SPA232D no on-> SIP-> FXS FXO caller ID

Similar Questions

• On cards FXO caller ID

  I need a little help from someone intelligent voice (or at least a little bit cleverer than me). I use the handler calls 7.1.3, with H.323 gateways.  Most of my entry doors are 2800 routers with PRI and DID blocks on them.  I have a special office that has just a few FXO cards with a few lines of JARS connected to them.  I almost never get dirty with this office, so I know very well the layout.  I'm much more familiar with my PRI gateways.

  So that's my problem.  I have a user who has a number, it is used for years.  This number is transmitted in our main number XXX-XXX-5155.  When he screams, he wants to display the number caller ID for a long time honey.  However, the caller ID shows POTS another line XXX-XXX-5740, when he emerges from this gateway.  If it goes out one of my PRI gateways, it shows the desired caller ID.  I'm trying to understand what dictates the caller ID.  I don't know if there is a setting in CUCM, a framework on the bridge, or just related to the line that uses the bridge for the tone to the outside, through the phone company.

  This is the base of my gateway config to this place (a VG200).

  voice class codec 1

  g711ulaw codec preference 1

  codec preference 2 g729r8

  !

  !

  !

  vocal h323 class 1

  H225 timeout tcp establish 3

  !

  !

  !

  !

  !

  interface FastEthernet0/0

  address IP X.X.112.3 255.255.255.0

  Speed 100

  full-duplex

  H323-gateway voip bind port X.X.112.3

  !

  IP classless

  no ip address of the http server

  enable IP pim Bennett

  !

  !

  Voice-port 1/0/0

  connection ERA opx 2501

  Description * XXX-5155 *.

  activation of the caller ID

  !

  Voice-port 1/0/1

  connection ERA opx 2501

  Description * XXX-0131 *.

  activation of the caller ID

  !

  Voice-port 1/1/0

  connection ERA opx 2501

  Description * XXX-4903 *.

  activation of the caller ID

  !

  Voice-port 1/1/1

  connection ERA opx 2501

  Description * XXX-5740 *.

  activation of the caller ID

  !

  voice pots Dial-peer 5

  preferably 3

  destination-model 9 t

  Setup progress_ind allow 3

  alert progress_ind activate 8

  port 1/0/0

  !

  Dial-peer voice 6 pots

  preference 2

  destination-model 9 t

  Setup progress_ind allow 3

  alert progress_ind activate 8

  port 1/0/1

  !

  Dial-peer voice 7 pots

  preference 1

  destination-model 9 t

  Setup progress_ind allow 3

  alert progress_ind activate 8

  port 1/1/0

  !

  Dial-peer voice 8 pots

  destination-model 9 t

  Setup progress_ind allow 3

  alert progress_ind activate 8

  port 1/1/1

  !

  Dial-peer voice 100 voip

  destination-model. T

  codec voice-class 1

  h323 voice-class 1

  session target ipv4:

  DTMF-relay h245 alphanumeric

  No vad

  !

  Dial-peer voice voip 101

  preference 1

  destination-model. T

  codec voice-class 1

  h323 voice-class 1

  session target ipv4:

  DTMF-relay h245 alphanumeric

  No vad

  !

  As you can see, the displayed caller ID is that of the line of pots voice-port 1/1/1.  2501 is a post internal which is attached to the phone from the receptionist to the same office.  In the call manager, I can go in the phone of that particular person and value its 'external phone number mask' XXX-XXX-0494, but if it goes out this VG2000, it will display XXX-XXX-5740.  Can anyone think of a way to spend the caller ID through the gateway?

  If there isn't a way, given the limitations of configuring FXO, maybe there is a way to convert his external number to a pots line, plug it directly into a FXO card and send it to its extension?  Then having its extension route its calls on this line of pots?

  Thanks in advance for your help.

  Hi Jake, there is no way to hide the identity of the appellant during the passage of the FXO port. The caller ID is showing as connected to port 1/1/1, because the voice of dial-peer a 8-0, preferably so the call will take this dial-peer as "primary".

  Gabriel.

• SIP URI B2B calls via CUCM and Highway

  Hi all

  Cannot find out how to configure the SIP URI from SX codecs, recorded calls on CUCM.

  I CUCM and IM & P 10.5.1 a pair of X8.5.2 expressway with two areas of course - for B2B calls, and the external jabber.

  I want to make calls of SX-codecs outside via SIP, by using the URI field, for example, [email protected] / * / or [email protected] / * /. How can I configure that?

  I already have the routing model, which is the SIP Trunk for Exp - C, for H323 calls by ip address. Is it possible to make a routing model to call the SIP URI?

  You can see my deployment in this image.

  fghj.PNG

  

  Best wishes

  Igor

  Hi Igor

  These guides will help you resolve your issues:

  http://www.Cisco.com/c/dam/en/us/TD/docs/voice_ip_comm/Expressway/config_guide/x8-5/Cisco-Expressway-SIP-trunk-to-unified-cm-deployment-guide-CUCM-8-9-10-and-x8-5.PDF

  https://supportforums.Cisco.com/ru/video/12362936

  BR Oleksandr

• Translate the name of SIP - UA number called

  Did anyone know, how to use the number called in SIP calls instead of the name of register SIP - UA.

  In my case A7302337 - authentication name, 7302338 - incoming called number.

  But, dial-peer unknown why use of A7302337.

  See below.

  Thanks in advance!

  C2901. Lawer #.

  C2901. Lawer #.

  C2901. Lawer #.

  C2901. Lawer #.

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

  Received:

  GUEST sip:[email protected]/ * /: SIP-5060/2.0

  Via: SIP/2.0/UDP 89.207.94.24:5060; branch = z9hG4bK464550; rport

  P-CGP-redirector: [email protected] / * /

  Record-Route:

  Record-Route:

  Via: SIP/2.0/UDP 89.207.94.25; branch = z9hG4bKac2045160544

  Max-Forwards: 69

  From: <> [email protected]/ * / >; tag = 1 c 2045153887

  To: <> [email protected]/ * /; user = phone >

  Call ID: [email protected]/ * /.

