Follow these steps to set up voice dial peers to support the local and remote stations. Not all possible commands are shown here. To learn more, see Voice, Video, and Home Applications Configuration Guide and Voice, Video, and Home Applications Command Reference for Cisco IOS Release 12.0.
| Step | Command | Purpose |
|---|---|---|
1. | Router# configure terminal | Enter global configuration mode. |
2. | Router(config)# dial-peer voice number pots | Enter dial-peer configuration mode and define a local dial peer that will connect to the plain old telephone service (POTS) network. number is one or more digits identifying the dial peer. Valid entries are from 1 to 2147483647. pots indicates a peer using basic telephone service. |
Router | Configure the dial peer's destination pattern so that the system can reconcile dialed digits with a telephone number. string is a series of digits that specify the E.164 or private dialing plan phone number. Valid entries are the digits 0 through 9 and the letters A through D. The plus symbol (+) is not valid. The following special characters can be entered:
When the timer (T) character is included at the end of the destination pattern, the system collects dialed digits as they are entered—until the interdigit timer expires (10 seconds, by default)—or the user dials the termination of end-of-dialing key (default is #). Note The timer character must be a capital T. | |
4. | Router(config-dialpeer)# prefix string | (Optional) Include a dial-out prefix that the system enters automatically instead of people dialing it. string is a value from 0 to 9, and you can use a comma (,) to indicate a pause. Note There are other digit manipulation commands available to handle such situations as prefixes for special services, ignoring some digits, and dialing into remote PBXs as though they are local. |
5. | Router(config-dialpeer)# port slot/port:ds0-group-no | This command associates the dial peer with a specific logical interface. slot is the router location where the voice port adapter is installed. Valid entries are from 0 to 3. port indicates the voice interface card location. Valid entries are 0 or 1. Each defined DS0 group number is represented on a separate voice port. This allows you to define individual DS0s on the digital T1/E1 card. |
6. | Router(config)# dial-peer voice number voip | Enter dial-peer configuration mode and define a remote VoIP dial peer. number is one or more digits identifying the dial peer. Valid entries are from 1 to 2147483647. voip indicates a VoIP peer using voice encapsulation on the IP network. |
Router | The voice-card configuration codec command sets the codec options that are available when you execute this command. See Step 2 of the "Configuring the DSPfarm Interface" section. If you do not set codec complexity, g729r8 with IETF bit-ordering is used. If you set codec complexity to high, the following options are available:
If you set codec complexity to medium, the following options are valid:
The optional bytes parameter sets the number of voice data bytes per frame. Acceptable values are from 10 to 240 in increments of 10 (for example, 10, 20, 30, and so on). Any other value is rounded down (for example, from 236 to 230). If you specify g729r8, then the IETF (Internet Engineering Task Force) bit-ordering is used. For interoperability with a Cisco 7200 series router running a Cisco IOS release prior to Release 12.0(5)T or12.0(4)XH, you must specify the additional key word pre-ietf after g729r8. | |
8. | Router(config | (Optional) This setting is enabled by default. It activates voice activity detection (VAD). VAD allows the system to reduce unnecessary voice transmissions caused by unfiltered background noise. |
9. | Router | (Optional) Dual-tone multifrequency (DTMF) describes the tone that sounds in response to a keypress on a touch-tone phone. DTMF tones are compressed at one end of a call and decompressed at the other end. If a low-bandwidth codec, such as a G.729 or G.723, is used, the tones can sound distorted. The dtmf-relay command transports DTMF tones generated after call establishment out-of-band by using a method that transmits with greater fidelity than is possible in-band for most low-bandwidth codecs. Without DTMF relay, calls established with low-bandwidth codecs may have trouble accessing automated phone menu systems, such as voicemail and interactive voice response (IVR) systems. A signaling method is supplied only if the remote end supports it, and the options are: Cisco proprietary (cisco-rtp), standard H.323 (h245-alphanumeric), and H.323 standard with signal duration (h245-signal). |
10. | Router | (Optional) Specify the transmission speed of a fax to be sent to this dial peer. disable turns off fax transmission capability, and voice specifies the highest possible fax speed supported by the voice rate. |
11. | Router | See Step 3 in this procedure. |
12. | Router
| Configure the IP session target for the dial peer. ipv4:destination-address indicates IP address of the dial peer. dns:host-name indicates that the domain name server will resolve the name of the IP address. Valid entries for this parameter are characters representing the name of the host device. There are also wildcards available for defining domain names with the keyword by using source, destination, and dialed information in the host name. For complete command syntax information, see Voice, Video, and Home Applications Command Reference for Cisco IOS Release 12.0. |
Follow the procedure below to verify dial-peer configuration. To learn more about these commands, see Voice, Video, and Home Applications Command Reference for Cisco IOS Release 12.0.
