{"id":131,"date":"2026-05-16T02:49:42","date_gmt":"2026-05-16T02:49:42","guid":{"rendered":"https:\/\/techgyan360.com\/blog\/?p=131"},"modified":"2026-05-31T16:21:01","modified_gmt":"2026-05-31T16:21:01","slug":"sip-trunk-dtmf-relay-cucm-mobility-interview-questions","status":"publish","type":"post","link":"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/","title":{"rendered":"Cisco IPT Advanced: SIP Trunks, DTMF Relay, MVA, and Call Park \u2013 50 Expert Q&#038;As"},"content":{"rendered":"<p class=\"ds-markdown-paragraph\"><span class=\"\">Welcome to\u00a0<\/span><strong><span class=\"\">Article 4<\/span><\/strong><span class=\"\">\u00a0of our Cisco IPT interview series. We have covered clustering, gatekeepers, ISDN, media resources, and SRST. Now, we tackle the protocols and features that dominate modern VoIP deployments \u2013\u00a0<\/span><strong><span class=\"\">SIP trunks, DTMF relay, mobility features (MVA\/SNR), and call park<\/span><\/strong><span class=\"\">.<\/span><\/p>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">These\u00a0<\/span><strong><span class=\"\">SIP trunk DTMF relay and CUCM mobility features<\/span><\/strong><span class=\"\">\u00a0questions will prepare you for real-world enterprise scenarios involving SIP providers, remote workers, and advanced call handling.<\/span><\/p>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Let\u2019s dive in.<\/span><\/p>\n<hr \/>\n<div id=\"ez-toc-container\" class=\"ez-toc-v2_0_84 counter-hierarchy ez-toc-counter ez-toc-grey ez-toc-container-direction\">\n<div class=\"ez-toc-title-container\">\n<p class=\"ez-toc-title\" style=\"cursor:inherit\">Table of Contents<\/p>\n<span class=\"ez-toc-title-toggle\"><a href=\"#\" class=\"ez-toc-pull-right ez-toc-btn ez-toc-btn-xs ez-toc-btn-default ez-toc-toggle\" aria-label=\"Toggle Table of Content\"><span class=\"ez-toc-js-icon-con\"><span class=\"\"><span class=\"eztoc-hide\" style=\"display:none;\">Toggle<\/span><span class=\"ez-toc-icon-toggle-span\"><svg style=\"fill: #999;color:#999\" xmlns=\"http:\/\/www.w3.org\/2000\/svg\" class=\"list-377408\" width=\"20px\" height=\"20px\" viewBox=\"0 0 24 24\" fill=\"none\"><path d=\"M6 6H4v2h2V6zm14 0H8v2h12V6zM4 11h2v2H4v-2zm16 0H8v2h12v-2zM4 16h2v2H4v-2zm16 0H8v2h12v-2z\" fill=\"currentColor\"><\/path><\/svg><svg style=\"fill: #999;color:#999\" class=\"arrow-unsorted-368013\" xmlns=\"http:\/\/www.w3.org\/2000\/svg\" width=\"10px\" height=\"10px\" viewBox=\"0 0 24 24\" version=\"1.2\" baseProfile=\"tiny\"><path d=\"M18.2 9.3l-6.2-6.3-6.2 6.3c-.2.2-.3.4-.3.7s.1.5.3.7c.2.2.4.3.7.3h11c.3 0 .5-.1.7-.3.2-.2.3-.5.3-.7s-.1-.5-.3-.7zM5.8 14.7l6.2 6.3 6.2-6.3c.2-.2.3-.5.3-.7s-.1-.5-.3-.7c-.2-.2-.4-.3-.7-.3h-11c-.3 0-.5.1-.7.3-.2.2-.3.5-.3.7s.1.5.3.7z\"\/><\/svg><\/span><\/span><\/span><\/a><\/span><\/div>\n<nav><ul class='ez-toc-list ez-toc-list-level-1 eztoc-toggle-hide-by-default' ><li class='ez-toc-page-1 ez-toc-heading-level-2'><a class=\"ez-toc-link ez-toc-heading-1\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#Part_1_SIP_Methods_and_Message_Responses_Q_151%E2%80%93160\" >Part 1: SIP Methods and Message Responses (Q 151\u2013160)<\/a><ul class='ez-toc-list-level-3' ><li class='ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-2\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#151_What_are_the_six_core_SIP_methods_requests\" >151. What are the six core SIP methods (requests)?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-3\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#152_What_is_the_purpose_of_the_ACK_method\" >152. What is the purpose of the ACK method?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-4\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#153_What_is_the_difference_between_BYE_and_CANCEL\" >153. What is the difference between BYE and CANCEL?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-5\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#154_What_are_the_six_classes_of_SIP_responses\" >154. What are the six classes of SIP responses?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-6\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#155_What_is_a_100_Trying_response\" >155. What is a 100 Trying response?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-7\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#156_What_is_a_180_Ringing_response\" >156. What is a 180 Ringing response?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-8\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#157_What_is_a_183_Session_Progress_response\" >157. What is a 183 Session Progress response?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-9\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#158_What_does_a_404_Not_Found_response_indicate\" >158. What does a 404 Not Found response indicate?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-10\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#159_What_does_a_486_Busy_Here_response_mean\" >159. What does a 486 Busy Here response mean?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-11\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#160_Scenario_You_receive_a_503_Service_Unavailable_from_a_SIP_trunk_provider_What_does_it_mean\" >160. Scenario: You receive a 503 Service Unavailable from a SIP trunk provider. What does it mean?<\/a><\/li><\/ul><\/li><li class='ez-toc-page-1 ez-toc-heading-level-2'><a class=\"ez-toc-link ez-toc-heading-12\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#Part_2_SIP_UA_and_Voice_Service_Configuration_Q_161%E2%80%93170\" >Part 2: SIP UA and Voice Service Configuration (Q 161\u2013170)<\/a><ul class='ez-toc-list-level-3' ><li class='ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-13\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#161_What_is_SIP_UA_User_Agent_configuration_mode\" >161. What is SIP UA (User Agent) configuration mode?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-14\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#162_What_does_the_registrar_command_do_in_SIP_UA\" >162. What does the\u00a0registrar\u00a0command do in SIP UA?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-15\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#163_What_does_the_sip-server_command_do\" >163. What does the\u00a0sip-server\u00a0command do?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-16\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#164_What_does_the_max-forwards_command_do\" >164. What does the\u00a0max-forwards\u00a0command do?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-17\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#165_What_SIP_UA_timers_can_be_adjusted\" >165. What SIP UA timers can be adjusted?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-18\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#166_What_is_redirect_ip2ip_in_voice_service_VoIP\" >166. What is\u00a0redirect ip2ip\u00a0in voice service VoIP?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-19\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#167_What_does_bind_control_source-interface_do\" >167. What does\u00a0bind control source-interface\u00a0do?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-20\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#168_What_does_bind_media_source-interface_do\" >168. What does\u00a0bind media source-interface\u00a0do?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-21\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#169_What_is_the_difference_between_session_transport_udp_and_tcp_for_SIP\" >169. What is the difference between\u00a0session transport udp\u00a0and\u00a0tcp\u00a0for SIP?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-22\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#170_Scenario_Your_SIP_gateway_sends_large_INVITE_messages_that_get_fragmented_How_do_you_fix\" >170. Scenario: Your SIP gateway sends large INVITE messages that get fragmented. How do you fix?<\/a><\/li><\/ul><\/li><li class='ez-toc-page-1 ez-toc-heading-level-2'><a class=\"ez-toc-link ez-toc-heading-23\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#Part_3_DTMF_Relay_Methods_Q_171%E2%80%93180\" >Part 3: DTMF Relay Methods (Q 171\u2013180)<\/a><ul class='ez-toc-list-level-3' ><li class='ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-24\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#171_Why_is_DTMF_relay_necessary\" >171. Why is DTMF relay necessary?