2901 Cisco Voice gateway + 9 CUCM DECT phones

Similar Questions

• Cisco Voice Gateway with FXO of Telco, operation support for IP phone

  Hi all

  A very quick question.

  Incoming calls via telephone in a voice gateway company ending on a Cisco 7941 G.

  Is it possible for a Cisco 7941 IP handsets or others to instagate actuation when the call is active, then causes the RTC line provide the new key, and then a Telco on via DTMF the combined IP instagate transfer to another number.

  This feature works on a PABX Ericcson previously!

  Is hookflask taken analog device support ONLY?

  I think that the GUY supports, anyone got it working on CallManager?

  Concerning

  Andrew

  Fix Andrew, actuation on IP phone is supported only with the MAN not with CUCM.

  HTH

  GP.

  Pls rate useful messages!

• Replacement of VG224 in Cisco Unified Call Manager (CUCM)

  Hello

  I'm about to replace some of the access points virtual (VG224) to one of my sites and have a few concerns:

  1. Copy configs of the old to the new VG224 will be sufficient?
  2. What needs to be done on the Cisco Unified Call Manager (CUCM) to complete the replacement VG224?
  3. Should I change the configs in Cisco Unified Call Manager if I change the IP address of the VG224 (VG224 migration to the new voice VLAN)?

  Thank you

  Yahya,

  Hope it goes well...

  V is standing for VOICE, not virtual. Vg224 is gateway224 of voice. The answers for your quastion will be...

  1 - Yes, it is sufficient.

  2 - CUCM administration page > device > gateway > find. Select your gateway, change the mac address (10 digits) of the old appliance with a new one.

  3. with the help of sccp Protocol, CUCM don't care about ip address, it's care only about mac address. If the answer is, NO.

  hope that could help

  Concerning

• Video support for voice gateway router can call?

  Hi all

  My gateway voice (Cisco 2921-V/K9) router to connect to the PSTN using SIP Trunk provider. CUCM 11 enter my ipphone and videoconferencing (SX20).

  If someone who using video-conferencing (SX20) to call my (SX20) videoconferencing via dial my number RTC, it only not Visual voice call of appeal.

  I have question: If the router voice gateway support video call?, if it can how to configure and there is an additional module?

  Thank you

  Cyriac

  Hi Cyriac,

  For a video call from point to point, you can see the "Configure the video telephony" section of the following link

  http://www.Cisco.com/c/en/us/TD/docs/voice_ip_comm/CUCM/Admin/9_1_1/CCMS...

  If video conferencing Setup check the following "feature Deprecation announces for video Conferencing and transcoding using PVDM3 on ISR G2 product Bulletin"

  http://www.Cisco.com/c/en/us/products/collateral/unified-communications/...

  Manish

• Why we use the Cisco Voice environment CUBE

  Hi all

  Hope you all are doing well.

  (1) can someone describer that's why we use CUBE in Cisco voice environment and this is the main purpose? Configure the steps?

  (2) what is allowed and what type of license that we use in the world of VOICE?

  (3) features and integration between the CUCM CUPS CUPS?

  Thank you

  Arjun keita

  This very open Q

  To read a bit of paper on the product

  http://www.Cisco.com/c/en/us/products/unified-communications/unified-BOR...

  and no further questions, see the Q & A

  https://supportforums.Cisco.com/document/69976/frequently-asked-question...

  BR
  Mamdouh

• Video conferencing does not work with the voice gateway PVDM3?

  Hello world

  I'll create a meetme conferencing. However, what config it use VGA, audio conference is only allowed

  Here's the material:

  • PVDM3-32 x 2
  • VIC2-2FXO
  • VIC3-2FXS/DON'T
  • 9971 ip phone x 3

  Here is the config VG (work),

  VG (config) #.
  !
  voice-card 0
  dsp-booking voice-service 6
  !
  dspfarm profile 1 homogeneous conference video
  Codec g729br8
  Codec g729r8
  Codec g729abr8
  Codec g729ar8
  Codec h264 cif-cadence 15 bitrate 128 Kbps
  maximum of participants at the conference 4
  maximum 1 x 1 layout
  maximum sessions 1
  associate the PCRS application
  !

