Calls H323

Similar Questions

• Configuration ExpresswayC/e 8.5 for B2B calls fails to external domains

  Hello

  I am trying to config Highway C/E for B2B scenario.

  The end user has no DNS Public so all the resolution is by its public IP address. In addition ExpresswayE has an IP address in the DMZ (single NIC option)

  I have the following problem:

  Call (h323 or URI) of the remote third listening outgoing calling for and after answering no video and voice.

  For incoming calls, the remote side to see the appeal abandoned, no dial tone and the premises of the party never listen to ringtone. In ExpresswayC I do not see

  the call attempt, I can see the call to ExpresswayE attempt.

  I followed a guide of configuration, but actually I'm not sure of the rule processing and research I have to config in highway C/E

  When there is no Public DNS.

  concerning

  You have inspection of SIP on your firewall? If so, try turning it off.

• Rogue SIP calls on C40

  Hello

  We have a videoconference system with the public IP address that is registered in the VCS-E received various rogue SIP calls.  All these SIP calls are aliases for figures 3-6 during 32sec each.  The calling address entering is: alias @ public IP address of our VC system. All calls are video calls to 384 k

  To avoid calls, I disabled the SIP option and since then we have not received one of these calls.  However, we need to connect to other devices using SIP.  Is there another way to stop these calls?

  Thanks in advance for your help.

  Amrit

  Make the following settings on your endpoints:

  • xConfiguration SIP ListenPort: Off
  • xConfiguration SIP profile 1 out: on

  See bug CSCue55239 for more details.

  You will also need to take steps to secure your VCS if you have not already, turning off UDP SIP SIP calls stop.  However, last year, we saw these calls come on H323 TCP, the only way to stop calls H323 is either secure endpoint behind your firewall and use a script CPL on the VCS.  See the analysis of sourceh323idcisco-incomingcalls on how to configure a CPL.

  FYI, the forums to search would be my first place to look, or search for bugs.  He is asked everywhere in the forums.

• H323 Keep-alives

  I have a client with calls H323 falling at the 1 hour mark. classic symptom of H323 or H225 timeout issues.

  Problem is that the customer has a network refuses (big client with many geographically distributed teams) to adjust the firewall of the team time-out values.

  Is there a way to Setup persistent on endpoints or the VCS?  Endpoints are mainly MXP always, slowly being changed for C-series.

  Thank you.

  Hi Anthony,.

  Cisco/Tandberg endpoints use H245 messages to keep alive the call, they send H245 keep alive messages every 30 seconds. However, in some cases, the firewall does not consider the Dungeon H.245 living messages and then it disconnects the call if no H225 keep alive messages are received during a certain period of time (in your case, 1 hour).

  The problem is, most of the Cisco/Tandberg endpoints cannot send H225 keep alive messages, only H245 messages (they use the roundTripDelayRequest and roundTripDelayResponse messages). Of the endpoints of TC, TC6 and versions later, Cisco has provided a keep alive feature H225, but the default time is 2 hours and that cannot be changed, so endpoints keep alive session every 2 hours.

  For the MXP endpoints, as far as I know, we're not support H225 keep alive, only H245.

  Therefore, you should really fix the configuration of your firewall, once you can make any configuration change in the endpoints to solve the problem.

  Take a peek at this defect associated with H225 keep alive question of the endpoints TC, note that Cisco suggests customers to make adjustments in the typology where the H323 (H225/H245) sessions are sent via or on the device firewall to handle the video traffic correctly.

  https://Tools.Cisco.com/bugsearch/bug/CSCub20591

  Another people in the community here has had the same problem, see:

  https://supportforums.Cisco.com/message/4039690?referring_site=BSS&channel=BDP#4039690

  https://supportforums.Cisco.com/message/3952227?referring_site=BSS&channel=BDP#3952227

  I hope this helps.

  Paulo Souza

  My answer was helpful? Please note the useful answers and do not forget to mark questions resolved as "responded."

• Cisco VCS VCSE H323 Support LAN2 missing

  Hello community,

  After the upgrade VCS to 8.7.1 I see below the H323 config just LAN1 is activated it. I use LAN2 for external connections. So, for this interface, that there is no possible connection with calls H323 1720 from the outside. Have they changed it? There are always two interfaces active in SIP.

  Now they dial-in can´t us on H323 like @.

