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
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 Tags: Cisco Support Getting below to ASSERT the message when you try to store a value in the PMString object. PMString::SetKey - call SetKey with a string that is not a translation No idea how can I avoid this message? Y at - it any changes required in the .fr file? Try using a Widestring instead, because you don't need translation. If you do not, try SetTranslated() SIP trunk CUBE with Callcentric - incoming unanswered call
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? voip phone service voice class codec 1 translation of the voice-rule 1 voice translation-profile IN Dial-peer voice 1 voip SIP - ua 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 RV016 incomming sip-applications Nice day I have a bad router Cisco RV016 framework to properly manage SIP applications. Here what I have: LAN - 10.1.0.0/24 10.1.0.1 - IP LAN of the RV016 10.1.0.2 - IP on the NETWORK adapter for the software IP - PBX "3CX Phone SYSTEM" RV016 have two interfaces configured, WAN to work with various Internet service providers. I also have a VoIP service provider somewhere through the internet, which is handlling the incoming and outgoing calls. For incomming calls VoIP to work properly, I did some port-forwarding rules: SIP - all connections inbound interface WAN and destination port 5060 are translated to the 10.1.0.2 RTP - incommming connections WAN interface and the destination ports 9000-9015 are translated to the 10.1.0.2 It works fine, but with a very unpleasant fact - everyone of the internet can send SIP Register requests to my IP - PBX, so long I have non-stop 'register' attempts from different IP addresses. I tried to do a few firewall access rules, but just RV016 ignores them, when Port-Forwarding rule is applied. In the previous solution, that I used, it was simple to make a rule to allow an incoming connection on the WAN port only from the IP address of a single source, but, unfortunately, RV016 doesn't have such a feature. Here, what is the question: What should I do? I can't leave the situation as it is, but I really want to change the router. Can someone help me please advice? Here is an example showing how to add access rules to the top of a port forwarding rule. When an access rule is set on a port forwarding rule (for example, the SSH service), you want to first add a deny rule to deny all IPs from the side WAN and then add an allow rule to allow a specific IP entering side WAN. Allow SSH WAN1 [specific IP] [private address] Deny SSH WAN1 everything [private address] 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:
v = 0 000268: 10:38:35.522 Oct 2 UTC: / / 698 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg: v = 0
000269: 10:38:35.522 Oct 2 UTC: / / 697 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg: 000270: 10:38:35.534 Oct 2 UTC: / / 698 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg: 000275: 10:38:38.422 Oct 2 UTC: / / 698 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg: v = 0 000276: Oct 2 UTC 10:38:38.426: / / 698 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg: 000277: 10:38:38.430 Oct 2 UTC: / / 697 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg:
v = 0 000278: Oct 2 UTC 10:38:38.442: / / 698 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg: 000279: Oct 2 UTC 10:38:38.922: / / 698 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg: v = 0 000280: Oct 2 UTC 10:38:38.926: / / 697 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg: v = 0
000281: 10:38:38.926 Oct 2 UTC: / / 698 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg:
000282: 10:38:38.934 Oct 2 UTC: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: 000283: 10:38:45.426 Oct 2 UTC: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: 000284: 10:38:45.430 Oct 2 UTC: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: 000285: 10:38:45.434 Oct 2 UTC: / / 698 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg:
000286: Oct 2 UTC 10:38:45.454: / / 698 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg:
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: 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
Then you must apply the sip profile at the foot of basketing od the call... IE cucm cubed Dial-peer telephony voip xxx Now, when the appeal is as a ringtone, the full display will be + 44... What was initially called number. Reject incoming calls on ios 10 not listed is not on the lock screen (Notification) Reject incoming calls when the phone is locked by pressing the power button / stop on ios 10 that the same call does not appear on the lock screen or (Notification) as a reminder... N ° it will not. He never has. you will see a missed call notification if you miss the call. If you actively reject the call, you don't have to Miss it. Access your recent calls list and call back them was from there. Language not displaying options do not Google Translate in Firefox Firefox is still having a problem displaying the languages in the drop-down control to Google Translate. This issue has been addressed in another thread, (now archived) here: https://support.Mozilla.org/en-us/questions/957611?ESAB=a & As = AAQ I still encounter the problem, and I created a SIMPLER example to illustrate the problem, here: http://www.Audiosparx.com/SA/testing/translate/test.htm What we have tried so far to solve: 1. we tried the reset Firefox option, which did