Troubleshooting Analog Voice Interfaces to the IP Network
If you are troubleshooting an analog connection, you must understand what type of circuit and interface your voice port is using. Analog voice port interfaces connect routers in packet-based networks to analog two-wire or four-wire analog circuits in telephony networks. Two-wire circuits connect to analog telephone or fax devices, and four-wire circuits connect to PBXs. Analog voice telephony interfaces include foreign exchange office (FXO), foreign exchange station (FXS), and receive and transmit (E&M). Direct Inward Dialing (DID) is a service offered by telephone companies that enables callers to dial directly to an extension on a PBX without the assistance of an operator or automated call attendant.
To troubleshoot analog voice interfaces, see the following sections:
If you are troubleshooting a connection to a PBX, you might find the PBX interoperability notes useful. These notes contain configuration information for Cisco gateways and several types of PBXs. To access these notes, use the following website:
FXS Interfaces
An FXS interface connects the router or access server to end-user equipment such as telephones, fax machines, and modems. The FXS interface supplies ring, voltage, and dial tone to the station and includes an RJ-11 connector for basic telephone equipment, keysets, and PBXs. In Figure 10, FXS signaling is used for end-user telephony equipment, such as a telephone or fax machine.
Figure 10 FXS Signaling Interfaces
If you are having trouble with an FXS port, check the following sections:
FXS Hardware Troubleshooting
An FXS interface connects directly to a standard telephone, fax machine, or similar device and supplies ring, voltage, and dial tone.
Troubleshoot FXS hardware by checking the following sections:
Software Compatibility
To ensure that your FXS card is compatible with your software, check the following:
•
For network modules inserted into Cisco 2600 series, Cisco 3600 series, and Cisco 3700 series, check the compatibility tables in the "Overview of Cisco Network Modules" chapter in the Cisco Network Modules Hardware Installation Guide.
For network modules inserted into Cisco 2600 series, Cisco 3600 series, and Cisco 3700 series, check the compatibility tables in the "Overview of Cisco Network Modules" chapter in the Cisco Network Modules Hardware Installation Guide.
•For interface cards inserted into Cisco 1600 series, Cisco 1700 series, Cisco 2600 series, Cisco 3600 series, Cisco 3700 series, and Cisco ICS 7750 platforms, check the compatibility tables in the "Overview of Cisco Interface Cards" chapter in the Cisco Interface Cards Installation Guide.
For interface cards inserted into Cisco 1600 series, Cisco 1700 series, Cisco 2600 series, Cisco 3600 series, Cisco 3700 series, and Cisco ICS 7750 platforms, check the compatibility tables in the "Overview of Cisco Interface Cards" chapter in the Cisco Interface Cards Installation Guide.Cabling
Two types of cabling are supported for Cisco FXS interfaces. They are described in the following sections:
Note For FXS connections, use a 2-wire (RJ-11) cable. A 4-wire cable can cause the second port to busy out.
For FXS connections, use a 2-wire (RJ-11) cable. A 4-wire cable can cause the second port to busy out.RJ-11 Connectors
The two-port and four-port FXS interface cards support the RJ-11 connector. Illustrations of the connector ports are shown in Figure 11 and Figure 12. Information about LEDs can be found in the "Connecting Voice Interface Cards to a Network" chapter of the Cisco Interface Card Hardware Installation Guide.
Figure 11 Two-Port FXS Card Front Panel
Figure 12 Four-Port FXS/DID Card Front Panel
For information about the VIC-2FXS interface card, refer to Understanding Foreign Exchange Station (FXS) Voice Interface Cards, document ID 7938.
RJ-21 Connectors on the High-Density Analog Telephony Network Module
The High-Density Analog Telephony network module supports an RJ-21 connector. This network module supports both FXS and FXO traffic. An illustration of the connector port is shown in Figure 13. Information about LEDs and pinouts can be found in the "Connecting High-Density Analog Telephony Network Modules to a Network" chapter of the Cisco Network Modules Hardware Installation Guide.