  Contact: <> [email protected]/ * / >

  CSeq: 1 INVITE

  Support: em, 100rel, timer, replaces, path, start of session, resource-priority

  Enable: REGISTER, OPTIONS, GUEST, ACK, CANCEL, BYE, NOTIFY, PRACK, REFER, INFO, REGISTER, update

  User-Agent: Audiocodes-Sip-gateway-Mediant 1000/v.5.20A.021.001

  Content-Type: application/sdp

  Content-Disposition: session

  Content-Length: 569

  v = 0

  o = AudiocodesGW 2045145418 2045145099 IN IP4 89.207.94.25

  s = phone calls

  c = IN IP4 89.207.94.25

  t = 0 0

  m = audio 6770 RTP / AVP 0 8 18 101

  c = IN IP4 89.207.94.25

  a = rtpmap:8 PCMA/8000

  a = rtpmap:0 PCMU/8000

  a G729/8000 rtpmap:18 =

  a = fmtp:18 annex b = No.

  a rtpmap:101 telephone-event/8000 =

  a = fmtp:101 0-15

  a = sendrecv

  a = ptime:20

  a = rtcp:6771 IN IP4 89.207.94.25

  m = image 6772 udptl t38

  c = IN IP4 89.207.94.25

  a = T38FaxMaxBuffer:1024

  a = T38FaxMaxDatagram:122

  a = T38FaxRateManagement:transferredTCF

  a = T38FaxUdpEC:t38UDPRedundancy

  a = T38FaxVersion:0

  a = T38MaxBitRate:14400

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Number = A7302337, called number = A7302337, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = A7302337

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  No outbound dial-peer does not; Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = A7302337, saf_enabled is 1 saf_dndb_lookup = 1, dp_result =-1

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

  Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

  Number = 89036223360, called number =, Voice-Interface = 0 x 0.

  Timeout = TRUE, Peer Encap Type = ENCAP_VOIP, Type of research peer = PEER_TYPE_VOICE,

  Peer Type Info = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

  Result = NO_MATCH(-1) after all rules attempt Match

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result =-1

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

  Number = 89036223360, called number =, Voice-Interface = 0 x 0.

  Timeout = TRUE, Peer Encap Type = ENCAP_VOIP, Type of research peer = PEER_TYPE_VOICE,

  Peer Type Info = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

  Result = NO_MATCH(-1) after all rules attempt Match

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result =-1

  May 17, 14:06:34: //-1/580225688208/DPM/dpAssociateIncomingPeerCore:

  Number = 89036223360, called number = A7302337, Voice-Interface = 0 x 0.

  Timeout = TRUE, Peer Encap Type = ENCAP_VOIP, Type of research peer = PEER_TYPE_VOICE,

  Peer Type Info = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/580225688208/DPM/dpAssociateIncomingPeerCore:

  Result = NO_MATCH(-1) after all rules attempt Match

  May 17, 14:06:34: //-1/580225688208/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result =-1

  May 17, 14:06:34: //-1/580225688208/DPM/dpMatchPeersCore:

  Number =, called number = A7302337, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/580225688208/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = A7302337

  May 17, 14:06:34: //-1/580225688208/DPM/dpMatchPeersCore:

  No outbound dial-peer does not; Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/580225688208/DPM/dpMatchSafModulePlugin:

  dialstring = A7302337, saf_enabled = 0, saf_dndb_lookup = 1, dp_result =-1

  May 17, 14:06:34: //-1/580225688208/DPM/dpMatchPeersMoreArg:

  Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/580225688208/DPM/dpMatchPeersCore:

  Number =, called number = A7302337, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/580225688208/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = A7302337

  May 17, 14:06:34: //-1/580225688208/DPM/dpMatchPeersCore:

  No outbound dial-peer does not; Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/580225688208/DPM/dpMatchSafModulePlugin:

  dialstring = A7302337, saf_enabled = 0, saf_dndb_lookup = 1, dp_result =-1

  May 17, 14:06:34: //-1/580225688208/DPM/dpMatchPeersMoreArg:

  Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Number = A7302337, called number = A7302337, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = A7302337

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  No outbound dial-peer does not; Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = A7302337, saf_enabled = 0, saf_dndb_lookup = 1, dp_result =-1

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

  Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

  Number = A7302337, called number =, Voice-Interface = 0 x 0.

  Timeout = TRUE, Peer Encap Type = ENCAP_VOIP, Type of research peer = PEER_TYPE_VOICE,

  Peer Type Info = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

  Result = NO_MATCH(-1) after all rules attempt Match

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result =-1

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

  Number = A7302337, called number =, Voice-Interface = 0 x 0.

  Timeout = TRUE, Peer Encap Type = ENCAP_VOIP, Type of research peer = PEER_TYPE_VOICE,

  Peer Type Info = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

  Result = NO_MATCH(-1) after all rules attempt Match

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result =-1

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Number =, called number = A7302337, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = A7302337

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  No outbound dial-peer does not; Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = A7302337, saf_enabled = 0, saf_dndb_lookup = 1, dp_result =-1

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

  Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/580225688208/DPM/dpMatchPeersCore:

  Number =, called number = A7302337, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/580225688208/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = A7302337

  May 17, 14:06:34: //-1/580225688208/DPM/dpMatchPeersCore:

  No outbound dial-peer does not; Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/580225688208/DPM/dpMatchSafModulePlugin:

  dialstring = A7302337, saf_enabled is 1 saf_dndb_lookup = 1, dp_result =-1

  May 17, 14:06:34: //-1/580225688208/DPM/dpMatchPeersMoreArg:

  Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Number =, called number = A7302337, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = A7302337

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  No outbound dial-peer does not; Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result =-1

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:

  Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Number =, called number = 101, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = 101

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Result = Success (0) after DP_MATCH_DEST

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result = 0

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:

  Result = Success (0)

  The outgoing dial matched host list:

  1: Tag dial-peer = 40002

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Number =, called number = A7302337, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = A7302337

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  No outbound dial-peer does not; Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result =-1

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:

  Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Number =, called number = 103, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = 103

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Result = Success (0) after DP_MATCH_DEST

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result = 0

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:

  Result = Success (0)

  The outgoing dial matched host list:

  1: Tag dial-peer = 40006

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Number =, called number = A7302337, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = A7302337

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  No outbound dial-peer does not; Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result =-1

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:

  Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Number =, called number = 104, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = 104

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Result = Success (0) after DP_MATCH_DEST

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result = 0

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:

  Result = Success (0)

  The outgoing dial matched host list:

  1: Tag dial-peer = 40004

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Number =, called number = A7302337, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = A7302337

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  No outbound dial-peer does not; Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result =-1

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:

  Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Number =, called number = 105, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = 105

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Result = Success (0) after DP_MATCH_DEST

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result = 0

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:

  Result = Success (0)

  The outgoing dial matched host list:

  1: Tag dial-peer = 40003

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Number =, called number = A7302337, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = A7302337

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  No outbound dial-peer does not; Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result =-1

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:

  Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Number =, called number = 106, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = 106

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Result = Success (0) after DP_MATCH_DEST

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result = 0

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:

  Result = Success (0)

  The outgoing dial matched host list:

  1: Tag dial-peer = 40001

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Number =, called number = A7302337, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = A7302337

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  No outbound dial-peer does not; Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result =-1

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:

  Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Number =, called number = 107, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = 107

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Result = Success (0) after DP_MATCH_DEST

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result = 0

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:

  Result = Success (0)

  The outgoing dial matched host list:

  1: Tag dial-peer = 40005

  May 17, 14:06:34: / / 206/580225688208/SIP/Msg/ccsipDisplayMsg:

  Envoy:

  SIP/2.0 100 trying

  Via: SIP/2.0/UDP 89.207.94.24:5060;branch=z9hG4bK464550;rport,SIP/2.0/UDP 89.207.94.25; branch = z9hG4bKac2045160544

  From: <> [email protected]/ * / >; tag = 1 c 2045153887

  To: <> [email protected]/ * /; user = phone >

  Date: Thu, 17 may 2012 14:06:34 GMT

  Call ID: [email protected]/ * /.

