SIP trunk CUBE with Callcentric - incoming unanswered call

Similar Questions

• CUCM 8.6 > SIP Trunk > CUBE > SIP Trunk - ringtone / connected display Party

  Hi community support

  When we get calls from an IP phone on a gateway H323 ISDN globalize us the number of models of translation so the display on the phone is the E164 number, for example + 442071234567. We want this even when the call is ringing / connected for us to use the view command no service additional h225 - prevent cid-update on the H323 gateway otherwise the alert screen is updated with the number of the corresponding when Dial peer ringtone and connected.

  When we get calls from an IP on a CUBE phone to SIP the number is still globalized at E164 in models of translation however once we hear ringback the display on the phone IP poster "private." Can someone please advise if there is an equivalent in SIP to the command no service additional h225 - prevent cid--update.

  These are the CUBE CCSIP debugging messages:

  000267: 10:38:35.506 Oct 2 UTC: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
  Received:
  GUEST sip:[email protected]/ * /: SIP-5060/2.0
  Via: SIP/2.0/TCP 172.20.44.104:5060; branch = z9hG4bK3e5ecc3bcebb92
  From: "Anonymous" <> [email protected] / * />; tag=25280178~17337b8f-a50d-45a4-8f3a-f81dd5cc611b-41234378
  To: <> [email protected]/ * / >
  Date: Friday, October 2, 2015 09:38:35 GMT
  Call ID: [email protected] / * /
  Supported: timer, resource-priority, replaces
  Min - SE: 1800
  User-Agent: Cisco - CUCM8.6
  Allow: PROMPT, OPTIONS, INFO, BYE, ACK, CANCEL, PRACK, UPDATE, VIEW, SUBSCRIBE, NOTIFY
  CSeq: INVITE 101
  Expires: 180
  Allow-events: presence, kpml
  Support: X-cisco-srtp-fallback,X-cisco-original-called
  Cisco-Guid: 1488217344-0000065536-0000054240-1747719340
  Session time-out: 1800
  P a claimed identity: 'CR Test MAN - 3022852' <> [email protected]/ * / >
  Privacy: id
  Remote-Party-ID: 'CR Test MAN - 3022852' <> [email protected]/ * / >; left = call; screen = yes; intimacy = uri
  Contact:
  Max-Forwards: 70
  Content-Type: application/sdp
  Content-Length: 219

  v = 0
  o = CiscoSystemsCCM-SIP 25280178 1 IN IP4 172.20.44.104
  s = call SIP
  c = IN IP4 172.20.255.249
  t = 0 0
  m = 30088 audio RTP/AVP 8 101
  a = rtpmap:8 PCMA/8000
  a = ptime:20
  a rtpmap:101 telephone-event/8000 =
  a = fmtp:101 0-15

  000268: 10:38:35.522 Oct 2 UTC: / / 698 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg:
  Envoy:
  GUEST sip:[email protected]/ * /: SIP-5060/2.0
  Via: SIP/2.0/UDP 172.23.255.99:5060; branch = z9hG4bK54613BD
  Remote-Party-ID: 'CR Test MAN - 3022852' <> [email protected]/ * / >; left = call; screen = yes; intimacy = full
  From: "anonymous" <> [email protected]/ * / >; tag = 2E995B0-BF7
  To: <> [email protected]/ * / >
  Date: Friday, October 2, 2015 09:38:35 GMT
  Call ID: [email protected] / * /
  Supported: 100rel, timer, resource-priority, replaces, sdp-anat
  Min - SE: 1800
  Cisco-Guid: 1488217344-0000065536-0000054240-1747719340
  User-Agent: Cisco-SIPGateway/IOS-15.4.3.M3
  Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, VIEW, SUBSCRIBE, NOTIFY, INFO, REGISTER
  CSeq: INVITE 101
  Time stamp: 1443778715
  Contact: <> [email protected]/ * /: 5060 >
  Expires: 180
  Allow-events: telephone-event
  Max-Forwards: 69
  Session time-out: 1800
  Content-Type: application/sdp
  Content-Disposition: session; handling = required
  Content-Length: 250

  v = 0
  o = CiscoSystemsSIP-GW-UserAgent 1367 2609 IN IP4 172.23.255.99
  s = call SIP
  c = IN IP4 172.23.255.99
  t = 0 0
  m = audio RTP/AVP 8 101 17778
  c = IN IP4 172.23.255.99
  a = rtpmap:8 PCMA/8000
  a rtpmap:101 telephone-event/8000 =
  a = fmtp:101 0-15
  a = ptime:20

  000269: 10:38:35.522 Oct 2 UTC: / / 697 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg:
  Envoy:
  SIP/2.0 100 trying
  Via: SIP/2.0/TCP 172.20.44.104:5060; branch = z9hG4bK3e5ecc3bcebb92
  From: "Anonymous" <> [email protected] / * />; tag=25280178~17337b8f-a50d-45a4-8f3a-f81dd5cc611b-41234378
  To: <> [email protected]/ * / >
  Date: Friday, October 2, 2015 09:38:35 GMT
  Call ID: [email protected] / * /
  CSeq: INVITE 101
  Allow-events: telephone-event
  Server: Cisco-SIPGateway/IOS-15.4.3.M3
  Content-Length: 0

  000270: 10:38:35.534 Oct 2 UTC: / / 698 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg:
  Received:
  SIP/2.0 100 trying
  Via: SIP/2.0/UDP 172.23.255.99:5060; branch = z9hG4bK54613BD
  From: "anonymous" <> [email protected]/ * / >; tag = 2E995B0-BF7
  To: <> [email protected]/ * / >; tag = gK0291ecd6
  Call ID: [email protected] / * /
  CSeq: INVITE 101
  Time stamp: 1443778715
  Content-Length: 0

