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CUBE-C2951 remote destination and wireless phones not working

Chris9811
Level 1
Level 1

Hello everybody,

I have had the problem for a few days that my remote destinations no longer work. If someone places a call to my directory number my cellphone should ring. But it doesnt ring, in the traces you can see that there is a 408 timeout - i dont understand this timeout?! The same strange thing happens to me with the 8821 wireless phones. I can receive a call and it works normal but if i try to place a call outbound it doesnt work?! Same 408 timeout as mentioned before with the remote destination.

 

Here comes my setup and my trace:

Cisco 2951 as CUBE /c2951-universalk9-mz.SPA.157-3.M5.bin/ - config:

 

voice service voip
 ip address trusted list
  ipv4 217.XX.68.150 255.255.255.255
  ipv4 217.XX.64.0 255.255.240.0
  ipv4 217.116.112.0 255.255.240.0
  ipv4 212.9.32.0 255.255.224.0
  ipv4 217.XX.77.0 255.255.255.0
 mode border-element 
 media statistics
 allow-connections sip to sip
 supplementary-service h450.12
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
 h323
  call service stop
 sip
  bind control source-interface GigabitEthernet0/1
  bind media source-interface GigabitEthernet0/0
  registrar server expires max 3600 min 1800
  asserted-id ppi
  early-offer forced
  no silent-discard untrusted
  midcall-signaling passthru
  g729 annexb-all
!
voice class codec 1
 codec preference 1 g711alaw
 codec preference 2 g722-64
 codec preference 3 g711ulaw
 codec preference 4 g729r8
 codec preference 5 g726r32
!
!
voice class sip-profiles 1
 request INVITE sip-header To modify "(<.*:)(.*@)" "<sip:22XXXt7@" 
 request ANY sip-header From modify "(<.*:)(.*@)" "<sip:22XXXt7@" 
 request REGISTER sip-header From modify "(<.*:)(.*@)" "<sip:22XXXt7@" 
 request INVITE sip-header P-Preferred-Identity modify "<sip:0(.*)@(.*)>" "<sip:49\1@\2>" 
!
!
!
!
!
voice translation-rule 3
 rule 1 /^0/ /0049/
 rule 2 /^0\(.*$\)/ /0049\1/
 rule 4 /^11\(.*$\)/ /004911\1/
!
voice translation-rule 4
 rule 1 /^\(78..$\)/ /493XXX1139\1/
!
voice translation-rule 11
 rule 1 /^/ // type unknown unknown
 rule 2 /^/ /49/ type national national
 rule 3 /^/ /0049/ type international international
!
voice translation-rule 12
 rule 1 /^493XXX1139\(....$\)/ /\1/
!
!
voice translation-profile INCOMING
 translate calling 11
 translate called 12
!
voice translation-profile OUTGOING
 translate calling 4
 translate called 3