Figure 13 High-Density Analog Telephony Network Module
Shutdown Port
If the port is not working, be sure the port is not shut down. Enter the show voice port command with the voice port number that you are troubleshooting. The output will tell you:
•If the voice port is up. If it is not, use the no shutdown command to make it active.
If the voice port is up. If it is not, use the no shutdown command to make it active.
•What parameter values have been set for the voice port, including default values (which do not appear in the output from the show running-config command). If these values do not match those of the telephony connection you are making, reconfigure the voice port.
What parameter values have been set for the voice port, including default values (which do not appear in the output from the show running-config command). If these values do not match those of the telephony connection you are making, reconfigure the voice port.Disabling a Port on a Multiple Port Card
If you shut down a port on a multiple-port card, you can disable all of the ports on that card. If only one port is bad and the others are working, in many cases you can disable the bad port and then use the working ports until a replacement arrives. To disable a bad port, use one of the following methods:
•On a Cisco universal gateway, such as the Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850, busy out the port using the busyoutcommand. This setting allows the port to be taken out of service without disrupting the Cisco IOS configuration. See the product documentation for details:
On a Cisco universal gateway, such as the Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850, busy out the port using the busyoutcommand. This setting allows the port to be taken out of service without disrupting the Cisco IOS configuration. See the product documentation for details:
•On other Cisco gateways, remove the port from the dial peer. Refer to "Dial Peer Features and Configuration" in the Dial Peer Configuration on Voice Gateway Routers document to configure the dial peer.
On other Cisco gateways, remove the port from the dial peer. Refer to "Dial Peer Features and Configuration" in the Dial Peer Configuration on Voice Gateway Routers document to configure the dial peer.Ring Voltage Problems
Telephone exchanges and FXSs need to supply DC battery and AC ringing to enable the connected telephone equipment to transmit speech energy and to power the telephone equipment's ringing device. This section discusses what voltages are supplied by various Cisco FXS interfaces and how to overcome some known issues regarding voltage levels.
Ringing Voltages
The industry standard for PBX and key systems requires that the ring detection circuit be able to detect a ringing signal as low as 40 Vrms. This voltage takes into account the effects of load and cabling voltage drop on a ringing signal generated from a central office (CO). Conversely, the CO (exchange) must supply ringing with enough power to drive the maximum load over the maximum cable length. In order to meet this requirement, a CO-based unit must present a ringing signal with an amplitude of approximately 85 to 100 Vrms. Cisco voice gateways are intended for use as on premise services (ONS) equipment that is colocated or fairly close to equipment that detects ringing, so it can therefore use a lower ringing voltage and still meet the 40 Vrms 5 Ringer Equivalence Number (REN) requirement.
Idle Battery Voltage
Cisco voice gateways were designed for ONS connections and by default the FXS interface supplies either -24 Vdc or -36 Vdc idle battery, whereas off premise services, such as a CO, would require voltages of -48 V because it might have to interconnect over much greater cable lengths. Certain Cisco FXS interfaces can be configured to supply higher voltages.
Idle Line Voltages
Table 24 shows idle line voltages supplied by various Cisco gateway FXS interfaces.
Ring Voltage Problems
Voltage problems can cause three types of problems:
Certain automated devices, such as fax machines, answer machines, multiline phones and voice mail systems, look at the line voltage in order to deduce if the line is busy or idle. If another device is off hook, then the line voltage drops, and the automated system does not answer or initiate a call. If the threshold being used is close to -24 V or higher, this can cause the device not to work as expected.
Certain phones might not ring when the default ring voltage and ring frequency are applied from the Cisco FXS interface.
Answering and Call Initiation Problems with Automated Telephony Devices
In voice port configuration mode, configure the idle-voltage command on the voice port of the FXS to increase idle battery voltage from -24 V to -48 V. The idle-voltage low setting designates -24 V and the idle-voltage high setting designates -48 V.
Note This option is not available on VG248, VIC-2FXS, and WS-x6624 FXS interfaces.