  Here is the config VG (failure),

  VG (config) #.
  !
  voice-card 0
  dsp-booking voice-service 6
  !
  dspfarm profile 1 homogeneous conference video
  Codec g729br8
  Codec g729r8
  Codec g729abr8
  Codec g729ar8
  Codec h264 vga-cadence 30 bitrate 1 MB/s
  maximum of participants at the conference 4
  maximum 1 x 1 layout
  maximum sessions 1
  associate the PCRS application
  !

  Here is the result of the test with different codec

  • OK - h264 qcif-cadence 15 bitrate 128 Kbps
  • OK - h264 cif-cadence 15 bitrate 128 Kbps
  • Fail - h264 cif-cadence 30 bitrate 704 Kbps
  • Fail - h264 vga-cadence 30 bitrate 1 MB/s

  When you use the Cisco.com DSP calculator ,
  1 profile of videoconferencing
  Session number: 1
  number of participants: 4
  Conference type: homogeneous - 1 class
  H.264 VGA: checked

  It returns:

  Allocation of Module DSP:
  Router Slot 0: 16 video (25%) 4 votes (7%) 44 available (68%)
  PVDM Slot 0/0: PVDM3-64

  So, why can't I use the VGA with PVDM3-64?

  Thanks in advance

  Sam

  Do not forget this:

  Announces Deprecation feature video conferencing and transcoding using PVDM3 on the product for the ISR G2 Bulletin

  http://www.Cisco.com/c/en/us/products/collateral/unified-communications/TDM-gateways/bulletin-C25-735945.html

• SIP Dialer with voice gateway using the FXO Ports

  Hello

  We have a laboratory UCCE consisting of an outgoing SIP dialer and a h323 2811 voice modem router with 2 FXO ports. The Dialer reserve an agent and sending the call to the bridge, but the call fails immediately with the bridge return a "404 not found" error to the Dialer. Call directly from a CUCM extension attempts to connect successfully through the FXO. Any thoughts?

  IOS Ver - T4 15.1 (3)

  UCCE worm - 8.5.4

  The gateway configuration:

  version 15.1
  horodateurs service debug datetime msec
  Log service timestamps datetime msec
  no password encryption service
  sequence numbers service
  !
  hostname LAB-RS-ING_VXML-GW
  !
  boot-start-marker
  boot-end-marker
  !
  !
  logging buffered 50000000
  no logging monitor