Important command output is shown in bold.
Enter the privileged EXEC show dial-peer voice command. The following text is sample output from the command for a POTS dial peer:
Router# show dial-peer voice 1
VoiceEncapPeer1
tag = 1, dest-pat = \Q+14085551000',
answer-address = \Q',
group = 0, Admin state is up, Operation state is down
Permission is Both,
type = pots, prefix = \Q',
session-target = \Q', voice-port =
Connect Time = 0, Charged Units = 0
Successful Calls = 0, Failed Calls = 0
Accepted Calls = 0, Refused Calls = 0
Last Disconnect Cause is "10"
Last Disconnect Text is ""
Last Setup Time = 0
The following text is sample output from the show dial-peer voice command for a VoIP dial peer:
Router# show dial-peer voice 10
VoiceOverIpPeer10
tag = 10, dest-pat = \Q',
incall-number = \Q+14087',
group = 0, Admin state is up, Operation state is down
Permission is Answer,
type = voip, session-target = \Q',
sess-proto = cisco, req-qos = bestEffort,
acc-qos = bestEffort,
fax-rate = voice, codec = g729r8,
Expect factor = 10,Icpif = 30, VAD = disabled, Poor QOV Trap = disabled,
Connect Time = 0, Charged Units = 0
Successful Calls = 0, Failed Calls = 0
Accepted Calls = 0, Refused Calls = 0
Last Disconnect Cause is "10"
Last Disconnect Text is ""
Last Setup Time = 0
This section presents some useful show and debugging commands for understanding, maintaining, and troubleshooting your configuration.
| Command | Purpose |
|---|---|
Router# show dialplan number number | Shows which dial-peer is matched by a called number. |
Router# show call active voice | Shows statistics for currently active voice calls. |
Router# show call active fax | Shows statistics for currently active fax calls. |
Router# show call history voice | Shows statistics on previous voice calls. |
Router# show call history fax | Shows statistics on previous fax calls. |
Router# show voice port | Shows the status of voice ports. See "Verifying Voice Ports". |
Router# show controller t1 slot/port | Shows the status of the T1 controller. See "Verifying Card Type and Controller Settings". |
Router# show controller e1 slot/port | Shows the status of the E1 controller. See "Verifying Card Type and Controller Settings". |
Router# debug vpm all | Debugs the T1/E1 signaling. |
Router# debug vtsp all | Debugs the digits received and sent. |
Router# debug voip ccapi inout | Debugs the call setup process. |
The balance of this section shows the output of the commands listed in Table 1.
This section illustrates some of the privileged EXEC show commands that are useful for analyzing your system. Note that important information appears in bold, and bold text preceded by the "<<" characters explains the process.
The show dialplan number command provides information about the dial peer associated with a specified dial-plan number. Notice that the dial peer is operational and that IP Precedence has been configured to the preferred setting of 5.
Cisco-router# show dialplan number 75435
Macro Exp.: ##75435
VoiceOverIpPeer70000
information type = voice,
tag = 70000, destination-pattern = \Q##7....',
answer-address = \Q', preference=0,
group = 70000, Admin state is up, Operation state is up,
incoming called-number = \Q', connections/maximum = 0/unlimited,
DTMF Relay = disabled,
application associated:
type = voip, session-target = \Qipv4:171.68.253.18',
technology prefix:
settlement: disabled
ip precedence = 5, UDP checksum = disabled,
session-protocol = cisco, req-qos = best-effort,
acc-qos = best-effort,
fax-rate = 14400, payload size = 20 bytes
codec = g729r8, payload size = 20 bytes,
Expect factor = 10, Icpif = 30,signaling-type = cas,
VAD = disabled, Poor QOV Trap = disabled,
Connect Time = 0, Charged Units = 0,
Successful Calls = 3, Failed Calls = 0,
Accepted Calls = 3, Refused Calls = 0,
Last Disconnect Cause is "10 ",
Last Disconnect Text is "normal call clearing.",
Last Setup Time = 344813.
