Turn on suggestions
Auto-suggest helps you quickly narrow down your search results by suggesting possible matches as you type.
Showing results for
- Cisco Community
- :
- About Michael Endler
Michael Endler
Beginner
I started with my first cisco certification 2009. My current main focus is all around voice:
Cisco (Callmanager, Unity, Presence, ContactCenter, Meetingplace, Attendant), Andtek, Caseris, C4B, Polycom
and more..
Cisco CCNP Voice
Cisco CCNP Datacenter
Recent Badges
Stats
Member since
09-26-2013
24
Posts
0
Helpful
0
Solutions
05-15-2019
03:02 AM
Hi Mike, ok i tested it by queuing a call to multiple CSQs, with diffrent priorities, that did not work. https://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-contact-center-express/63710-multi-csqs-call.html But with seperated pilot numbers to UCCX it is working as you said. Thank you and best regards Michael
... View more
05-07-2019
02:02 AM
Hi Mike, i thought calls in queue A and will be sorted by priority, same in queue B. When it is given to agent, he will see both queues, but here it will be FIFO. So you say, an agent would get both queues, but priority would be still in use, even if prioritized call is later arriving?
... View more
05-06-2019
05:07 AM
Hi Everyone, i currently have a situation, where i do not understand, how a call will be treated correctly. We have several calls and less agents, so we try to queue calls but prioritize some specific. Here the situation: An agent is assigned to several queues. I assume the call will be treated FIFO, if no priority is given. I assume the call will be prioritized per queue But what is if i prioritize a call in one queue and our agent is part of two queues? If all agents are not ready, will call3 (see below) be given by priority, even if two other calls in both queues are earlier available? For example: Call1, Queue A, 9:00 a.m., priority= low Call2, Queue B, 9:02 a.m., priority= low Call3, Queue A, 9:03 a.m., priority= high Best Regards Michael
... View more
Labels:
Created by
Michael Endler
in IP Telephony and Phones
in IP Telephony and Phones
06-15-2015
03:20 AM
06-15-2015
03:20 AM
Hello, the error message '/usr/local/sip/.security/proxy/trust-certs/.0' can only be fixed by cisco TAC. They have to go on the CLI with root access to delete a position in the trust store, which starts with .0 There is no workaround you can fullfill, like renewing certificats or resarting drf services Regards Michael
... View more
Created by
Michael Endler
in Collaboration, Voice and Video Documents
in Collaboration, Voice and Video Documents
06-10-2015
03:20 AM
06-10-2015
03:20 AM
Hello Wolfgang, you have to configure a gateway, which offers the desired fxs-ports: Callmanager | Devices | Gateway | add gateway Gateway= Gateway which is in use (show version -> for exampel Cisco 2951) Protocoll= SCCP Mac Address (Last 10 Characters)= MAC of your interface (show int gi0/0) For example: sh int gi0/0 GigabitEthernet0/0 is up, line protocol is up Hardware is CN Gigabit Ethernet, address is d072.dc97.e000 (bia d072.dc97.e000) Mac Address= 72DC97E000 Then the biggest issue you can face, is when you not save during your mudule selections (Configured Slots, VICs and Endpoints) Which port you have to configure you can see by using sh inventory For example: sh inv NAME: "3rd generation four port FXS DID voice interface daughtercard on Slot 0 SubSlot 3", DESCR: "3rd generation four port FXS DID voice interface daughtercard" PID: VIC3-4FXS/DID , VID: V03 , SN: FOC18153SNP Port = 0/3/X Module in Slot 0= NM-4VWIC-MBRD Subunit 3= VIC3-4FXS/DID-SCCP Rate if usefull! Best Regards Michael
... View more
06-22-2014
06:42 AM
Is available, i installed yesterday the app from itunes store (version= 3.0.09440)
... View more
Created by
Michael Endler
in Mobile Applications Discussions
in Mobile Applications Discussions
06-22-2014
06:39 AM
06-22-2014
06:39 AM
Is again available, i installed it yesterday.