dial-peer voice 1 voip
 description *Outbound to Sipgate*
 translation-profile outgoing OUTGOING
 destination-pattern 0T
 session protocol sipv2
 session target ipv4:217.XX.68.150
 session transport udp
 voice-class codec 1  
 voice-class sip asserted-id ppi
 voice-class sip profiles 1
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 2 voip
 description *Incoming from Sipgate*
 translation-profile incoming INCOMING
 preference 1
 service session
 session protocol sipv2
 session target ipv4:217.XX.68.150
 session transport udp
 incoming called-number .T
 voice-class codec 1  
 voice-class sip asserted-id ppi
 voice-class sip profiles 1
 dtmf-relay rtp-nte
 ip qos dscp cs5 media
 ip qos dscp cs4 signaling
 no vad
!
dial-peer voice 3 voip
 description *Outbound to Sipgate*
 translation-profile outgoing OUTGOING
 destination-pattern +T
 session protocol sipv2
 session target ipv4:217.XX.68.150
 session transport udp
 voice-class codec 1  
 voice-class sip asserted-id ppi
 voice-class sip profiles 1
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 10 voip
 description CUCM to Cube
 session protocol sipv2
 session target sip-server
 incoming called-number 0T
 voice-class codec 1  
 no voice-class sip asserted-id
 voice-class sip bind media source-interface GigabitEthernet0/1
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 11 voip
 description CUCM to Cube
 session protocol sipv2
 session target sip-server
 incoming called-number +49T
 voice-class codec 1  
 no voice-class sip asserted-id
 voice-class sip bind media source-interface GigabitEthernet0/1
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 30 voip
 description CUBE to CUCM
 destination-pattern 78..
 session protocol sipv2
 session target ipv4:192.168.83.200
 voice-class codec 1  
 no voice-class sip asserted-id
 voice-class sip bind media source-interface GigabitEthernet0/1
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 31 voip
 description CUBE to CUCM
 destination-pattern 78..
 session protocol sipv2
 session target ipv4:192.168.83.201
 voice-class codec 1  
 no voice-class sip asserted-id
 voice-class sip bind media source-interface GigabitEthernet0/1
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 32 voip
 description CUBE to CUCM
 destination-pattern 78..
 session protocol sipv2
 session target ipv4:192.168.77.201
 voice-class codec 1  
 no voice-class sip asserted-id
 voice-class sip bind media source-interface GigabitEthernet0/1
 dtmf-relay rtp-nte
 no vad
!
!
sip-ua 
 credentials username 22XXXt7 password 7 realm 217.10.68.150
 keepalive target ipv4:217.XX.68.150
 authentication username 22XXXt7 password 7
 no remote-party-id
 retry invite 2
 retry response 3
 retry bye 2
 retry cancel 2
 retry register 10
 timers connect 100
 timers register 250
 timers keepalive active 100
 registrar ipv4:217.XX.68.150 expires 3600
 sip-server ipv4:217.XX.68.150
 presence enable

 

CUCM 12.0.1.23900-9

Trunks.pngroutepattern.pngCalling Search Space.pngCS Config.png2020-05-25 10_35_58-Phone Configuration.pngMOB_RDP.pngRDP.png

 First trace:

 

May 25 08:12:03.370: //3579/3DF80E3390C4/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.83.254:5060;branch=z9hG4bK8075C0
From: "036482374326" <sip:036482374326@217.10.68.150>;tag=4F38520-21AA
To: <sip:7811@192.168.83.200>
Date: Mon, 25 May 2020 08:12:03 GMT
Call-ID: 400F6363-9D9611EA-90CC9AC8-546A44B2@192.168.83.254
CSeq: 101 INVITE
Allow-Events: presence
Content-Length: 0


Timestamp: 3673239123370
UTC Timestamp:3673239123370

May 25 08:12:03.394: //3579/3DF80E3390C4/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.83.254:5060;branch=z9hG4bK8075C0
From: "036482374326" <sip:036482374326@217.10.68.150>;tag=4F38520-21AA
To: <sip:7811@192.168.83.200>;tag=33755~badef395-9309-48d9-8bf8-030362ac44b2-19489562
Date: Mon, 25 May 2020 08:12:03 GMT
Call-ID: 400F6363-9D9611EA-90CC9AC8-546A44B2@192.168.83.254
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Allow-Events: presence
Server: Cisco-CUCM12.0
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-ID: 00000000000000000000000000000000;remote=5a25609346e650b1b89bec77e38e129f
P-Preferred-Identity: "Mr. XXX" <sip:7811@192.168.83.200>
Contact: <sip:7811@192.168.83.200:5060>
Content-Length: 0


Timestamp: 3673239123394
UTC Timestamp:3673239123394

May 25 08:12:03.394: //3577/3DF80E3390C4/SIP/Msg/ccsipDisplayMsg:
Sent: 
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 217.10.68.150;branch=z9hG4bK5dbb.3f5ed151baaa1ce19daa5e1aa4b8787a.0,SIP/2.0/UDP 172.20.40.8;branch=z9hG4bK5dbb.f2929beb70159fa048795107c9d9120e.0,SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK5dbb.5702be3bd00b028b10c11ddddb7b5559.0,SIP/2.0/UDP 212.9.44.6:5060;branch=z9hG4bK39b08e53
From: "036482374326" <sip:036482374326@sipconnect.sipgate.de>;tag=as09d979ac
To: <sip:0049366511397811@sipconnect.sipgate.de>;tag=4F38540-246D
Date: Mon, 25 May 2020 08:11:59 GMT
Call-ID: 259399cd0686c844702e00c51f7160b9@sipconnect.sipgate.de
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
P-Preferred-Identity: <sip:7811@192.168.83.254>
Contact: <sip:49366511397811@192.168.83.254:5060>
Record-Route: <sip:217.10.68.150;lr;ftag=as09d979ac>,<sip:172.20.40.8;lr>,<sip:217.10.68.137;lr;ftag=as09d979ac>
Server: Cisco-SIPGateway/IOS-15.7.3.M5
Session-ID: 00000000000000000000000000000000;remote=5a25609346e650b1b89bec77e38e129f
Content-Length: 0