  Can we do a downgrade and restore the config? Is there a known change? Nothing in the Release Notes.

  Thanks for help.

  Have you tried to call your VCS-E from the outside and to see if it's actually not accessible, or by using SSH check the State of H323 with command: xStatus H323?

  I saw no bug on this topic.  I would downgrade to the previous version you are running, if the second address lan reappears, Sue to get it looked at TAC.  It could be a bug or just a cosmetic problem.

• Public IP address on VC selector

  Dear,

  Attached are the newspapers, (do not consider the time in the newspapers that the sx20 has not been configured with the good times, but newspapers are 5 hrs return)

  I'm calling from SX20 that is configured with the public IP address to another device of tandberg, which is also on the public ip address but I couldn't connect.

  I tried several public IP according to the link, but no luck, https://supportforums.cisco.com/discussion/11499281/ip-adress-test

  To use the independent SX20 I make you have more SX20 configured on the Web Interface:

  Configuration-> Advance Configuration-> H323-> profile-> Callsetup Mode 1: Live

  Configuration-> Advance Configuration-> H323-> profile-> PortAllocation 1: static

  can someone help me understand the issue lies where.

  Thank you

  Just as I suggested in my previous post, you need a videoconferencing (Polycom PVX or Realpresence for desktop PC) application/software running on your laptop. Your port LAN for laptop computer using LAN cable connect it directly to the port LAN SX20. Configure the SX20 and portable with a static ip address in the same network segment to communicate as a peer to peer.

  Laptop IP: 192.168.1.10/24

  SX20 IP: 192.168.1.11/24

  Your own laptop with Polycom software launched, you should be able to dial the IP address of your SX20 and call H323 successful. SX20, you should be able to call the laptop using his IP address as well.

  It is a way to test if your SX20 codec works with the basic configuration to call H323 via IP address.

  You can download the Polycom PVX or Polycom RealPresence of Polycom Web site.

  Kind regards

  Acevirgil

• Problema con el modo TMS provisioning Extension

  Hola, No tengo soporte por lo than no puedo open el contrato a caso en tac

  Tengo los back following problems:

  1 uando yo activo el modo TMS provisioning, me dejan funcionar las llamadas sip, estos his los erroresque me appears:

  VCS-E cuando el are the llamada desde Internet toward the Red international.

  2013-09 - 05T 11: 50:38 - 04:30

  "" "" "" "TVCS: event = 'Search is complete" reason ="authorization not valid - insufficient privilege" Service = "H323" type-aliases-Src ="E164" CBC-alias = '7449"Dst-alias-type ="H323"Dst-alias ="anthony_accardi"call-number ="1a069dfa-1647-11e3-86f9-0010f328943a"Tag ="1a069f44-1647-11e3-b22f-0010f328943a"detail ="found: fake, searchtype:ARQ"Level ="1"elements UTCTime = '2013-09-05 16:20:38, 670"

  In el VCS - C the llamada cuando es desde red Interna toward Internet:

  2013-09 - 05T 11: 53:31 - 04:30

  "" "" "" "TVCS: event = 'Search is complete" reason ="prohibited" Service = "H323" type-aliases-Src ="E164" CBC-alias = '7429"Dst-alias-type ="H323"Dst-alias ="vianyfel_cordaro"call-number ="812a5198-1647-11e3-ba89-0010f325da04"Tag ="812a52e2-1647-11e3-93c9-0010f325da04"detail ="found: fake, searchtype:ARQ"Level ="1"elements UTCTime = '2013-09-05 16:23:31, 687"

  2013-09 - 05T 11: 53:31 - 04:30

  "" "" "TVCS: Event = 'research has attempted" Service ="H323" CBC-alias-type = "E164" CBC-alias ='7429"Dst-alias-type ="H323"Dst-alias ="vianyfel_cordaro"call-number ="812a5198-1647-11e3-ba89-0010f325da04"Tag ='812a52e2-1647-11e3-93c9-0010f325da04" detail = "searchtype:ARQ" Level = "1" elements UTCTime ='2013-09-05 16:23:31, 680"