Matched: ##75435 Digits: 3
Target: ipv4:171.68.253.18
The show call active voice command displays information about a current call:
Cisco-router# show call active voice GENERIC: SetupTime=94523746 ms Index=448 PeerAddress=##73072 PeerSubAddress= PeerId=70000 PeerIfIndex=37 LogicalIfIndex=0 ConnectTime=94524043 DisconectTime=94546241 CallOrigin=1 ChargedUnits=0 InfoType=2 TransmitPackets=6251 TransmitBytes=125020 ReceivePackets=3300 ReceiveBytes=66000 VOIP: ConnectionId[0x142E62FB 0x5C6705AF 0x0 0x385722B0] RemoteIPAddress=171.68.235.18 RemoteUDPPort=16580 RoundTripDelay=29 ms SelectedQoS=best-effort tx_DtmfRelay=inband-voice SessionProtocol=cisco SessionTarget=ipv4:171.68.235.18 OnTimeRvPlayout=63690 GapFillWithSilence=0 ms GapFillWithPrediction=180 ms GapFillWithInterpolation=0 ms GapFillWithRedundancy=0 ms HiWaterPlayoutDelay=70 ms LoWaterPlayoutDelay=30 ms ReceiveDelay=40 ms LostPackets=0 ms EarlyPackets=1 ms LatePackets=18 ms VAD = disabled CoderTypeRate=g729r8 CodecBytes=20 cvVoIPCallHistoryIcpif=0 SignalingType=cas
The show call history voice command shows statistics about previous calls:
sb1pbx-voip# show call history voice GENERIC: SetupTime=94893250 ms Index=450 PeerAddress=##52258 PeerSubAddress= PeerId=50000 PeerIfIndex=35 LogicalIfIndex=0 DisconnectCause=10 DisconnectText=normal call clearing. ConnectTime=94893780 DisconectTime=95015500 CallOrigin=1 ChargedUnits=0 InfoType=2 TransmitPackets=32258 TransmitBytes=645160 ReceivePackets=20061 ReceiveBytes=401220 VOIP: ConnectionId[0x142E62FB 0x5C6705B3 0x0 0x388F851C] RemoteIPAddress=171.68.235.18 RemoteUDPPort=16552 RoundTripDelay=23 ms SelectedQoS=best-effort tx_DtmfRelay=inband-voice SessionProtocol=cisco SessionTarget=ipv4:171.68.235.18 OnTimeRvPlayout=398000 GapFillWithSilence=0 ms GapFillWithPrediction=1440 ms GapFillWithInterpolation=0 ms GapFillWithRedundancy=0 ms HiWaterPlayoutDelay=97 ms LoWaterPlayoutDelay=30 ms ReceiveDelay=49 ms LostPackets=1 ms EarlyPackets=1 ms LatePackets=132 ms VAD = disabled CoderTypeRate=g729r8 CodecBytes=20 cvVoIPCallHistoryIcpif=0 SignalingType=cas
This section illustrates some of the EXEC mode debug commands that are useful when analyzing and troubleshooting your system. Note that important information appears in bold, and bold text preceded by the "<<" characters explains the process.
The debug vpm all command displays information that allows you to troubleshoot T1/E1 signaling:
Cisco-router# debug vpm all Apr 19 19:18:54 PDT: htsp_process_event: [1/0/16, 1.4 , 34] em_onhook_offhookem_offhookem_onhookhtsp_setup_ind << port goes offhook Apr 19 19:18:54 PDT: htsp_process_event: [1/0/16, 1.5 , 8] Apr 19 19:19:01 PDT: htsp_process_event: [1/0/16, 1.5 , 10] htsp_alert_notify Apr 19 19:19:01 PDT: htsp_process_event: [1/0/16, 1.5 , 11] Apr 19 19:19:02 PDT: htsp_process_event: [1/0/16, 1.5 , 11] Apr 19 19:19:15 PDT: htsp_process_event: [1/0/16, 1.5 , 22] em_offhook_onhookem_stop_timers em_onhook << port goes onhook Apr 19 19:19:15 PDT: htsp_process_event: [1/0/16, 1.4 , 7] em_onhook_releaseem_onhook
The debug vtsp all command displays information that allows you to troubleshoot digits received and sent on a call:
cisco-router# debug vtsp all Apr 19 19:21:55 PDT: dsp_cp_tone_on: [1/0:1 (9502)] packet_len=30 channel_id=1 packet_id=72 tone_id=3 n_freq=2 freq_of_first=350 freq_of_second=440 amp_of_first=4000 amp_of_second=4000 direction=1 on_time_first=65535 off_time_first=0 on_time_second=65535 off_time_second=0 << providing dialtone Apr 19 19:21:59 PDT: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_BEGIN: digit=2,rtp_timestamp=0xF2D37240 act_report_digit_begin Apr 19 19:22:00 PDT: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_OFF: digit=2, duration=102act_report_digit_end Apr 19 19:22:00 PDT: dsp_cp_tone_off: [1/0:1 (9502)] packet_len=8 channel_id=1 packet_id=71 Apr 19 19:22:00 PDT: vtsp_timer: 34838705 Apr 19 19:22:00 PDT: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_BEGIN: digit=3,rtp_timestamp=0xF2D37240 act_report_digit_begin Apr 19 19:22:00 PDT: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_OFF: digit=3, duration=92act_report_digit_end Apr 19 19:22:00 PDT: dsp_cp_tone_off: [1/0:1 (9502)] packet_len=8 channel_id=1 packet_id=71 Apr 19 19:22:00 PDT: vtsp_timer: 34838724 Apr 19 19:22:00 PDT: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_BEGIN: digit=1,rtp_timestamp=0xF2D37240 act_report_digit_begin Apr 19 19:22:00 PDT: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_OFF: digit=1, duration=92act_report_digit_end Apr 19 19:22:00 PDT: dsp_cp_tone_off: [1/0:1 (9502)] packet_len=8 channel_id=1 packet_id=71 Apr 19 19:22:00 PDT: vtsp_timer: 34838744 Apr 19 19:22:00 PDT: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_BEGIN: digit=9,rtp_timestamp=0xF2D37240 act_report_digit_begin Apr 19 19:22:00 PDT: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_OFF: digit=9, duration=102act_report_digit_end Apr 19 19:22:00 PDT: dsp_cp_tone_off: [1/0:1 (9502)] packet_len=8 channel_id=1 packet_id=71 Apr 19 19:22:00 PDT: vtsp_timer: 34838768 Apr 19 19:22:00 PDT: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_BEGIN: digit=8,rtp_timestamp=0xF2D37218 act_report_digit_begin Apr 19 19:22:00 PDT: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT_OFF: digit=8, duration=107act_report_digit_end *** The Caller dialed the digits 23198 ***
The debug voip ccapi inout EXEC command traces the execution path through the call control API, which serves as the interface between the call-session application and the underlying network-specific software.
During the capabilities exchange shown in the command output, both sides agree on what compression to use, and the debug voip ccapi inout output helps you determine what each side is negotiating.
You can use the output from this command to understand how calls are being handled by the router. This command shows how a call flows through the system. By using this debug level, you can see the call setup and teardown operations performed on both the telephony and network call legs:
cisco-router# debug voip ccapi inout
Apr 19 19:23:11 PDT: sess_appl: ev(19=CC_EV_CALL_SETUP_IND), cid(9504), disp(0) << a
new call is originating
Apr 19 19:23:11 PDT: ccCallSetContext (callID=0x2520, context=0x61C0806C)
Apr 19 19:23:11 PDT: ccCallSetupAck (callID=0x2520)
Apr 19 19:23:11 PDT: ccGenerateTone (callID=0x2520 tone=8) << dialtone
Apr 19 19:23:18 PDT: cc_api_call_digit_begin (vdbPtr=0x61A1B1B4, callID=0x2520,
digit=2, flags=0x1, timestamp=0xCE2796D1, expiration=0x0) << digit 2 received
Apr 19 19:23:18 PDT: sess_appl: ev(10=CC_EV_CALL_DIGIT_BEGIN), cid(9504), disp(0)