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-25\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#172_What_is_RTP-NTE_RFC_2833_DTMF_relay\" >172. What is RTP-NTE (RFC 2833) DTMF relay?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-26\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#173_What_is_SIP-NOTIFY_DTMF_relay\" >173. What is SIP-NOTIFY DTMF relay?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-27\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#174_What_is_SIP-INFO_DTMF_relay\" >174. What is SIP-INFO DTMF relay?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-28\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#175_What_is_KPML_KPML_DTMF_relay\" >175. What is KPML (KPML) DTMF relay?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-29\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#176_How_do_you_configure_DTMF_relay_on_a_SIP_dial_peer\" >176. How do you configure DTMF relay on a SIP dial peer?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-30\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#177_Scenario_SIP_to_SCCP_calls_have_DTMF_failures_IVR_doesnt_recognize_digits_Why\" >177. Scenario: SIP to SCCP calls have DTMF failures (IVR doesn&#8217;t recognize digits). Why?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-31\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#178_What_is_the_notify_telephone-event_max-duration_command\" >178. What is the\u00a0notify telephone-event max-duration\u00a0command?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-32\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#179_Scenario_DTMF_digits_are_recognized_but_repeated_multiple_times_How_to_fix\" >179. Scenario: DTMF digits are recognized but repeated multiple times. How to fix?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-33\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#180_What_is_the_default_DTMF_method_if_no_dtmf-relay_is_configured\" >180. What is the default DTMF method if no\u00a0dtmf-relay\u00a0is configured?<\/a><\/li><\/ul><\/li><li class='ez-toc-page-1 ez-toc-heading-level-2'><a class=\"ez-toc-link ez-toc-heading-34\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#Part_4_SIP_Trunk_Configuration_%E2%80%93_CUCM_4x_vs_5x_Q_181%E2%80%93185\" >Part 4: SIP Trunk Configuration \u2013 CUCM 4.x vs 5.x (Q 181\u2013185)<\/a><ul class='ez-toc-list-level-3' ><li class='ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-35\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#181_How_do_you_configure_a_SIP_trunk_in_CUCM_4x\" >181. How do you configure a SIP trunk in CUCM 4.x?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-36\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#182_How_does_SIP_trunk_configuration_differ_in_CUCM_5x\" >182. How does SIP trunk configuration differ in CUCM 5.x+?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-37\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#183_Why_did_CUCM_4x_require_an_MTP_on_SIP_trunks\" >183. Why did CUCM 4.x require an MTP on SIP trunks?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-38\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#184_Scenario_Your_SIP_trunk_works_for_voice_but_fails_for_fax_G711_pass-through_What_is_wrong\" >184. Scenario: Your SIP trunk works for voice but fails for fax (G.711 pass-through). What is wrong?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-39\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#185_What_is_toll_bypass_using_SIP\" >185. What is toll bypass using SIP?<\/a><\/li><\/ul><\/li><li class='ez-toc-page-1 ez-toc-heading-level-2'><a class=\"ez-toc-link ez-toc-heading-40\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#Part_5_Mobile_Voice_Access_MVA_and_SNR_Q_186%E2%80%93195\" >Part 5: Mobile Voice Access (MVA) and SNR (Q 186\u2013195)<\/a><ul class='ez-toc-list-level-3' ><li class='ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-41\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#186_What_is_Mobile_Voice_Access_MVA\" >186. What is Mobile Voice Access (MVA)?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-42\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#187_How_does_MVA_hairpinning_work\" >187. How does MVA hairpinning work?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-43\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#188_What_is_the_VXML_URL_used_for_MVA\" >188. What is the VXML URL used for MVA?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-44\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#189_How_do_you_configure_a_dial_peer_for_MVA_on_an_H323_gateway\" >189. How do you configure a dial peer for MVA on an H.323 gateway?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-45\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#190_What_is_Single_Number_Reach_SNR_Mobile_Connect\" >190. What is Single Number Reach (SNR) \/ Mobile Connect?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-46\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#191_What_is_the_difference_between_MVA_and_SNR\" >191. What is the difference between MVA and SNR?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-47\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#192_What_is_the_BHCA_formula_for_SNR_remote_destinations\" >192. What is the BHCA formula for SNR remote destinations?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-48\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#193_Scenario_A_user_reports_that_MVA_authentication_fails_PIN_not_recognized\" >193. Scenario: A user reports that MVA authentication fails (PIN not recognized).<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-49\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#194_What_DTMF_digits_are_available_during_an_MVA_call_for_call_control\" >194. What DTMF digits are available during an MVA call for call control?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-50\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#195_Scenario_MVA_call_connects_but_there_is_no_dial_tone_after_PIN_entry\" >195. Scenario: MVA call connects but there is no dial tone after PIN entry.<\/a><\/li><\/ul><\/li><li class='ez-toc-page-1 ez-toc-heading-level-2'><a class=\"ez-toc-link ez-toc-heading-51\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#Part_6_Call_Park_Meet-Me_Conference_Custom_Ringtones_Q_196%E2%80%93200\" >Part 6: Call Park, Meet-Me Conference, Custom Ringtones (Q 196\u2013200)<\/a><ul class='ez-toc-list-level-3' ><li class='ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-52\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#196_What_is_Call_Park\" >196. What is Call Park?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-53\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#197_What_is_Directed_Call_Park_How_is_it_different\" >197. What is Directed Call Park? How is it different?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-54\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#198_What_is_the_difference_between_Ad-Hoc_and_Meet-Me_Conference\" >198. What is the difference between Ad-Hoc and Meet-Me Conference?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-55\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#199_How_do_you_add_a_custom_ringtone_to_a_Cisco_IP_Phone\" >199. How do you add a custom ringtone to a Cisco IP Phone?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-56\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#200_Final_Scenario_Call_Park_works_but_Directed_Call_Park_fails_What_do_you_check\" >200. Final Scenario: Call Park works but Directed Call Park fails. What do you check?<\/a><\/li><\/ul><\/li><li class='ez-toc-page-1 ez-toc-heading-level-2'><a class=\"ez-toc-link ez-toc-heading-57\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#Part_7_Advanced_Topics_%E2%80%93_SAF_WNIC2_card_type_ISDN_troubleshooting_Q_201%E2%80%93210\" >Part 7: Advanced Topics \u2013 SAF, WNIC2, card type, ISDN troubleshooting (Q 201\u2013210)<\/a><ul class='ez-toc-list-level-3' ><li class='ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-58\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#201_What_is_SAF_Service_Advertisement_Framework\" >201. What is SAF (Service Advertisement Framework)?