  2013-09 - 05T 11: 53:23 - 04:30

  "" "" "" "TVCS: event = 'Search is complete" reason ="prohibited" Service = "H323" type-aliases-Src ="E164" CBC-alias = '7429"Dst-alias-type ="H323"Dst-alias ="vianyfel_cordaro"call-number ="7c9181c4-1647-11e3-bda8-0010f325da04"Tag ="7c918304-1647-11e3-865b-0010f325da04"detail ="found: fake, searchtype:ARQ"Level ="1"elements UTCTime = '2013-09-05 16:23:23, 974"

  Cuando el mod TMS Agent activated todo works correctamente esta

  2 Cuando cambio al modo TMS Provisioning yo puedo provisionar equipos desde red interna pero desde internet. Y lo mas than me floor are el jabber video, este are el error q appears:

  013 09 - 05 T 11: 07:42 - 04:30	"" "" TVCS: elements UTCTime = '2013-09-05 15:37:42, 263"Module ="network.sip"Level = 'INFO': Src - ip ="192.168.0.252"Src-port ="25084"detail = 'receive the Request OPTIONS = method, Request-URI = sip: 192.168.0.250:7001; transport = tls, [email protected] / * /"
  2013-09 - 05T 11: 07:42 - 04:30

  "" TVCS: elements UTCTime = '2013-09-05 15:37:42, 261"Module ="network.sip"Level ="DEBUG": Dst - ip ="192.168.0.252"Dst-port ="25084"
  SIPMSG:
  | SIP/2.0 401 Unauthorized
  Via: SIP/2.0/TLS 192.168.0.252:5061; branch = z9hG4bK4de281330ed1277914e57a4bb98ac81416134; received = 192.168.0.252; rport = 25084
  Call ID: [email protected]/ * /.
  CSeq: 38570 OPTIONS
  Starting at: ; tag = 21e96c96b3f9a439
  To: ; tag = ba0e03ca2f6b3957
  Server: TANDBERG/4120 (X7.2.1)
  WWW-Authenticate: Digest realm = "TraversalZone", nonce = "b40cb8278b4a11da992154324161d566d2b57bac3d83c5c518c4528c790d", opaque = "AQAAAN1NC9IHdFS3kNJ3Q6UX2JiBXhut", stale = FALSE, algorithm = MD5, qop = "auth".
  Content-Length: 0

  |

  2013-09 - 05T 11: 07:42 - 04:30

  "" "" TVCS: elements UTCTime = '2013-09-05 15:37:42, 261"Module ="network.sip"Level = 'INFO': Dst - ip ="192.168.0.252"Dst-port ="25084"detail ="sending = 401, method = OPTIONS, To = sip response Code: 192.168.0.250:7001, [email protected] / * /"

  2013-09 - 05T 11: 07:42 - 04:30

  "" TVCS: elements UTCTime = '2013-09-05 15:37:42, 261"Module ="network.sip"Level ="DEBUG": Src - ip ="192.168.0.252"Src-port ="25084"
  SIPMSG:
  | Sip OPTIONS: 192.168.0.250:7001; transport = tls SIP/2.0
  Via: SIP/2.0/TLS 192.168.0.252:5061; branch = z9hG4bK4de281330ed1277914e57a4bb98ac81416134; received = 192.168.0.252; rport = 25084
  Call ID: [email protected]/ * /.
  CSeq: 38570 OPTIONS
  Starting at: ; tag = 21e96c96b3f9a439
  TO:
  Max-Forwards: 0
  User-Agent: TANDBERG/4120 (X7.2.1)
  Support: com.tandberg.vcs.resourceusage
  Content-Type: text/xml
  Content-Length: 250

  25075024960|

  2013-09 - 05T 11: 07:42 - 04:30

  "" "" TVCS: elements UTCTime = '2013-09-05 15:37:42, 261"Module ="network.sip"Level = 'INFO': Src - ip ="192.168.0.252"Src-port ="25084"detail = 'receive the Request OPTIONS = method, Request-URI = sip: 192.168.0.250:7001; transport = tls, [email protected] / * /"

  2013-09 - 05T 11: 07:36 - 04:30

  "" "" "TVCS: elements UTCTime = '2013-09-05 15:37:36, 757" Module ="network.tcp" Level = "DEBUG": Src - ip = "10.10.10.1" Src-port ="10191" Dst - ip = "10.10.10.10" Dst-port ='5060"detail = 'TCP connection is closed"