  CSeq: 1 INVITE

  Allow-events: telephone-event

  Server: Cisco-SIPGateway/IOS-15.2.3.T

  Content-Length: 0

  May 17, 14:06:34: / / 206/580225688208/SIP/Msg/ccsipDisplayMsg:

  Envoy:

  SIP/2.0 404 not found

  Via: SIP/2.0/UDP 89.207.94.24:5060;branch=z9hG4bK464550;rport,SIP/2.0/UDP 89.207.94.25; branch = z9hG4bKac2045160544

  From: <> [email protected]/ * / >; tag = 1 c 2045153887

  To: <> [email protected]/ * /; user = phone >; tag = 29C6FC-1848

  Date: Thu, 17 may 2012 14:06:34 GMT

  Call ID: [email protected]/ * /.

  CSeq: 1 INVITE

  Allow-events: telephone-event

  Server: Cisco-SIPGateway/IOS-15.2.3.T

  Reason: Q.850; cause = 1

  Content-Length: 0

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

  Received:

  SIP ACK:[email protected]/ * /: SIP-5060/2.0

  P-CGP-redirector: [email protected] / * /

  Via: SIP/2.0/UDP 89.207.94.24:5060; branch = z9hG4bK464550; rport

  Max-Forwards: 69

  From: <> [email protected]/ * / >; tag = 1 c 2045153887

  To: <> [email protected]/ * /; user = phone >; tag = 29C6FC-1848

  Call ID: [email protected]/ * /.

  CSeq: 1 ACK

  Content-Length: 0

  Try to use a voice translation like this rule:

  translation of the voice-rule 1

  rule 1 /A7302337/ /7302338/

  voice translation-profile XLATE_CALLED

  translate 1 called

  If you want to use the To header instead the SIP URI, you must use the CUBE.

  See this article: http://tblog.cisco.be/2011/02/17/cube-conditional-sip-profiles/

  Kind regards.

• SPA232D problems integrating the POTS via FXO

  I just bought a SPA232D and I have problems with the initial configuration with the POTS via the FXO line.  I'm back to the original factory configuration for partition the problem.  If I connect a phone to the fixed telephone line work and turn the SP232D off, all phone funtion works, inbound and outbound more voicemail from the central server by the PSTN.

  If I power up the SPA232D with the line FXO connected to the POTS RJ11 and FXS line phone, I can make outgoing calls through the PSTN.  However the incoming calls to the PSTN not sound and the service of voicemail never folds on the Central POTS but disconnects.  Sometimes I see the hook out in the status page.

  In the State of the engine, if I register a DECT handset and set it to the PSTN and do to the PSTN by default, I make a call through the PSTN, it sounds and I get a successful connection.  I can also make outgoing calls through the PSTN.  However, if I do not answer the call voicemail service never folds on the Central POTS but disconnects.

  Clearly, I have good location of the SPA232D with the POTS (US, Verizon (formerly Frontier), considered an old switch of POTS, not a great line, Upstate New York).  But since it works when it is off, the basics seem to be there.

  Does anyone have advice.  I can't find any close correspondence on the resources of the web.

  Thank you

  Dan Davis

  The notice just found elsewhere - it may or may not help:

  1. You can set this to 'no' (No. When disabled, PSTN calls will not be auto-répondu by the SPA.
  2. "The SPA-3000 is two have independent in a box." For 'line 1 to @gw0' and ' line PSTN Ring Thru for line 1 "feature, the SPA-3000 places in fact call VoIP on one side of the box to the other."

  In addition, read page 210 of the Cisco SPA232D enhanced mobility phone adapter Administration Guide

  It describes the process of appeal RTC / 1 a little curve.

  From the pages 209-210, I wish that:

  1. ring of appeal incomming thru online FXS FXO
  2. If not caught in time, it is autoanswered by ATA. He began to play the tone. The caller must enter target number via DTMF
  3. target number is handled by the default dial plan

  If no target number not entered until volume dial timeouted, the call is completed.

  It seems to comply with your observation.

  Then try to PSTN-to-VoIP Gateway allow no value and try again.

• SIP trunk CUBE with Callcentric - incoming unanswered call

  I'm doing some tests with a Sip trunk with a provider called Callcentric.
  It is a CUBE scenario. I use a SIP to the CUCM trunk.
  
  I have a Cisco Callmanager and a virtual router 7200 (GNS3) as a gateway (C7200-ADVENTERPRISEK9-M), Version 12.4 (24).

  A CPIC connected to Callmanager, I call out to PSTN and it works perfectly.

  When I do an incoming call to the PSTN to the softphone destination, she sounds, but when I press the answer button, the call is not connected and the analog phone in RTC can listen back ringtone.

  Do you have any idea what it could be?
   

  Some relevant configurations:
   

  voip phone service
  allow sip to sip connections
  Fax protocol cisco
  SIP
  interface FastEthernet0/0 source control binding
  bind media source interface FastEthernet0/0
  Registrar Server

  voice class codec 1
  g711ulaw codec preference 1

  translation of the voice-rule 1
  rule 1 / ^ 8 / /0056/
  !
  voice translation-rule 2
  rule 1 5.0 / /17772114zzz/
  !
  voice translation-rule 3
  rule 1 /17772114zzz/ /500/

   

  voice translation-profile IN
  definition of 3 called
  !
  FLIGHT voice translation-profile
  definition of call 2
  translate 1 called

  Dial-peer voice 1 voip
  CALLCENTRIC description
  entrants IN translation-profile
  translation-profile outgoing OUT
  destination-model 8.T
  codec voice-class 1
  session protocol sipv2
  session target sip-Server
  incoming called-number 17772114zzz
  SIP DTMF-relay-notify rtp - nte
  !
  Dial-peer voice 2 voip
  CUCM description
  destination-model 500
  media stream-autour
  codec voice-class 1
  session protocol sipv2
  session target ipv4:192.168.10.116
  incoming called number 8.T
  SIP DTMF-relay-notify rtp - nte

  SIP - ua
  credentials of username, password 7 17772114zzz 094F471B100928 callcentric.com Kingdom
  authentication username 17772114zzz password 7 121A0C051B0803 callcentric.com Kingdom
  no remote-party-id
  Registrar dns:callcentric.com expires 3600
  DNS:callcentric.com SIP server
  Home-Office

  
  Thank you guys.
   