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-59\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#202_What_is_WNIC2\" >202. What is WNIC2?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-60\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#203_How_do_you_set_the_card_type_for_E1\" >203. How do you set the card type for E1?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-61\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#204_Scenario_show_isdn_status_shows_Layer_2_as_TEI_ASSIGNED_not_MULTIPLE_FRAME_ESTABLISHED_How_do_you_fix\" >204. Scenario:\u00a0show isdn status\u00a0shows Layer 2 as TEI_ASSIGNED (not MULTIPLE_FRAME_ESTABLISHED). How do you fix?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-62\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#205_What_do_SABME_messages_in_debug_isdn_q921_indicate\" >205. What do SABME messages in\u00a0debug isdn q921\u00a0indicate?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-63\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#206_What_is_the_desired_state_for_ISDN_Layer_2\" >206. What is the desired state for ISDN Layer 2?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-64\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#207_What_are_common_causes_of_clocking_problems_on_T1E1_PRI\" >207. What are common causes of clocking problems on T1\/E1 PRI?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-65\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#208_How_do_you_isolate_a_clocking_problem\" >208. How do you isolate a clocking problem?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-66\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#209_What_is_the_debug_isdn_q921_command_used_for\" >209. What is the\u00a0debug isdn q921\u00a0command used for?<\/a><\/li><li class='ez-toc-page-1 ez-toc-heading-level-3'><a class=\"ez-toc-link ez-toc-heading-67\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#210_Final_advanced_scenario_T1_PRI_Layer_1_is_ACTIVE_Layer_2_is_MULTIPLE_FRAME_ESTABLISHED_but_calls_fail_with_%E2%80%9CUnallocated_Number%E2%80%9D_What_is_wrong\" >210. Final advanced scenario: T1 PRI Layer 1 is ACTIVE, Layer 2 is MULTIPLE_FRAME_ESTABLISHED, but calls fail with &#8220;Unallocated Number&#8221;. What is wrong?<\/a><\/li><\/ul><\/li><li class='ez-toc-page-1 ez-toc-heading-level-2'><a class=\"ez-toc-link ez-toc-heading-68\" href=\"https:\/\/techgyan360.com\/blog\/sip-trunk-dtmf-relay-cucm-mobility-interview-questions\/#Conclusion\" >Conclusion<\/a><\/li><\/ul><\/nav><\/div>\n<h2><span class=\"ez-toc-section\" id=\"Part_1_SIP_Methods_and_Message_Responses_Q_151%E2%80%93160\"><\/span><span class=\"\">Part 1: SIP Methods and Message Responses (Q 151\u2013160)<\/span><span class=\"ez-toc-section-end\"><\/span><\/h2>\n<h3><span class=\"ez-toc-section\" id=\"151_What_are_the_six_core_SIP_methods_requests\"><\/span><span class=\"\">151. What are the six core SIP methods (requests)?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<ol start=\"1\">\n<li>\n<p class=\"ds-markdown-paragraph\"><strong><span class=\"\">INVITE<\/span><\/strong><span class=\"\">\u00a0\u2013 Initiates a call session.<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><strong><span class=\"\">ACK<\/span><\/strong><span class=\"\">\u00a0\u2013 Confirms final response to INVITE.<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><strong><span class=\"\">BYE<\/span><\/strong><span class=\"\">\u00a0\u2013 Terminates an existing session.<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><strong><span class=\"\">CANCEL<\/span><\/strong><span class=\"\">\u00a0\u2013 Cancels a pending INVITE (before final response).<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><strong><span class=\"\">REGISTER<\/span><\/strong><span class=\"\">\u00a0\u2013 Registers a user agent&#8217;s location with a SIP server.<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><strong><span class=\"\">OPTIONS<\/span><\/strong><span class=\"\">\u00a0\u2013 Queries server\/capabilities (no session established).<\/span><\/p>\n<\/li>\n<\/ol>\n<h3><span class=\"ez-toc-section\" id=\"152_What_is_the_purpose_of_the_ACK_method\"><\/span><span class=\"\">152. What is the purpose of the ACK method?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">ACK confirms that the final response (e.g., 200 OK) to an INVITE was received. If the calling party includes a session description in the ACK, that description overrides the one in the INVITE.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"153_What_is_the_difference_between_BYE_and_CANCEL\"><\/span><span class=\"\">153. What is the difference between BYE and CANCEL?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<ul>\n<li>\n<p class=\"ds-markdown-paragraph\"><strong><span class=\"\">BYE<\/span><\/strong><span class=\"\">\u00a0\u2013 Ends an\u00a0<\/span><strong><span class=\"\">established<\/span><\/strong><span class=\"\">\u00a0session (after 200 OK is sent).<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><strong><span class=\"\">CANCEL<\/span><\/strong><span class=\"\">\u00a0\u2013 Cancels a\u00a0<\/span><strong><span class=\"\">pending<\/span><\/strong><span class=\"\">\u00a0INVITE (before final response). CANCEL does not affect completed requests.<\/span><\/p>\n<\/li>\n<\/ul>\n<h3><span class=\"ez-toc-section\" id=\"154_What_are_the_six_classes_of_SIP_responses\"><\/span><span class=\"\">154. What are the six classes of SIP responses?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<div class=\"ds-scroll-area ds-scroll-area--show-on-focus-within _1210dd7 c03cafe9\">\n<div class=\"ds-scroll-area__gutters\">\n<div class=\"ds-scroll-area__horizontal-gutter\"><\/div>\n<div class=\"ds-scroll-area__vertical-gutter\"><\/div>\n<\/div>\n<table>\n<thead>\n<tr>\n<th><span class=\"\">Class<\/span><\/th>\n<th><span class=\"\">Range<\/span><\/th>\n<th><span class=\"\">Description<\/span><\/th>\n<\/tr>\n<\/thead>\n<tbody>\n<tr>\n<td><span class=\"\">1xx<\/span><\/td>\n<td><span class=\"\">100\u2013199<\/span><\/td>\n<td><span class=\"\">Provisional (progress, e.g., 100 Trying, 180 Ringing)<\/span><\/td>\n<\/tr>\n<tr>\n<td><span class=\"\">2xx<\/span><\/td>\n<td><span class=\"\">200\u2013299<\/span><\/td>\n<td><span class=\"\">Success (e.g., 200 OK)<\/span><\/td>\n<\/tr>\n<tr>\n<td><span class=\"\">3xx<\/span><\/td>\n<td><span class=\"\">300\u2013399<\/span><\/td>\n<td><span class=\"\">Redirection (e.g., 302 Moved Temporarily)<\/span><\/td>\n<\/tr>\n<tr>\n<td><span class=\"\">4xx<\/span><\/td>\n<td><span class=\"\">400\u2013499<\/span><\/td>\n<td><span class=\"\">Client Failure (e.g., 404 Not Found, 486 Busy)<\/span><\/td>\n<\/tr>\n<tr>\n<td><span class=\"\">5xx<\/span><\/td>\n<td><span class=\"\">500\u2013599<\/span><\/td>\n<td><span class=\"\">Server Failure (e.g., 503 Service Unavailable)<\/span><\/td>\n<\/tr>\n<tr>\n<td><span class=\"\">6xx<\/span><\/td>\n<td><span class=\"\">600\u2013699<\/span><\/td>\n<td><span class=\"\">Global Failure (e.g., 603 Decline)<\/span><\/td>\n<\/tr>\n<\/tbody>\n<\/table>\n<\/div>\n<h3><span class=\"ez-toc-section\" id=\"155_What_is_a_100_Trying_response\"><\/span><span class=\"\">155. What is a 100 Trying response?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">A provisional response sent by the next-hop server to indicate that the INVITE was received and processing is in progress. It stops retransmissions of the INVITE.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"156_What_is_a_180_Ringing_response\"><\/span><span class=\"\">156. What is a 180 Ringing response?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Sent by the called party&#8217;s user agent to indicate that the phone is ringing.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"157_What_is_a_183_Session_Progress_response\"><\/span><span class=\"\">157. What is a 183 Session Progress response?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Used to send early media (e.g., ringback tones, announcements) before the call is answered. Can contain SDP for media establishment.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"158_What_does_a_404_Not_Found_response_indicate\"><\/span><span class=\"\">158. What does a 404 Not Found response indicate?