  2013-09 - 05T 11: 07:36 - 04:30

  "" TVCS: elements UTCTime = '2013-09-05 15:37:36, 641"Module ="network.sip"Level ="DEBUG": Dst - ip ="10.10.10.1"Dst-port ="10191"
  SIPMSG:
  | SIP/2.0 404 not found
  Via: SIP/2.0/TCP 201.210.111.54:2379; branch = z9hG4bK5fc6a3c5021e3557216ef01c2434fb00.1; received = 10.10.10.1; rport = 10191; DefaultZone = ingress-box
  Call ID: [email protected]/ * /.
  CSeq: 301 SUBSCRIBE
  From: <> [email protected] / * />; tag = 2991aa56d191ede3
  To: <> [email protected] / * />; tag = c4114db76ace49d8
  Server: TANDBERG/4120 (X7.2.1)
  WARNING: 200.11.230.253:5060 399 'political response '.
  Content-Length: 0

  |

  2013-09 - 05T 11: 07:36 - 04:30

  "" "" TVCS: elements UTCTime = '2013-09-05 15:37:36, 641"Module ="network.sip"Level = 'INFO': Dst - ip ="10.10.10.1"Dst-port ="10191"detail = 'send = 404, method = SUBSCRIBE, To = sip response Code: [email protected] / * /, [email protected] / * /"

  2013-09 - 05T 11: 07:36 - 04:30

  "" TVCS: elements UTCTime = '2013-09-05 15:37:36, 638"Module ="network.sip"Level ="DEBUG": Src - ip ="10.10.10.1"Src-port ="10191"
  SIPMSG:
  | Sip SUBSCRIBE:[email protected] / * / SIP/2.0
  Via: SIP/2.0/TCP 201.210.111.54:2379; branch = z9hG4bK5fc6a3c5021e3557216ef01c2434fb00.1; received = 10.10.10.1; rport = 10191
  Call ID: [email protected]/ * /.
  CSeq: 301 SUBSCRIBE
  Contact: <> [email protected]/ * /: 2379; transport = tcp >
  From: <> [email protected] / * />; tag = 2991aa56d191ede3
  To: <> [email protected] / * />
  Max-Forwards: 70
  Directions:
  User-Agent: TANDBERG/774 (4.6.3.17194 PCS) - Windows
  Expires: 300
  Event: ua-profile;model=movi;vendor=tandberg.com;profile-type=user;version=4.6.3.17194;clientid="S-1-5-21-1078081533-484061587-725345543";connectivity=1
  Accept: application/pidf + xml
  Content-Length: 0

  Este are el modelo authenticacion as uso y por tanto esa manera config:

  

  Config en VCS Expressway:

  TMS Agent Legacy mode

  Search rule:

  local area-no domain

  Any

  Any

  NO.

  Alias matching

  Regex

  (. +) @domain.com. *.

  Replace

  Continue

  LocalZone.GetDaylightChanges

  local area full URL

  Any

  Any

  NO.

  Alias matching

  Regex

  (. +) @domain.com. *.

  Leave

  Continue

  LocalZone.GetDaylightChanges

  Search of covered area rule

  Any

  Any

  NO.

  Any alias

  Continue

  TraversalZone

  Search for DNS zone rule

  Any

  AllZones

  NO.

  Alias matching

  Regex

  (?. *@%localdomains%.*$).*)

  Leave

  Continue

  DNSZone

  Transform

  Transform the alis destinations to URL

  ([^@]*)

  Regex

  Replace

  ------[email protected] / * /

  Presence PUA - on

  Presence server - off

  CONTROL VCS:

  TMS Extension commissioning of fashion

  Search rule

  local area-no domain

  Any

  Any

  NO.

  Alias matching

  Regex

  (. +) @domain.com. *.

  Replace

  Continue

  LocalZone.GetDaylightChanges

  local area full URL

  Any

  Any

  NO.

  Alias matching

  Regex

  (. +) @domain.com. *.

  Leave

  Continue

  LocalZone.GetDaylightChanges

  Search of covered area rule

  Any

  Any

  NO.

  Any alias

  Continue

  TraversalZone

  External IP address search rule

  Any

  Any

  NO.

  Any IP address

  Continue

  TraversalZone

  Transform

  Transform the alis destinations to URL

  ([^@]*)

  Regex

  Replace

  ------[email protected] / * /

  PUA - on

  presence server - on

  No tengo appeal policy

  Por favor ayudenem a worm as me falta?