This option is not available on VG248, VIC-2FXS, and WS-x6624 FXS interfaces.Ringing Problems
Phone manufacturers sometimes use frequency filters known as antitinkle circuits to prevent ringer devices from sounding while the user is dialing. Sometimes it is necessary to adjust the frequency of the ring to suit the connected device.
Configure the ring frequency for Cisco modular access routers by issuing the following command:
Router(config-voiceport)# ring frequency ?
25 ring frequency 25 Hertz
50 ring frequency 50 Hertz
Configure the ring frequency for the Cisco IAD2400 platform by issuing the following command:
Router(config-voiceport)# ring frequency ?
20 ring frequency 20 Hertz
30 ring frequency 30 Hertz
To prevent ringer devices from sounding, you can also provide a voltage threshold so that the lower voltages, which can be produced during dialing, are ignored. Increasing the voltage can overcome this.
Configure the DC offset voltage on Cisco IAD2400 series routers by issuing the following command:
Router(config-voiceport)# ring dc-offset ?
10-volts Ring DC offset 10 volts
20-volts Ring DC offset 20 volts
24-volts Ring DC offset 24 volts
Note This command sequence can be used only for Cisco IAD2400 series routers. The 24-V ring DC offset setting is available for Cisco IOS 12.2(11)T and later releases.
This command sequence can be used only for Cisco IAD2400 series routers. The 24-V ring DC offset setting is available for Cisco IOS 12.2(11)T and later releases.FXS Ring Failure in the United Kingdom
A telephone approved for the United Kingdom might fail to ring when connected to a Cisco FXS port. The failure results from a physical interoperability issue and is independent of Cisco hardware or software. British Telecom did not implement RJ-11 type connectors when it adopted plug-and-socket connection methodology. RJ-11 connectors allow parallel connectivity for the transmission path and the ringer circuit. They were not used because older telephones needed to have their ringer circuits connected in series due to a requirement for high current.
Outside the United Kingdom, ringer circuitry is self-contained in each phone. The U.K. implementation puts the capacitor, which provides the AC ring path, and the antitinkle feature (prevents the bell or ringer from sounding when pulse dialing is used) externally in the first socket, connected to the local loop.
In the United Kingdom, certain British Approval Board for Telecommunications (BABT) telephones fail to ring when they are connected to FXS ports on Cisco voice-enabled routers and switches. Outgoing calls can be made and voice communication in both directions can be established. However, incoming calls do not ring the telephone. These telephones functioned correctly before they were connected to the FXS ports.
Because a proprietary connection system is implemented, you must use an adapter to connect the telephone to an FXS port. The adapter must be amaster that contains the capacitor, or the telephone fails to ring.
For a schematic and more information, refer to Understanding Why Telephones in the United Kingdom Connected to Cisco FXS Interfaces May Fail to Ring, document ID 25800.
Unbreakable Dial Tone
A common problem encountered in a VoIP network is being unable to break dial tone. The router seizes a line on the local PBX but when digits are dialed, the dial tone stays. The calling party is unable to pass the dual-tone multifrequency (DTMF) tones or digits to the terminating device, resulting in callers being unable to dial the desired extension or interact with a device that needs DTMF tones such as a voice mail or an interactive voice response (IVR) application. This problem can result from a number of sources, for example:
•DTMF tones are not passed.
DTMF tones are not passed.
•DTMF tones are not understood.
DTMF tones are not understood.
•DTMF tones are too distorted to be understood.
DTMF tones are too distorted to be understood.
•Other signaling and cabling issues occur.
Other signaling and cabling issues occur.
Make sure the dial type is set as DTMF on both the router and the PBX. The FXS port does not pass on the digits; therefore, this setting is not available on an FXS port. However, this setting can be changed on FXO and E&M ports:
Router(config-voiceport)# dial-type ?
dtmf touch-tone dialer
mf mf-tone dialer
pulse pulse dialer
For more information, refer to Inability To Break Dialtone in a Voice over IP Network, document ID 22376.
No LED When Phone Off the Hook
Verify if you have an analog or digital card. If you have an analog card like the VIC-2FXS or the VIC2-4FXS, you might have one of the following problems:
•The port is in a shutdown state.