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">The requested user or domain does not exist at the server. Usually means a dial plan mismatch or unregistered endpoint.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"159_What_does_a_486_Busy_Here_response_mean\"><\/span><span class=\"\">159. What does a 486 Busy Here response mean?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">The called party is currently busy and cannot accept the call. The endpoint is reachable but not available.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"160_Scenario_You_receive_a_503_Service_Unavailable_from_a_SIP_trunk_provider_What_does_it_mean\"><\/span><span class=\"\">160. Scenario: You receive a 503 Service Unavailable from a SIP trunk provider. What does it mean?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><strong><span class=\"\">Answer:<\/span><\/strong><span class=\"\">\u00a0The provider&#8217;s server is overloaded or undergoing maintenance. The call should be retried later or routed via an alternate trunk (PSTN backup).<\/span><\/p>\n<hr \/>\n<h2><span class=\"ez-toc-section\" id=\"Part_2_SIP_UA_and_Voice_Service_Configuration_Q_161%E2%80%93170\"><\/span><span class=\"\">Part 2: SIP UA and Voice Service Configuration (Q 161\u2013170)<\/span><span class=\"ez-toc-section-end\"><\/span><\/h2>\n<h3><span class=\"ez-toc-section\" id=\"161_What_is_SIP_UA_User_Agent_configuration_mode\"><\/span><span class=\"\">161. What is SIP UA (User Agent) configuration mode?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Entered via\u00a0<\/span><code>sip-ua<\/code><span class=\"\">\u00a0command on a Cisco IOS gateway. It controls SIP client behavior including registration, timers, NAT traversal, and retry counts.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"162_What_does_the_registrar_command_do_in_SIP_UA\"><\/span><span class=\"\">162. What does the\u00a0<\/span><code>registrar<\/code><span class=\"\">\u00a0command do in SIP UA?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">It allows a gateway to register E.164 numbers of non-SIP phones (analog FXS, etc.) with a SIP registrar or proxy server. Example:<\/span><\/p>\n<div class=\"md-code-block md-code-block-light\">\n<div class=\"md-code-block-banner-wrap\">\n<div class=\"md-code-block-banner md-code-block-banner-lite\">\n<div class=\"_121d384\">\n<div class=\"d2a24f03\"><span class=\"d813de27\">text<\/span><\/div>\n<\/div>\n<\/div>\n<\/div>\n<pre>registrar ipv4:10.30.25.250 tcp expires 3600<\/pre>\n<\/div>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Secondary registrar can be configured for redundancy.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"163_What_does_the_sip-server_command_do\"><\/span><span class=\"\">163. What does the\u00a0<\/span><code>sip-server<\/code><span class=\"\">\u00a0command do?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Specifies the name or IP address of a SIP server (usually a proxy). When configured, dial peers can use\u00a0<\/span><code>session target sip-server<\/code><span class=\"\">\u00a0instead of explicit IP addresses.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"164_What_does_the_max-forwards_command_do\"><\/span><span class=\"\">164. What does the\u00a0<\/span><code>max-forwards<\/code><span class=\"\">\u00a0command do?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Limits the maximum number of hops a SIP request can traverse (default 70). Each proxy decrements the value. When it reaches 0, the request is discarded \u2013 prevents infinite loops.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"165_What_SIP_UA_timers_can_be_adjusted\"><\/span><span class=\"\">165. What SIP UA timers can be adjusted?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<ul>\n<li>\n<p class=\"ds-markdown-paragraph\"><code>timers connect<\/code><span class=\"\">\u00a0\u2013 Wait for 200 OK to ACK (default 500ms)<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><code>timers disconnect<\/code><span class=\"\">\u00a0\u2013 Wait for 200 OK to BYE (default 500ms)<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><code>timers expires<\/code><span class=\"\">\u00a0\u2013 Valid time for an INVITE (default 180 seconds)<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><code>timers trying<\/code><span class=\"\">\u00a0\u2013 Wait for 100 Trying response (default 500ms)<\/span><\/p>\n<\/li>\n<\/ul>\n<h3><span class=\"ez-toc-section\" id=\"166_What_is_redirect_ip2ip_in_voice_service_VoIP\"><\/span><span class=\"\">166. What is\u00a0<\/span><code>redirect ip2ip<\/code><span class=\"\">\u00a0in voice service VoIP?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">It allows SIP calls to be &#8220;hairpinned&#8221; \u2013 received on one VoIP dial peer and sent out another VoIP dial peer. Essential for gateway-to-gateway toll bypass.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"167_What_does_bind_control_source-interface_do\"><\/span><span class=\"\">167. What does\u00a0<\/span><code>bind control source-interface<\/code><span class=\"\">\u00a0do?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Sets the source IP address for SIP control signaling (not media). Example:<\/span><\/p>\n<div class=\"md-code-block md-code-block-light\">\n<div class=\"md-code-block-banner-wrap\">\n<div class=\"md-code-block-banner md-code-block-banner-lite\">\n<div class=\"_121d384\">\n<div class=\"d2a24f03\"><span class=\"d813de27\">text<\/span><\/div>\n<\/div>\n<\/div>\n<\/div>\n<pre>bind control source-interface Loopback0<\/pre>\n<\/div>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">This ensures consistent source IP for SIP messages, critical for NAT and firewall traversal.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"168_What_does_bind_media_source-interface_do\"><\/span><span class=\"\">168. What does\u00a0<\/span><code>bind media source-interface<\/code><span class=\"\">\u00a0do?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Sets the source IP address for RTP media streams. If not configured, the gateway uses the outgoing interface IP.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"169_What_is_the_difference_between_session_transport_udp_and_tcp_for_SIP\"><\/span><span class=\"\">169. What is the difference between\u00a0<\/span><code>session transport udp<\/code><span class=\"\">\u00a0and\u00a0<\/span><code>tcp<\/code><span class=\"\">\u00a0for SIP?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<ul>\n<li>\n<p class=\"ds-markdown-paragraph\"><strong><span class=\"\">UDP<\/span><\/strong><span class=\"\">\u00a0\u2013 Default. Lightweight but unreliable; no message fragmentation.<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><strong><span class=\"\">TCP<\/span><\/strong><span class=\"\">\u00a0\u2013 Reliable; used when SIP messages exceed MTU (fragmentation avoidance). Can be globally set with\u00a0<\/span><code>session transport tcp<\/code><span class=\"\">\u00a0or dynamically with\u00a0<\/span><code>transport switch udp tcp<\/code><span class=\"\">.<\/span><\/p>\n<\/li>\n<\/ul>\n<h3><span class=\"ez-toc-section\" id=\"170_Scenario_Your_SIP_gateway_sends_large_INVITE_messages_that_get_fragmented_How_do_you_fix\"><\/span><span class=\"\">170. Scenario: Your SIP gateway sends large INVITE messages that get fragmented. How do you fix?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><strong><span class=\"\">Answer:<\/span><\/strong><span class=\"\">\u00a0Configure\u00a0<\/span><code>transport switch udp tcp<\/code><span class=\"\">\u00a0under voice service VoIP \u2192 SIP. This forces the gateway to switch from UDP to TCP when a SIP message exceeds the MTU (usually 1500 bytes).<\/span><\/p>\n<hr \/>\n<h2><span class=\"ez-toc-section\" id=\"Part_3_DTMF_Relay_Methods_Q_171%E2%80%93180\"><\/span><span class=\"\">Part 3: DTMF Relay Methods (Q 171\u2013180)<\/span><span class=\"ez-toc-section-end\"><\/span><\/h2>\n<h3><span class=\"ez-toc-section\" id=\"171_Why_is_DTMF_relay_necessary\"><\/span><span class=\"\">171. Why is DTMF relay necessary?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Low-bandwidth codecs (G.729, G.723) compress voice and can distort or completely lose DTMF tones (touch tones). DTMF relay sends tone information\u00a0<\/span><strong><span class=\"\">out-of-band<\/span><\/strong><span class=\"\">\u00a0or using special packets.