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- About filiberto.aguirre
01-18-2019
filiberto.aguir
Member since
06-05-2006
Explorer
Created by
filiberto.aguir
in Collaboration, Voice and Video Documents
01-11-2019
11:52 AM
01-11-2019
11:52 AM
Hi
Muthurani , we are running CUCM 11.5 we can not see the same options as you are showing on RTMT
, No callmanager option and no SIP tra.
ce option are shown. The option available are Voice/Video .
I need to get the ladder diagram for sip calls. What could it be the reason?
... View more
Created by
filiberto.aguir
in IP Telephony and Phones
01-11-2019
09:43 AM
01-11-2019
09:43 AM
End user has CUCM 11.5 and ISR 4321, currently for PSTN calls there is E1 with digital Trunk connected to ISR4321. In short time end user will disconnect digital trunk and use SIP trunk, so now we are carrying out test with a TEST sip trunk connected to the same ISR. We configured the TEST SIP trunk on ISR , and config on CUCM the SIP trunk to ISR for outgoing calls and works fine, calls are completed without problems. But for incoming calls we are receiving error code 503 Service Unavailable after checking some discussions we found that CUBE option must be enable and configure some dial-peers, we tried a similar config but we still received error 503. We enable CUBE on ISR and follow this guide.
https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/117300-configure-cube-00.html
There must be some additional config on ISR and CUCM so that incoming calls on SIP Trunk can be completed when
on the same ISR there is working a H323 trunk to CUCM?
Regards
... View more
Labels:
Created by
filiberto.aguir
in IP Telephony and Phones
11-15-2018
12:16 PM
11-15-2018
12:16 PM
Hi Hilario,
I attach the traces for ext 1548 in CUCM , and a debug in CME for remote site for ext 5122.
The behaivor is... when the user in remote site (CME sip phone extension 5122) dial to main office (CUCM ext 1548)
the user (remote site) listen ring back tone but when the user in main office off-hook a busy tone is listenin remote site.
regards
... View more
Created by
filiberto.aguir
in IP Telephony and Phones
11-15-2018
09:42 AM
11-15-2018
09:42 AM
Hi,
we have the following scenario:
- Remote site
CME 10.5 SIP phones / SCCP phones
- Main Office
CUCM 11.5 SIP phones / SCCP phones
- Remote office connect to Main office by VPN
- In the Main office there was a CME but was replace by the current CUCM 11.5 last weekend.
- With CME in main office there was no problem, but after the installation of CUCM this issue was detected.
The calls from remote site with SCCP phones to Main Office work fine and from Main office to remote side also.
The calls from remote site with SIP phones to Main Office cannot be completed but calls from main office to remote site with SIP phones work well.
Currently this is the dial-peer set in remote office (CME).With this we can dial from remote site to main office just with SCCP phones.
dial-peer voice 198 voip description *** to Main office **** destination-pattern 15.. session target ipv4:X.X.X.X (CUCM) voice-class codec 1 dtmf-relay h245-alphanumeric
We also tried with SIP option but for this case did not work for SCCP and SIP phones ! dial-peer voice 1181 voip description *** to Main office **** destination-pattern 10.. session protocol sipv2 session target ipv4:X.X.X.X (cucm) voice-class codec 1 voice-class sip dtmf-relay force rtp-nte voice-class sip pass-thru content sdp dtmf-relay rtp-nte no vad
any suggestion, recommendation ?
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Labels:
Created by
filiberto.aguir
in Contact Center
07-06-2018
02:01 PM
07-06-2018
02:01 PM
Hi Anju,
Could you clear the issue for the certificate?
I have a similar issue with SM 11.6 , but in our case we set the chat plugin on wordpress web site.
regards
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Created by
filiberto.aguir
in IP Telephony and Phones
05-13-2018
03:28 PM
05-13-2018
03:28 PM
The problem was cleared after disabling SIP inspection in both FW (fortinet) that handle the VPN between main site and branch.
regards
... View more
Created by
filiberto.aguir
in IP Telephony and Phones
05-07-2018
11:44 AM
05-07-2018
11:44 AM
Hi Dennis,
I attach the some screenshoot for failed a success call and the sh run of the GW.
we have tried with codec on dial-peer an region . the trace between success an failed call are very similar. The difference is:
- incoming call from PSTN on Main Office --> AA (unity main office) --->dialed by menu of AA ip ext of branch --> call Ok. The trace shows calling number and called number.