  000275: 10:38:38.422 Oct 2 UTC: / / 698 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg:
  Received:
  SIP/2.0 183 during the Session
  Via: SIP/2.0/UDP 172.23.255.99:5060; branch = z9hG4bK54613BD
  From: "anonymous" <> [email protected]/ * / >; tag = 2E995B0-BF7
  To: <> [email protected]/ * / >; tag = gK0291ecd6
  Call ID: [email protected] / * /
  CSeq: INVITE 101
  Contact: <> [email protected]/ * /: 5060 >
  Allow: INVITE, ACK, CANCEL, BYE, REGISTER, REFER, INFO, SUBSCRIBE, NOTIFY, PRACK, UPDATE, OPTIONS, MESSAGE, PUBLISH
  Need: 100rel
  RSeq: 26192
  Content-Length: 235
  Content-Disposition: session; treatment required =
  Content-Type: application/sdp

  v = 0
  o = 10406 9594 Sonus_UAC IN IP4 192.168.200.29
  s = SIP multimedia tools
  c = IN IP4 192.168.200.4
  t = 0 0
  m = 21708 audio RTP/AVP 8 101
  a = rtpmap:8 PCMA/8000
  a rtpmap:101 telephone-event/8000 =
  a = fmtp:101 0-15
  a = sendrecv
  a = ptime:20

  000276: Oct 2 UTC 10:38:38.426: / / 698 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg:
  Envoy:
  SIP PRACK:[email protected]/ * /: SIP-5060/2.0
  Via: SIP/2.0/UDP 172.23.255.99:5060; branch = z9hG4bK54824BE
  From: "anonymous" <> [email protected]/ * / >; tag = 2E995B0-BF7
  To: <> [email protected]/ * / >; tag = gK0291ecd6
  Date: Friday, October 2, 2015 09:38:35 GMT
  Call ID: [email protected] / * /
  CSeq: 102 PRACK
  Grid: 26192 101 INVITE
  Allow-events: telephone-event
  Max-Forwards: 70
  Content-Length: 0

  000277: 10:38:38.430 Oct 2 UTC: / / 697 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg:
  Envoy:
  SIP/2.0 183 during the Session
  Via: SIP/2.0/TCP 172.20.44.104:5060; branch = z9hG4bK3e5ecc3bcebb92
  From: "Anonymous" <> [email protected] / * />; tag=25280178~17337b8f-a50d-45a4-8f3a-f81dd5cc611b-41234378
  To: <> [email protected]/ * / >; tag = 2E9A10C-CB9
  Date: Friday, October 2, 2015 09:38:35 GMT
  Call ID: [email protected] / * /
  CSeq: INVITE 101
  Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, VIEW, SUBSCRIBE, NOTIFY, INFO, REGISTER
  Allow-events: telephone-event
  Remote-Party-ID: <> [email protected]/ * / >; left = called; screen = yes; intimacy = full
  Contact: <> [email protected]/ * /: 5060; transport = tcp >
  Supported: sdp-anat
  Server: Cisco-SIPGateway/IOS-15.4.3.M3
  Content-Type: application/sdp
  Content-Disposition: session; handling = required
  Content-Length: 250

  v = 0
  o = CiscoSystemsSIP-GW-UserAgent 7824 1232 IN IP4 172.23.255.99
  s = call SIP
  c = IN IP4 172.23.255.99
  t = 0 0
  m = audio RTP/AVP 8 101 17776
  c = IN IP4 172.23.255.99
  a = rtpmap:8 PCMA/8000
  a rtpmap:101 telephone-event/8000 =
  a = fmtp:101 0-15
  a = ptime:20

  000278: Oct 2 UTC 10:38:38.442: / / 698 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg:
  Received:
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 172.23.255.99:5060; branch = z9hG4bK54824BE
  From: "anonymous" <> [email protected]/ * / >; tag = 2E995B0-BF7
  To: <> [email protected]/ * / >; tag = gK0291ecd6
  Call ID: [email protected] / * /
  CSeq: 102 PRACK
  Content-Length: 0

  000279: Oct 2 UTC 10:38:38.922: / / 698 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg:
  Received:
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 172.23.255.99:5060; branch = z9hG4bK54613BD
  From: "anonymous" <> [email protected]/ * / >; tag = 2E995B0-BF7
  To: <> [email protected]/ * / >; tag = gK0291ecd6
  Call ID: [email protected] / * /
  CSeq: INVITE 101
  Accept: application/sdp, application/isup, application/dtmf, dtmf-relay application, multipart/mixed
  Contact: <> [email protected]/ * /: 5060 >
  Allow: INVITE, ACK, CANCEL, BYE, REGISTER, REFER, INFO, SUBSCRIBE, NOTIFY, PRACK, UPDATE, OPTIONS, MESSAGE, PUBLISH
  Require: timer
  Supported: timer
  Session time-out: 1800; recycling = uac
  Content-Length: 235
  Content-Disposition: session; treatment required =
  Content-Type: application/sdp

  v = 0
  o = 10406 9594 Sonus_UAC IN IP4 192.168.200.29
  s = SIP multimedia tools
  c = IN IP4 192.168.200.4
  t = 0 0
  m = 21708 audio RTP/AVP 8 101
  a = rtpmap:8 PCMA/8000
  a rtpmap:101 telephone-event/8000 =
  a = fmtp:101 0-15
  a = sendrecv
  a = ptime:20

