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Calls from an analog FXS port through a CCME router over SIP trunk have a one-way audio issue

Core Issue

The issue is the inability to make calls from a Foreign Exchange Station (FXS) port through a Cisco CallManager Express Session Initiation Protocol (SIP) gateway over an SIP trunk. Calls from the analog phone through the Cisco CallManager Express router out over the SIP trunk have one-way audio. The Cisco CallManager Express analog caller cannot hear audio from the destination phone. However, registered IP phones on the Cisco CallManager Express gateway work, and the analog phones can even call the IP phones on the LAN.

This is the call flow for this scenario:

Analog phone > FXS > Cisco CallManager Express Router >(Internet Service Provider [ISP] cloud > ISP SIP Proxy Server) > ROW

Calls through the same SIP trunk/network from a local IP phone work fine. Additionally, a call from an analog phone on the FXS port to a local IP phone works. Hairpinned analog calls FXS-FXO also work fine.


To overcome this problem, configure the sip bind {control I media}command under voice service voip configuration mode to use the external interface with public address.

For more information about how to bind the control interface, refer to Configuring SIP Bind Features.

Problem Type

Call connection (Connection quality: One-way audio, no busy tone, dropped calls, etc.)

Call control software (CallManager, CallManager Express, ICS7750, SRST, SS7 call agents)

Gateway or module

Call Control

CallManager express

Call Connection

One-way audio

Failure Type

Cannot connect to server, disconnect, register, synchronize, communicate, etc.

End Where Problem is Heard or Device Type Involved

Analog phone

Signaling Protocol


TDM Signaling Type


SIP Features

SIP gateway

SIP trunk

Voice Gateways

IOS gateways
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Last update:
‎06-22-2009 04:05 PM
Updated by:
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