  !
  No aaa new-model
  no location network-clock-participate 1
  !
  voice-card 0
  !
  voice-card 1
  dspfarm
  !
  dot11 syslog
  IP source-route
  !
  !
  IP cef
  !
  !
  !
  IP host mediaserver 10.15.80.83
  No ipv6 cef
  Authenticated MultiLink bundle-name Panel
  !
  !
  !
  !
  !
  !
  Circuits-port FXO group
  hunting-sequential diagram
  !
  !
  !
  voip phone service
  No IP trust to authenticate
  h323 connections allow h323
  allow connections h323 to SIP
  allow connections sip h323
  allow sip to sip connections
  no service additional sip refer
  service additional media - renegotiate
  signs before any
  H323
  Modem passthrough codec g711ulaw nse
  SIP
  interface FastEthernet0/0 source control binding
  bind media source interface FastEthernet0/0
  header-passage
  !
  voice class codec 1
  g711ulaw codec preference 1
  preferably 5 codec g729r8
  !
  vocal h323 class 1
  H225 timeout tcp establish 3
  !
  !
  !
  !
  translation-article 99 of the voice
  rule 1 / ^ 4 / /334/
  !
  !
  voice translation-profile PROFILE_TO_CVP
  translate called 99
  !
  !
  Crypto pki token removal timeout default 0
  !
  !
  !
  !
  Archives
  The config log
  hidekeys
  !
  !
  !
  !
  !
  !
  !
  interface Loopback0
  255.255.255.255 IP address 172.20.1.1
  !
  interface FastEthernet0/0
  IP 255.255.255.0
  automatic duplex
  automatic speed
  !
  interface FastEthernet0/1
  IP 255.255.255.0
  Shutdown
  automatic duplex
  automatic speed
  !
  !
  Router eigrp 15
  10.0.0.0 network
  Auto-resume
  !
  IP forward-Protocol ND
  !
  no ip address of the http server
  no ip http secure server
  !
  IP route 0.0.0.0 0.0.0.0 10.x.x.x
  !
  !
  !
  !
  control plan
  !
  !
  voice-port 0/0/0
  Ports FXO 1 trunk group
  surveillance cut dualtone Mid-communication
  output attenuation - 3
  no echo - cancel enable
  No non-linear
  No vad
  broadcast-maximum: 250
  broadcast / nominal 200
  minimum of playout / high-delay
  broadcast-delay mode fixed
  call waiting times - disconnect 5
  timeouts wait-version 5
  connection ÉRA 4930
  4930 description
  !
  voice-port 1/0/0
  2 ports FXO trunk-group
  surveillance cut dualtone Mid-communication
  output attenuation - 3
  no echo - cancel enable
  No non-linear
  No vad
  broadcast-maximum: 250
  broadcast / nominal 200
  minimum of playout / high-delay
  broadcast-delay mode fixed
  call waiting times - disconnect 5
  timeouts wait-version 5
  connection ÉRA 4198
  4198 description
  !
  voice-port 0/0/2
  !
  voice-port 0/0/3
  !
  !
  !
  profile MGCP default
  !
  !
  voice POTS dial-peer 1
  incoming called-number.
  direct line to inside
  !
  Dial-peer voice 2 voip
  session protocol sipv2
  incoming called-number.
  codec voice-class 1
  no class voice sip reset timer expires 183
  DTMF-relay h245 alphanumeric
  No vad
  !
  Dial-peer voice voip 4198
  destination-model 4198
  session target ipv4:10.15.80.81
  codec voice-class 1
  h323 voice-class 1
  voice-class sip rel1xx taken in charge "100rel.
  no class voice sip reset timer expires 183
  DTMF-relay h245 alphanumeric
  No vad
  !
  voice pots Dial-peer 301
  trunkgroup FXO Ports
  Description * MIA local calls *.
  destination-model 305 [2-9]...
  alert progress_ind activate 8
  progress_ind enable progress 8
  direct line to inside
  prefix 305
  !
  Dial-peer voice voip 4930
  destination-model 4930
  session target ipv4:10.15.80.81
  codec voice-class 1
  h323 voice-class 1
  voice-class sip rel1xx taken in charge "100rel.
  no class voice sip reset timer expires 183
  DTMF-relay h245 alphanumeric
  No vad
  !
  Dial-peer voice voip 7000
  Description of Agents
  destination-model 7...
  session protocol sipv2
  session target ipv4:10.15.80.81
  codec voice-class 1
  voice-class sip rel1xx taken in charge "100rel.
  no class voice sip reset timer expires 183
  !
  Dial-peer voice voip 9999
  Phone agent test description
  destination-model 9999
  session protocol sipv2
  session target ipv4:10.15.80.81
  codec voice-class 1
  voice-class sip rel1xx taken in charge "100rel.
  !
  Dial-peer voice voip 1444
  Phone agent test description
  destination-model 1444
  session protocol sipv2
  session target ipv4:10.15.80.81
  codec voice-class 1
  voice-class sip rel1xx taken in charge "100rel.
  !
  voice pots Dial-peer 303
  trunkgroup FXO Ports
  Description * LD calls *.
  destination-pattern 1 [2-9]...
  alert progress_ind activate 8
  progress_ind enable progress 8
  direct line to inside
  Forward-digit 11
  1 prefix
  !
  voice pots Dial-peer 302
  trunkgroup FXO Ports
  Description * MIA local calls *.
  destination-model 786 [2-9]...
  alert progress_ind activate 8
  progress_ind enable progress 8
  direct line to inside
  prefix 786
  !
  voice pots Dial-peer 9
  destination-model 91 [2-9]...
  alert progress_ind activate 8
  progress_ind enable progress 8
  direct line to inside
  port 0/0/0
  Forward-digit 11
  !
  !
  Dial-peer voice 1400 voip
  Description of Agents
  destination-model 14...
  session protocol sipv2
  session target ipv4:10.15.80.81
  codec voice-class 1
  voice-class sip rel1xx taken in charge "100rel.
  !
  !
  !
  !
  !
  Line con 0
  line to 0
  line vty 0 4
  opening of session
  transport of entry all
  !
  No Scheduler allocate
  end

