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- About Mohammed Alkiswani
07-26-2019
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04-16-2012
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Re: MRA and split domains
in Other Collaboration Subjects
12-18-2017
10:15 AM
12-18-2017
10:15 AM
For the SRV Records, you create it because Jabber client will take the domain part from JID (User1@external.com) which used in the discovery page on Jabber client, and use it to find CUCM using srv lookup. So in the above example Jabber client will take (external.com), and will do DNS lookup for the srv records.
So If you want to use (external.com) as JID in your deployment you have to create this zone in internal DNS, but you can use another domain for JID in your deployment.
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Re: Adding New DIDs to gateway that is ...
in IP Telephony and Phones
12-18-2017
09:36 AM
12-18-2017
09:36 AM
MGCP VGW is configured from CUCM, so on CUCM go to (Device --> Gateway), and search for this Gateway, you will find it with its hostname, in this VGW configuration, set the "Significant digits" to 4 for both T1s (T1 0/1/0, and T1 0/1/1), so now you can handle this DID on CUCM as 4 digit, for example if your DID was (064679857), you have to handle it as (9857), you can give this (9857) directly to IP Phone as DN, or you can create a translation pattern to translate this (9857) to any DN you want.
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Re: MRA and split domains
in Other Collaboration Subjects
12-18-2017
09:18 AM
12-18-2017
09:18 AM
You need to create (external.com) on the internal DNS server, to create srv record, and also you will add a record for Expressway E on this domain, because on the expressway-C zone configuration, you will point it public FQDN of Expressway E (ExpE.external.com).
And as mentioned above, you will create srv records (_cisco-uds._tcp.external.com, _cuplogin._tcp.external.com), on the internal DNS server, so you have to create this zone on the internal DNS server.
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12-15-2016
03:20 AM
I want to ask if CUCM support custom API? in other word, I want to develop a web service and upload/deploy it in CUCM.
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Labels:
01-19-2015
02:31 AM
Hi All, We want to setup SPAN, and its source is a physical port on the catalyst switch and the destination is a VM on a Hyper-V host, the hyper-V host has 4 NICs and we can not Dispense about any NIC. And I want to ask how can we do this without a dedicated NIC for SPAN traffic? Can we pass the SPAN traffic with the Host traffic or we have to dedicate a NIC for SPAN traffic?
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- Tags:
- span
Labels:
Created by
Mohammed Alkiswani
in IP Telephony and Phones
in IP Telephony and Phones
10-16-2014
06:37 AM
10-16-2014
06:37 AM
post the output from "debug tftp event/ debug tftp packet", when you reset the phone, and lets see what happened, trust me the above command is very helpful when troublshooting CME phone registration.
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Created by
Mohammed Alkiswani
in IP Telephony and Phones
in IP Telephony and Phones
10-16-2014
06:28 AM
10-16-2014
06:28 AM
Check that you configure the correct voice port module on CUCM, "show inventory/ show diag" will help you. What is the voice port that you try to register. Provide The Whole Show Runn. and The Below Show Commands: - show ccm-manager - show mgcp - show mgcp endpoint - show mgcp connection - show mgcp statistics Shutdown and Enable the Voice Ports.
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Created by
Mohammed Alkiswani
in IP Telephony and Phones
in IP Telephony and Phones
01-28-2014
10:42 PM
01-28-2014
10:42 PM
My problem not on the FXO, my problem on the CUCM and IP Phones, that the phone still connected after the FXO return to On-Hook for 30 Seconds. for example if i shutdown the FXO will it in active call the IP Phone will still in the connected state for 30 seconds, and the user hear a tone at this period.
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Created by
Mohammed Alkiswani
in IP Telephony and Phones
in IP Telephony and Phones
01-28-2014
11:37 AM
01-28-2014
11:37 AM
Hi All, we have Cisco 2801 VGW with 2 analog lines connected to it (FXO Card), and it is confiugred as H323 to CUCM 8.5, and we have aproblem that when the PSTN caller hangup, the FXO go On-Hook, but the phone still in connected state for 30 seconds, and this case happened also when i shudown the FXO port while it on active call.