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"172_What_is_RTP-NTE_RFC_2833_DTMF_relay\"><\/span><span class=\"\">172. What is RTP-NTE (RFC 2833) DTMF relay?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">In-band method where DTMF tones are sent as special RTP Named Telephony Event packets instead of encoded voice. SDP negotiates the payload type. Most common and interoperable method.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"173_What_is_SIP-NOTIFY_DTMF_relay\"><\/span><span class=\"\">173. What is SIP-NOTIFY DTMF relay?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Out-of-band method. DTMF digits are sent in SIP NOTIFY messages. Negotiated via\u00a0<\/span><code>Call-Info<\/code><span class=\"\">\u00a0header in INVITE\/200 OK. Preferred for SCCP to SIP interworking.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"174_What_is_SIP-INFO_DTMF_relay\"><\/span><span class=\"\">174. What is SIP-INFO DTMF relay?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Out-of-band method using SIP INFO messages. Always enabled on Cisco gateways. Good for interoperability but less common than NOTIFY.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"175_What_is_KPML_KPML_DTMF_relay\"><\/span><span class=\"\">175. What is KPML (KPML) DTMF relay?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Keg Pad Markup Language \u2013 XML-based method where the phone sends digit information in SUBSCRIBE\/NOTIFY messages. More efficient for long digit strings (e.g., IVR menu traversal).<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"176_How_do_you_configure_DTMF_relay_on_a_SIP_dial_peer\"><\/span><span class=\"\">176. How do you configure DTMF relay on a SIP dial peer?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<div class=\"md-code-block md-code-block-light\">\n<div class=\"md-code-block-banner-wrap\">\n<div class=\"md-code-block-banner md-code-block-banner-lite\">\n<div class=\"_121d384\">\n<div class=\"d2a24f03\"><span class=\"d813de27\">text<\/span><\/div>\n<\/div>\n<\/div>\n<\/div>\n<pre>dial-peer voice 100 voip\r\n dtmf-relay sip-notify rtp-nte<\/pre>\n<\/div>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">This tells the gateway to prefer SIP-NOTIFY, but fall back to RTP-NTE if the remote side doesn&#8217;t support NOTIFY.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"177_Scenario_SIP_to_SCCP_calls_have_DTMF_failures_IVR_doesnt_recognize_digits_Why\"><\/span><span class=\"\">177. Scenario: SIP to SCCP calls have DTMF failures (IVR doesn&#8217;t recognize digits). Why?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><strong><span class=\"\">Answer:<\/span><\/strong><span class=\"\">\u00a0SCCP uses out-of-band DTMF only. SIP trunk between CUCM and the gateway may be using in-band (RTP-NTE) by default. In CUCM 4.x, you\u00a0<\/span><strong><span class=\"\">must<\/span><\/strong><span class=\"\">\u00a0enable an MTP on the SIP trunk to translate between in-band and out-of-band. CUCM 5.x+ can negotiate without MTP if both sides support out-of-band.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"178_What_is_the_notify_telephone-event_max-duration_command\"><\/span><span class=\"\">178. What is the\u00a0<\/span><code>notify telephone-event max-duration<\/code><span class=\"\">\u00a0command?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Sets the maximum duration (in milliseconds) between NOTIFY messages for a single DTMF event. Default 2000ms (2 seconds). The lower value between two SIP peers is used.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"179_Scenario_DTMF_digits_are_recognized_but_repeated_multiple_times_How_to_fix\"><\/span><span class=\"\">179. Scenario: DTMF digits are recognized but repeated multiple times. How to fix?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><strong><span class=\"\">Answer:<\/span><\/strong><span class=\"\">\u00a0The DTMF relay method may be sending duplicate events. Check\u00a0<\/span><code>dtmf-relay<\/code><span class=\"\">\u00a0configuration. If using RTP-NTE, ensure the gateway is not also sending in-band tones (disable\u00a0<\/span><code>dtmf-relay cisco-rtp<\/code><span class=\"\">). Adjust timers if using NOTIFY.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"180_What_is_the_default_DTMF_method_if_no_dtmf-relay_is_configured\"><\/span><span class=\"\">180. What is the default DTMF method if no\u00a0<\/span><code>dtmf-relay<\/code><span class=\"\">\u00a0is configured?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">In-band \u2013 DTMF tones are sent as normal voice audio through the codec. This fails with G.729 and is unreliable with G.711 in noisy environments.<\/span><\/p>\n<hr \/>\n<h2><span class=\"ez-toc-section\" id=\"Part_4_SIP_Trunk_Configuration_%E2%80%93_CUCM_4x_vs_5x_Q_181%E2%80%93185\"><\/span><span class=\"\">Part 4: SIP Trunk Configuration \u2013 CUCM 4.x vs 5.x (Q 181\u2013185)<\/span><span class=\"ez-toc-section-end\"><\/span><\/h2>\n<h3><span class=\"ez-toc-section\" id=\"181_How_do_you_configure_a_SIP_trunk_in_CUCM_4x\"><\/span><span class=\"\">181. How do you configure a SIP trunk in CUCM 4.x?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<ol start=\"1\">\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Device \u2192 Trunk \u2192 Add a New Trunk \u2192 SIP Trunk.<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Device Protocol automatically set to SIP.<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Enter destination address (IP of remote gateway\/proxy).<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><strong><span class=\"\">MTP Required<\/span><\/strong><span class=\"\">\u00a0\u2013 Must be checked (to translate DTMF between SCCP and SIP).<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Destination Port \u2013 default 5060.<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><strong><span class=\"\">Incoming Port<\/span><\/strong><span class=\"\">\u00a0\u2013 Must be unique for each signaling interface in CUCM 4.x.<\/span><\/p>\n<\/li>\n<\/ol>\n<h3><span class=\"ez-toc-section\" id=\"182_How_does_SIP_trunk_configuration_differ_in_CUCM_5x\"><\/span><span class=\"\">182. How does SIP trunk configuration differ in CUCM 5.x+?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<ul>\n<li>\n<p class=\"ds-markdown-paragraph\"><strong><span class=\"\">MTP Required<\/span><\/strong><span class=\"\">\u00a0\u2013 Not automatically checked; may not be needed if endpoints negotiate out-of-band DTMF.<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><strong><span class=\"\">Incoming Port<\/span><\/strong><span class=\"\">\u00a0\u2013 Multiple trunks can share the same incoming port (5060).<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">New fields: Presence Group, SIP Trunk Security Profile, SIP Profile, SUBSCRIBE Calling Search Space.<\/span><\/p>\n<\/li>\n<\/ul>\n<h3><span class=\"ez-toc-section\" id=\"183_Why_did_CUCM_4x_require_an_MTP_on_SIP_trunks\"><\/span><span class=\"\">183. Why did CUCM 4.x require an MTP on SIP trunks?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">SCCP phones use out-of-band DTMF. SIP trunks default to in-band DTMF (RTP-NTE). The MTP translated between the two. CUCM 5.x+ improved SIP stack to negotiate out-of-band without MTP.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"184_Scenario_Your_SIP_trunk_works_for_voice_but_fails_for_fax_G711_pass-through_What_is_wrong\"><\/span><span class=\"\">184. Scenario: Your SIP trunk works for voice but fails for fax (G.711 pass-through). What is wrong?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><strong><span class=\"\">Answer:<\/span><\/strong><span class=\"\">\u00a0SIP trunks often need\u00a0<\/span><code>dtmf-relay<\/code><span class=\"\">\u00a0configured on the gateway side. For fax, ensure\u00a0<\/span><code>fax protocol pass-through g711ulaw<\/code><span class=\"\">\u00a0and disable VAD (no vad). Also verify that the SIP trunk is not forcing an MTP that breaks T.38 or pass-through.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"185_What_is_toll_bypass_using_SIP\"><\/span><span class=\"\">185. What is toll bypass using SIP?