Apr 19 19:23:18 PDT: ssa: cid(9504)st(0)oldst(0)cfid(-1)csize(0)in(1)fDest(0)
Apr 19 19:23:18 PDT: ssaIgnore cid(9504), st(0),oldst(0), ev(10)
Apr 19 19:23:18 PDT: cc_api_call_digit (vdbPtr=0x61A1B1B4, callID=0x2520, digit=2,
duration=102)
Apr 19 19:23:18 PDT: sess_appl: ev(9=CC_EV_CALL_DIGIT), cid(9504), disp(0)
Apr 19 19:23:18 PDT: ssa: cid(9504)st(0)oldst(0)cfid(-1)csize(0)in(1)fDest(0)
Apr 19 19:23:18 PDT: cc_api_call_digit_begin (vdbPtr=0x61A1B1B4, callID=0x2520,
digit=3, flags=0x1, timestamp=0xCE2796D1, expiration=0x0)
Apr 19 19:23:18 PDT: sess_appl: ev(10=CC_EV_CALL_DIGIT_BEGIN), cid(9504), disp(0)
Apr 19 19:23:18 PDT: ssa: cid(9504)st(0)oldst(0)cfid(-1)csize(0)in(1)fDest(0)
Apr 19 19:23:18 PDT: ssaIgnore cid(9504), st(0),oldst(0), ev(10)
Apr 19 19:23:18 PDT: cc_api_call_digit (vdbPtr=0x61A1B1B4, callID=0x2520, digit=3,
duration=102) << digit 3 received
Apr 19 19:23:18 PDT: sess_appl: ev(9=CC_EV_CALL_DIGIT), cid(9504), disp(0)
Apr 19 19:23:18 PDT: ssa: cid(9504)st(0)oldst(0)cfid(-1)csize(0)in(1)fDest(0)
Apr 19 19:23:18 PDT: cc_api_call_digit_begin (vdbPtr=0x61A1B1B4, callID=0x2520,
digit=1, flags=0x1, timestamp=0xCE2796D1, expiration=0x0)
Apr 19 19:23:18 PDT: sess_appl: ev(10=CC_EV_CALL_DIGIT_BEGIN), cid(9504), disp(0)
Apr 19 19:23:18 PDT: ssa: cid(9504)st(0)oldst(0)cfid(-1)csize(0)in(1)fDest(0)
Apr 19 19:23:18 PDT: ssaIgnore cid(9504), st(0),oldst(0), ev(10)
Apr 19 19:23:18 PDT: cc_api_call_digit (vdbPtr=0x61A1B1B4, callID=0x2520, digit=1,
duration=92) << digit 1 received
Apr 19 19:23:18 PDT: sess_appl: ev(9=CC_EV_CALL_DIGIT), cid(9504), disp(0)
Apr 19 19:23:18 PDT: ssa: cid(9504)st(0)oldst(0)cfid(-1)csize(0)in(1)fDest(0)
Apr 19 19:23:18 PDT: cc_api_call_digit_begin (vdbPtr=0x61A1B1B4, callID=0x2520,
digit=9, flags=0x1, timestamp=0xCE2796B9, expiration=0x0)
Apr 19 19:23:18 PDT: sess_appl: ev(10=CC_EV_CALL_DIGIT_BEGIN), cid(9504), disp(0)
Apr 19 19:23:18 PDT: ssa: cid(9504)st(0)oldst(0)cfid(-1)csize(0)in(1)fDest(0)
Apr 19 19:23:18 PDT: ssaIgnore cid(9504), st(0),oldst(0), ev(10)
Apr 19 19:23:18 PDT: cc_api_call_digit (vdbPtr=0x61A1B1B4, callID=0x2520, digit=9,
duration=105) << digit 9 received
Apr 19 19:23:18 PDT: sess_appl: ev(9=CC_EV_CALL_DIGIT), cid(9504), disp(0)
Apr 19 19:23:18 PDT: ssa: cid(9504)st(0)oldst(0)cfid(-1)csize(0)in(1)fDest(0)
Apr 19 19:23:18 PDT: cc_api_call_digit_begin (vdbPtr=0x61A1B1B4, callID=0x2520,
digit=8, flags=0x1, timestamp=0xCE279691, expiration=0x0)
Apr 19 19:23:18 PDT: sess_appl: ev(10=CC_EV_CALL_DIGIT_BEGIN), cid(9504), disp(0)
Apr 19 19:23:18 PDT: ssa: cid(9504)st(0)oldst(0)cfid(-1)csize(0)in(1)fDest(0)
Apr 19 19:23:18 PDT: ssaIgnore cid(9504), st(0),oldst(0), ev(10)
Apr 19 19:23:18 PDT: cc_api_call_digit (vdbPtr=0x61A1B1B4, callID=0x2520, digit=8,
duration=100) << digit 8 received
Apr 19 19:23:18 PDT: sess_appl: ev(9=CC_EV_CALL_DIGIT), cid(9504), disp(0)
Apr 19 19:23:18 PDT: ssa: cid(9504)st(0)oldst(0)cfid(-1)csize(0)in(1)fDest(0)
Apr 19 19:23:18 PDT: ssaSetupPeer cid(9504) peer list: tag(20000)
Apr 19 19:23:18 PDT: ssaSetupPeer cid(9504), destPat(23198), matched(1), prefix(),
peer(61C04464) << matched dial-peer 20000 voip
Apr 19 19:23:18 PDT: peer_tag=20000 << matched dial-peer voip 20000
Apr 19 19:23:18 PDT: ccIFCallSetupRequest: (vdbPtr=0x61A25524, dest=, callParams
<< voip call setup
