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No Audio when forwarding calls out FXO

Unanswered Question
Oct 15th, 2010
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We have a number of DID's associated with a PRI at one of our sites.  These are assigned to internal DNs with translation patterns.  Calls coming into these DID's work fine.

I have one customer at a second office who tried to forward calls to their DID to a cell number.  The calls appear to foward - the target phone rings, however when the call is answered neither party have any audio.  The calling party also does not hear any ringing.

The forwarded calls are coming in from the PRI but going out through a local FXO.  If I force the forwarded calls to go back out the PRI with a route pattern, both caller and called party can hear each other and the forwarding is successful.

If I revert back to the calls going out via FXO, there is no audio.

Normal dialed calls out these FXO ports work without issue, and internal forwarded calls are also fine.

In summary:

PRI @ Site A ---> WAN --->  3825 @ Site B --->  Call Forwarded ---> FXO on 3825 @ Site B ---> Call goes through but no audio.

PRI @ Site A ---> WAN ---> 3825 @ Site B ---> Call Forwarded ---> WAN ---> PRI @ Site A ---> Call goes through and audio is fine.

Any ideas?

CUCM 7.1.3

The voice gateways for both the PRI ard the FXO are both 3825's.

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frlindse Fri, 10/15/2010 - 11:16
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Could you get a couple of copies of "show call active voice brief"

(to see if counters are incrementing) from both gateways while the call is up?  Are these MGCP or H.323 gateways?


Steven Holl Fri, 10/15/2010 - 11:49
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To start, bring a call up with the issue and get this output while the call is up:

sh call active voice br

Get it from each side.  Look at the tx and rx fields to see if RTP is being transmitted and received.  It may be a routing/network issue.

Depending what we see there, you may want to also get signaling debugs.  What VoIP protocol are you running?

rcoote5902_2 Fri, 10/15/2010 - 12:11
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TX and RX are incrementing on both sides.  These are MGCP gateways.

rcoote5902_2 Fri, 10/15/2010 - 12:14
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I just tried something.  We are using G.711 for internal calls, but site to site using G.729 to save on wan bandwidth.  I switched to G.711 and we now have audio.

frlindse Fri, 10/15/2010 - 12:34
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Aaahhh... Transcoding. That's one effect of codec mismatch, the other is the call drops when the person answers.

Frank Lindsey

Engineer.customer Support

Cisco Unified Communication Infrastructure

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Phone: +1 469 255 6556

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Cisco Systems, Inc.

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