... View more
Created by
Michael Endler
in Collaboration, Voice and Video Documents
in Collaboration, Voice and Video Documents
06-06-2014
03:02 AM
06-06-2014
03:02 AM
Great HowTo! I had an issue to understand the command "destination uri" and then i found your howto faster then a cisco guide
... View more
Created by
Michael Endler
in IP Telephony and Phones
in IP Telephony and Phones
04-30-2014
02:56 AM
04-30-2014
02:56 AM
Hello Community, i stock in configuration and need assistance. My callflow: Telco – PRI – GW – H323– CUCM – SIP – ATA187 – Modem Voicegateway (Version 15.3(2)T) + CUCM (Version= 8.6) + ATA187 (Version= 9.2.3.1) The modem connection is still not working. What is still to configure on the voicegateway? modem passthrough? Regards Michael ATA 187 Configuration: Fax Mode= T.38 Fax Relay Fax Error Correction Mode Override= Off Maximum Fax Rate= 14000bps Impedance= 900Ohms complex Gateway Configuration: voice service voip ip address trusted list ipv4 172.30.50.1 ipv4 172.30.50.2 ipv4 172.30.50.3 ipv4 172.30.50.4 ipv4 172.30.50.5 allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip no supplementary-service h225-notify cid-update fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none sip bind control source-interface GigabitEthernet0/0 bind media source-interface GigabitEthernet0/0 registrar server expires max 600 min 60 ! dial-peer voice 1 pots translation-profile incoming INCOMING_PSTN incoming called-number . direct-inward-dial ! dial-peer voice 30 voip description OUTGOING_CUCM destination-pattern [1-9].. session target ipv4:172.20.60.12 voice-class codec 1 dtmf-relay h245-alphanumeric fax-relay ecm disable fax-relay sg3-to-g3 fax nsf 000000 no vad
... View more
Labels:
Created by
Michael Endler
in IP Telephony and Phones
in IP Telephony and Phones
02-20-2014
06:15 AM
02-20-2014
06:15 AM
Hi, i managed to modify the session progress sip header: Regards Michael voice class sip-profiles 800 response 183 sip-header TO copy " response 183 sip-header Remote-Party-ID modify "
... View more
Created by
Michael Endler
in IP Telephony and Phones
in IP Telephony and Phones
02-03-2014
04:11 AM
02-03-2014
04:11 AM
Hello Manish, thank you for the quick answer, which seems to be correct. It looks like this is not done by a simple, additional parameter on the voicegateway. I think this is again a thing what cisco should fullfill -> manipulation in both directions Can i change the mandatory fields on callmanager, when incoming ? Thanks Michael
... View more
Created by
Michael Endler
in IP Telephony and Phones
in IP Telephony and Phones
02-03-2014
03:04 AM
02-03-2014
03:04 AM
Hi, i facing an issue with jabber. My issue is coming from the voice gateway. I get from the provider a SIP/2.0 100 Trying to the gateway. The Phonenumber get sent to the callmanager via SIP/2.0 183 Session Progress. May current thinking is the usage of voice class sip-profiles, can i use that like a translation-rule? How can i change the incoming Remote-Party-ID-field + Contact -field in the sip header? voice class sip-profiles 500 request INVITE sip-header P-Asserted-Identity modify "<sip:(.*)@(.*)>" "<sip:00496925771065@192.168.50.48>" request INVITE sip-header P-Asserted-Identity add "P-Asserted-Identity: <sip:00496925771065@192.168.50.48>" request INVITE sip-header Remote-Party-ID modify "<sip:(.*)@(.*)>" "<sip:00496925771065@192.168.50.48>" Debugs: 6175365: Feb 2 17:58:00.412: //1315268/2AB9E1800000/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/TCP 192.168.50.53:5060;branch=z9hG4bK130a9371280707 From: "Test IT-DS" <sip:1065@192.168.50.53>;tag=9230525~8f30a84d-ce6a-4ef2-86e4-7e1d4e6fe6fc-40971876 To: <sip:005123520542@192.168.50.59> Date: Sun, 02 Feb 2014 16:58:00 GMT Call-ID: 2ab9e180-2ee17918-113bf8-35401bac@192.168.50.53 CSeq: 101 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-15.2.2.T2 Content-Length: 0 ! 6175373: Feb 2 17:58:00.412: //1315268/2AB9E1800000/CCAPI/cc_api_display_ie_subfields: ccCallSetupRequest: cisco-username=1065 ----- ccCallInfo IE subfields ----- cisco-ani=00496925771065 dest=00495123520542 ! 6175398: Feb 2 17:58:00.416: //1315269/000000000000/SIP/Info/ccsip_get_int_type_frm_set_mode_ev: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.50.59:5060;branch=z9hG4bK176B16173E From: "Test IT-DS" <sip:00496925771065@192.168.50.59>;tag=3BABBE1C-319 To: <sip:00495123520542@82.211.220.200>;tag=gK02dcc866 Call-ID: 1C7A118-8B6211E3-AB6BF30F-EC3C4587@192.168.50.59 CSeq: 101 INVITE Timestamp: 1391360280 Content-Length: 0 ! 6175768: Feb 2 17:58:04.240: //1315268/2AB9E1800000/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 183 Session Progress Via: SIP/2.0/TCP 192.168.50.53:5060;branch=z9hG4bK130a9371280707 From: "Test IT-DS" <sip:1065@192.168.50.53>;tag=9230525~8f30a84d-ce6a-4ef2-86e4-7e1d4e6fe6fc-40971876 To: <sip:005123520542@192.168.50.59>;tag=3BABCD0C-634 Date: Sun, 02 Feb 2014 16:58:00 GMT Call-ID: 2ab9e180-2ee17918-113bf8-35401bac@192.168.50.53 CSeq: 101 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: <ip:00495123520542@192.168.50.59>;party=called;screen=no;privacy=off Contact: <sip:00495123520542@192.168.50.59:5060;transport=tcp>