   

  Hello

  Please configure the sip requires low profile to remove the header and apply this profile in the dial-peer 1 outgoing.

  SIP-class voice profiles 1

  response header 200 sip requires DELETE

  If this does not work under Dial-peers, try to apply globally.

  voip phone service

  SIP

  SIP profiles 1

  Suresh

  Please note all useful posts

• SPA9000 Setup problem with FXS ports

  SPA9000 PBX, configured with two VoIP providers implemented voice-> 1/2 line parameters. A telephone is connected to the FXS 1 (or 2 FXS) port and able to choose the right VoIP provider when making outgoing calls.

  SPA9000 question is that once the outgoing call is established, does not meet the other numbers composed for the extension of the called party number. Same question when you dial a number where voice prompt responds to the call, offering the choice of activities, to choose the specific number. None of the input selection works as nothing is written on the dial.

  Located in the Canada. VoIP providers: freephoneline.ca & callcentric.com

  SPA9000 Setup:

  Software version: 6.1.5; Hardware version: 1.0.5(a)

  Voice-> SIP-> PBX parameters-> call routing rule: (<:L1>9xx. |) <:L2>8xx.)

  Voice-> SIP-> settings-> telephone numbering Plan PBX phone: ([89], S 11, 0 [3469] |) [89] 1 [2-9] xxxxxxxxxS0 | [89], 011xx. | ([89], xx.)
  Voice-> FXS 1 / 2-> Dial Plan: ([89], S 11, 0 [3469] |) [89] 1 [2-9] xxxxxxxxxS0 | [89], 011xx. | ([89], xx.)

  Voice-> line 1-> Dial Plan: (<>xx.) [Smiley here shows insetad of: and >]
  Voice-> 2-> Dial Plan line: (<>xx.) [Smiley here shows insetad of: and >]

  Voice-> regional-> ringtones of appellate courts:
  Tone: 350@-19,440@-19;10(*/0/1+2)
  Second tone: 420@-19,520@-19;10(*/0/1+2)
  Outside the tone: 420@-16;10(*/0/1)
  Prompt tone: 520@-19,620@-19;10(*/0/1+2)
  Busy signal: 480@-19,620@-19;10(.5/.5/1+2)
  Reorder tone: 480@-19,620@-19;10(.25/.25/1+2)
  Off-hook warning: 480@-10,620@0;10(.125/.125/1+2)
  Back ringtone: 440@-19,480@-19;* (4/2/1 + 2)
  Confirm tone: 600@-16;1(.25/.25/1)
  SIT1 Tone: 985@-16,1428@-16,1777@-16;20(.380/0/1,.380/0/2,.380/0/3,0/4/0)
  T2is Tone: 914@-16,1371@-16,1777@-16;20(.274/0/1,.274/0/2,.380/0/3,0/4/0)
  SIT3 Tone: 914@-16,1371@-16,1777@-16;20(.380/0/1,.380/0/2,.380/0/3,0/4/0)
  SIT4 Tone: 985@-16,1371@-16,1777@-16;20(.380/0/1,.274/0/2,.380/0/3,0/4/0)
  Tone MWI: 350@-19,440@-19;2(.1/.1/1+2);10(*/0/1+2)
  Cfwd-tonalite: 350@-19,440@-19;2(.2/.2/1+2);10(*/0/1+2)
  Holding tone: 600@-19;*(.1/.1/1,.1/.1/1,.1/9.5/1)
  Tone of the Conference: 350@-19;20(.1/.1/1,.1/9.7/1)
  Ensure the Indication of the ringtone: 397@-19,507@-19;15(0/2/0,.2/.1/1,.1/2.1/2)
  Featured call tone: 350@-16;*(.1/.1/1)

  Hi @sglumac, it seems that your SPA9000 is a Cisco product. Please repost this at this link: https://supportforums.cisco.com/ for solutions.

• SPA112 not answering calls

  Hi all

  I have a problem with Cisco SPA112 ATA. I can make calls without problem. When someone calls the ATA configured number, analog phone attached to rings of the ATA, but when I answer, I don't hear anything. The calling party infiltrates the waiting tone.

  ATA debugging log when this happens is presented below. I modified the phone number, the call is initiated from.

  Any ideas what is the cause?

  Kind regards

  Tom

  NEW_CALL_STATE(), dial 0: former State = CC_CST_IDLE, new State CC_CST_RINGING
  CID_initGen 8
  [0]:CID_initGen() Cid > won 0 delay 2200 phone 501 501 * name *.
  RTP_nextMediaPort(), port = 16442
  RTP_nextMediaPort(), rc = 16440
  AUD_allocCallObj() call (0x1b1d40)
  [0:0] DIAL the ALLOC AUD (port = 16440)
  +++ SIP_lineCcCmdProc CC_CMD_ACCEPT AUD_startRtpRx
  [AUD] AUD_startRtpRx (0x1b1d40) cover 0
  Local loop mode: No. Type: No.
  Remote loopback mode: No. Type none.
  Configuration of the RTP channels: udp_no_checksum 0, sysmmetric_rtp 0, tos 0xb8, cos mlb 6, 0.
  uchConnectEpToNode(), connection VoIP EP 0 node 0
  uchEnableEchoCan(), cover 0 EP 2 turn on
  uchEnableModemCall() Modem call state (0) no change
  UCH sync parameter hold off time is 70
  Succeed together QoS
  uchSetGTD(), disable GTD 1 FXS
  State GTD uchSetGTD() not change
  uchSetDTMFMute(), DISABLE
  cordless_start_rtp(), chan: 0 ip: (null) remote port: 0 local: 16440 rx: 1 ipt:0 ptime:30 bInMdmPasstru:0
  From RTP only Rx.
  Decision-making related to 16440 port 14.
  Remote IP/port: 0.0.0.0:0
  SDP codec list (internal pt): 0 18 8 134 136
  List of load RX: PCMU/8000 (0) G.729/8000 (18) PCMA/8000 (8) NSE/8000 (100) encaprtp/8000 (112)
  Set RTP_SESSION_OPT_DTMF
  VAD = 0
  RTP configuration:
  audio_mode RTP_MODE_REC_ONLY, media_loop_level RTP_LOOP_LEVEL_NONE, dtmf2833numEndPakcets 3, opt 0 x 0
  Codec: length 30, 101, 101 tx_pt_event rx_pt_event, tx_pt 0
  RX [0] 0 PCMU/8000, rx [1] 18 G.729/8000, rx [2] 8 PCMA/8000
  RX [3] 100 NSE/8000, rx [4] 112 encaprtp/8000, rx [5] - 1
  Stem: 180ms max, min 60ms, adapt 1
  RTP channel 0 is blank: 1.
  # rtp seq number is 668
  RTP session 0 has begun
  [AUD] RTP Rx upward
  Pol CID:OnHookTx
  [0] CID CID_ST_POLREV_POST_DELAY
  uchPlayDTASTone(), playuchSetMute(), SWITCH, ret = 0
  [0] CID CID_ST_CAS
  [0] CID CID_ST_CAS_POST_DELAY
  uchDisplayCIDFSK(), EP 2 0 95 buflen cover costs 60 generals SZ_MAX_USERDATA 200 won 0
  uchDisplayCIDFSK(), FSK Caller ID standard is 0 (bell 202)
  uchDisplayCIDFSK(), MarkFreq SeizeFreq 0 x 16 0xc
  [0] CID Start DTMF/FSK, CID_ST_ACTIVE
  uchAppCb(), 49 received EP 2 0 cover event
  receive CH_ASYNC_CIT_TRANSMITTED
  [0] CID CID: FACT
  [0] CID CID_ST_ACTIVE_POST_DELAY
  [0] CID CID_ST_IDLE
  CID:ring now
  CC_eventProc(), event: CC_EV_CID_DONE (0 x 51), cover: 0, rating: 0, par2: (none)
  AUD_ccEventProc: event 81 vid 0 by 0 x 0 par2 0x0
  SLIC_startRing of State 0 ts 0x1af54con 2000 stop 4000 len 60000
  [0] ring cad event 0-pol 0
  uchSetMute(), DISABLE, ret = 0
  [0] ring cad event 1-pol 0