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Labels:
Created by
Mohammed Alkiswani
in IP Telephony and Phones
in IP Telephony and Phones
01-28-2014
11:25 AM
01-28-2014
11:25 AM
anyone can help?
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Created by
Mohammed Alkiswani
in IP Telephony and Phones
in IP Telephony and Phones
01-25-2014
04:37 AM
01-25-2014
04:37 AM
we have 2801 router that connected with 2 anloga lines (FXO Card), but now we have a problem with disconnect problem, the phone still connected after the PSTN Caller Disconnect, and our policy tell that the agents shouldn't end the call, so we want when the PSTN caller end the call we need the phone to return to idle state. and according for the below link, i configure the custom disconnect tone, and in the attachment you can find tow disconnect tones. http://ciscoflair.blogspot.com/2009/05/cisco-fxo-disconnect-issue.html the configuration of the custom disconnect tone voice class dualtone-detect-params 1 cadence-variation 3 voice class custom-cptone Custom1 dualtone disconnect frequency 420 cadence 245 255 245 255 245 255 and the below you can find the configuration of the voice port, and you can find the debug vpm port 0/3/0, debug vpm signal, for a call that disconnected immediately, and for a call the take long time to disconnected. this is the configuration of the voice port: voice-port 0/3/0 supervisory disconnect dualtone mid-call supervisory custom-cptone Custom1 supervisory dualtone-detect-params 1 no battery-reversal cptone NL timeouts call-disconnect 5 timeouts wait-release 5 timing hookflash-out 50 timing guard-out 300 caller-id enable caller-id alerting line-reversal caller-id alerting dsp-pre-allocate this is the debug of a call that is disconnected immediately: Jan 25 11:48:32.262: [0/3/0] htsp_dsm_feature_notify_cb returns 2 id=DSM_FEATURE_SM_CALLERID_RX Jan 25 11:48:32.262: htsp_process_event: [0/3/0, FXOLS_ONHOOK, E_HTSP_CALLERID_RX_DONE] Jan 25 11:48:32.262: [0/3/0] htsp_stop_caller_id_rx. message length 11 Jan 25 11:48:32.262: [0/3/0] htsp_dsm_close_done Jan 25 11:48:34.026: htsp_process_event: [0/3/0, FXOLS_ONHOOK, E_DSP_SIG_0000]fxols_onhook_ringing Jan 25 11:48:34.026: htsp_timer - 125 msec Jan 25 11:48:34.154: htsp_process_event: [0/3/0, FXOLS_WAIT_RING_MIN, E_HTSP_EVENT_TIMER]fxols_wait_ring_min_timer Jan 25 11:48:34.154: htsp_timer - 10000 msec Jan 25 11:48:35.305: htsp_process_event: [0/3/0, FXOLS_RINGING, E_DSP_SIG_0100] Jan 25 11:48:35.305: fxols_ringing_not Jan 25 11:48:35.305: htsp_timer_stop Jan 25 11:48:35.305: htsp_timer_stop3 htsp_setup_ind Jan 25 11:48:35.305: [0/3/0] get_fxo_caller_id:Caller ID received. Message type=129 length=11 checksum=00 Jan 25 11:48:35.309: [0/3/0] Caller ID String 44 30 36 35 36 37 39 31 34 31 Jan 25 11:48:35.309: [0/3/0] get_fxo_caller_id calling num=065679141 calling name= calling time=01/25 13:48 Jan 25 11:48:35.313: htsp_process_event: [0/3/0, FXOLS_WAIT_SETUP_ACK, E_HTSP_SETUP_ACK] Jan 25 11:48:35.313: fxols_wait_setup_ack: Jan 25 11:48:35.313: [0/3/0] set signal state = 0xC timestamp = 0fxols_check_auto_call