Timestamp: 3673239123394
UTC Timestamp:3673239123394

May 25 08:12:05.394: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
INVITE sip:+4917XXXX9872@192.168.83.254:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.83.200:5060;branch=z9hG4bK2e2957760b30
From: "036482374326" <sip:036482374326@192.168.83.200>;tag=33761~badef395-9309-48d9-8bf8-030362ac44b2-19489563
To: <sip:+4917XXXX9872@192.168.83.254>
Date: Mon, 25 May 2020 08:12:05 GMT
Call-ID: 6a3afe00-ecb17dd5-1fc2-c853a8c0@192.168.83.200
Supported: timer,resource-priority,replaces
Min-SE:  1800
User-Agent: Cisco-CUCM12.0
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=MIXED
Session-ID: 5a25609346e650b1b89bec77e38e129f;remote=00000000000000000000000000000000
Cisco-Guid: 1782251008-0000065536-0000000150-3360925888
Session-Expires:  1800
Diversion: "Mr. XXX" <sip:7811@192.168.83.200>;reason=follow-me;privacy=off;screen=yes
P-Preferred-Identity: "036482374326" <sip:036482374326@192.168.83.200>
Contact: <sip:036482374326@192.168.83.200:5060;transport=tcp>
Max-Forwards: 65
Content-Type: application/sdp
Content-Length: 311
v=0
o=CiscoSystemsCCM-SIP 33761 1 IN IP4 192.168.83.200
s=SIP Call
c=IN IP4 192.168.83.254
b=TIAS:64000
b=AS:64
t=0 0
m=audio 18430 RTP/AVP 0 8 18 101
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15


Timestamp: 3673239125394
UTC Timestamp:3673239125394

May 25 08:12:05.402: //3581/6A3AFE000000/SIP/Msg/ccsipDisplayMsg:
Sent: 
INVITE sip:+4917XXXX9872@217.10.68.150:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.83.254:5060;branch=z9hG4bK80825E7
From: "036482374326" <sip:22XXXXt7@217.10.68.150>;tag=4F38D18-219A
To: <sip:22XXXXt7@217.10.68.150>
Date: Mon, 25 May 2020 08:12:05 GMT
Call-ID: 41461754-9D9611EA-90D29AC8-546A44B2@192.168.83.254
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 1782251008-0000065536-0000000150-3360925888
User-Agent: Cisco-SIPGateway/IOS-15.7.3.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1590394325
Contact: <sip:036482374326@192.168.83.254:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 64
P-Preferred-Identity: "036482374326" <sip:4936482374326@192.168.83.254>
Diversion: "Mr. XXX"<sip:7811@192.168.83.200>;privacy=off;reason=follow-me;screen=yes
Session-ID: 5a25609346e650b1b89bec77e38e129f;remote=00000000000000000000000000000000
Session-Expires:  1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 322
v=0
o=CiscoSystemsSIP-GW-UserAgent 7662 6281 IN IP4 192.168.83.254
s=SIP Call
c=IN IP4 PUBLIC IP
t=0 0
m=audio 18434 RTP/AVP 0 8 18 101
c=IN IP4 PUBLIC IP
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20


Timestamp: 3673239125402
UTC Timestamp:3673239125402

May 25 08:12:05.402: //3580/6A3AFE000000/SIP/Msg/ccsipDisplayMsg:
Sent: 
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.83.200:5060;branch=z9hG4bK2e2957760b30
From: "036482374326" <sip:036482374326@192.168.83.200>;tag=33761~badef395-9309-48d9-8bf8-030362ac44b2-19489563
To: <sip:+4917XXXX9872@192.168.83.254>
Date: Mon, 25 May 2020 08:12:05 GMT
Call-ID: 6a3afe00-ecb17dd5-1fc2-c853a8c0@192.168.83.200
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.7.3.M5
Session-ID: 00000000000000000000000000000000;remote=5a25609346e650b1b89bec77e38e129f
Content-Length: 0