The port is in a shutdown state.
•The port is in a park state.
The port is in a park state.
•The port is bad.
The port is bad.
If you have a digital card like the NM-2V, you might have bad DSPs.
Use the following procedure if there is no LED when your phone is off hook:
Step 1 Check the cable to make sure that it is RJ-11 with two pins for the FXS port.
Check the cable to make sure that it is RJ-11 with two pins for the FXS port.
Step 2 Test the LED using a different phone.
Test the LED using a different phone.
Step 3 Check your Cisco IOS version to make sure that the feature set is either IP Plus or Enterprise Plus.
Check your Cisco IOS version to make sure that the feature set is either IP Plus or Enterprise Plus.
Step 4 If Steps 1 to 3 do not work, replace the voice interface card (VIC).
If Steps 1 to 3 do not work, replace the voice interface card (VIC).FXO Interfaces
An FXO interface is used for trunk, or tie line, connections to a PSTN CO or to a PBX that does not support E&M signaling (when local telecommunications authority permits). This interface is of value for off-premises station applications. Figure 14 shows an FXS connection to a telephone and an FXO connection to the PSTN at the far side of a WAN.
Figure 14 FXS and FXO Signaling Interfaces
If you are having trouble with an FXO port, check the following sections:
FXO Hardware Troubleshooting
An FXO interface is used for trunk connections. Troubleshoot FXO hardware by checking the following sections:
Software Compatibility
To ensure that your card is compatible with your software, check the following:
•For network modules inserted into Cisco 2600 series, Cisco 3600 series, and Cisco 3700 series routers, refer to the compatibility tables in the"Overview of Cisco Network Modules" chapter in the Cisco Network Modules Hardware Installation Guide.
For network modules inserted into Cisco 2600 series, Cisco 3600 series, and Cisco 3700 series routers, refer to the compatibility tables in the"Overview of Cisco Network Modules" chapter in the Cisco Network Modules Hardware Installation Guide.
•For interface cards inserted into Cisco 1600 series, Cisco 1700 series, Cisco 2600 series, Cisco 3600 series, Cisco 3700 series, and Cisco ICS 7750 platforms, refer to the compatibility tables in the "Overview of Cisco Interface Cards" chapter in the Cisco Interface Cards Installation Guide.
For interface cards inserted into Cisco 1600 series, Cisco 1700 series, Cisco 2600 series, Cisco 3600 series, Cisco 3700 series, and Cisco ICS 7750 platforms, refer to the compatibility tables in the "Overview of Cisco Interface Cards" chapter in the Cisco Interface Cards Installation Guide.Cabling
Two types of cabling are supported for Cisco FXO interfaces. They are described in the following sections:
Note For FXO connections, use a 2-wire (RJ-11) cable. A 4-wire cable can cause the second port to busy out.
For FXO connections, use a 2-wire (RJ-11) cable. A 4-wire cable can cause the second port to busy out.RJ-11 Connectors
The two-port and four-port FXO interface cards support the RJ-11 connector. Illustrations of the connector ports are shown in Figure 15 and Figure 16. Information about LEDs can be found in the "Connecting Voice Interface Cards to a Network" chapter of the Cisco Interface Card Hardware Installation Guide.
Figure 15 Two-Port FXO Card Front Panel
Figure 16 Four-Port FXO Card Front Panel
RJ-21 Connectors on the High-Density Analog Telephony Network Module
The High-Density Analog Telephony network module supports an RJ-21 connector. This network module supports both FXS and FXO traffic. An illustration of the connector port is shown in Figure 17. Information about LEDs and pinouts can be found in the "Connecting High-Density Analog Telephony Network Modules to a Network" chapter of the Cisco Network Modules Hardware Installation Guide.
Figure 17 High-Density Analog Telephony Network Module
Shutdown Port
If the port is not working, be sure the port is not shut down. Enter the show voice port command with the voice port number that you are troubleshooting. The output will tell you:
•If the voice port is up. If it is not, use the no shutdown command to make it active.