Jan 25 11:48:35.321: htsp_process_event: [0/3/0, FXOLS_PROCEEDING, E_HTSP_PROCEEDING]fxols_offhook_proc Jan 25 11:48:35.321: htsp_timer - 120000 msechtsp_alert_notify Jan 25 11:48:35.389: htsp_process_event: [0/3/0, FXOLS_PROCEEDING, E_HTSP_ALERT]fxols_offhook_alerthtsp_call_service_msghtsp_call_service_msg not EFXS (2) Jan 25 11:48:35.509: htsp_call_bridged invokedhtsp_call_service_msghtsp_call_service_msg not EFXS (2) Jan 25 11:48:35.513: htsp_process_event: [0/3/0, FXOLS_PROCEEDING, E_HTSP_CONNECT]fxols_offhook_connect Jan 25 11:48:35.513: htsp_timer_stop Jan 25 11:48:35.521: htsp_process_event: [0/3/0, FXOLS_CONNECT, E_HTSP_VOICE_CUT_THROUGH]fxols_connect_proc_voice Jan 25 11:48:35.585: htsp_process_event: [0/3/0, FXOLS_CONNECT, E_DSP_SIG_0110]fxols_rvs_battery Jan 25 11:48:35.585: htsp_timer_stop2 sh voice po su IN OUT PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC =============== == ============ ===== ==== ======== ======== == 0/3/0 -- fxo-ls up up idle off-hook y 0/3/1 -- fxo-ls up dorm idle on-hook y 50/0/1 1 efxs up up on-hook idle y 50/0/1 2 efxs up up on-hook idle y PWR FAILOVER PORT PSTN FAILOVER PORT ================= ================== Jan 25 11:48:43.857: htsp_process_event: [0/3/0, FXOLS_CONNECT, E_DSP_SUP_DISCONNECT]fxols_conn_sup_disc Jan 25 11:48:43.857: htsp_timer2 - 5000 msec Madaba_Maint# Jan 25 11:48:48.856: htsp_process_event: [0/3/0, FXOLS_CONNECT, E_HTSP_EVENT_TIMER2]fxols_disc_confirm Jan 25 11:48:48.856: htsp_timer_stop Jan 25 11:48:48.856: htsp_timer_stop2 Jan 25 11:48:48.856: htsp_timer_stop3 Jan 25 11:48:48.860: htsp_timer_stop3 Jan 25 11:48:48.876: htsp_process_event: [0/3/0, FXOLS_REMOTE_RELEASE, E_HTSP_RELEASE_REQ]fxols_offhook_release Jan 25 11:48:48.876: htsp_timer_stop Jan 25 11:48:48.876: htsp_timer_stop2 Jan 25 11:48:48.876: htsp_timer_stop3 Jan 25 11:48:48.876: [0/3/0] set signal state = 0x4 timestamp = 0 Jan 25 11:48:48.876: htsp_timer - 300 msec Jan 25 11:48:49.148: htsp_process_event: [0/3/0, FXOLS_GUARD_OUT, E_DSP_SIG_0110] Jan 25 11:48:49.176: htsp_process_event: [0/3/0, FXOLS_GUARD_OUT, E_HTSP_EVENT_TIMER]fxols_guard_out_timeout Jan 25 11:48:49.176: fxols_dsp_prealloc_clid_wait. Line reversal alerting DSP preallocation done Jan 25 11:48:49.176: [0/3/0] htsp_start_caller_id_rx:ETSI_DTMF Jan 25 11:48:49.176: htsp_start_caller_id_rx create dsp_stream_manager Jan 25 11:48:49.176: htsp_process_event: [0/3/0, FXOLS_ONHOOK, E_DSP_SIG_0100] Jan 25 11:48:49.176: [0/3/0] htsp_dsm_create_success returns 1 Madaba_Maint#sh voice po su IN OUT PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC =============== == ============ ===== ==== ======== ======== == 0/3/0 -- fxo-ls up dorm idle on-hook y 0/3/1 -- fxo-ls up dorm idle on-hook y 50/0/1 1 efxs up up on-hook idle y 50/0/1 2 efxs up up on-hook idle y this is a debug for a call that is take a long time to disconnected: Jan 25 11:49:42.267: [0/3/0] htsp_dsm_feature_notify_cb returns 2 id=DSM_FEATURE_SM_CALLERID_RX Jan 25 11:49:42.267: htsp_process_event: [0/3/0, FXOLS_ONHOOK, E_HTSP_CALLERID_RX_DONE] Jan 25 11:49:42.267: [0/3/0] htsp_stop_caller_id_rx. message