  000280: Oct 2 UTC 10:38:38.926: / / 697 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg:
  Envoy:
  SIP/2.0 200 OK
  Via: SIP/2.0/TCP 172.20.44.104:5060; branch = z9hG4bK3e5ecc3bcebb92
  From: "Anonymous" <> [email protected] / * />; tag=25280178~17337b8f-a50d-45a4-8f3a-f81dd5cc611b-41234378
  To: <> [email protected]/ * / >; tag = 2E9A10C-CB9
  Date: Friday, October 2, 2015 09:38:35 GMT
  Call ID: [email protected] / * /
  CSeq: INVITE 101
  Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, VIEW, SUBSCRIBE, NOTIFY, INFO, REGISTER
  Allow-events: telephone-event
  Remote-Party-ID: <> [email protected]/ * / >; left = called; screen = no; intimacy = off
  Contact: <> [email protected]/ * /: 5060; transport = tcp >
  Supported: replaces
  Supported: sdp-anat
  Server: Cisco-SIPGateway/IOS-15.4.3.M3
  Session time-out: 1800; recycling = uac
  Require: timer
  Supported: timer
  Content-Type: application/sdp
  Content-Disposition: session; handling = required
  Content-Length: 250

  v = 0
  o = CiscoSystemsSIP-GW-UserAgent 7824 1232 IN IP4 172.23.255.99
  s = call SIP
  c = IN IP4 172.23.255.99
  t = 0 0
  m = audio RTP/AVP 8 101 17776
  c = IN IP4 172.23.255.99
  a = rtpmap:8 PCMA/8000
  a rtpmap:101 telephone-event/8000 =
  a = fmtp:101 0-15
  a = ptime:20

  000281: 10:38:38.926 Oct 2 UTC: / / 698 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg:
  Envoy:
  SIP ACK:[email protected]/ * /: SIP-5060/2.0
  Via: SIP/2.0/UDP 172.23.255.99:5060; branch = z9hG4bK549768
  From: "anonymous" <> [email protected]/ * / >; tag = 2E995B0-BF7
  To: <> [email protected]/ * / >; tag = gK0291ecd6
  Date: Friday, October 2, 2015 09:38:35 GMT
  Call ID: [email protected] / * /
  Max-Forwards: 70
  CSeq: 101 ACK
  Allow-events: telephone-event
  Content-Length: 0

  000282: 10:38:38.934 Oct 2 UTC: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
  Received:
  SIP ACK:[email protected]/ * /: 5060; transport = tcp SIP/2.0
  Via: SIP/2.0/TCP 172.20.44.104:5060; branch = z9hG4bK3e5ece183b5108
  From: "Anonymous" <> [email protected] / * />; tag=25280178~17337b8f-a50d-45a4-8f3a-f81dd5cc611b-41234378
  To: <> [email protected]/ * / >; tag = 2E9A10C-CB9
  Date: Friday, October 2, 2015 09:38:35 GMT
  Call ID: [email protected] / * /
  Max-Forwards: 70
  CSeq: 101 ACK
  Allow-events: presence, kpml
  Privacy: id
  Content-Length: 0

  000283: 10:38:45.426 Oct 2 UTC: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
  Received:
  SIP BYE:[email protected]/ * /: 5060; transport = tcp SIP/2.0
  Via: SIP/2.0/TCP 172.20.44.104:5060; branch = z9hG4bK3e5ed866258a9
  From: "Anonymous" <> [email protected] / * />; tag=25280178~17337b8f-a50d-45a4-8f3a-f81dd5cc611b-41234378
  To: <> [email protected]/ * / >; tag = 2E9A10C-CB9
  Date: Friday, October 2, 2015 09:38:35 GMT
  Call ID: [email protected] / * /
  User-Agent: Cisco - CUCM8.6
  Max-Forwards: 70
  Privacy: id
  P a claimed identity: 'CR Test MAN - 3022852' <> [email protected]/ * / >
  CSeq: 102 BYE
  Reason: Q.850; cause = 16
  Content-Length: 0

  000284: 10:38:45.430 Oct 2 UTC: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
  Envoy:
  SIP/2.0 200 OK
  Via: SIP/2.0/TCP 172.20.44.104:5060; branch = z9hG4bK3e5ed866258a9
  From: "Anonymous" <> [email protected] / * />; tag=25280178~17337b8f-a50d-45a4-8f3a-f81dd5cc611b-41234378
  To: <> [email protected]/ * / >; tag = 2E9A10C-CB9
  Date: Friday, October 2, 2015 09:38:45 GMT
  Call ID: [email protected] / * /
  Server: Cisco-SIPGateway/IOS-15.4.3.M3
  CSeq: 102 BYE
  Reason: Q.850; cause = 16
  P-RTP-Stat: PS = 351, OS = 56160, PR = 342, OR = 54720, PL = 0, JI = 0, THE = 0, 6 =
  Content-Length: 0

  000285: 10:38:45.434 Oct 2 UTC: / / 698 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg:
  Envoy:
  SIP BYE:[email protected]/ * /: SIP-5060/2.0
  Via: SIP/2.0/UDP 172.23.255.99:5060; branch = z9hG4bK54A13C3
  From: "anonymous" <> [email protected]/ * / >; tag = 2E995B0-BF7
  To: <> [email protected]/ * / >; tag = gK0291ecd6
  Date: Friday, October 2, 2015 09:38:35 GMT
  Call ID: [email protected] / * /
  User-Agent: Cisco-SIPGateway/IOS-15.4.3.M3
  Max-Forwards: 70
  Time stamp: 1443778725
  CSeq: 103 BYE
  Reason: Q.850; cause = 16
  P-RTP-Stat: PS = 342, OS = 54720, PR = 480 OR = 76800, PL = 0, JI = 0, THE = 0, 6 =
  Content-Length: 0

  000286: Oct 2 UTC 10:38:45.454: / / 698 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg:
  Received:
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 172.23.255.99:5060; branch = z9hG4bK54A13C3
  From: "anonymous" <> [email protected]/ * / >; tag = 2E995B0-BF7
  To: <> [email protected]/ * / >; tag = gK0291ecd6
  Call ID: [email protected] / * /
  CSeq: 103 BYE
  Content-Length: 0

  Carl Ratcliffe

  Preston-Lancashire-England

  Carl,

  Understand why you are getting the private sector in State of ringtone will lead us to solve this problem much more efficiently.