  Gracias

  Hi, I do not read Spanish, but I read the newspapers.

  Why don't you use the domain "domain.com"? Could you change this to your own domain name?

  You call H323 to SIP? If you call of H323 and don't add not the domain (if you call SIP), you will not achieve this SIP unit, except if you add a search (or transformation) rule to add the field.

  Additionally, make sure that you have the possibility of the installed key (116341G 00) interoperability, and the interoperability that you allow users ("Registered only" or "On") on your VCS.

  Hope this helps,

  Arne

• Nuisance H323 call SX20

  Hello

  I'm having a problem with a SX20 device that keeps the subject of unwelcome h323 calls, no IP address appears and it just says: "cisco".  We received more than 100 of these calls.  As a work around, I blocked all the h323 on the firewall and permitted in a few IP of known addresses we use.

  I tried to look in the logs of the SX20 to see if I could find the address called IP that was spam so I could block but I couldn't find something useful in there.  Is there a better way all about that now I need to manually add an IP address on the firewall, whenever they want to compose a new person?

  Thank you

  Hello!

  Yes, I noticed that in the newspapers of VCS as well.

  Before the common analyses were on sip/5060/udp.

  The same question that you might have in SIP is expoded to the public, in particular the udp port, but there are also scanning on tcp and at some point there will be TLS as well.

  Its like spam, spammers adopt.

  For now, I would say that, as you did to put behind a firewall and not allow that you really want to have disclosure of sources with (which might not be ideal or even possible) or use a call control as vcs or cucm + highway from the outset.

  If you have your firewall from the outset, you can log the ip addresses trying to do h323 and sip connections, you can try to block. But its also while playing hide and seek, there is no common ips used for scanning, and anyway, you don't know where they scan from, often compromised systems are used for these scans.

  Some firewalls have features of geo - ip, which you could block the continents / countries / regions that you do not contact.

  In any case, consider using a call control and admission control there.

  Please rate this announcement using the stars below.

• SX80 h323 call question

  Hello

  I have a SX80 with software version THIS 8.2.2 on my call manager v11.5 via sip.

  I can call all internal extensions and can call external pstn.

  It's more the CV out there uses H.323 so that when I try to call and conference H.323 via ip call is rejected, after having checked the call log, it's using sip and H.323 not although h.323 is enabled and the direct value (no watchman),

  to work around this problem, I have informed my client to add the address called as a contact and manually select the Protocol of control for this contact.

  But this isn't a solution for my client, because it is possible from the web interface and users of the Conference are high-level management and do not have the time to do so.

  Please I need solution because it is not acceptable to me. and yes I have tried to use x * x * x * x or [email protected] / * / with no use using all the don't SIP not h.323

  Best regards

  Since the touch screen you can add h323: in the IP Address, this will force endpoint to compose using H323.

  Example = h323:1.1.1.1

  Other, you must deploy servers of VCS/Highway to the interoperability of SIP H323, http://www.cisco.com/c/en/us/support/docs/unified-communications/telepre... please check

  Kind regards

• Endpoint collaboration - Protocol call by default Auto - does not seek after failing with SIP H323

  Hi all

  Hoping someone in the community may be able to explain how the "auto" setting for the default calling Protocol?

  My understanding is that, when together, he should try to SIP and H323 - however, it seems to just try SIP then stop.

  I've seen CSCum27379, but could not know if it was fixed in the end.

  Context:

  We are currently move to SIP from H323, and have all items on the same VCS cluster endpoint (Polycom HDX & Cisco) double (first H323 / by default). The problem is that some people dial still using IP addresses, however our new external gateway and bridge services cannot be connected to the use of SIP (non-interoperable).

  The Polycom HDXs are ok, they'll just try a protocol after the other.

  The Cisco seems to be stuck on one, so either not will have access to the new services (except preconfigured in the address book), or will have all their IP address interoperability of point to point calls.

  Is there a way to direct address IP SIP work while composition on the VCS?

  I was also hoping to create a search rule that blackholed IP SIP addressed calls would force the reversal, but did not: (.)

  As always, appreciate any guidance you can provide.

  Rich

  This bug link you to, CSCum27379, there is no solution for it that its status is "completed".

