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- About asilveira
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Member since
01-19-2006
15
Posts
1
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0
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Created by
asilveira
in IP Telephony and Phones
in IP Telephony and Phones
02-05-2013
08:57 AM
02-05-2013
08:57 AM
Hi everyone! Does anyone have either information or experience about SNR - Single Number Reach (Remote Destination Profile) not seeing routes learned from SAF/CCD? I have an environment with four clusters (CUCM 8.6) that exchange routes between them through SAF/CCD. In one of them there are RDP associated with extensions. The Remote Destination numbers are supposed to use the routes learned from SAF/CCD. When I call directly to the number configured on the Remote Destination, the call proceeds normally. When I call an extension associated with a RDP, the call does not proceeds. The basic verification were already done: the CSS and Rerouting CSS of the RDP has the partition of the routes learned from SAF/CCD. My question is: Is SNR supposed to see and use the routes learned via SAF/CCD or not? Regards! André Silveira
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02-05-2013
06:14 AM
Hi Erik, Below are the versions of the systems: CUCM 8.6 (System version: 8.6.2.21900-5) Cisco Unity Connection 8.0 (Version 8.0.1.10000-272). Single machine (not cluster). One more detail: Cisco Unity Connection is integrated with two systems. The first one is a CUCM 8.0 and the integration was done through SCCP. This integration is working fine and is defined as the default TRAP phone system for Unity Connection. The second one (I have made it two days ago) is with a CUCM 8.6 through SIP. Everything worked correctly except for the MWI. This is the problem we are discussing here. Regards, André
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02-04-2013
11:28 AM
Hi Erik, I have followed all instructions on the "Setting Up a Cisco Unified Communications Manager 8.x SIP Trunk Integration with Cisco Unity Connection" manual below. All three fields were checked and the SIP Trunk Security Profile applied on the SIP Trunk configuration. http://www.cisco.com/en/US/docs/voice_ip_comm/connection/8x/integration/guide/cucm_sip/cucintcucmsip060.html The problem seems to be on the Unity side. Analyzing the trace files of CUCM, there is no SIP message arriving from Unity Connection. Regards, André Silveira
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02-04-2013
09:08 AM
Hi Islam The voice-mail is there in the voice-mail box (Unity Connection). I can hear it when I call Unity through IP Phone. Regards, André Silveira
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02-04-2013
08:57 AM
Hi Islam, This test would be good if the integration were done through SCCP. With SIP integration there is no MWI On or Off extensions to call. Regards, André
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02-04-2013
08:49 AM
Hello folks, I have integrated CUCM 8.6 with Cisco Unity Connection 8.0 through SIP . Everything worked fine excetp for the MWI. They do not turn either on or off. The message in the trace is the same above: 12:43:14.970 |29542,,,-1,-1,Failed to set message waiting lamp for Promon Logicalis, ext 7069999, switch CORP-01. TECHNICAL DETAILS: SetMWI failed with [0x80004005; E_FAIL]. Does anyone know where could be the problem? Regards, André Silveira
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Re: S_R2_COLLECT and R2_Q421_IDLE on se...
in Other Collaboration Subjects
11-22-2010
10:42 AM
11-22-2010
10:42 AM
Hi Eduardo, Could you tell me how you solved this problem? Currently I have a very similar problem with the E1 circuit on the gateway. Thanks, André
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Created by
asilveira
in Other Collaboration Subjects
in Other Collaboration Subjects
11-22-2010
10:38 AM
11-22-2010
10:38 AM
Hi Felipe, Following is the partial show version output. I have checked the show log output, but there was no logs related to this problem. Thanks! Cisco IOS Software, 2800 Software (C2800NM-ADVENTERPRISEK9-M), Version 12.4(15)T8, RELEASE SOFTWARE (fc3) System image file is "flash:c2800nm-adventerprisek9-mz.124-15.T8.bin" Cisco 2811 (revision 53.51) with 243712K/18432K bytes of memory. Processor board ID FTX1024A3NT 3 FastEthernet interfaces 3 Serial(sync/async) interfaces 1 Channelized E1/PRI port 1 Virtual Private Network (VPN) Module 2 Voice FXO interfaces DRAM configuration is 64 bits wide with parity enabled. 239K bytes of non-volatile configuration memory. 62720K bytes of ATA CompactFlash (Read/Write)
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Created by
asilveira
in Other Collaboration Subjects
in Other Collaboration Subjects
11-19-2010
06:34 AM
11-19-2010
06:34 AM