  If we look at the 183 Session progress sent to CUCM...

  +++ See here this part called privay = full (meaning private)

  Envoy:
  SIP/2.0 183 during the Session
  Via: SIP/2.0/TCP 172.20.44.104:5060; branch = z9hG4bK3e5ecc3bcebb92
  -
  Remote-Party-ID: <> [email protected]/ * / >; left = called; screen = yes; intimacy = full

  Now that we know this, I would take a different approach to solve this rather than disable remote-party id in total...

  Here is my proposed solution...

  SIP-class voice profiles 3
  answer 183 sip remote-Party-ID header change "" "<>[email protected]"/ * / > ""

  Then you must apply the sip profile at the foot of basketing od the call... IE cucm cubed

  Dial-peer telephony voip xxx
  profiles of sip voice-class 3

  Now, when the appeal is as a ringtone, the full display will be + 44... What was initially called number.

• Incoming SIP - SP CUBE is not of translations

  Perplexed as to why the incoming calls from SIP service provider do not correspond to the translation in CUBE

  I have a number presented on the incoming CUBE SIP trunk and need to get rid of the figures for the last 3 numbers to present to the CUCM.  The test voice translation works, but it seems that the incoming number provided by the supplier is not hit or corresponding to the translation rule.

  Incoming dial peer config:

  Dial-peer voice voip 60
  Description incoming PSTN (elite) to the CUBE
  translation-profile entering EliteSIP-DDI-numbers-inbound
  session protocol sipv2
  incoming called number 44239...
  codec voice-class 1
  DTMF-relay rtp - nte sip-kpml
  No vad

  Profile and set the configuration of translation

  voice translation rule 44239
  rule 1 / ^ 442392006.
  rule 2 / ^ \+442392006/ / /.
  !
  !
  voice translation-profile EliteSIP-DDI-numbers-inbound
  definition of 44239 called

  The result of the translation:

  Matched with rule 2
  Original number: + 442392006339 translated number: 339
  Number of origin type: no number translation type: no
  Original number plan: no number plan translated: no

  BE6000S #test voice translation rule 44239 442392006339
  Matched with rule 1
  Original number: 442392006339 translated number: 339
  Number of origin type: no number translation type: no
  Original number plan: no number plan translated: no

  The translation of debugging output:

  Voice translation of BE6000S #debug
  VoIP translation rule debugging is enabled
  BE6000S #.
  SIP: Attempt to analyze the attribute not supported at the level of the media
  SIP: Attempt to analyze the attribute not supported at the level of the media
  065139: June 7 23:35:29.157: //-1/5A562434A112/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack = 0x3F5552E8; Count = 1
  065140: June 7 23:35:29.157: //-1/5A562434A112/RXRULE/regxrule_stack_pop_callinfo_internal: infonum = 0x421F2934
  065141: June 7 23:35:29.161: //-1/5A562434A112/RXRULE/regxrule_stack_push_RegXruleNumInfo_internal: stack = 0x3F5552E8; Count = 1
  065142: June 7 23:35:29.161: //-1/5A562434A112/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack = 0x3F5552E8; Count = 1
  065143: June 7 23:35:29.161: //-1/5A562434A112/RXRULE/regxrule_stack_pop_callinfo_internal: infonum = 0x421F2934
  065144: June 7 23:35:29.161: //-1/5A562434A112/RXRULE/regxrule_stack_push_RegXruleNumInfo_internal: stack = 0x3F5552E8; Count = 1
  065145: June 7 23:35:29.165: //-1/xxxxxxxxxxxx/RXRULE/sed_subst: no match! number = matchPattern = id; [; ] * replacePattern$ id =
  065146: June 7 23:35:32.157: //-1/5A562434A112/RXRULE/regxrule_stack_pop_callinfo_internal: infonum = 0x0
  065147: June 7 23:35:32.169: //-1/5A562434A112/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack = 0x3F5552E8; Count = 1
  065148: June 7 23:35:32.169: //-1/5A562434A112/RXRULE/regxrule_stack_pop_callinfo_internal: infonum = 0x421F2934

  Debug messages ccsip just to make sure the call come and the DNIS format (btw - which bit of the track to show the DNIS?)

  BE6000S #debug ccsip messages
  Call SIP tracing messages is enabled
  BE6000S #.
  065149: June 7 23:38:16.925: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
  Received:
  GUEST sip:[email protected]/ * /: SIP-5060/2.0
  Record-Route:
  Via: SIP/2.0/UDP 217.68.246.241:5060; branch = z9hG4bKe4be.24390fd700572c75f3247fa6444e9fcc.0
  Max-Forwards: 16
  To: <> [email protected]/ * /: 5060 >
  From: <> [email protected]/ * / >; tag = as6b74b830
  Call ID: [email protected]/ * /: 5050
  Contact: <> [email protected]/ * /: 5060 >
  CSeq: INVITE 102
  User-Agent: Elite hosted voice
  Date: Tuesday, June 7, 2016 23:38:14 GMT
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  Supported: replaces, timer
  X voipnow-did: + 442392006339
  X voipnow-extension: 0071 * 001
  X voipnow pbx: 3a5b131e3e
  X voipnow-infrastructureid: 92f21508
  X voipnow-did: + 442392006339
  Content-Type: application/sdp
  Content-Length: 520

  Ideas?