  There is a specific order in which the endpoint will attempt to establish a call and depends on what protocol endpoint is saved for use, pg 84 SX20 Administrator's Guide :

  Allows the automatic selection of the memorandum of Appeal based on what protocols are available. If several protocols are available, is of the order of priority: 1) SIP; (2) H323; (3) H320. If the system is unable to register, self-selection chooses H323

  Endpoint won't try the first Protocol in the list and then switch to the other, he tries only one he can and that's all.  If you subscribed using SIP, but you need to make a call to a registered endpoint H323, you must set the Protocol to call to serve as a H323 or allow interoperability on the VCS to allow SIP-H323 calls.

• Cisco Telepresence SX20 H323 autonomous "no video call.

  This system can receive calls IP to external systems. But when I dial to test outgoing calls (IP calls) see the other side of the call to come and he received but no video/audio on their side. On the side of the appellant, I receive the following message if poster on the screen "no video in" then the call ends. The only messages in the log are below.

  Log messages:

  WARNING: Default call Protocol

  The default call protocol is set to H323, but the system is not registered on this Protocol. The system isn't able to make all the calls. Please change the default call Protocol, verify that Protocol is enabled, or check the configuration of H323.

  Caution: setting up an IPv6 address

  The IPv6 address configuration is statically defined for an invalid address (or mask, gateway or subnet). Enable the DHCP, or enter a valid static IPv6 assignment

  Oh yes, whenever you deal with video, any type of inspection of the firewall must be disabled because it could cause a delay or simply no video sent/received.

• Sx20 does not seek to make the IP to call with H323, after the first attempt failed with SIP

  Hi all

  We use the Sx20 SIP with CUCM registered devices and also register H323 with VCS - c... There are also a few registered with VCS - E external SX20.

  Sx20 > CUCM > VCS - C > VCS-E > Sx20

  the defaultcall Protocol is selected Auto (SIP, H323) on devices SX20.

  SX20 on cucm cannot call IP when defaultcall is selected as auto, it is able to call IP through VSC - C only when defaultcall is selected H323 on Sx20.

  We cannot change the defaultcall as H323 because we cannot make a call since sx20 on CUCM sx20 on vcs - e

  someone has an idea to make the two whole scenerio? Perhaps there is a feature to do this on sx20?

  your response will be appreciated.

  Thank you.

  Take a look at these examples of configuration - Dial IP addresses registered to CUCM endpoints with VCS/highway. This would work in your environment.

  Kind regards

  Acevirgil

• Missing call to E20 in H323 call control buttons

  Hello, everyone runs into a problem with the E20 videophone, where if it's in a H323 call hold and transfer buttons do not appear?  It seems as if make or receive a call from H323 in this case.  SIP works very well. I'm under TE4.1.1 in this environment.

  Sigh... do not currently implemented.

  Link to RN:

  http://www.Cisco.com/en/us/docs/Telepresence/endpoint/software/TE4/Release_notes/cisco_ip_video_phone_e20_te_software_release_notes_te4.PDF

• H323 incoming call drops after 5 minutes each time

  Hi people,

  Can you shed any light on this please?

  H.323 incoming call (numbering by IP)-> external F5-> Fortinet Firewall-> internal F5-> internal firewall-> VCS - E-> VCS - C-> autoattendent MCU

  The call drops after 5 minutes into the call, every time. We tried H.323 call to an internal endpoint, it falls as well. Looks like there is a value of the time-out period defined somewhere on firewall he or F5?

  Any suggestion is appreciated!

  concerning

  Chenzhe

  I hope this helps:

  http://www.Cisco.com/c/en/us/TD/docs/Telepresence/infrastructure/article...

  1. Term applied to network connections through the firewall limits. For example, a Cisco PIX firewall can have a timeout of the form control:

     timeout conn 1:00:00 udp 0:02:00 h225 1:00:00 h323 2:00:00

     (i.e. a list of names of Protocol recognized each followed by a timeout in hours, minutes, and seconds). This example requires a limit of 2 hours on the H.323 connections; However, it also imposes limits on the other protocols, which would also affect a video call (UDP and H225). Several different network protocols are involved in an IP video call. The call being demolished may cause a delay applied to all of them.

• can't make an external call with CuCM and H323 gateway

  Hi experts, I have a problem with CuCM and h323 Gateway. I use a 2811 router to make a gateway H323, CuCM server and Switch layer 2 build a VoIP LAB, this my network:

  

  and my router configuration:

  Current configuration: 2335 bytes

  !

  version 12.4

  horodateurs service debug datetime msec

  Log service timestamps datetime msec

  no password encryption service

  !

  router host name

  !

  boot-start-marker

  boot-end-marker

  !

  forest-meter operation of syslog messages

  !