If the voice port is up. If it is not, use the no shutdown command to make it active.
•What parameter values have been set for the voice port, including default values (these values do not appear in the output from the show running-config command). If these values do not match those of the telephony connection you are making, reconfigure the voice port.
What parameter values have been set for the voice port, including default values (these values do not appear in the output from the show running-config command). If these values do not match those of the telephony connection you are making, reconfigure the voice port.Disabling a Port on a Multiple Port Card
If you shut down a port on a multiple-port card, you can disable all of the ports on that card. If only one port is bad and the others are working, in many cases you can disable the bad port and use the working ports until a replacement arrives. To disable a bad port, use one of the following methods:
•On a Cisco universal gateway, such as the Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850, busy out the port using the busyoutcommand. This allows the port to be taken out of service without disrupting the Cisco IOS configuration. Refer to the product documentation for details:
On a Cisco universal gateway, such as the Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850, busy out the port using the busyoutcommand. This allows the port to be taken out of service without disrupting the Cisco IOS configuration. Refer to the product documentation for details:
–For Cisco AS5350 and Cisco AS5400 universal gateways, refer to the "Managing and Troubleshooting the Universal Port Card" chapter in theCisco AS5350 and Cisco AS5400 Universal Gateway Software Configuration Guide.
For Cisco AS5350 and Cisco AS5400 universal gateways, refer to the "Managing and Troubleshooting the Universal Port Card" chapter in theCisco AS5350 and Cisco AS5400 Universal Gateway Software Configuration Guide.
–For Cisco AS5800 access servers, refer to the Managing and Troubleshooting NextPort Services on the AS5800 feature document.
For Cisco AS5800 access servers, refer to the Managing and Troubleshooting NextPort Services on the AS5800 feature document.
–For Cisco AS5850 universal gateways, refer to the Managing Port Services on the Cisco AS5850 Universal Gateway feature document.
For Cisco AS5850 universal gateways, refer to the Managing Port Services on the Cisco AS5850 Universal Gateway feature document.
•On other Cisco gateways, remove the port from the dial peer. Refer to "Dial Peer Features and Configuration" in the Dial Peer Configuration on Voice Gateway Routers document to configure the dial peer.
On other Cisco gateways, remove the port from the dial peer. Refer to "Dial Peer Features and Configuration" in the Dial Peer Configuration on Voice Gateway Routers document to configure the dial peer.FXO Disconnect Failure
When loop-starting signaling is used, an FXO interface looks like a phone to the switch that it is connecting to. The FXO interface closes the loop to indicate off hook. The switch always provides a battery so there is no disconnect supervision from the switch side. Because a switch expects a phone user or modem to hang up the phone when the call is terminated on either side, it also expects the FXO port on the router to hang up. However, the FXO port expects the switch to tell it when to hang up. Because the port relies on the switch, there is no guarantee that a near- or far-end FXO port will disconnect the call once either end of the call has hung up.
The most common symptoms of this problem are phones that continue to ring when the caller has cleared, or FXO ports that remain busy after the previous call should have been cleared.
To troubleshoot this problem, refer to Understanding FXO Disconnect Problem, document ID 8120.
Troubleshooting FXO Answer and Disconnect Supervision
This section describes troubleshooting the FXO Answer and Disconnect Supervision feature for analog FXO voice ports. This feature applies to analog FXO voice ports with loop-start signaling connected to PSTNs, PBXs, or key systems.
The FXO Answer and Disconnect Supervision feature enables analog FXO ports to monitor call-progress tones and to monitor voice and fax transmissions returned from a PBX or from the PSTN.
Answer supervision can be accomplished in two ways: by detecting battery reversal, or by detecting voice, fax, or modem tones. If an FXO voice port is connected to the PSTN and battery reversal is supported, use the battery reversal method. Voice ports that do not support battery reversal must use the answer supervision method, in which answer supervision is triggered when the DSP detects voice, modem, or fax transmissions. Configuring answer supervision automatically enables disconnect supervision; however, you can configure disconnect supervision separately if answer supervision is not configured.