Hi all, I am experiencing a problem with an E1 controller installed on a Cisco 2811 gateway. Since people make calls through the E1, some of the calls do not end correctly and the timeslots used by those calls get stucked with S_R2_COLLECT R2_Q421_IDLE status. From the gateway perspective those timeslots are unused and so free for new calls, but when it try to use them for new calls they are rejected. Does anyone know how could I solve this problem, please? Following are the controller configuration and the "show voice call summary" command output. Thanks a lot! ! controller E1 1/0/0 framing NO-CRC4 line-termination 75-ohm ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled ani cas-custom 0 country brazil metering debounce-time 20 seizure-ack-time 78 double-answer category 2 answer-signal group-b 1 ! voice-port 1/0/0:0 translation-profile incoming CONVERTE-ANI-ENTRADA echo-cancel coverage 32 cptone BR timeouts interdigit 4 ! RCISPO01#sh voice call summary PORT CODEC VAD VTSP STATE VPM STATE ============== ========= === ==================== ====================== 1/0/0:0.1 - - - R2_Q421_IDLE 1/0/0:0.2 - - - R2_Q421_IDLE 1/0/0:0.3 - - - R2_Q421_IDLE 1/0/0:0.4 - - - R2_Q421_IDLE 1/0/0:0.5 g711ulaw n S_R2_COLLECT R2_Q421_IDLE 1/0/0:0.6 - - - R2_Q421_IDLE 1/0/0:0.7 - - - R2_Q421_IDLE 1/0/0:0.8 - - - R2_Q421_IDLE 1/0/0:0.9 g711ulaw n S_R2_COLLECT R2_Q421_IDLE 1/0/0:0.10 - - - R2_Q421_IDLE 1/0/0:0.11 - - - R2_Q421_IDLE 1/0/0:0.12 - - - R2_Q421_IDLE 1/0/0:0.13 - - - R2_Q421_IDLE 1/0/0:0.14 - - - R2_Q421_IDLE 1/0/0:0.15 - - - R2_Q421_IDLE 1/0/0:0.17 - - - R2_Q421_IDLE 1/0/0:0.18 - - - R2_Q421_IDLE 1/0/0:0.19 - - - R2_Q421_IDLE 1/0/0:0.20 - - - R2_Q421_IDLE 1/0/0:0.21 g711ulaw n S_R2_COLLECT R2_Q421_IDLE 1/0/0:0.22 - - - R2_Q421_IDLE 1/0/0:0.23 g711ulaw n S_R2_COLLECT R2_Q421_IDLE 1/0/0:0.24 - - - R2_Q421_IDLE 1/0/0:0.25 g711ulaw n S_R2_COLLECT R2_Q421_IDLE 1/0/0:0.26 - - - R2_Q421_IDLE 1/0/0:0.27 - - - R2_Q421_IDLE 1/0/0:0.28 - - - R2_Q421_IDLE 1/0/0:0.29 g711ulaw n S_R2_COLLECT R2_Q421_IDLE 1/0/0:0.30 - - - R2_Q421_IDLE 1/0/0:0.31 - - - R2_Q421_IDLE
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Labels:
09-02-2009
12:29 PM
In addition to the previous information, the DSP of IP Phone may be used as a Conference Bridge to mix three G.711 audio flows, only when the Barge function is used.
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Created by
asilveira
in IP Telephony and Phones
in IP Telephony and Phones
09-02-2009
12:21 PM
09-02-2009
12:21 PM
Following are some interesting information about how to do that. Good luck! http://www.voip-info.org/wiki/index.php?page=Asterisk%20phone%20cisco%2079xx http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP
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Created by
asilveira
in Unified Communications Infrastructure
07-02-2014
07-02-2014
Hi everyone,I'm trying to register a Polycom video endpoint on CUCM v
8.6. I had success using Third...
Created by
asilveira
in Collaboration Applications
04-05-2013
04-05-2013
Hello Jaime,I have a similar scenario, but with the difference that the
contacts were added from Out...
Created by
asilveira
in Unified Communications Infrastructure
04-04-2013
04-04-2013
Thank you for sharing.
Created by
asilveira
in Unified Communications Infrastructure
04-03-2013
04-03-2013
Hi Aldred,Did you solve the problem? I have exactly the same scenario
and the same problem. Could yo...
Created by
asilveira
in IP Telephony and Phones
02-05-2013
02-05-2013
Hi everyone!Does anyone have either information or experience about SNR
- Single Number Reach (Remot...
02-05-2013
Hi Erik,Below are the versions of the systems:CUCM 8.6 (System version:
8.6.2.21900-5)Cisco Unity Co...
02-04-2013
Hi Erik,I have followed all instructions on the "Setting Up a Cisco
Unified Communications Manager 8...
02-04-2013
Hi IslamThe voice-mail is there in the voice-mail box (Unity
Connection). I can hear it when I call ...
02-04-2013
Hi Islam,This test would be good if the integration were done through
SCCP. With SIP integration the...
02-04-2013
Hello folks,I have integrated CUCM 8.6 with Cisco Unity Connection 8.0
through SIP. Everything worke...
Created by
asilveira
in Other Collaboration Subjects
11-22-2010
11-22-2010
Hi Eduardo,Could you tell me how you solved this problem? Currently I
have a very similar problem wi...
Created by
asilveira
in Other Collaboration Subjects
11-22-2010
11-22-2010
Hi Felipe,Following is the partial show version output. I have checked
the show log output, but ther...
Created by
asilveira
in Other Collaboration Subjects
11-19-2010
11-19-2010
Hi all,I am experiencing a problem with an E1 controller installed on a
Cisco 2811 gateway. Since pe...
09-02-2009
In addition to the previous information, the DSP of IP Phone may be used
as a Conference Bridge to m...
Created by
asilveira
in IP Telephony and Phones
09-02-2009
09-02-2009
Following are some interesting information about how to do that. Good
luck!http://www.voip-info.org/...
Public Statistics
| Date Registered | 01-19-2006 12:39 PM |
| Date Last Visited | 08-18-2017 04:01 AM |
| Total Messages Posted | 15 |
| Total Helpful Votes Received | 1 |