... View more
Labels:
Created by
Michael Endler
in IP Telephony and Phones
in IP Telephony and Phones
01-15-2014
04:38 AM
01-15-2014
04:38 AM
Hello Will, i have successfully tested it with prefixes for incoming calls from cucm and cutting the prefix for outgoing to the provider :-) It is not the easiest way, but a working one. Best Regards Michael
... View more
Created by
Michael Endler
in IP Telephony and Phones
in IP Telephony and Phones
01-09-2014
07:05 AM
01-09-2014
07:05 AM
Hi, i have attached my current idea with the prefix. My thinking is that this is choosing the proper interface, but have to test it Regards Michael The config is not covering every single parameter, like the ports or destination
... View more
Created by
Michael Endler
in IP Telephony and Phones
in IP Telephony and Phones
01-04-2014
05:20 AM
01-04-2014
05:20 AM
Hi, Thank you for helping. I now that i hit here an overlapping pattern what gives me the configuration issue. But technically asked, how can i have two mainnumbers and same accesscode, but have a routing depends on the mainnumber? I would think on MGCP, because of directly handeling the interfaces, but this must be solvable by H323?! I tried it with a prefix, but this is not helpfull. As soon as the setup from callmanager reaches the voicegateway, i only can use a matching destination pattern, what activates a translationrule. My thinking was to seperate both mainnumbers, but for incoming calls this is not a problem, only for outgoing. Peer my understanding, the call is coming from callmanager. This is matching an incoming dialpeer, where i can use translationrules. The same behavior will apply for outgoing dialpeer to phoneprovider. My thinking and still my hope, is to use a prefix for all incoming calls from callmanager, which matches the proper outgoing dialpeer. I have to share my configuration, later. Besides direct inward dialing is necessary, otherwise you have two stage dialing. Sent from Cisco Technical Support iPad App
... View more
Created by
Michael Endler
in Collaboration Applications
09-30-2019
09-30-2019
Hi, we have a request if it is possible to share destop in jabber 12.6,
without accepting a presence...
05-15-2019
Hi Mike, ok i tested it by queuing a call to multiple CSQs, with
diffrent priorities, that did not
w...
05-07-2019
Hi Mike, i thought calls in queue A and will be sorted by priority, same
in queue B.When it is given...
05-06-2019
Hi Everyone,i currently have a situation, where i do not understand, how
a call will be treated corr...
Created by
Michael Endler
in Unified Communications Infrastructure
10-09-2017
10-09-2017
Hi, this feature “utils uccx icd clid enable”. is not helping in a way,
where you have to contact pe...
06-15-2015
Hello,the error message '/usr/local/sip/.security/proxy/trust-certs/.0'
can only be fixed by cisco T...
Created by
Michael Endler
in Collaboration, Voice and Video Documents
06-10-2015
06-10-2015
Hello Wolfgang,you have to configure a gateway, which offers the desired
fxs-ports:Callmanager | Dev...
08-14-2014
Hi, i have open a Cisco TAC case and we figured out that later
callmanagers have more options for th...
06-30-2014
I cannot insert a @-symbol or more then one dot in a routepattern, so
far it is not working.The reas...
Created by
Michael Endler
in Unified Communications Infrastructure
06-30-2014
06-30-2014
Hi,one of our customers had the same issue and we could solve it. They
had an outage with the cisco ...
06-26-2014
Hello Jaime,thank you for the quick answer. Then my last concerns are,
how to use Sip RP.They will s...
Created by
Michael Endler
in Collaboration Applications
06-25-2014
06-25-2014
Hi,we have a callmanager with a polycom videoconferencing solution.The
polycom system is attached vi...
Created by
Michael Endler
in Collaboration, Voice and Video Documents
06-06-2014
06-06-2014
Great HowTo! I had an issue to understand the command "destination uri"
and then i found your howto ...
Public Statistics
| Date Registered | 09-26-2013 12:30 AM |
| Date Last Visited | 10-02-2019 02:03 AM |
| Total Messages Posted | 24 |
Helpful Votes Given To