Disconnect supervision can be configured to detect call-progress tones sent by the PBX or PSTN (for example, busy, reorder, out-of-service, number-unavailable), or to detect any tone received (for example, busy tone or dial tone). When an incoming call ends, the DSP detects the associated call-progress tone, causing the analog FXO voice port to go on-hook.
This section provides solutions to problems that you might encounter when implementing the
FXO Answer and Disconnect Supervision feature.
FXO Answer and Disconnect Supervision feature.
Typical problems with the answer supervision feature are as follows:
•Call-progress tones such as ringback are not heard by the calling party.
Call-progress tones such as ringback are not heard by the calling party.
If any call legs have IVR configured, ensure that the IVR version is 2.0.
•Ringback timer is not initiated or ringback is not detected.
Ringback timer is not initiated or ringback is not detected.
–The wrong call-progress tone (cptone) command is configured on the voice port.
The wrong call-progress tone (cptone) command is configured on the voice port.
–The wrong DTMF detection parameters are configured.
The wrong DTMF detection parameters are configured.
–Custom call-progress tones are assigned to the voice port but ringback tone has not been configured; in this case, the default behavior is not to detect any ringback tones.
Custom call-progress tones are assigned to the voice port but ringback tone has not been configured; in this case, the default behavior is not to detect any ringback tones.
•Answer supervision is not triggered.
Answer supervision is not triggered.
Answer supervision—either by battery-reversal detection or by call-progress tone detection—is not configured on the voice-port in use.
•Excessive delay before answer supervision is activated.
Excessive delay before answer supervision is activated.
The level on the sensitivity parameter in the supervisory answer dualtone command is set too low. Configure the sensitivity for high.
If incorrect disconnect cause codes are reported, check the following:
•The values configured for custom call-progress tones could be incorrect.
The values configured for custom call-progress tones could be incorrect.
•Overlapping detection frequencies might have been incorrectly specified in the voice class created by the voice class dualtone-detect-paramscommand. For example if the freq-max-deviation parameter is configured to be 20 Hz, and the busy and reorder parameters are set for frequencies 350 and 370 respectively, the voice port cannot detect the reorder tone, resulting in an incorrect disconnect cause code.
Overlapping detection frequencies might have been incorrectly specified in the voice class created by the voice class dualtone-detect-paramscommand. For example if the freq-max-deviation parameter is configured to be 20 Hz, and the busy and reorder parameters are set for frequencies 350 and 370 respectively, the voice port cannot detect the reorder tone, resulting in an incorrect disconnect cause code.
Note If the frequencies and cadences (including error deviations as defined in the voice class dualtone-detect-params command) are the same for multiple call-progress tones, the order of detection is as follows: busy, reorder, number-unobtainable, out-of-service, disconnect.
If the frequencies and cadences (including error deviations as defined in the voice class dualtone-detect-params command) are the same for multiple call-progress tones, the order of detection is as follows: busy, reorder, number-unobtainable, out-of-service, disconnect.
If calls are not billed correctly, it might be that answer supervision is not being triggered. For answer supervision to be triggered, voice, fax, or data tones originating at the called-party end must be detected.
To configure the FXO Supervisory Disconnect Tone, refer to the Voice Port Configuration document.
Monitoring and Maintaining FXO Answer and Disconnect Supervision
To monitor the status of the FXO Answer and Disconnect Supervision feature, use the show voice port command, which causes the FXO voice port to be monitored. The following table illustrates the use of the show voice port command for monitoring voice port 1/1/0.
Unbreakable Dial Tone
A common problem encountered in a VoIP network is being unable to break dial tone. The router puts a seizure on the local PBX but when digits are dialed, the dial tone stays. The calling party is unable to pass the DTMF tones or digits to the terminating device, resulting in callers being unable to dial the desired extension or interact with the device that needs DTMF tones, such as a voice mail or IVR application. Here are some possible causes of the problem:
•DTMF tones not being sent.
DTMF tones not being sent.
•DTMF tones not being understood.