  SIP_tsCreateClient(), 1671, uiTmrF = 1600, SIP_TMR_F_INIT = 1600
  SIP_tsClientEventProc(Event: 28) State = 1
  SIP_regTsEventProc(Event: 28)
  SIP_regTsEventProc(Event: 32)
  Detached TS
  SIP_tsClientEventProc(Event: 3) status = 3
  [0] ring cad event 0-pol 0
  [0] ring cad event 1-pol 0
  SIP_tsClientEventProc(Event: 9) status = 3
  [0] off hook
  CC_eventProc(), event: CC_EV_USR_OFFHOOK (0x2), cover: 0, rating: 0, par2: (none)
  AUD_ccEventProc: event 2 vid 0 by 0 x 0 par2 0x0
  sysstatus_set_led_status_payton(), led_id: 1, statusCode:7, systemEvent: 0 x 100095
  callEventProcTable [5] is cepRingingProc
  cepRingingProc (lid = 0, call = 0x17f9f4, event = 11 (CC_EV_USR_ANSWER), by = 0, par2 = (Nile))
  NEW_CALL_STATE(), dial 0: former State = CC_CST_RINGING, new State CC_CST_ANSWERING
  SLIC_stopRing
  [0] ring cad event 2-pol 0
  SLIC_stopRing
  SLIC_stopTone
  uchStopVoipTone(), stop Voip your EP 3
  Start the timer G and H (e = 22)
  TMR shot G
  TMR shot G
  TMR shot G
  SIP_sessTsEventProc(Event:27)
  Departure TmrJ
  SIP_sessDlgEventProc: event: 45 (SIP_EV_DLG_BYED), ucState: 0
  Entered SIP_releaseAudioResources() #!
  Statistics of calls asking...
  Stats of RTP TX updated for channel 0
  Stats of RTP RX updated for channel 0
  Call the updated statistics.
  AUD_releaseCallObj() call (0x1b1d40)
  [AUD] AUD_stopRtpTx (0x1b1d40)
  cordless_stop_rtp_tx(), channel 0.
  Channel RTP not in Tx. Nothing can stop!
  Channel RTP not in Tx. Nothing can stop!
  [AUD] RTP low Tx
  [AUD] AUD_stopRtpRx (0x1b1d40)
  cordless_stop_rtp_rx(), channel 0.
  RTP channel 0 ranging from Rx to idle.
  RTP configuration:
  audio_mode RTP_MODE_INACTIVE, media_loop_level RTP_LOOP_LEVEL_NONE, dtmf2833numEndPakcets 3, opt 0 x 0
  Codec: length 30, 101, 101 tx_pt_event rx_pt_event, tx_pt 0
  RX [0] 0 PCMU/8000, rx [1] 18 G.729/8000, rx [2] 8 PCMA/8000
  RX [3] 100 NSE/8000, rx [4] 112 encaprtp/8000, rx [5] - 1
  Stem: 180ms max, min 60ms, adapt 1
  RTP channel 0 is now inactive.
  [AUD] RTP down
  [AUD] AUD_releaseRtp (0x1b1d40)
  cordless_stop_rtp(), freeing the channel RTP: 0
  cordless_stop_rtp(), RTP session 0 arrested succussfully
  uchRelChanAndEP (0, 3)
  uchDisconnectEpFromNode(), disconnection VoIP EP 0 node 0
  [AUD] RTP channel released
  [0:0] call of Rel AUD
  SIP_releaseAudioResources(), CC_lineIsIdle (0) = 0, gAudLine [0] .bIvr = 0, AUD_relUchNode?
  Output SIP_releaseAudioResources() #!
  CC_eventProc(), event: CC_EV_SIG_CALL_ENDED (0 x 34), cover: 0, rating: 6, par2: (none)
  AUD_ccEventProc: event 52 vid 0 by 0 x 6 par2 0x0
  callEventProcTable [7] is cepAnsweringProc
  cepAnsweringProc (lid = 0, call = 0x17f9f4, event = 52 (CC_EV_SIG_CALL_ENDED), by = 6, par2 = (Nile))
  CC: finished
  NEW_CALL_STATE(), dial 0: former State = CC_CST_ANSWERING, new State CC_CST_INVALID
  Statr CPC CC_EV_TMR_INVALID timer
  SLIC_stopRing
  SLIC_stopRing
  SLIC_stopTone
  # 1248 RTP_SEQ_NUM_EVT
  [0] on the hook
  CC_eventProc(), event: CC_EV_USR_ONHOOK (0x1), cover: 0, rating: 0, par2: (none)
  AUD_ccEventProc: event 1 vid 0 by 0 x 0 par2 0x0
  sysstatus_set_led_status_payton(), led_id: 1, statusCode:1, systemEvent: 0 x 100093
  callEventProcTable [6] is cepInvalidProc
  cepInvalidProc (lid = 0, call = 0x17f9f4, event = 1 (CC_EV_USR_ONHOOK), by = 0, par2 = (Nile))
  callEventProcTable [0] is cepIdleProc
  cepIdleProc (lid = 0, call = 0x17fc14, event = 1 (CC_EV_USR_ONHOOK), by = 0, par2 = (Nile))
  cepIdleProc(), lid = 0
  [IVR_eventProc] evt 1 lid 0
  callEventProcTable [6] is cepInvalidProc
  cepInvalidProc (lid = 0, call = 0x17f9f4, event = 10 (CC_EV_USR_ENDCALL), by = 0, par2 = (Nile))
  NEW_CALL_STATE(), dial 0: former State = CC_CST_INVALID, new State CC_CST_IDLE
  [AUD] Release knot UCH for AUD_LINE 0.
  uchDisableNode(), node released 0 ret = 0
  [AUD] Node UCH released 0.
  SLIC_stopRing
  SLIC_stopTone
  Succeed together QoS
  TMR shot G

