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.
Resolution
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 expressCall Connection
One-way audioFailure Type
Cannot connect to server, disconnect, register, synchronize, communicate, etc.End Where Problem is Heard or Device Type Involved
Analog phoneSignaling Protocol
SIPTDM Signaling Type
Analog-FXSSIP Features
SIP gateway
SIP trunk
Voice Gateways
IOS gateways