  Dear MEP,

  I think that if you add + to incoming called number, it should solve the problem as provider sends with +.

  Incoming called number + 44239...

  Also run dialpeer voip debug to see dial-peers are put in correspondence on incoming direction of ITSP.thanks

• CUCM v8.5 with 3 SIP Trunks to the Lync Server - Route algorithm of Distribution for the Group

  I CUCM connected to three different Lync server via 3 different SIP trunks.

  RG is composed of the following elements:

  LYNC SIP TRUNK 1 (1.1.1.1)

  LYNC SIP TRUNK 2 (2.2.2.2)

  LYNC SIP TRUNK 3 (3.3.3.3)

  The route group was built with "Top Down" the algorithm of distribution. The first SIP trunk knows congestion and some calls are never routed to secondary and tertiary SIP trunks.

  Based on all the forum posts I've seen - it seems that I have to configure the algorithm of group distribution of ranges as 'circular '.

  If I change the algorithm group of "Circular" lines - can I expect the following results:

  1. first call will go through LYNC SIP TRUNK 1

  2. second call will cross LYNC SIP TRUNK 2

  3. third call will cross LYNC SIP TRUNK 3

  When I change the algorithm of distribution of the route to the 'circular' group and click 'SAVE', I am invited on "RESET".  This service assigns to the existing calls through SIP Trunks?

  TIA,

  Amir

  Hello

  the circular algorithm will work the way you mentioned, but I suggest to try and make sure that is not perform integration

  to reset the SIP trunk actually active calls should not be made because the gose media directly between endpoint unles syou use something like trust rely wher evous enforce calls to go to the SIP

  Therefore, in most cases is should be fine just try configuration at the time of the calls will face service disruption

  hope this helps

• Problem with a SIP Trunk between VCS and CUCM

  Hello world

  I created a SIP Trunk between control VCS (X7.1) and a CUCM 8.6.

  I followed this Cisco_VCS_Cisco_Unified_Communications_Manager_Deployment_Guide_CUCM_6-1_7_8_and_X7 - 0.pdf deployment guide.

  The trunk is in place, but I have the following problem: I can ring since an EX90 recorded on the VCS for a registered on the CUCM 9951 but when I take the call, I have a busy signals (audio and video). When I on the EX90 9951 ring, I have the following message on VC: can not reach the contact.

  I have a 503 service unavailable status on the call history and "call rejected" on the event log.

  My transformation is OK and the call is sent to my neighbor CUCM zone

  You have an idea to solve this problem?

  Thanks for your help

  Best regards

  Bruno Z.

  Hello

  Yes, it's true! It seems at first glance a problem with codec negotiation when I see the reason 47 code.

  in any case set the region to G.711 trunk phone in case if you have different device pool and the trunk region and ip-phones.

  make G.711 end to end and test again.

  Thank you

  Alok

• Caller ID of VCS Jabber Video on SIP Trunk on the phone

  Hello, Netpros.

  I am trying to find a way to show a caller E.164 ID when a call to a video client Jabber a UCM or a PSTN phone.

  VCSC/E 7.1; 13.2 TMS, configured TMSPE.

  We have a VCSC > SIP Trunk > UCM (8,6) > H.323 GW w / PRI voice configured for the composition of and the PSTN destination.  We do this, so the Jabber client video can be used as a softphone, and so we can have transparent FindMe to voice and SNR to the video in either sense.  In TMS, we have a model of implementation configured with an address of the video, ILA and unit address.  He pulls ad office phone number for video of Jabber users.

  From what I read here:

  http://www.cisco.com/en/US/docs/telepresence/infrastructure/vcs/config_guide/Cisco_VCS_FindMe_Express_Deployment_Guide_X7-1.pdf, we should be able to see the caller ID when you call send out an ISDN GW.

  «This section describes how to use FindMe with calls are routed through an ISDN gateway (for example, when you call a mobile phone, or any other destination accessible ISDN).» If VCS has Caller ID (Applications > FindMe > Configuration) defined to use the ID FindMe, the identification of the appellant presented will be the user E.164 phone number. »

  UCM SIP config is vanilla, by

  ttp: / /www.tandberg.com/collateral/documentation/Deployment_Guides/Cisco_VCS_Ci...

  However, when a video call Jabber is directed to a University Complutense of MADRID registered phone, we get "unknown caller".  When to call us directly to the PSTN, we see that the BTN from the PRI.

  What is strange, is that if we call another client of Jabber of VCS (with all the same problems), directly into a phone UCM via Internet on URI (for example [email protected] / * /) through the VCSE, we see the caller ID name.

  Thank you, all!

  Joshua

  Hi Joshua,.

  At present the VCS will only send caller E.164 id to a GW of H.323 ISDN, which is registered for VCS.

  There a feature request in the order book to define if E.164 Caller ID or name/Video address on a per-zone basis. This will also be sove the question when the call is to go on a SIP connection, as in this case, a remote ISDN GW.

  Thank you

  Guy

• Configure (fxs) analog phones with Sip trunk

  Hi all

  IS - this configuration FXS possiple to knit on SIP TRUNK? I HAVE 16 FXS ports and voip gateway 2921 cisco.

  Configuring fxs to work with her?

  any help thanks.

  Is there a PBX at stake here, i.e. CUCM or CME?

  In both cases, you have set good dial-peers to point to the FXS ports to match the destination of the SIP trunk, which is pretty simple.