  SIP_tsCreateClient(), 1671, uiTmrF = 1600, SIP_TMR_F_INIT = 1600
  SIP_tsClientEventProc(Event: 28) State = 1
  SIP_regTsEventProc(Event: 28)
  SIP_regTsEventProc(Event: 32)
  Detached TS
  SIP_tsClientEventProc(Event: 3) status = 3
  TMR shot G
  TP Analyzer error: 34
  SIP_tsClientEventProc(Event: 9) status = 3
  TMR shot G
  TMR shot G
  TMR shot G

  16:31:23 [0]Off Hook16:31:23 NEW_CALL_STATE(), call 0: old state = CC_CST_RINGING, new state CC_CST_ANSWERING16:31:49 SIP_sessDlgEventProc: event: 45(SIP_EV_DLG_BYED), ucState: 0

  Phone go off the hook. Setup of the call must be made here. I guess that the anti-terrorism Act has sent 200 OK "response to confirmation by peers. It has not been recognized, call was BYEd peer (after 20 sec timeout).

  The disconnection happens indeed on the remote side but after 'no answer'.

  No way to have two the same call disconnection. Mobile phone received 'no response' for a reason. And your ATA did not sign-out request (well, I guess with any SIP connects available). The disconnection was initiated by an outside entity to you. Yes, it can be caused by some session parameters used by the anti-terrorism Act, but with no available SIP newspaper, we have little chance to guess the true cause. My best advice is (assuming that you are unable to capture the SIP packets now) - call your support operator for help. I suspect that the next PBX (from your side) launches the disconnection so logs of your operator should disclose the cause in its entirety.

• CallManager Express SIP trunk problem

  Hi all here.

  I have a problem with my SIP trunk CallManager Express (version 10.0). On my site already configured trunk SIP between CME and CUE. I have configured DN for SIP phone and SIP phones recorded on CME successfully, but... As soon as SIP phone registered on CME, DIAL-PEER for CME and CUE connection changes. Especially 'session target ipv4' automatically changes to my IP phone SIP and all calls go to my SIP phone and does not reach the auto attendant.

  How can I solve this problem?

  Hello

  Following is the error of CUCM;

  WARNING: 399 UnicompCM "cannot find a Device Manager for the request received on port 55549 leave 192.168.10.2.

  This probably means that CME use 192.168.10.2 for source packages SIP CUCM however SIP trunk in CUCM is not directed to this IP address, it is useful to point out to another IP interface of the GUY and therefore dismiss this appeal.

  Can you please check what is the IP of CMF address you configured on the SIP trunk in CUCM?

  -Vivek

• SPA232D network connectivity

  I have a SPA232D that I want to connect to a pre-existing network of SOHO, I looked for the network interfaces, I have a request.

  The current network has a router running NAT.

  What I expected to be able to do was to just add this phone to my LAN and use the DHCP server on my existing router to provide the address (I use DHCP to all my devices in LAN, but reservations to provide compatible addresses).

  It seems that it is not possible to configure the 232D SPA to use a DHCP server on the LAN interface.

  As I see it, I have two options:

  (1) connect the SPA232D to my LAN via its Ethernet port and configure a static address

  (2) connect the SPA232D to my LAN via its Internet port and use my DHCP server (then select "Remote" so I can manage the SPA)

  Will there be operational or functional problems with option 2.

  Bridge mode (as opposed to the NAT mode) has no importance with or the other of these two connectivity options.

  Use the WAN of SPA232D with DHCP port, leave the unused LAN port. Router/bridge mode not is not relevant then, let him therefore in default state.

  While SIP is passing NAT there may be operational and functional issues. It depends on type of NAT, SIP ALG put implemented (or not) on the NAT device and the General configuration of both SPA232D as remote SIP proxy.

• BFCP of MCU for encrypted calls

  Hello

  I know that it has been discussed before on the CUCM especially earlier this year:

  https://supportforums.Cisco.com/message/4001153#4001153

  We have a series using VCS (7.2), conductor (2.2) and MCU (4.4) where we need to content SIP on encrypted calls for dual display using BFCP systems, because the driver does NOT.  It is not acceptable to disable encryption on the MCU for all types of calls as a workaround.

  The 4.4 release notes:

  Binary control protocol floor on encrypted calls:

  The transmission of content SIP of the MCU by using the binary protocol (BFCP) ground control is not supported on encrypted calls. To allow content to pass on SIP calls in a separate main video channel, you must disable encryption on the MCU or target endpoint.

  Is anyone aware of a fix for this?

  Thank you very much

  Trevyn

  I think that the objective of this implementation will be 4.5.  The date from which this version will be public is (interim) August, but the SW not EFT, so you may register and test if you wish.

  VR

  Patrick

• History of EX60 using redial missed call fails

  All,

  I have a scenario where I have a (home network) VCS-E <-> <->VCS - C <->EX60 EX60. We have MSDS, the two units registered, if I dial e164 of external or internal device call is successful, BUT if the external unit lack an inner calling and I try to use the history tab and call the internal unit, my call fails. Here's what the I see in the VCS-E. The VCS - C is below as well (I removed ome IP and information area)

  VCS-E

  2013 09-13 T 11: 20:26 - 05:00 tvcs: event = "Call rejected" Service = "SIP" Src - ip = 'external users IP' Src-port = "55448" CBC-alias-type = Src-alias "SIP" = "sip:[email protected] / * /" Dst-alias-type = "SIP" Dst-alias = "sip:[email protected] / * /Tag"Call-serial-number="5a259a0e-1c90-11e3-b878-0050568000d1" = "5a259bd0-1c90-11e3-a19e-0050568000d1" detail = 'Temporarily unavailable' Protocol = "TLS" response code "480" = Level = '1' elements UTCTime = "2013-09-13 16:20:26, 897" "

  "2013 09-13 T 11: 20:26 - 05:00 tvcs: event = 'search completed" reason = "Temporarily unavailable" Service = "SIP" type-aliases-Src = Src-alias "SIP" = "[email protected] / * /"Dst-alias-type = "SIP" Dst-alias ="sip:[email protected] / * /Tag"Call-serial-number="5a259a0e-1c90-11e3-b878-0050568000d1" = "5a259bd0-1c90-11e3-a19e-0050568000d1" detail = "" found: false searchtype:INVITE, ' Level = '1' elements UTCTime = "2013-09-13 16:20:26, 897" "