  No aaa new-model

  !

  !

  !

  dot11 syslog

  IP source-route

  !

  !

  IP cef

  DHCP excluded-address 192.168.1.1 IP 192.168.1.10

  !

  IP VOICE dhcp pool

  network 192.168.1.0 255.255.255.0

  option 150 ip 192.168.1.2

  default router 192.168.1.1

  !

  !

  No ipv6 cef

  !

  Authenticated MultiLink bundle-name Panel

  !

  !

  voip phone service

  h323 connections allow h323

  allow connections h323 to SIP

  allow connections sip h323

  allow sip to sip connections

  service additional h450.12

  Fax protocol cisco

  !

  !

  !

  vocal h323 class 1

  H225 timeout tcp establish 3

  !

  !

  !

  translation of the voice-rule 1

  rule 1 / ^ 9.

  !

  !

  voice translation-profile PSTN

  translate 1 called

  !

  !

  voice-card 0

  !

  !

  application

  Service flash aa1: its - cisco.2.0.2.0.tcl

  paramspace English language en

  paramspace English index 1

  Param aa-driver 0839959204

  paramspace location English flash:

  prefix English paramspace

  !

  !

  !

  Archives

  The config log

  hidekeys

  !

  !

  interface FastEthernet0/0

  IP 192.168.1.1 255.255.255.0

  automatic duplex

  automatic speed

  H323-gateway voip interface

  port of link voip H323-gateway 192.168.1.1

  !

  interface FastEthernet0/1

  no ip address

  Shutdown

  automatic duplex

  automatic speed

  !

  IP forward-Protocol ND

  no ip address of the http server

  no ip http secure server

  !

  !

  !

  control plan

  !

  !

  !

  voice-port 0/0/0

  cover of the echo - cancel 32

  connection ERA opx 3100

  Description 08395959204

  activation of the caller ID

  !

  voice-port 1/0/0

  !

  voice-port 0/0/2

  !

  voice-port 0/0/3

  !

  !

  !

  1010 Dial-peer voice pots

  Service aa1

  incoming called number 08395959204

  direct line to inside

  port 0/0/0

  !

  voice pots Dial-peer 2020

  PSTN description

  translation-profile outgoing PSTN

  destination-model 9 t

  Setup progress_ind allow 3

  progress_ind enable progress 8

  port 0/0/0

  Forward-digits all the

  !

  Dial-peer voice 10 voip

  destination-model 1...

  Setup progress_ind allow 3

  progress_ind enable progress 8

  h323 voice-class 1

  session target ipv4:192.168.1.2

  DTMF-relay h245 alphanumeric

  Codec g711ulaw

  No vad

  !

  !

  !

  !

  Line con 0

  line to 0

  line vty 0 4

  Cisco password

  opening of session

  !

  Scheduler allocate 20000 1000

  end

  I have configured the gateway h323 and attach files to router parttern on follow CUCM

  What is the problem and how to solve?

  Thanks for all the help.

  Your DP pointing CUCM is wrong, or your ÉRA OPX, depending on what you need.

  voice-port 0/0/0

  cover of the echo - cancel 32

  connection ERA opx 3100

  Description 08395959204

  activation of the caller ID

  Dial-peer voice 10 voip

  destination-model 1...

  Setup progress_ind allow 3

  progress_ind enable progress 8

  h323 voice-class 1

  session target ipv4:192.168.1.2

  DTMF-relay h245 alphanumeric

  Codec g711ulaw

  No vad

  Your DP is waiting for 1... and you send 3100, they must match.

  You might need to change the significantt numbers and entering CSS, but only you can know

  For outgoing, you are throwing 9 predot, but again, expect it in your RFP, again, change as you need.

  voice pots Dial-peer 2020

  PSTN description

  translation-profile outgoing PSTN

  destination-model 9 t

  Setup progress_ind allow 3

  progress_ind enable progress 8

  port 0/0/0

  Forward-digits all the

  You cannot transfer the numbers all send you also 9 for each call

  Change your configuration you want / need.

  HTH

  Java

  If it helps, please note

  www.Cisco.com/go/pdihelpdesk