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Toll bypass sends calls between two sites over the IP network (SIP between routers) instead of the PSTN, avoiding long-distance charges. Configure VoIP dial peers on both routers pointing to each other&#8217;s IP addresses with\u00a0<\/span><code>session protocol sipv2<\/code><span class=\"\">.<\/span><\/p>\n<p><strong>Check<\/strong> &#8211; <a href=\"https:\/\/tools.kloudmint.com\/aws-cloud-tools\/aws-policy-generator.php\">AWS Policy Generator Tool<\/a><\/p>\n<hr \/>\n<h2><span class=\"ez-toc-section\" id=\"Part_5_Mobile_Voice_Access_MVA_and_SNR_Q_186%E2%80%93195\"><\/span><span class=\"\">Part 5: Mobile Voice Access (MVA) and SNR (Q 186\u2013195)<\/span><span class=\"ez-toc-section-end\"><\/span><\/h2>\n<h3><span class=\"ez-toc-section\" id=\"186_What_is_Mobile_Voice_Access_MVA\"><\/span><span class=\"\">186. What is Mobile Voice Access (MVA)?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">MVA is a CUCM feature that allows a PSTN user (e.g., mobile phone) to dial into the office, authenticate with a PIN, and receive an internal dial tone. The mobile phone then acts like an internal IP phone \u2013 can dial extensions, transfer calls, place calls that appear from the desk phone.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"187_How_does_MVA_hairpinning_work\"><\/span><span class=\"\">187. How does MVA hairpinning work?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">The PSTN call comes into the voice gateway, is sent to CUCM, authenticated via VXML, and then CUCM sends a new call leg back out the same gateway to the final destination. The carrier may charge for both legs (hairpinning fee). Negotiate this with your telco.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"188_What_is_the_VXML_URL_used_for_MVA\"><\/span><span class=\"\">188. What is the VXML URL used for MVA?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">CUCM serves an IVR script (VXML) that prompts the user for PIN. Example configuration on the gateway:<\/span><\/p>\n<div class=\"md-code-block md-code-block-light\">\n<div class=\"md-code-block-banner-wrap\">\n<div class=\"md-code-block-banner md-code-block-banner-lite\">\n<div class=\"_121d384\">\n<div class=\"d2a24f03\"><span class=\"d813de27\">text<\/span><\/div>\n<\/div>\n<\/div>\n<\/div>\n<pre>application\r\n service frog http:\/\/142.2.64.11:8080\/ccmivr\/pages\/IVRMainpage.vxml<\/pre>\n<\/div>\n<h3><span class=\"ez-toc-section\" id=\"189_How_do_you_configure_a_dial_peer_for_MVA_on_an_H323_gateway\"><\/span><span class=\"\">189. How do you configure a dial peer for MVA on an H.323 gateway?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<div class=\"md-code-block md-code-block-light\">\n<div class=\"md-code-block-banner-wrap\">\n<div class=\"md-code-block-banner md-code-block-banner-lite\">\n<div class=\"_121d384\">\n<div class=\"d2a24f03\"><span class=\"d813de27\">text<\/span><\/div>\n<\/div>\n<\/div>\n<\/div>\n<pre>dial-peer voice 20 voip\r\n service frog\r\n session target ipv4:142.2.64.11\r\n incoming called-number 99992222   ! MVA DID number\r\n codec g711ulaw\r\n no vad<\/pre>\n<\/div>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">The\u00a0<\/span><code>service frog<\/code><span class=\"\">\u00a0command links the dial peer to the VXML application.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"190_What_is_Single_Number_Reach_SNR_Mobile_Connect\"><\/span><span class=\"\">190. What is Single Number Reach (SNR) \/ Mobile Connect?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">SNR allows incoming calls to a desk phone to simultaneously ring up to four remote destinations (mobile phone, home phone). The user can answer from any device. Call appears to come from the desk phone (Caller ID preservation).<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"191_What_is_the_difference_between_MVA_and_SNR\"><\/span><span class=\"\">191. What is the difference between MVA and SNR?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<div class=\"ds-scroll-area ds-scroll-area--show-on-focus-within _1210dd7 c03cafe9\">\n<div class=\"ds-scroll-area__gutters\">\n<div class=\"ds-scroll-area__horizontal-gutter\"><\/div>\n<div class=\"ds-scroll-area__vertical-gutter\"><\/div>\n<\/div>\n<table>\n<thead>\n<tr>\n<th><span class=\"\">Feature<\/span><\/th>\n<th><span class=\"\">SNR (Mobile Connect)<\/span><\/th>\n<th><span class=\"\">MVA (Mobile Voice Access)<\/span><\/th>\n<\/tr>\n<\/thead>\n<tbody>\n<tr>\n<td><span class=\"\">Direction<\/span><\/td>\n<td><span class=\"\">Inbound only (calls to desk phone ring mobile)<\/span><\/td>\n<td><span class=\"\">Outbound only (mobile calls out using desk identity)<\/span><\/td>\n<\/tr>\n<tr>\n<td><span class=\"\">Authentication<\/span><\/td>\n<td><span class=\"\">None (mobile is pre-registered)<\/span><\/td>\n<td><span class=\"\">PIN required<\/span><\/td>\n<\/tr>\n<tr>\n<td><span class=\"\">Hairpinning<\/span><\/td>\n<td><span class=\"\">Yes (call comes in, goes out to mobile)<\/span><\/td>\n<td><span class=\"\">Yes (call comes in, goes out to destination)<\/span><\/td>\n<\/tr>\n<\/tbody>\n<\/table>\n<\/div>\n<h3><span class=\"ez-toc-section\" id=\"192_What_is_the_BHCA_formula_for_SNR_remote_destinations\"><\/span><span class=\"\">192. What is the BHCA formula for SNR remote destinations?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Each remote destination adds 0.5 call legs per call. Formula:<\/span><\/p>\n<div class=\"md-code-block md-code-block-light\">\n<div class=\"md-code-block-banner-wrap\">\n<div class=\"md-code-block-banner md-code-block-banner-lite\">\n<div class=\"_121d384\">\n<div class=\"d2a24f03\"><span class=\"d813de27\">text<\/span><\/div>\n<\/div>\n<\/div>\n<\/div>\n<pre>Remote Destination BHCA = 0.5 \u00d7 (Number of users) \u00d7 (User BHCA)<\/pre>\n<\/div>\n<p class=\"ds-markdown-paragraph\"><strong><span class=\"\">Example:<\/span><\/strong><span class=\"\">\u00a0300 users, each 5 BHCA, each with 1 remote destination:<\/span><\/p>\n<ul>\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Remote BHCA = 0.5 \u00d7 (300 \u00d7 5) = 750<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Total system BHCA = (300 \u00d7 5) + 750 = 2250<\/span><\/p>\n<\/li>\n<\/ul>\n<h3><span class=\"ez-toc-section\" id=\"193_Scenario_A_user_reports_that_MVA_authentication_fails_PIN_not_recognized\"><\/span><span class=\"\">193. Scenario: A user reports that MVA authentication fails (PIN not recognized).<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><strong><span class=\"\">Answer:<\/span><\/strong><span class=\"\">\u00a0Check:<\/span><\/p>\n<ul>\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">User has a PIN configured in CUCM (not just a password).<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">User has &#8220;Enable Mobile Voice Access&#8221; checked in Mobility settings.<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">The MVA DID number in clusterwide parameters matches the inbound dial peer.<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">CSS on the MVA route pattern allows access to internal partitions.<\/span><\/p>\n<\/li>\n<\/ul>\n<h3><span class=\"ez-toc-section\" id=\"194_What_DTMF_digits_are_available_during_an_MVA_call_for_call_control\"><\/span><span class=\"\">194. What DTMF digits are available during an MVA call for call control?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<div class=\"ds-scroll-area ds-scroll-area--show-on-focus-within _1210dd7 c03cafe9\">\n<div class=\"ds-scroll-area__gutters\">\n<div class=\"ds-scroll-area__horizontal-gutter\"><\/div>\n<div class=\"ds-scroll-area__vertical-gutter\"><\/div>\n<\/div>\n<table>\n<thead>\n<tr>\n<th><span class=\"\">Action<\/span><\/th>\n<th><span class=\"\">Digit<\/span><\/th>\n<\/tr>\n<\/thead>\n<tbody>\n<tr>\n<td><span class=\"\">Hold call<\/span><\/td>\n<td><span class=\"\">*81<\/span><\/td>\n<\/tr>\n<tr>\n<td><span class=\"\">Exclusive hold<\/span><\/td>\n<td><span class=\"\">*82<\/span><\/td>\n<\/tr>\n<tr>\n<td><span class=\"\">Resume call<\/span><\/td>\n<td><span class=\"\">*83<\/span><\/td>\n<\/tr>\n<tr>\n<td><span class=\"\">Transfer<\/span><\/td>\n<td><span class=\"\">*84<\/span><\/td>\n<\/tr>\n<tr>\n<td><span class=\"\">Conference<\/span><\/td>\n<td><span class=\"\">*85<\/span><\/td>\n<\/tr>\n<\/tbody>\n<\/table>\n<\/div>\n<h3><span class=\"ez-toc-section\" id=\"195_Scenario_MVA_call_connects_but_there_is_no_dial_tone_after_PIN_entry\"><\/span><span class=\"\">195. Scenario: MVA call connects but there is no dial tone after PIN entry.