  Do you have installation trunks in call to the dialer Manager?

• Number of conversion rules in a voice gateway

  I'm an old dog in the world of UC, and the latest versions of IOS had a limitation on the number of voice translation rules, you might have in a profile of translation.

  Anyone know what the limit is for operation 3925 15.1 serial code? I have a customer with a special condition where I may need more than one hundred rules in a profile and won't interrupt the gateway with a test if I already know what the limit is.

  Thanks, Jeff

  Hello, Jeff

  15.1 (4) M, which has 100 rules

  Router (config) #voice translation-rule 1
  Router (cfg-translation-rule) #rule?
  <1-100>Tag translation rule

  Sent by Cisco Support technique iPhone App

• MGCP router/gateway in CUCM update

  I may have been looking in all the wrong ways, but am not finding something that fits

  We have replaced just 10 28XX gateways with bridges 29erxx. They all run MGCP. While they all work in their prior CUCM configuration with the 28XX model shown, I like to keep everything up-to-date.

  Is it possible to make a change to the model of router without the construction of a new and delete the old, or what it is for me?

  No way to change the model, you must recreate them using the new ISR G2 model.

  HTH

  Java

  If it helps, please note

  www.Cisco.com/go/pdihelpdesk

• Want to know primary and secondary configuration to the call, Manager with the voice gateways

  Hi all

  Hope you all are doing well, I wanted to know that we have two other PRI service provider and we want one of them are primary and secondary schools on the other. We have two supplier dedicated 4-4 finish lines. Please provide me with the part of configuration that are required in this case and how to re a SP for primary education and another SP for secondary lines.

  Thank you

  Arjun keita

  Hello Arjun,

  For the full bridge configuration you can check the guides below, but for the PRI redundancy, you create dial-peers and specify the feedback:

  https://www.Google.co.in/URL?SA=t&source=Web&RCT=j&URL=http: / / www.cisco...

  Voice POTS dial-peer 1

  Destination-pettern 0 t

  Port 0/0/0:15

  Preference 0

  Dial-peer voice 2 pots

  Port 0/0/1:15

  Preference 1

  Dest-model 0 t

  Dial-peer voice 3 pots

  Description incoming only

  Incoming called-number. T

  Direct inward dial

  Aseem

  (Please rate if useful)

• Creation of trunk of Cisco 6513 to Cisco SG300 - 10 p for Shoretel phones

  I plugged a new Cisco SG300 - 10 p in an access on our Cisco 6513 port, which is in vtp mode.  I think I will need to create a trunk port of the Cisco SG300 - 10 p 6513, to carry out my office data vlan 1 and my new vlan 112 shoretel VOIP.  I believe that some how all ports are in mode trunk on the default sg300.  I have attached a picture of what it looks like on the management area of vlan sg300.  For some reason any I can plug 3 phones in the sg300 currently just plugged in the 6513 access port and one of the 3 phones come with the vlan voip good 112 and goes into the service very good.  The other 2 phones come but show no service, until I closed the port on sg300 for these other 2 phones and then put the ports back up, then the phones go up.  All of this without going through the port on the Cisco 6513 as a trunk port, it is only now as an access port vlan 1 data and vlan 112 voip vlan.