  2013 09-13 T 11: 20:06 - 05:00 tvcs: Event = Service "Call tried" = "SIP" Src - ip = "IP Address of external users" Src-port = "55448" CBC-alias-type = Src-alias "SIP" = "sip:[email protected] / * /" Dst-alias-type = Dst-alias "SIP" = "sip:[email protected] / * /Tag"Call-serial-number="5a259a0e-1c90-11e3-b878-0050568000d1" = "5a259bd0-1c90-11e3-a19e-0050568000d1" Protocol = "TLS" Auth = 'YES' level = '1' elements UTCTime = "2013-09-13 16:20:06, 456" "

  "2013 09-13 T 11: 20:06 - 05:00 tvcs: Event = 'attempt of search" Service = "SIP" type-aliases-Src = Src-alias "SIP" = "[email protected] / * /"Dst-alias-type = Dst-alias "SIP" ="sip:[email protected] / * /"Call-serial-number="5a259a0e-1c90-11e3-b878-0050568000d1" Tag = "5a259bd0-1c90-11e3-a19e-0050568000d1" details = "searchtype:INVITE" level = "1" elements UTCTime = "2013-09-13 16:20:06, 455'"

  VCS - C

  "2013 09-13 T 11: 21:29 - 05:00 tvcs: event ="Call rejected"Service ="H323"Src - ip = 'internal IP address' Src-port ="2776"CBC-alias-type ="H323"CBC-alias ="[email protected] / * /"CBC-alias-type ="E164"CBC-alias ="6666"Dst-alias-type ="H323"Dst-alias ='[email protected] / * /"Call-serial-number="855035a4-1c90-11e3-9ec1-0050568000c5" Tag = "7f42aade-1c90-11e3-865f-0050568000d1" Protocol = "TCP" code-response = "Temporarily unavailable" level = '1' elements UTCTime = "2013-09-13 16:21:29, 044" "

  "2013 09-13 T 11: 21:29 - 05:00 tvcs: Event = 'search completed" reason = "Temporarily unavailable" Service = "H323" CBC-alias-type = "H323" CBC-alias = "[email protected] / * /" Src-alias-type = "E164" CBC-alias = "6666" Dst-alias-type = "H323" Dst-alias ="[email protected] / * /Tag"Call-serial-number="855035a4-1c90-11e3-9ec1-0050568000c5" = "7f42aade-1c90-11e3-865f-0050568000d1" detail = "" found: false searchtype:Setup, ' Level = '1' elements UTCTime = "2013-09-13 16:21:29, 043" "

  "2013 09-13 T 11: 21:18 - 05:00 tvcs: event ="Attempt of search"Service ="H323"type-aliases-Src ="H323"CBC-alias ="[email protected] / * /"type-aliases-Src ="E164"CBC-alias ="6666"Dst-alias-type ="H323"Dst-alias ="[email protected] / * /"Call-serial-number="855035a4-1c90-11e3-9ec1-0050568000c5"Tag ="7f42aade-1c90-11e3-865f-0050568000d1"details ="searchtype:Setup"level ="1"= UTCTime elements"2013-09-13 16:21:18, 944""

  "2013 09-13 T 11: 21:18 - 05:00 tvcs: Event = Service"Call tried"="H323"Src - ip = 'internal IP address' Src-port ="2776"CBC-alias-type ="H323"CBC-alias ="[email protected] / * /"CBC-alias-type ="E164"CBC-alias ="6666"Dst-alias-type ="H323"Dst-alias ='[email protected] / * /"Call-serial-number="855035a4-1c90-11e3-9ec1-0050568000c5" Tag = "7f42aade-1c90-11e3-865f-0050568000d1" Protocol = "TCP" Auth = "YES" level = "1" = UTCTime elements "2013-09-13 16:21:18, 943" "

  "2013 09-13 T 11: 21:18 - 05:00 tvcs: event ="Research Service completed "="H323"CBC-alias-type ="H323"CBC-alias ="[email protected] / * /"Dst-alias-type ="H323"Dst-alias ="[email protected] / * /Tag "Call-serial-number="855035a4-1c90-11e3-9ec1-0050568000c5"="7f42aade-1c90-11e3-865f-0050568000d1"detail =" "found: true, searchtype:LRQ ' routed call = 'YES' level = '1' elements UTCTime =" 2013-09-13 16:21:18, 923""

  "2013 09-13 T 11: 21:18 - 05:00 tvcs: Event ="Attempt to search"Service ="H323"CBC-alias-type ="H323"CBC-alias ="[email protected] / * /"Dst-alias-type ="H323"Dst-alias ="[email protected] / * /"Call-serial-number="855035a4-1c90-11e3-9ec1-0050568000c5"Tag ="7f42aade-1c90-11e3-865f-0050568000d1"details ="searchtype:LRQ"level ="1"= UTCTime elements '2013-09-13 16:21:18, 875'"

  Because I'm fairly new troubleshooting these untis, I did not enter all other newspapers. It almost seems that I have problems with SIP to H323 calls. I continue to dig, but I was wondering if one of you could point me in the right direction.

  Thank you

  Greg

  A few additional remarks.

  * Please describe your deployment a little better (network and telepresene config... (also understood what Paulo asked).

  * have you bandwidth restrictions, by default the links, call policies or other things up?

  * Happens all the time or random / sometimes it works /...?

  * Looks like the return call was a sip call. Is the intention, which was calling returend origin, now

  What is the default call protocoll endpoint and progress areas correctly?

  * If you use the e164 numbers and only the vcs I would use either h323 as the default calling Protocol

  or see you get enum upward and running though a call can stay in the Protocol, all the time.

  * have you checked that all the areas are properly?

  * No ALG / Inspection or other layer3 h323/sip stuff active?

  * Search rules ok?

  * all used identifications?

  * registration of endpoints ok?

  * using a cluster? We noticed a bug that is still present in X7.2.2 (even if it is said that it's been fixed).

  If the call to the number e a164 hits VCS where endpoint is NOT registered, the call will fail.

  It could thus help us to see some screenshots of the inscriptions and the search history and details of the search.

  Please remember useful frequency responses and identify useful or correct answers.