length 11 Jan 25 11:49:42.271: [0/3/0] htsp_dsm_close_done Jan 25 11:49:43.999: htsp_process_event: [0/3/0, FXOLS_ONHOOK, E_DSP_SIG_0000]fxols_onhook_ringing Jan 25 11:49:43.999: htsp_timer - 125 msec Jan 25 11:49:44.127: htsp_process_event: [0/3/0, FXOLS_WAIT_RING_MIN, E_HTSP_EVENT_TIMER]fxols_wait_ring_min_timer Jan 25 11:49:44.127: htsp_timer - 10000 msec Jan 25 11:49:45.279: htsp_process_event: [0/3/0, FXOLS_RINGING, E_DSP_SIG_0100] Jan 25 11:49:45.279: fxols_ringing_not Jan 25 11:49:45.279: htsp_timer_stop Jan 25 11:49:45.279: htsp_timer_stop3 htsp_setup_ind Jan 25 11:49:45.279: [0/3/0] get_fxo_caller_id:Caller ID received. Message type=129 length=11 checksum=00 Jan 25 11:49:45.279: [0/3/0] Caller ID String 44 30 36 35 36 37 39 31 34 31 Jan 25 11:49:45.279: [0/3/0] get_fxo_caller_id calling num=065679141 calling name= calling time=01/25 13:49 Jan 25 11:49:45.283: htsp_process_event: [0/3/0, FXOLS_WAIT_SETUP_ACK, E_HTSP_SETUP_ACK] Jan 25 11:49:45.283: fxols_wait_setup_ack: Jan 25 11:49:45.287: [0/3/0] set signal state = 0xC timestamp = 0fxols_check_auto_call Jan 25 11:49:45.291: htsp_process_event: [0/3/0, FXOLS_PROCEEDING, E_HTSP_PROCEEDING]fxols_offhook_proc Jan 25 11:49:45.291: htsp_timer - 120000 msechtsp_alert_notify Jan 25 11:49:45.379: htsp_process_event: [0/3/0, FXOLS_PROCEEDING, E_HTSP_ALERT]fxols_offhook_alerthtsp_call_service_msghtsp_call_service_msg not EFXS (2) Jan 25 11:49:45.495: htsp_call_bridged invokedhtsp_call_service_msghtsp_call_service_msg not EFXS (2) Jan 25 11:49:45.503: htsp_process_event: [0/3/0, FXOLS_PROCEEDING, E_HTSP_CONNECT]fxols_offhook_connect Jan 25 11:49:45.503: htsp_timer_stop Jan 25 11:49:45.507: htsp_process_event: [0/3/0, FXOLS_CONNECT, E_HTSP_VOICE_CUT_THROUGH]fxols_connect_proc_voice Jan 25 11:49:45.559: htsp_process_event: [0/3/0, FXOLS_CONNECT, E_DSP_SIG_0110]fxols_rvs_battery Jan 25 11:49:45.559: htsp_timer_stop2 sh voice po su IN OUT PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC =============== == ============ ===== ==== ======== ======== == 0/3/0 -- fxo-ls up up idle off-hook y 0/3/1 -- fxo-ls up dorm idle on-hook y 50/0/1 1 efxs up up on-hook idle y 50/0/1 2 efxs up up on-hook idle y Jan 25 11:50:30.387: htsp_timer_stop3 Jan 25 11:50:30.399: htsp_process_event: [0/3/0, FXOLS_CONNECT, E_HTSP_RELEASE_REQ]fxols_offhook_release Jan 25 11:50:30.399: htsp_timer_stop Jan 25 11:50:30.399: htsp_timer_stop2 Jan 25 11:50:30.399: htsp_timer_stop3 Jan 25 11:50:30.399: [0/3/0] set signal state = 0x4 timestamp = 0 Jan 25 11:50:30.399: htsp_timer - 300 msec Jan 25 11:50:30.671: htsp_process_event: [0/3/0, FXOLS_GUARD_OUT, E_DSP_SIG_0110] Jan 25 11:50:30.699: htsp_process_event: [0/3/0, FXOLS_GUARD_OUT, E_HTSP_EVENT_TIMER]fxols_guard_out_timeout Jan 25 11:50:30.699: fxols_dsp_prealloc_clid_wait. Line reversal alerting DSP preallocation done Jan 25 11:50:30.699: [0/3/0] htsp_start_caller_id_rx:ETSI_DTMF Jan 25 11:50:30.699: htsp_start_caller_id_rx create dsp_stream_manager Jan 25 11:50:30.699: htsp_process_event: [0/3/0, FXOLS_ONHOOK, E_DSP_SIG_0100] Jan 25 11:50:30.699: [0/3/0] htsp_dsm_create_success returns 1 So what can we do? what is the wronge of my configuration?