not solve the problem The strange thing is that it works very well on a system running Windows 7, and it worked for two Firefox, 23, 24 and 25, which were upgraded to the order. But it does not work on an another high-end system nine also runs Windows 7 and 25 of Firefox. It works very well on Google Chrome. Internet Explorer does not yet display control to translate! Something changed recently potentially with Google translation which is the cause of break. I'm not the first to report, although he seemed to have resolved for other people in some cases, without an explantation. A customer has reported this problem and we can reproduce it on a Windows 7 system, but also is not another system running Windows 7. Any help that you can offer to fix it for Firefox users would be much appreciated. Note that Google translation element requires Flash work, so if it does not work so make sure you who have installed the Flash plugin and you also authorize this plugin. You can check for problems with the latest version of the Flash plugin and try these: How to turn on when it is not automatically translate? How turn on when it is not automatically enabled to translate? Visited a blog (RSS feeds) that did not Translate function. Make use of the Google Translator Toolbar Otherwise, use Firefox Addons Several incoming clones with the FGV not reentrant timer Hello Attached image illustrates the idea. So I throw clone1 (which is some reentrant VI), in this clone I have not reentrant timer but it is not reentrant in the perspective of clone 1 only. 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I suggest that repost you your Question in Exchange for TechNet Forums. https://social.technet.Microsoft.com/forums/Exchange/en-us/home?category=ExchangeServer Or here: https://social.technet.Microsoft.com/forums/Exchange/en-us/home?Forum=exchangesvrgeneral TechNet Server forums. http://social.technet.Microsoft.com/forums/WindowsServer/en-us/home?category=WindowsServer See you soon. I have Verizon FIOS. They block incoming traffic on port 443. I have an ISA Server, but were unsuccessful, redirect to 442 on the listener interface and bypass at 443 internally. Thank you! Victor Hello Take a look at this thread who can answer your specific question. http://social.Microsoft.com/forums/en-us/whssoftware/thread/8c2238da-c4c7-4777-8CE9-b1bb506daf0f/ However, if it is not, I would say that to post your question in the forum below. http://social.Microsoft.com/forums/en-us/whshardware/threads Update incoming e-mail from customers not not to date calls We have a situation when a client sends an update to a call via incoming e-mail, the call is not updated. It works well for creating a new call and also if an agent sends an email via cmail incoming update appealed but if the same email from a customer of incoming e-mail is simply ignored. Everyone has to this problem. It worked in Version 8.0, but since the upgrade to V9.0 le05 it does not work. Had a look and found a bug existing about this (call 171735 iTracker), but according to the story, it should have been resolved in le05. 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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: 520Similar Questions
I have a Cisco Callmanager and a virtual router 7200 (GNS3) as a gateway (C7200-ADVENTERPRISEK9-M), Version 12.4 (24).
allow sip to sip connections
Fax protocol cisco
SIP
interface FastEthernet0/0 source control binding
bind media source interface FastEthernet0/0
Registrar Server
g711ulaw codec preference 1
rule 1 / ^ 8 / /0056/
!
voice translation-rule 2
rule 1 5.0 / /17772114zzz/
!
voice translation-rule 3
rule 1 /17772114zzz/ /500/
definition of 3 called
!
FLIGHT voice translation-profile
definition of call 2
translate 1 called
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
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.
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
answer 183 sip remote-Party-ID header change "
profiles of sip voice-class 3
2 uninstall, reinstall, which did not solve the problem
I want to create a launcher of timer. So I created simple timer based on a functional global variable and tried to launch several clones of her home "timer Launcher. The thing is that the timer is not reentrant (each FGV I saw was not reentrant) and I do not know how to create such a pitcher.
I created an application Multicurrency, consisting of 2 cubes. The first cube has only the default dimensions, the second cube additional custom two-dimensional. Custom2 stores members 'Amount' and 'Description', CUSTOM1 stores data for transactions (T1, T2, etc.).
Cube 2 there is a form that uses an account shared between cubes ("Transactions") with cube 2 as plan source type. Thus, in cube 2 data is stored for a combination of account "Opérations", "T1" (Custom1) and 'Amount' (Custom2).
Cube 1 there is a form that shows the account 'Operations' (read only due to the plan of different source type), however after I entered some data for the form assigned to cube2, train in cube1 is giving me all the data.
I checked the essbase outline and cube 1 account 'Opérations' has the value dynamic Calc and formula automatically genereated during refresh XREF. For cube 2, this account is set to store.
It seems that I'm missing something, however, I am not in a position to define what is originally the function XREF does not give results.
The gurus of the Hyperion, you help me?
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