• Questions call non-Traversal

  Hi all

  I implement some new devices of telepresence:
  Site has
  VCS-control (X8.1)
  VCS-Expressway (X8.1)
  MCU 5310 (4.4)
  C40 codec (7.0.2)

  Site B
  (7.0.2) codec C20
  The site has different subnet

  I have enabled H.323 & SIP on the VCS - C and tested several calls between Site A & b. whenever calls go out H.323, SIP, or SIP to H.323 from the C40 for the C20 or vice versa, that there is no problem. Both sites receive video and audio.
  Whenever I force the same protocol. That is to say, H.323 to SIP for SIP or H.323 and making the same calls, the site B (C20) is not video of the Site (C40). However the C40 receives both video & audio of the C20. What is interesting, is that if I do the same calls of the MCU, which is also in Site A, then there is no problem. So the problem is between the C40 and C20.
  Since I'm not physcally on sites because I do this execution remotely, I can count only on the stats I get from the GUI. I have attached a photo taken of the customer from the screen of the C20.

  Calls:
  Calls crossing
  C40 - C20 H.323 - SIP - call ok
  C20 - C40 H.323 - SIP - call ok
  C40 - C20 SIP - H.323 - call ok
  C20 - C40 SIP - H.323 - call ok
  MCU - C20 H.323 - H.323 - call ok

  Non-Traversal calls
  C40 - C20 H.323 - H.323 - no video received the C20
  C40 - C20 SIP - SIP - no video received the C20
  C20 - C40 H.323 - H.323 - no video received the C20
  C20 - C40 SIP - SIP - no video received the C20

  This suggests that there is only a problem on the C20 in non-traversal calls. I don't want to use licensed traveled between these 2 points of termination.

  Thank you for your help. Any input would be greatly appreciated.

  -Greg

  When you make an H323, SIP call, all media is 'mandated' by the VCS, so they would still work even if they cannot ping each other.  When you do a non-traversal call, media flows directly between the endpoints.

  Ping a C-Series, connect to the codec via SSH (same credentials as the Web browser), and then type systemtools network ping 10.1.1.1

  Where "10.1.1.1" is the IP address of the remote end point.

• SE connect - FMG - SIP confusion

  Hi all

  We try to universal voice configuration for our project Connect.

  The Cisco Call Recorder communicates with the server to connect, but we struggle to Connect and FMG to cooperate.

  On the same host as Connect (it had been on its own server, although the behavior was apparently the same), we installed the FMG.

  When you create a new provider to Test Dial-In Audio the process times out.

  The core00.log contains the following:

  [Startup output]

  #Start - date: 2012-06-22 09:12:52

  #Software: flash version gateway multimedia 2,0,1,15

  #Fields: COMMENT THREAD for the DATE TYPE LEVEL

  2012-06 - 22:09:12:52.654 Server INFO CORE 10096 reading configurations

  2012-06 - 22:09:12:52.654 ERROR CORE 10096 license key is not valid. activating Developer Edition limited.

  2012-06 - 22:09:12:52.654 DEBUG CORE 10096 log initialized to 1 level

  2012-06 - 22:09:12:52.655 DEBUG CORE 10096 loading Plugins.

  2012-06 - 22:09:12:52.669 DEBUG CORE 10096 FMG started.

  2012-06 - 22:09:12:52.669 WARNING CONTROL 9144 no remote HTTP host or the pass not specified in fmsmg.xml, exit discoverHttp

  2012-06 - 22:09:12:52.669 INFO Reinitializing server.

  2012 06 - 22:09:12:52.674 DEBUG SIP 9844 loading quality level for device config = Tandberg990MXP of C:\Breeze\8.2.0.1\Flash Media Gateway\conf\videoqualitylevels.xml

  2012 06 - 22:09:12:52.674 DEBUG SIP 9844 loading quality level for device config = TandbergEdge95MXP of C:\Breeze\8.2.0.1\Flash Media Gateway\conf\videoqualitylevels.xml

  2012 06 - 22:09:12:52.674 DEBUG SIP 9844 loading quality level for device config = GenericVtcDevice of C:\Breeze\8.2.0.1\Flash Media Gateway\conf\videoqualitylevels.xml

  2012-06 - 22:09:12:52.675 DEBUG SIP 9844 local Interface: [host Connect Server IP] will be used to dest ip: [host SIP server IP]

  2012-06 - 22:09:12:52.675 DEBUG SIP 9844 sip domain address not defined...

  2012-06 - 22:09:12:52.676 INFO FMS detected IPv6 protocol stack!

  2012-06-config INFO FMS 22::09:12:52.677 < NetworkingIPv6 enable = false >

  2012-06 - 22:09:12:52.677 INFO FMS running in IPv4 protocol stack mode!

  2012-06 - 22:09:12:52.677 host INFO: connect1 IPv4: [host server IP Connect]

  2012-06 - 22:09:12:52.677 Server INFO from...

  2012-06 - 22:09:12:53.682 INFO server started (C:\Breeze\8.2.0.1\Flash Media Gateway\conf\http.xml).

  2012-06 - 22:09:12:54.925 DEBUG SIP 9844 error: can't resolve address (reserved) no error: 11001

  2012-06 - 22:09:12:54.925 DEBUG SIP 9844 local Interface: [host Connect Server IP] will be used for ip dest:

  2012-06 - 22:09:12:54.925 DEBUG SIP 9844 sip domain address not defined...

  2012-06 - 22:09:12:54.982 DEBUG SIP 9844 departure to the ping profile: fmgInternalUse

  2012-06 - 22:09:12:54.982 DEBUG SIP 9844 departure to the ping profile: sipGateway

  2012-06 - 22:09:12:57.232 DEBUG SIP 9844 error: can't resolve address (reserved) no error: 11001

  2012-06 - 22:09:12:57.232 parameter DEBUG SIP 9844 outgoing call Details: Addr: reserved, port: 5060

  2012-06 - 22:09:12:59.482 DEBUG SIP 9844 error: can't resolve address (reserved) no error: 11001

  2012-06 - 22:09:12:59.482 DEBUG SIP 9844 local Interface: [host Connect Server IP] will be used to dest ip: [host server IP Connect]

  2012-06 - 22:09:12:59.485 DEBUG SIP 9844 Registering State for profile: fmgInternalUse

  2012-06 - 22:09:12:59.492 ERROR SIP 9844 registration failed for profile - fmgInternalUse!

  [output end]

  Any help to solve this problem would be greatly appreciated.

  Thanks in advance,

  GW.

  The issue seems to have been incorrectly set for the FMG Server fully qualified domain name.

• Access telephone IP SRP547W of line through Vlan VoIP Wifi No.

  The SRP547W supported the creation of VLAN wireless voice and data.

  Can I set a phone IP from Wifi to connect to the SRP547W voice Wifi Vlan and have the RPS to associate with line 1 (instead of a standard telephone connected to port FXO 1 line - without additional hardware)?

  Hi Gary,.

  Sorry, this is not possible. There are no AU SIP port FXO on the SRP540, just an internal mechanism to connect ports FXS internal for incoming and outgoing calls.

  Kind regards

  Andy