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Labels:
01-23-2014
07:07 AM
The problem was on the Agent PC that was block the traffic from the VGW to the CIPC.
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Created by
Mohammed Alkiswani
in IP Telephony and Phones
in IP Telephony and Phones
01-22-2014
12:12 AM
01-22-2014
12:12 AM
yes, and we configure the core-1 as the root bridge, and you can see the sh spanning-tree & sh runn in the attachment, but there is a strange issue on the switches, the issue is the switches see non physical neighbor in the show cdp neighbor you can see that on Agents_sw_1.txt, and i am sure that there no hub between them, and thank you.
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01-21-2014
11:45 PM
This is our network topology, and we are not configured the QoS because all calls on the lan, and the calls dropped in time that there few number of calls.
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01-21-2014
11:08 PM
So the Problem is in the network, and there is no prolem in the configuration. and if the problem in the network, why the problem happened in some calls, and how to know where is the problem, and kindly note that there in no firewall between the CUCM, VGW, and Agents.
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12-18-2017
For the SRV Records, you create it because Jabber client will take the
domain part from JID (User1@e...
Created by
Mohammed Alkiswani
in IP Telephony and Phones
12-18-2017
12-18-2017
MGCP VGW is configured from CUCM, so on CUCM go to (Device --> Gateway),
and search for this Gateway...
12-18-2017
You need to create (external.com) on the internal DNS server, to create
srv record, and also you wil...
11-07-2017
you have to use postman, you have to use the put method in postman, and
in the attachment you can fi...
12-15-2016
I want to ask if CUCM support custom API? in other word, I want to
develop a web service and upload/...
01-19-2015
Hi All,We want to setup SPAN, and its source is a physical port on the
catalyst switch and the desti...
10-16-2014
post the output from "debug tftp event/ debug tftp packet", when you
reset the phone, and lets see w...
10-16-2014
Check that you configure the correct voice port module on CUCM, "show
inventory/ show diag" will hel...
Created by
Mohammed Alkiswani
in IP Telephony and Phones
01-28-2014
01-28-2014
My problem not on the FXO, my problem on the CUCM and IP Phones, that
the phone still connected afte...
Created by
Mohammed Alkiswani
in IP Telephony and Phones
01-28-2014
01-28-2014
Hi All,we have Cisco 2801 VGW with 2 analog lines connected to it (FXO
Card), and it is confiugred a...
01-25-2014
we have 2801 router that connected with 2 anloga lines (FXO Card), but
now we have a problem with di...
01-22-2014
yes, and we configure the core-1 as the root bridge, and you can see the
sh spanning-tree & sh runn ...
Public Statistics
| Date Registered | 04-16-2012 04:37 AM |
| Date Last Visited | 07-26-2019 12:12 AM |
| Total Messages Posted | 39 |
