


so you mean do we need to upgrade the firmware?


We are not facing the issue with 4331 router.


"The gateway requires you to change the endpoint details when you transfer the call." how to change the endpoint details you mean. I didnt get that point.
So according to you, is Gateway still sending RTP packet to the first phone instead of transferred phone?

Before we try with the firmware, could you please check a few more points as others mentioned in the post? 


  1. Send send-receive SDP in mid-call INVITE  option in Trunk SIP profile - check the current state, test the call with enabling this if it is unchecked. 
  2. Check the SIP global configuration in the VG try with the configuring "mid-call signalling passthrough" under sip. 

If the calls are still not working, please share the running configuration from the VG and the debugs output.




When you transfer a call in CUCM the call is first muted, meaning an Invite (or you may call it a re-Invite) is sent to the gateway stopping media flow ("a=inactive").  Then when the transfer goes through another (re)Invite is sent with the new end point address and port.  I can't remember whether that sets media active at the same time, or whether that's a third (re)Invite.

The point about mid-call passthrough is whether these changes are passed through to the service provider.  In most cases you don't want to, as the change in endpoint within your network is not relevant to them, and it works perfectly well if their call leg continues with two-way audio while this is happening.

So according to you , rtp packets are send to first phone instead of transferred phone? or as you mentioned when the transfer happens, when another (re)Invite is sent with the new end point address and port Gateway keep mom without knowing what to do or where to send and drops the packets there?

I'm not quite clear what you're asking, but if it is a Transferred call as opposed to diverted, then yes RTP is set to the first phone because at that stage the system has no way of knowing that the call is going to be transferred.  Then went the transfer completes, the re-Invite to the gateway changes in internal end point to the new phone, and RTP now passes directly between those.

Can you grab some gateway debugs, and paste them here as a text file?  "debug ccsip mess" and "debug isdn q931"?

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