• Source IP address of the originating call to sip trunk

  I'm unable to set address ip source from trunk sip call. I read somewhere that the source ip address of the call must be the ip address that call device registered to. I also checked "run in all active nodes such as suggested by cisco. We have 3 Sub cluster and 1 pub however still call orgianate to a particular Sub.

  I also undergoes several changes i - e DP and list route but think not worked for me. Can someone help to sortout this problem?

  Let's say if your ip phone registered in sub - and "run on all nodes" enabled, then the source ip address is always sup - A s ip address regardless of the device pool config on the sip trunk.

  Could please explain you the device pool config in phones IP & SIP trunk?

  you have enabled the "run on all nodes" for a list of course also?

  Suresh

  Please note all useful messages.

• is it possible, end point of video recorded in CUCM call standalon SX10 via SIP Trunk?

  Dear all,

  Need help :)

  1.

  is it possible, end point of video recorded in CUCM call standalon SX10 via SIP Trunk?

  my customers will not buy TP - lic :)

  Topology:

  8900 (jabber) - CUCM-(SIP) - SX10

  2.

  Customer, have MCU machine.

  If not possible. is it possible, meeting point of MCU use?

  Topology:

  8900 (jabber) - CUCM - MCU - SX10

  1. Yes, but it is not that simple and straightforward that the SX10 can be reached by calling is the public IP address and CUCM does not support the IP address numbering is not a SIP address format...

  2. Yes, using the MCU would be much simpler, easier and, in my opinion, a better solution to be implemented. Here is an example of how a stand-alone SX10 could dial the IP address of the MCU standard automatic.

  You can also assign a SIP URI address to the meeting point which the SX10 can connect to.

  /Jens

  Please note the answers and score the questions as "answered" as appropriate.

• Sip Trunk design question

  Hello

  I have a requirement to pass an h323 to SIP environment environment. I'm looking for good practices, especially around security. I have 2 servers CUCM (8.5) in cities separated for redundancy. I have also 2 voice gateways which, at the present time, h323 to the PSTN, are each located in different cities.

  My requirements are:

  1. create a sip trunk instead of the supplier of the use of PRI.

  2 If the Wan link fails on a gateway provider, router replacing in the other location should be able to receive installation messages and if a user connects via extension mobility, should be able to answer the call.

  Is there a simplified design docos on for this? I hesitate to create a SIP trunk directly to the supplier for safety, thus thinking to end the call on the routers of voice with the CUBE. I am sure that it is managed from the factory and would appreciate comments.

  See you soon!

  Pieter

  Simple answer use ALWAYS the CUBE.  With IOS 15.1 T and more you have security against fraud free of charge that you can use to restrict which can address IP contacted the CUBE, that's all you need.

  HTH,

  Chris

• DTMF in SIP Trunk problem

  Hello

  I have a problem in case of detection of the DTMF

  We have a SIP of the ITSP Trunk and everything is ok except DTMF.

  The sip trunk is between ITSP and router 3945

  ITSP <->3945 <->CUCM 10.5

  I have try all method such as rtp - nte, h245 alphanumeric and h245-signal, info-sip, sip-kpml,... in line-peer to ITSPs

  ITSP say that he send with standard RFC2833 dtmf (it equals to rtp-net, I know), but when I am 'debug ccsip message', the results show the dtmf with rfc2833 sends us

  16 August 13:52:28.991: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
  Received:
  INVITE sip: [email protected] / * /: 5060. user = phone SIP/2.0
  Via: SIP/2.0/UDP 10.105.40.34:5060; branch = z9hG4bKejhh8jixkobhnb7ykvi87vuj8
  Call ID: [email protected]/ * /.
  From: sip: [email protected] / * />; tag = c3cx3vcw-CC-40
  To: sip: [email protected] / * /; user = phone >
  CSeq: 1 INVITE
  Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTRY, PRACK, INFO SUBSCRIBE, NOTIFY, updates, MESSAGE, see
  Max-Forwards: 69
  Supported: 100rel, timer
  User-Agent: Huawei SoftX3000 V300R010
  Session time-out: 300
  Min - SE: 90
  Contact: sip: [email protected] / * /: 5060; user = phone >
  Content-Length: 374
  Content-Type: application/sdp
  v = 0
  o = HuaweiSoftX3000 11042757 11042757 IN IP4 10.105.40.34
  s = call Sip
  c = IN IP4 10.105.40.34
  t = 0 0
  m = audio 27762 RTP / AVP 8 0 18 4 2 98 99 102
  a = rtpmap:8 PCMA/8000
  a = rtpmap:0 PCMU/8000
  a G729/8000 rtpmap:18 =
  a = rtpmap:4 G723/8000
  a = rtpmap:2 G726-32/8000
  a = rtpmap:98 G726-40/8000
  a = rtpmap:99 G726-32/8000
  a = rtpmap:102 G726-24/8000
  a = ptime:20
  a = fmtp:18 annex b = No.
  It is a message to guest (with sdp) of ITSP
  As you can see the line with red color must have a code with number of 101 but rather a code with number of 18
  In my "ccsip media debug' output show that the method of negotiation between me and itsp 'incoming voice. '
  It's my router config:
  voip phone service
  No IP trust to authenticate
  allow connections h323 to SIP
  allow connections sip h323
  allow sip to sip connections
  SIP
  interface FastEthernet0/0/1 source control binding
  bind media source interface FastEthernet0/0/1
  min - to 300 session expires-300
  !