<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><strong><span class=\"\">Answer:<\/span><\/strong><span class=\"\">\u00a0Check:<\/span><\/p>\n<ul>\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">The route pattern for MVA DID has &#8220;Provide Outside Dial Tone&#8221; checked.<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">The gateway has\u00a0<\/span><code>voice service voip<\/code><span class=\"\">\u00a0with\u00a0<\/span><code>allow-connections h323 to h323<\/code><span class=\"\">\u00a0(for H.323 hairpinning).<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">The CSS after authentication includes access to route patterns.<\/span><\/p>\n<\/li>\n<\/ul>\n<hr \/>\n<h2><span class=\"ez-toc-section\" id=\"Part_6_Call_Park_Meet-Me_Conference_Custom_Ringtones_Q_196%E2%80%93200\"><\/span><span class=\"\">Part 6: Call Park, Meet-Me Conference, Custom Ringtones (Q 196\u2013200)<\/span><span class=\"ez-toc-section-end\"><\/span><\/h2>\n<h3><span class=\"ez-toc-section\" id=\"196_What_is_Call_Park\"><\/span><span class=\"\">196. What is Call Park?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">A feature that allows a user to &#8220;park&#8221; an active call into a virtual slot (extension). The phone displays the slot number. Any other phone can dial that slot number to retrieve the call.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"197_What_is_Directed_Call_Park_How_is_it_different\"><\/span><span class=\"\">197. What is Directed Call Park? How is it different?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<div class=\"ds-scroll-area ds-scroll-area--show-on-focus-within _1210dd7 c03cafe9\">\n<div class=\"ds-scroll-area__gutters\">\n<div class=\"ds-scroll-area__horizontal-gutter\"><\/div>\n<div class=\"ds-scroll-area__vertical-gutter\"><\/div>\n<\/div>\n<table>\n<thead>\n<tr>\n<th><span class=\"\">Feature<\/span><\/th>\n<th><span class=\"\">Call Park<\/span><\/th>\n<th><span class=\"\">Directed Call Park<\/span><\/th>\n<\/tr>\n<\/thead>\n<tbody>\n<tr>\n<td><span class=\"\">Slot selection<\/span><\/td>\n<td><span class=\"\">System chooses first available<\/span><\/td>\n<td><span class=\"\">User dials a specific slot<\/span><\/td>\n<\/tr>\n<tr>\n<td><span class=\"\">Retrieval method<\/span><\/td>\n<td><span class=\"\">Dial the slot number<\/span><\/td>\n<td><span class=\"\">Dial retrieval prefix + slot number<\/span><\/td>\n<\/tr>\n<tr>\n<td><span class=\"\">Security<\/span><\/td>\n<td><span class=\"\">Low (anyone can retrieve)<\/span><\/td>\n<td><span class=\"\">Higher (need to know prefix)<\/span><\/td>\n<\/tr>\n<\/tbody>\n<\/table>\n<\/div>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Retrieval prefix is configured separately (e.g., *51). User parks call to slot &#8220;01&#8221;. Retrieving user dials *5101.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"198_What_is_the_difference_between_Ad-Hoc_and_Meet-Me_Conference\"><\/span><span class=\"\">198. What is the difference between Ad-Hoc and Meet-Me Conference?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<div class=\"ds-scroll-area ds-scroll-area--show-on-focus-within _1210dd7 c03cafe9\">\n<div class=\"ds-scroll-area__gutters\">\n<div class=\"ds-scroll-area__horizontal-gutter\"><\/div>\n<div class=\"ds-scroll-area__vertical-gutter\"><\/div>\n<\/div>\n<table>\n<thead>\n<tr>\n<th><span class=\"\">Feature<\/span><\/th>\n<th><span class=\"\">Ad-Hoc<\/span><\/th>\n<th><span class=\"\">Meet-Me<\/span><\/th>\n<\/tr>\n<\/thead>\n<tbody>\n<tr>\n<td><span class=\"\">Initiation<\/span><\/td>\n<td><span class=\"\">Active user adds participants<\/span><\/td>\n<td><span class=\"\">All users dial a common bridge number<\/span><\/td>\n<\/tr>\n<tr>\n<td><span class=\"\">Bridge duration<\/span><\/td>\n<td><span class=\"\">Ends when initiator leaves<\/span><\/td>\n<td><span class=\"\">Stays up until last participant leaves<\/span><\/td>\n<\/tr>\n<tr>\n<td><span class=\"\">Scheduling<\/span><\/td>\n<td><span class=\"\">Impromptu<\/span><\/td>\n<td><span class=\"\">Scheduled in advance<\/span><\/td>\n<\/tr>\n<tr>\n<td><span class=\"\">Use case<\/span><\/td>\n<td><span class=\"\">Quick 3-way call<\/span><\/td>\n<td><span class=\"\">Regular team meeting<\/span><\/td>\n<\/tr>\n<\/tbody>\n<\/table>\n<\/div>\n<h3><span class=\"ez-toc-section\" id=\"199_How_do_you_add_a_custom_ringtone_to_a_Cisco_IP_Phone\"><\/span><span class=\"\">199. How do you add a custom ringtone to a Cisco IP Phone?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><strong><span class=\"\">Step 1:<\/span><\/strong><span class=\"\">\u00a0Convert MP3\/WAV to CCITT u-Law, 8,000 kHz, 8-bit, no header (use Sound Recorder or SOX).<\/span><br \/>\n<strong><span class=\"\">Step 2:<\/span><\/strong><span class=\"\">\u00a0Save as\u00a0<\/span><code>.raw<\/code><span class=\"\">\u00a0file (max 1.4 MB).<\/span><br \/>\n<strong><span class=\"\">Step 3:<\/span><\/strong><span class=\"\">\u00a0Copy to CUCM TFTP server path (<\/span><code>C:\\Program Files\\Cisco\\TFTPPath<\/code><span class=\"\">).<\/span><br \/>\n<strong><span class=\"\">Step 4:<\/span><\/strong><span class=\"\">\u00a0Edit\u00a0<\/span><code>Ringlist.xml<\/code><span class=\"\">\u00a0(add\u00a0<\/span><code>&lt;DisplayName&gt;<\/code><span class=\"\">\u00a0and\u00a0<\/span><code>&lt;FileName&gt;<\/code><span class=\"\">\u00a0entries).<\/span><br \/>\n<strong><span class=\"\">Step 5:<\/span><\/strong><span class=\"\">\u00a0Save and restart TFTP service. Phones download new ringlist on next reboot.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"200_Final_Scenario_Call_Park_works_but_Directed_Call_Park_fails_What_do_you_check\"><\/span><span class=\"\">200. Final Scenario: Call Park works but Directed Call Park fails. What do you check?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><strong><span class=\"\">Answer:<\/span><\/strong><span class=\"\">\u00a0Verify:<\/span><\/p>\n<ul>\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Directed Call Park slots are configured (Call Routing \u2192 Directed Call Park).<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Retrieval prefix is set (e.g., *51) and does not overlap with other patterns.<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">The user&#8217;s phone CSS includes the partition containing the Directed Call Park slots.<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">The user dials the retrieval prefix\u00a0<\/span><strong><span class=\"\">before<\/span><\/strong><span class=\"\">\u00a0the slot number.<\/span><\/p>\n<\/li>\n<\/ul>\n<hr \/>\n<h2><span class=\"ez-toc-section\" id=\"Part_7_Advanced_Topics_%E2%80%93_SAF_WNIC2_card_type_ISDN_troubleshooting_Q_201%E2%80%93210\"><\/span><span class=\"\">Part 7: Advanced Topics \u2013 SAF, WNIC2, card type, ISDN troubleshooting (Q 201\u2013210)<\/span><span class=\"ez-toc-section-end\"><\/span><\/h2>\n<h3><span class=\"ez-toc-section\" id=\"201_What_is_SAF_Service_Advertisement_Framework\"><\/span><span class=\"\">201. What is SAF (Service Advertisement Framework)?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">SAF is a network protocol modeled after\u00a0<\/span><strong><span class=\"\">EIGRP<\/span><\/strong><span class=\"\">\u00a0(Cisco proprietary). It enables sharing of call routing data between call agents (CUCM, CUCME, CUBE, SRST, IOS Voice Gateways). It advertises and requests routing data to\/from the SAF network.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"202_What_is_WNIC2\"><\/span><span class=\"\">202. What is WNIC2?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">WNIC2 is a voice interface card (port adapter) that can be configured for\u00a0<\/span><strong><span class=\"\">E1 or T1<\/span><\/strong><span class=\"\">\u00a0connectivity. There is no default card type \u2013 you\u00a0<\/span><strong><span class=\"\">must<\/span><\/strong><span class=\"\">\u00a0specify using the\u00a0<\/span><code>card type<\/code><span class=\"\">\u00a0command. Without it, the port adapter is non-functional and does not appear in\u00a0<\/span><code>show<\/code><span class=\"\">\u00a0commands.