- - incoming call from PSTN on branch Office --> AA (unity main office) --->dialed by menu of AA ip ext of branch --> call FAILED. Trace shows just calling number.
any suggestion?
... View more
Created by
filiberto.aguir
in IP Telephony and Phones
05-06-2018
02:15 PM
05-06-2018
02:15 PM
Hi Dennis,
The issue was solved, but now we detected the following issue.
When the incoming call from PSTN (sip trunk) is received in branch office then redirected to AA (unity connection on main office) an then via the menu of AA the ip extension of branch office is dialed
when the called party answer nobody can hear each other.
For the moment we have tried changing codec on regions but not succes up to now.
... View more
Created by
filiberto.aguir
in IP Telephony and Phones
05-05-2018
04:58 PM
05-05-2018
04:58 PM
Hi,
End user has CUCM 11.5, Unity connection 11.5, and voice gateway 2921; all in main office. There is a branch office with voice GW 4351 config as SRST that communicate with main office via VPN. All the ip phones are 78xx. In both voice GW there is a SIP trunk with PSTN.
- calls between IP extension main office and branch office complete well
- calls from PSTN (sip trunk) to main office complete well.
- calls from PSTN (sip trunk) to branch office complete well.
- calls from PSTN (sip trunk) to IP extension on main office via AA complete well.
The issue:
when an incoming call from PSTN on main office and goes to menu of AA (unity connection) and dials an extension number that correspond to branch office the call is dropped.
We have tried with differentes codecs but not success
Any suggestion?
regards
... View more
Created by
filiberto.aguir
in Switching
04-23-2018
09:54 AM
04-23-2018
09:54 AM
Hi CodeBarbarian,
Thanks for your explanation.
I have just tried botton reset but no success to recovered PoE on switch.
when this switch is connected to IX5200, the first time setup guide state " the switch download the config.." , this mean that a special config is downloading to switch from IX5200?.
I tried to paired a touch panel but no success.
regards
... View more
Created by
filiberto.aguir
in Switching
04-22-2018
07:07 PM
04-22-2018
07:07 PM
Hi,
I would like to know if there is a difference when we reset to factory default a switch WS-C2960C-8PC-L with Reset button instead of Mode button when is used for IX5200.
Troubleshooting guide for IX5200 just shows press Mode button as an option.
I tried to reset to factory default the switch with mode button but no success,I want to try Reset button as the second option.
regards
... View more
05-21-2017
07:42 PM
Hi Mario,
I have the same problem with 2960X running version 152-2.E5 and 152-2.E6 (recommended version) when we connect a meraki AP MR53-HW (Power over Ethernet: 37 - 57 V - 802.3at required).
Tomorrow I will try with the command power inline port 2x-mode .
regards
... View more
04-05-2017
02:44 PM
Hi Manish,
Thanks for your recomendation.
Both options Bind interface and codec/region are config.
any other suggestion?
UDP for SIP shouldn´t be open?
CME#sh control-plane host open-ports Active internet connections (servers and established) Prot Local Address Foreign Address Service State tcp *:2000 10.10.20.42:49167 Cisco Call manager serve ESTABLIS tcp *:2000 10.10.20.39:49159 Cisco Call manager serve ESTABLIS tcp *:22 *:0 SSH-Server LISTEN tcp *:23 *:0 Telnet LISTEN tcp *:2000 10.10.20.32:49160 Cisco Call manager serve ESTABLIS tcp *:2000 10.10.20.24:35033 Cisco Call manager serve ESTABLIS tcp *:2000 10.10.20.36:49193 Cisco Call manager serve ESTABLIS tcp *:2000 10.10.20.36:49192 Cisco Call manager serve ESTABLIS tcp *:2000 10.10.20.30:49172 Cisco Call manager serve ESTABLIS tcp *:80 *:0 HTTP CORE LISTEN tcp *:80 *:0 HTTP CORE LISTEN tcp *:2000 10.10.20.1:48156 Cisco Call manager serve ESTABLIS tcp *:48156 10.10.20.1:2000 SCCP Application ESTABLIS tcp *:23 50.50.50.2:3933 Telnet ESTABLIS tcp *:443 *:0 HTTP CORE LISTEN tcp *:443 *:0 HTTP CORE LISTEN tcp *:1720 *:0 H.225 LISTEN tcp *:2000 10.10.20.1:63332 Cisco Call manager serve ESTABLIS tcp *:63332 10.10.20.1:2000 SCCP Application ESTABLIS tcp *:2000 *:0 Cisco Call manager serve LISTEN tcp *:5060 *:0 SIP LISTEN tcp *:2000 10.10.20.55:49152 Cisco Call manager serve ESTABLIS tcp *:2000 10.10.20.38:49173 Cisco Call manager serve ESTABLIS udp *:67 *:0 DHCPD Receive LISTEN udp *:69 *:0 TFTP-Server LISTEN udp *:123 *:0 NTP LISTEN