  Dial-peer voice 2 voip---> router CUCM and vice versa
  translation-profile outgoing toos
  destination-model 42584...
  session protocol sipv2
  session target ipv4:10.20.30.70
  Codec g711ulaw
  DTMF-relay rtp - nte
  !
  VoIP voice 10 Dial - peer---> router for ITSP and vice versa
  destination-model. T
  session protocol sipv2
  session target ipv4:10.105.40.34
  incoming called-number. T
  DTMF-relay rtp - nte
  Codec g711ulaw
  I have configured cucm with a sip section to my favorite router with active PSG and RFC2833
  BUT HE DIDN'T THERE WAS NO DETECTION OF DTMF IN INCOMING CALLS AND OUTGOING
  I even test dial-position 10 without any method of dtmf-relay because I want to configure it with the incoming voice method, but it does not work
  I change the codec but does not solve the problem
  There is an interesting point and that is, if use Elastix 3945 of the router and configure dtmf among elastix and itsp as "inbound" method and method dtmf among elastix and cucm as 'rfc2833' everything is OK (ITSP<--Inbound--> Elastix <--rfc2833-->CUCM)
  Please give me a solution to solve the problem between Cisco 3945 and ITSP
  Concerning

  It would be more accurate to say that the problem has been bypassed by using a transcoding resource. The correct resolution here would be to open a ticket to trouble with the ITSP. I agree that the PROMPT message you show doesn't include advertising RFC2833 in the SDP offer. It is a problem that only the carrier can rectify.

• CUCME no calls incoming, outgoing calls okay

  Hello everyone,

  I'm setting up a CUCME with SIP trunk, I can make calls outside, but I can´t receive everything from the outside, it's my second time as a SIP configuration

  I ve use debug command voice dialpeer all to check was happening, but I can´t find the problem.

  This is my config:

  IP server host sip - A.B.C.D

  !

  voip phone service

  list of approved IP addresses

  IPv4 A.B.C.D 255.255.255.252

  !

  translation of the voice-rule 1

  rule 1 / 325277\ (\) / / 1\1 /.

  !

  voice translation-profile IN

  translate 1 called

  !

  Dial-peer voice 1 voip

  Description * incoming SIP trunk call *.

  entrants IN translation-profile

  session protocol sipv2

  session target sip-Server

  incoming called-number.

  codec voice-class 1

  voice-class sip dtmf-relay rtp - nte force

  DTMF-relay rtp - nte

  No vad

  !

  ePhone-dn 1

  number 100

  Description of RECEPTION

  !

  ePhone 2

  address Mac YYYY. BENAMER. CCBC

  ePhone-model 1

  type 7942

  Keep-Conference

  button 1:1

  NOTE: The IP address are hidden, just for safety

  Here is the output from my debug/tests:

  voice translation rule 1 32527700 #test

  Matched with rule 1

  Original number: 32527700 translated number: 100

  Number of origin type: no number translation type: no

  Original number plan: no number plan translated: no

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Number = 32527700, called number = 32527700, Peer Info Type = DIALPEER_INFO_SPEECH

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = 32527700

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:

  Dial String = 32527700, expanded String = 32527700 number = 32527700T

  Timeout = TRUE, incoming = FALSE, Peer Info Type = DIALPEER_INFO_SPEECH

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:

  Result =-1

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

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

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = 32527700, saf_enabled = 1 saf_dndb_lookup = 1, dp_result =-1

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

  Result = NO_MATCH(-1)

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

  Number = 59513212, 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

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

  Associate the rule of = DP_MATCH_ANSWER; Number = 59513212

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:

  Incoming = TRUE, expand = FALSE

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:

  Dial String =, expanded string =, number of calls = 59513212T

  Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:

  Result =-1

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:[email protected]/ * /.

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

  Associate the rule of = DP_MATCH_ORIGINATE; Number = 59513212

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:

  Incoming = TRUE, expand = FALSE

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:

  Dial String =, expanded string =, number of calls = 59513212T

  Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:

  Result =-1

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:[email protected]/ * /.

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

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

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

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

  * Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeer:[email protected]/ * /.

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:

  Number = 59513212, called number = 32527700, 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

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:

  Associate the rule of = DP_MATCH_VIA_URI; URI = SIP:A.B.C.D:5060

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:

  Incoming = TRUE, expand = FALSE

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

  Dial String =, expanded string =, number of calls =

  Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

  Result =-1

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:[email protected]/ * /.

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:

  Associate the rule of = DP_MATCH_REQUEST_URI; URI = sip:[email protected]/ * /: 5060; user = phone

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:

  Incoming = TRUE, expand = FALSE

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

  Dial String =, expanded string =, number of calls =

  Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

  Result =-1

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:[email protected]/ * /.

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:

  Associate the rule of = DP_MATCH_TO_URI; URI = sip:[email protected]/ * /; user = phone

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:

  Incoming = TRUE, expand = FALSE

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

  Dial String =, expanded string =, number of calls =

  Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

  Result =-1

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:[email protected]/ * /.

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:

  Associate the rule of = DP_MATCH_FROM_URI; URI = sip:[email protected]/ * /; user = phone

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:

  Incoming = TRUE, expand = FALSE

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

  Dial String =, expanded string =, number of calls =

  Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

  Result =-1

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:[email protected]/ * /.

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:

  Associate the rule of = DP_MATCH_INCOMING_DNIS; Called number = 32527700

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:

  Incoming = TRUE, expand = FALSE

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:

  Dial String = 32527700, expanded String = 32527700 number =

  Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/MatchNextPeer:

  Result = Success (0); Incoming dial-peer = 1 is set in correspondence

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:[email protected]/ * /.

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:

  Result = Success (0) after DP_MATCH_INCOMING_DNIS; Incoming dial-peer = 1

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchSafModulePlugin:

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

  * Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerSPI:[email protected]/ * /.

  Can someone help me?

  Thanks in advance!

  I looked on the other leg of the SIP messages appeal, here's the fault for the where incoming call is being failed because the session timer is too small, has received from the SBC (provider)

  Call ID:

  Call ID: [email protected]/ * /.