DTMF tones not being understood.
•DTMF tones too distorted to be understood.
DTMF tones too distorted to be understood.
•Other signaling and cabling issues.
Other signaling and cabling issues.
Make sure the dial type is set as DTMF on both the router and the PBX. The FXS port does not pass on the digits, therefore this setting is not available on an FXS port. However, this setting can be changed on FXO and E&M ports:
Router(config-voiceport)# dial-type ?
dtmf touch-tone dialer
mf mf-tone dialer
pulse pulse dialer
For more information, refer to Inability To Break Dialtone in a Voice over IP Network, document ID 22376.
Troubleshooting Caller ID Problems
Several debugs can be used to troubleshoot the Caller ID feature on the routers. The voice port module (VPM) signaling debugs, such as the debug vpm signal command, track the standard debugs with Caller ID feature turned on. These debugs are analyzed from the perspective of the terminating router and its FXO port; the caller ID is sent from this end. The following example shows an FXO port receiving caller ID. In this example, the phone sends the caller ID to the FXO port.
Nov 20 10:40:15.861 EST: [1/0/0] htsp_start_caller_id_rx Nov 20 10:40:15.861 EST: [1/0/0] htsp_set_caller_id_rx:BELLCORE Nov 20 10:40:15.861 EST: htsp_timer - 10000 msec Nov 20 10:40:17.757 EST: [1/0/0, FXOLS_RINGING, E_DSP_SIG_0100] Nov 20 10:40:17.757 EST: fxols_ringing_not Nov 20 10:40:17.761 EST: htsp_timer_stop Nov 20 10:40:17.761 EST: htsp_timer - 10000 msec Nov 20 10:40:18.925 EST: [1/0/0] htsp_stop_caller_id_rx Nov 20 10:40:21.857 EST: [1/0/0, FXOLS_RINGING, E_DSP_SIG_0000] Nov 20 10:40:23.857 EST: [1/0/0, FXOLS_RINGING, E_DSP_SIG_0100] Nov 20 10:40:23.857 EST: fxols_ringing_not Nov 20 10:40:23.861 EST: htsp_timer_stop htsp_setup_ind Nov 20 10:40:23.861 EST: [1/0/0] get_fxo_caller_id:Caller ID received. Message type=128 length=31 checksum=74 Nov 20 10:40:23.861 EST: [1/0/0] Caller ID String 80 1C 01 08 31 31 32 30 31 35 34 30 02 07 35 35 35 31 32 31 32 07 07 4F 75 74 73 69 64 65 74 Nov 20 10:40:23.865 EST: [1/0/0] get_fxo_caller_id calling num=5551212 calling name=Outside calling time=11/20 15:40 Nov 20 10:40:23.869 EST: [1/0/0, FXOLS_WAIT_SETUP_ACK, E_HTSP_SETUP_ACK] Nov 20 10:40:23.873 EST: fxols_wait_setup_ack: Nov 20 10:40:23.873 EST: [1/0/0] set signal state = 0xC
timestamp = 0 Nov 20 10:40:23.985 EST: [1/0/0, FXOLS_PROCEEDING, E_DSP_SIG_0100] fxols_proceed_clear Nov 20 10:40:23.985 EST: htsp_timer_stop2 Nov 20 10:40:24.097 EST: [1/0/0, FXOLS_PROCEEDING, E_DSP_SIG_0110] fxols_rvs_battery Nov 20 10:40:24.097 EST: htsp_timer_stop2 Nov 20 10:40:24.733 EST: [1/0/0, FXOLS_PROCEED_RVS_BT, E_HTSP_PROCEEDING] fxols_offhook_proc Nov 20 10:40:24.733 EST: htsp_timer - 120000 msec Nov 20 10:40:24.745 EST: [1/0/0, FXOLS_PROCEED_RVS_BT, E_HTSP_VOICE_CUT_THROUGH] fxols_proc_voice
In this example, everything was working fine and both Name and Number Display were properly delivered to the phone. In the two scenarios below, the calling number is missing in one case, and the name display is missing in the other.
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