CME#
regards
... View more
Created by
filiberto.aguir
in IP Telephony and Phones
04-04-2017
05:53 PM
04-04-2017
05:53 PM
Hi,
End user has a Branch office with CME 10.5 with ip phones 7821 ,8945 (SIP) , 7911 (SCCP) , HQ CUCM 8.6 with 8945 (SIP), 7911 (SCCP)
Connection between them is via VPN - SITE to SITE
For dialing between offices we configured the next
Branch Office
dial-peer voice 1400 voip description SIP TRUNK to HQ destination-pattern 1[4-5].. session protocol sipv2 session target ipv4:192.168.1.93 voice-class sip dtmf-relay force rtp-nte dtmf-relay rtp-nte codec g711ulaw no vad
HQ SIP TRUNK to CME
Result of test
SCCP phone Branch ---> SCCP Phone HQ Completed SCCP Phone HQ ---> SCCP phone Branch Completed
SIP phone Branch ----> SCCP Phone HQ dropped after 20 sec SCCP Phone HQ ------> SIP phone Branch Busy tone
Any suggestion?
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Labels:
Created by
Evan Ruiz
in IP Telephony and Phones
06-28-2017 08:43 AM
06-28-2017 08:43 AM
i am having issues random call recording failures to a trunk we have to
zoom call rec. Attached are ...
Created by
Evan Ruiz
in IP Telephony and Phones
02-21-2017 01:23 PM
02-21-2017 01:23 PM
How can we filter or remove the prefix of +9 when transferring a call?
Currently when we transter a ...
Created by
Evan Ruiz
in Physical Security
02-17-2017 07:29 AM
02-17-2017 07:29 AM
I have a client who is experiencing issues accessing his VSOM from web
browser. I can access the ser...
Created by
Evan Ruiz
in IP Telephony and Phones
02-15-2017 08:26 AM
02-15-2017 08:26 AM
i have a customer who has sent this request to me. // // Details,
Edit, New if (serviceTicketEdit....
Created by
Evan Ruiz
in Contact Center
08-25-2014 12:27 PM
08-25-2014 12:27 PM
We are trying to publish RDweb site through our ASA VPN portal. I get
the errors i have attached whe...
04-22-2014 03:13 PM
Sam now we are having issues with the Cisco voicemail.aef and
voicemailqueueing.aef not delivering v...
04-17-2014 07:31 AM
We created a UCCX script designed to allow callers to hit option 2
record a message instead of waiti...
11-14-2013 07:10 AM
If I give a user in UCCX reporting capability will they only have access
to view agent statistics fo...
Created by
Evan Ruiz
in IP Telephony and Phones
10-30-2013 12:44 PM
10-30-2013 12:44 PM
it uses a wlan with auto(AKM) and a WPA pre-shared key.
Created by
Evan Ruiz
in IP Telephony and Phones
10-30-2013 12:25 PM
10-30-2013 12:25 PM
yes that is correct we utilize a profile to setup wireless when we
program the device via USB.
Created by
Evan Ruiz
in IP Telephony and Phones
10-22-2013 08:27 AM
10-22-2013 08:27 AM
I have a 7925 that flash authorization required. It doesn't have any
UCCX services subscribed to it ...
09-17-2013 10:44 AM
Thanks Chris after looking closer all new application weren't
registering in call manager correctly....
09-13-2013 02:03 PM
I am currently trying to start a new queue in UCCX 8.5. I have installed
a new application along wit...
Created by
Evan Ruiz
in Unified Communications Infrastructure
04-02-2013 03:25 PM
04-02-2013 03:25 PM
Is it possible to have some agents be standard licensed and other agents
enhanced? If i upload an en...
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| Date Registered | 06-05-2006 02:25 PM |
| Date Last Visited | 01-18-2019 04:40 PM |
| Total Messages Posted | 659 |
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