  INVITE RECEIEVED SBC - SIP

  * Jan 29 16:53:19: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

  Received:

  GUEST sip: 32527700 @(WAN): 5060; user = phone SIP/2.0

  Via: SIP/2.0/UDP (SIP_SERVER): 5060; Branch = z9hG4bK776928550196f0d843ca0b092

  Call ID: [email protected]/ * /.

  From:; tag = 6e8b9968-CC-25

  TO:

  CSeq: 1 INVITE

  Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTRY, PRACK, INFO SUBSCRIBE, NOTIFY, updates, MESSAGE, see

  Max-Forwards: 70

  Supported: 100rel, timer

  User-Agent: Huawei SoftX3000 V300R601

  Session time-out: 300

  Min - SE: 90

  Contact:

  Content-Length: 376

  Content-Type: application/sdp

  v = 0

  o = HuaweiSoftX3000 4507886 4507886 IN IP4 (SIP_SERVER)

  s = call Sip

  c = IN IP4 (SIP_SERVER)

  t = 0 0

  m = audio RTP 11554 / AVP 8 0 18 4 2 98 98 98

  a = rtpmap:8 PCMA/8000

  a = rtpmap:0 PCMU/8000

  a G729/8000 rtpmap:18 =

  a = rtpmap:4 G723/8000

  a = rtpmap:2 G726-32/8000

  a = rtpmap:98 G726-40/8000

  a = rtpmap:98 G726-32/8000

  a = rtpmap:98 G726-24/8000

  a = ptime:20

  a = fmtp:18 annex b = No.

  In response to GUY sends 422

  Envoy:

  SIP/2.0 422 Session Timer too small

  Via: SIP/2.0/UDP (SIP_SERVER): 5060; Branch = z9hG4bK776928550196f0d843ca0b092

  From:; tag = 6e8b9968-CC-25

  Up to:; tag = 4CD1E84-2094

  Date: Wednesday, January 29, 2014 22:53:19 GMT

  Call ID: [email protected]/ * /.

  CSeq: 1 INVITE

  Allow-events: telephone-event

  Min - SE: 1800

  Server: Cisco-SIPGateway/IOS-15.2.4.M

  Content-Length: 0

  According to rfc

  If the Session time-out interval is too low for a proxy (i.e., lower)

  that the value Min - SE that the proxy would argue), the

  Proxy denies the request with a 422 response.  This response

  contains a header field in Min - TO identify the minimum session

  meantime, she is ready to support.  The UAC will try again, this time

  including the header of Min - SE in the query field.  The header field

  contains the largest header field Min - SE that he observed in all 422

  responses received previously.  In this way, the minimum timer meets the

  constraints of all proxies on the way.

  http://www.Cisco.com/en/us/docs/iOS/voice/SIP/configuration/guide/sip_cg-msg_tmr_rspns.html#wp1056968

  Response message 422

  If the value of the Session header expires is too small, the UAS or proxy refuses the call with a response message 422 Session Timer too small . With 422 response, the proxy or the SAMU message includes a header of Min - SE, indicating the value of minimum session, he can accept. UAC can then try again the appeal with a higher value of session timer.

  If a 422-response message is received after a GUEST query, the UAC can again INVITE him.

  There is two way to fix this

  1) asked the SBC (your SIP provider) the value change and the value of standards send the SIP invite session expires

  (2) change the value of the Min - SE on the CME on demand

  Run this Global Config on CME

  voip phone service

  allow sip to sip connection

  SIP

  90 min - to

  BR,

  Nadeem

  Please note all the useful post.

• Incoming PSTN call to fail UCCX

  Hello

  I have no call flows as below

  RTC - sip - CUBE (SP edition) - sip - CUCM - CTI Port - CCX

  When RTC called the number and it should trigger CCX application to read the message BUT rather we transport message ' xxx notifies you that the number tried to call has been disconnected. "

  Track of the CCM shows 404 not found but DNA shows the works of appeal show. Why?

  CCM traces and traces of CCX engine is attached.

  CM version is 10.5

  Version of the CCX is 10.5

  call: + 61423348232

  call: + 61730842590 (DN 41490 is CTI RP CCM registry).

  DN 40700 is CTI port.

  CTI port 40700 and CTI RP 41490 are attributed to jtapi user application in CUCM

  FH

  Make sure that the SIP trunk CSS lists partition associated with the ports above the Point of road ITCS ITCS even if the call goes to the CTI RP you need access to CTI ports for the call to connect.

• CUCM SIP-trunk multiple Ports

  I would like to know if it is possible to create a Script of SIP standardization in CUCM which change the Destination SIP port based on the phone which makes the phone.

  Current issue:

  The ITSP provider asked that I send calls from different regions to the same address IP SBC but a different port.

  Example:

  Phone (A) makes a call

  RP - RL - RG---> SIP Trunk (5060)---> CUBE (5060<-->5001)---> ITSP1

  Telephone (B) makes a call

  RP - RL - RG---> SIP Trunk (5060)--->---> CUBE ITSP1 (5060<-->5002)

  I can do it on the CUBE, but which require several dial-Exchange and SIP profiles, is it possible in the CUCM to change the sip by device-pool port?.

  Or is there another I can match and send calls via the same destination with different dial-peer sip ports.?

  Thank you.

  Zakiab,

  You can do this, its all just impossible. Your signage ports are defined not on the endpoints, but on the B2BUA (CUCM or CUBE). The CUBE, you have the most flexibility, because you can change the port of signs based on your dial-peers. However, as you pointed out quite rightly, need you an insurmountable amount of dial-peers to have this for each endpoint. So I suggest tell you your ITSP to change their design or moving to a new.

• 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