={called=23198, calling=+9.......T, fdest=0, voice_peer_tag=20000}, mode=0x0)
Apr 19 19:23:18 PDT: ccCallSetContext (callID=0x2521, context=0x61C12E18)
Apr 19 19:23:18 PDT: ccCallProceeding (callID=0x2520, prog_ind=0x0)
Apr 19 19:23:19 PDT: cc_api_call_alert(vdbPtr=0x61A25524, callID=0x2521, prog_ind=0x88,
sig_ind=0x1)
Apr 19 19:23:19 PDT: sess_appl: ev(7=CC_EV_CALL_ALERT), cid(9505), disp(0)
Apr 19 19:23:19 PDT: ssa:
cid(9505)st(1)oldst(0)cfid(-1)csize(0)in(0)fDest(0)-cid2(9504)st2(1)oldst2(0)
Apr 19 19:23:19 PDT: ccCallAlert (callID=0x2520, prog_ind=0x88, sig_ind=0x1)
Apr 19 19:23:19 PDT: ccConferenceCreate (confID=0x61A21670, callID1=0x2520,
callID2=0x2521, tag=0x0)
Apr 19 19:23:19 PDT: cc_api_bridge_done (confID=0x33, srcIF=0x61A25524,
srcCallID=0x2521, dstCallID=0x2520, disposition=0, tag=0x0)
Apr 19 19:23:19 PDT: cc_api_bridge_done (confID=0x33, srcIF=0x61A1B1B4,
srcCallID=0x2520, dstCallID=0x2521, disposition=0, tag=0x0)
Apr 19 19:23:19 PDT: cc_api_caps_ind (dstVdbPtr=0x61A25524, dstCallId=0x2521, sr
<< negotiating capabilities with the remote VoIP gateway
Apr 19 19:23:36 PDT: sess_appl: ev(8=CC_EV_CALL_CONNECTED), cid(9505), disp(0)
Apr 19 19:23:36 PDT: ssa:
cid(9505)st(4)oldst(1)cfid(51)csize(0)in(0)fDest(0)-cid2(9504)st2(4)oldst2(4)
<< the VoIP call is connected
Apr 19 19:23:54 PDT: sess_appl: ev(12=CC_EV_CALL_DISCONNECTED), cid(9505),disp(0)
<< the VoIP call is disconnected
Apr 19 19:23:54 PDT: ccCallDisconnect (callID=0x2520, cause=0x10 tag=0x0)
<< the VoIP call is disconnected by cause_code 0x10
SEE BELOW FOR OSP - SETTLEMENT STATEMENTS
Configuring the Gateway
The following table shows the commands required to configure the Cisco AS5300 router for the UNI-OSP feature.
| Command | Purpose | |
|---|---|---|
Step 1 | Router# configure terminal | Enters the global configuration mode. |
Step 2 | Router(config)# settlement number | Enters the Settlement mode and configure the Settlement provider number. The settlement command puts you in the Settlement command mode. |
Step 3 | Router(config-settlement# type osp | Configures the Settlement provider type. In Cisco IOS Release 12.0(4)XH, OSP is the only type available. |
Step 4 | Router(config-settlement# url <server url> | This step can be repeated if the settlement provider has more than one service point. |
Step 5 | Router(config-settlement)# no shutdown | Bring up the settlement provider. |
Use the show settlement command to verify your configuration.
For UNI-OSP settlement, configure the URL of the settlement server and indicate that the settlement
type is issuing-osp, as in the following example.
Router# settlement 0 Router# type uni-osp Router# url 172.100.100.1
For the POTS dial peer, configure it like the example below. This example shows how to configure a
destination number of 1000, where 1000 is the route point/DNIS of a device connected to the PBX to
which the T1 line is attached:
dial-peer voice 111 pots destination pattern 1000 application session port 0:D session target settlement:0 dial-peer voice 222 voip incoming called-number 1000 codec g711ulaw application session
When the H.323 call "setup" message arrives, it should have the destination number (DNIS) of "1000".
The Cisco AS5300 will then handle it as a settlement call, and route an authorization request to the
OSP server.