  My question is, should I put the cisco 6513 in trunk mode and the sg300 will attempt to become the server in vtp and ruin my entire network.  This is what scares me, because I've heard the horror stories of what happens.  My other question is if I have to put the port in trunk mode on him going the sg300 6513, it causes all future phones at the same time without problems?  What would be the cause of 1 of the 3 phones to come as they do and 2 others to come after stop int and put it up?

  Thanks Dave

  Double post.

  Go HERE.

• WAP4410N existing other than Cisco network gateway

  Hello

  I was wondering if someone can quickly answer my question?

  I am trying to use my WAP4410N to create a wireless bridge to an existing network that does not contain any Cisco kit. After reading the manual, I tried the bridge WDS wireless solution, which is my router wireless D-Link & MAC address, but there is no option to put in network WPA2 key, I guess I misunderstood how this feature works, & maybe it works with other products of Cisco wireless?

  Thanks in advance to anyone who can shed light on my question,

  Andy

  Sent by Cisco Support technique iPhone App

  HI Andy,.

  Noramlly that configure you WDS Bridge would need to implement WPA 2 personal option under the wireless part, and then click Security.

  Since the SSID and security must be the same, is that were you would enter information. As to make this set your work with a D-Link wireless router, you'll never miss that probably the question were that this wireless device must support WDS. Even then you might also encounter a question having no match wireless chipsets as well.

  More than likely that this configuration will not work for you. I advise to use an other WAP4410n to establish the connection. Rely on what you are wanting to do a WET200 may also work for you since its just fill a wireless network and only use WDS, but I have never tested this connection with a D-link router so I can't 100% guaranteed if this implementation will work as well or not.

  Hope this helps and let me know if you have any questions.

  Thank you

  -Clayton Sill

• Cisco ATA 187 - unsubscribe CUCM 10.5

  Hello

  I have some problems with my Cisco ATA 187. They keep opt-out at any time. If stop you the Interface on the switch and not stop, it gets saved comes right away. According to cdp, unit is to ping requests all the time, only the record with the callmanager seems to be a problem.

  The versions are:

  • The application load ID APP187.9 - 2-3-1
  • ID startup load 1.8.0.6
  • SW_Version ID 187, 9-2-3-1

  External engineer told me that this series contains some sort of bug, but if I assign/configure the 2nd Port on that one, he shouldn't be hung up. I also tried this, but without success.

  Any ideas? :)

  Thanks and greetings

  Marc

  The second port number not be applied to the 186 s ATA, not 187 s.

  There is a special version of genius of the firmware for the ATA187 which fixes a lot of bugs, but is only available from TAC. We have cmterm-ata187.9-2-3-1-es9.cop.sgn and this makes the 187 s has * a lot * more stable.

  GTG

• Cisco SPA122 gateway analog SNMP MIB

  Hello

  Anyone know where I can get the list of OID or MIB for analog gateway SPA122. I would like to be able to query the State of the line and recording.

  Thank you

  Ronald

  The entire OID tree on SPA122 contain approximately 4000 points about 25 MIB (or more).

  But I'm not sure there is information that you want.

  It seems that the SNMP protocol to provide standard information to the underlying operating system (Linux). Line status and registration is maintained by the speech application running on it and such request provide no SNMP information as far as I know.

  I saved whole SNMP tree in State of rest and again during a call. It seems not to have difference related to the State of the line.

• Problem on Cisco VSMS Gateway

  Hello

  I implement VSMS and already added to VSOM. Every time I change the VSMS via VSOM setting or via the Management Console, the bridge of VSMS always disappear and need to re - put the IP of the gateway. The VSMS and VSOM are not co-located.

  How to solve the problem?

  Thank you.

  More than likely, Yes.  Given the second card (eth1) defined as DHCP is probably the substitution of the static gateway that tries to be configured on eth0.  I don't think that this is even a supported configuration.  I would try certainly eth1 setting to disabled, or if she has a legitimate use which requires its static configuration with no default gateway configured.  At least as a first test.