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"203_How_do_you_set_the_card_type_for_E1\"><\/span><span class=\"\">203. How do you set the card type for E1?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<div class=\"md-code-block md-code-block-light\">\n<div class=\"md-code-block-banner-wrap\">\n<div class=\"md-code-block-banner md-code-block-banner-lite\">\n<div class=\"_121d384\">\n<div class=\"d2a24f03\"><span class=\"d813de27\">text<\/span><\/div>\n<\/div>\n<\/div>\n<\/div>\n<pre>card type e1 0<\/pre>\n<\/div>\n<h3><span class=\"ez-toc-section\" id=\"204_Scenario_show_isdn_status_shows_Layer_2_as_TEI_ASSIGNED_not_MULTIPLE_FRAME_ESTABLISHED_How_do_you_fix\"><\/span><span class=\"\">204. Scenario:\u00a0<\/span><code>show isdn status<\/code><span class=\"\">\u00a0shows Layer 2 as TEI_ASSIGNED (not MULTIPLE_FRAME_ESTABLISHED). How do you fix?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><strong><span class=\"\">Step 1:<\/span><\/strong><span class=\"\">\u00a0Verify\u00a0<\/span><code>isdn switch-type<\/code><span class=\"\">\u00a0matches telco (e.g.,\u00a0<\/span><code>primary-5ess<\/code><span class=\"\">,\u00a0<\/span><code>primary-ni<\/code><span class=\"\">).<\/span><br \/>\n<strong><span class=\"\">Step 2:<\/span><\/strong><span class=\"\">\u00a0Verify\u00a0<\/span><code>pri-group timeslots<\/code><span class=\"\">\u00a0is correct.<\/span><br \/>\n<strong><span class=\"\">Step 3:<\/span><\/strong><span class=\"\">\u00a0Ensure D-channel interface (<\/span><code>serial x:23<\/code><span class=\"\">) is not shutdown.<\/span><br \/>\n<strong><span class=\"\">Step 4:<\/span><\/strong><span class=\"\">\u00a0Check cabling and clocking.<\/span><br \/>\n<strong><span class=\"\">Step 5:<\/span><\/strong><span class=\"\">\u00a0Contact telco to verify their side is up.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"205_What_do_SABME_messages_in_debug_isdn_q921_indicate\"><\/span><span class=\"\">205. What do SABME messages in\u00a0<\/span><code>debug isdn q921<\/code><span class=\"\">\u00a0indicate?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">SABME (Set Asynchronous Balanced Mode Extended) indicates Layer 2 is trying to reinitialize. If seen repeatedly, the link is unstable. Causes:<\/span><\/p>\n<ul>\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Wrong switch-type<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Clocking mismatch<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Physical layer issues (cabling, distance)<\/span><\/p>\n<\/li>\n<\/ul>\n<h3><span class=\"ez-toc-section\" id=\"206_What_is_the_desired_state_for_ISDN_Layer_2\"><\/span><span class=\"\">206. What is the desired state for ISDN Layer 2?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><strong><span class=\"\">MULTIPLE_FRAME_ESTABLISHED<\/span><\/strong><span class=\"\">\u00a0\u2013 Means Layer 2 frames are being exchanged and initialization is complete.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"207_What_are_common_causes_of_clocking_problems_on_T1E1_PRI\"><\/span><span class=\"\">207. What are common causes of clocking problems on T1\/E1 PRI?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<ul>\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Incorrect DSU\/CSU configuration<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Cables longer than 50 feet (unshielded)<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Poor patch panel connections<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Multiple cables connected in series<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Loopback left enabled on the interface<\/span><\/p>\n<\/li>\n<\/ul>\n<h3><span class=\"ez-toc-section\" id=\"208_How_do_you_isolate_a_clocking_problem\"><\/span><span class=\"\">208. How do you isolate a clocking problem?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<ol start=\"1\">\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Perform local loopback test (router looped to itself \u2013 errors stop? Problem is external).<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Perform remote loopback test (telco loops the line \u2013 errors continue? Problem is telco or CSU\/DSU).<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Use\u00a0<\/span><code>show controllers t1<\/code><span class=\"\">\u00a0to check for alarms (LOS, LOF, RAI, AIS).<\/span><\/p>\n<\/li>\n<\/ol>\n<h3><span class=\"ez-toc-section\" id=\"209_What_is_the_debug_isdn_q921_command_used_for\"><\/span><span class=\"\">209. What is the\u00a0<\/span><code>debug isdn q921<\/code><span class=\"\">\u00a0command used for?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">Displays data link layer (Layer 2) access procedures on the D-channel. Shows SABME, RR (Receiver Ready), RNR (Receiver Not Ready) messages. Essential for troubleshooting PRI\/BRI bring-up.<\/span><\/p>\n<h3><span class=\"ez-toc-section\" id=\"210_Final_advanced_scenario_T1_PRI_Layer_1_is_ACTIVE_Layer_2_is_MULTIPLE_FRAME_ESTABLISHED_but_calls_fail_with_%E2%80%9CUnallocated_Number%E2%80%9D_What_is_wrong\"><\/span><span class=\"\">210. Final advanced scenario: T1 PRI Layer 1 is ACTIVE, Layer 2 is MULTIPLE_FRAME_ESTABLISHED, but calls fail with &#8220;Unallocated Number&#8221;. What is wrong?<\/span><span class=\"ez-toc-section-end\"><\/span><\/h3>\n<p class=\"ds-markdown-paragraph\"><strong><span class=\"\">Answer:<\/span><\/strong><span class=\"\">\u00a0Layer 1 and 2 are fine. Layer 3 is the issue. Possible causes:<\/span><\/p>\n<ul>\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">The called number is not in the allowed range (telco side).<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">The gateway is not sending the correct number of digits.<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">The\u00a0<\/span><code>isdn incoming-called-number<\/code><span class=\"\">\u00a0or dial peer matching is stripping digits incorrectly.<\/span><\/p>\n<\/li>\n<li>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">The telco has not activated the DID range for that PRI.<\/span><br \/>\n<span class=\"\">Use\u00a0<\/span><code>debug isdn q931<\/code><span class=\"\">\u00a0to see the exact Setup message and called number being sent.<\/span><\/p>\n<\/li>\n<\/ul>\n<hr \/>\n<h2><span class=\"ez-toc-section\" id=\"Conclusion\"><\/span><span class=\"\">Conclusion<\/span><span class=\"ez-toc-section-end\"><\/span><\/h2>\n<p class=\"ds-markdown-paragraph\"><span class=\"\">These 60 questions (151\u2013210) on\u00a0<\/span><strong><span class=\"\">SIP trunk DTMF relay and CUCM mobility features<\/span><\/strong><span class=\"\">\u00a0cover SIP signaling, DTMF relay methods, MVA\/SNR configuration, call park types, and advanced ISDN troubleshooting. You now have the knowledge to configure SIP trunks, deploy mobility features, and diagnose Layer 2\/3 ISDN issues.<\/span><\/p>\n<p><strong>Also Check<\/strong> &#8211; <a href=\"https:\/\/techgyan360.com\/blog\/top-50-amazon-connect-interview-questions-and-answers-for-2026\/\">Top 50 Amazon Connect Interview Questions and Answers for 2026<\/a><\/p>\n","protected":false},"excerpt":{"rendered":"<p>Welcome to\u00a0Article 4\u00a0of our Cisco IPT interview series. We have covered clustering, gatekeepers, ISDN, media resources, and SRST. Now, we tackle the protocols and features that dominate modern VoIP deployments&hellip;<\/p>\n","protected":false},"author":1,"featured_media":133,"comment_status":"closed","ping_status":"closed","sticky":false,"template":"","format":"standard","meta":{"footnotes":""},"categories":[20,14],"tags":[31,21],"class_list":["post-131","post","type-post","status-publish","format-standard","has-post-thumbnail","hentry","category-cisco-contact-centre-hub","category-interview-questions-and-answers","tag-cisco-ipt-contact-center","tag-cisco-ipt-interview-questions-and-answers"],"yoast_head":"<!-- This site is optimized with the Yoast SEO plugin v27.7 - https:\/\/yoast.com\/product\/yoast-seo-wordpress\/ -->\n<title>Cisco IPT Advanced: SIP Trunks, DTMF Relay, MVA, and Call Park \u2013 50 Expert Q&amp;As - TechGyan360.Com<\/title>\n<meta name=\"description\" content=\"Master SIP trunk DTMF relay and CUCM mobility features for your CCIE Voice or IPT interview. 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