Timestamp: 3673239125402
UTC Timestamp:3673239125402

May 25 08:12:05.430: //3581/6A3AFE000000/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.83.254:5060;rport=64340;received=PUBLIC IP;branch=z9hG4bK80825E7
From: "036482374326" <sip:22XXXXt7@217.10.68.150>;tag=4F38D18-219A
To: <sip:22XXXXt7@217.10.68.150>;tag=ca40b14695a455acdff6b4fb6f6d0a8e.0c69
Call-ID: 41461754-9D9611EA-90D29AC8-546A44B2@192.168.83.254
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="sipconnect.sipgate.de", nonce="Xst/AV7LfdVcD+rDi7NMLoGm6bp5oxs3"
Content-Length: 0


Timestamp: 3673239125430
UTC Timestamp:3673239125430

May 25 08:12:05.430: //3581/6A3AFE000000/SIP/Msg/ccsipDisplayMsg:
Sent: 
ACK sip:+4917XXXX9872@217.10.68.150:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.83.254:5060;branch=z9hG4bK80825E7
From: "036482374326" <sip:22XXXXt7@217.10.68.150>;tag=4F38D18-219A
To: <sip:22XXXXt7@217.10.68.150>;tag=ca40b14695a455acdff6b4fb6f6d0a8e.0c69
Date: Mon, 25 May 2020 08:12:05 GMT
Call-ID: 41461754-9D9611EA-90D29AC8-546A44B2@192.168.83.254
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Session-ID: 5a25609346e650b1b89bec77e38e129f;remote=cbe7beebbc5c55f7b773fcdd7c16604a
Content-Length: 0


Timestamp: 3673239125430
UTC Timestamp:3673239125430

May 25 08:12:05.430: //3581/6A3AFE000000/SIP/Msg/ccsipDisplayMsg:
Sent: 
INVITE sip:+4917XXXX9872@217.10.68.150:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.83.254:5060;branch=z9hG4bK8095A4
From: "036482374326" <sip:22XXXXt7@217.10.68.150>;tag=4F38D18-219A
To: <sip:22XXXXt7@217.10.68.150>
Date: Mon, 25 May 2020 08:12:05 GMT
Call-ID: 41461754-9D9611EA-90D29AC8-546A44B2@192.168.83.254
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 1782251008-0000065536-0000000150-3360925888
User-Agent: Cisco-SIPGateway/IOS-15.7.3.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Timestamp: 1590394325
Contact: <sip:036482374326@192.168.83.254:5060>
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="22XXXXt7",realm="sipconnect.sipgate.de",uri="sip:+4917XXXX9872@217.10.68.150:5060",response="2f48fb412c4bfcee1e1177c2687fd50d",nonce="Xst/AV7LfdVcD+rDi7NMLoGm6bp5oxs3",algorithm=md5
Max-Forwards: 64
P-Preferred-Identity: "036482374326" <sip:4936482374326@192.168.83.254>
Diversion: "Mr. XXX"<sip:7811@192.168.83.200>;privacy=off;reason=follow-me;screen=yes
Session-ID: 5a25609346e650b1b89bec77e38e129f;remote=00000000000000000000000000000000
Session-Expires:  1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 322
v=0
o=CiscoSystemsSIP-GW-UserAgent 7662 6281 IN IP4 192.168.83.254
s=SIP Call
c=IN IP4 PUBLIC IP
t=0 0
m=audio 18434 RTP/AVP 0 8 18 101
c=IN IP4 PUBLIC IP
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20


Timestamp: 3673239125430
UTC Timestamp:3673239125430

May 25 08:12:05.930: //3581/6A3AFE000000/SIP/Msg/ccsipDisplayMsg:
Sent: 
INVITE sip:+4917XXXX9872@217.10.68.150:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.83.254:5060;branch=z9hG4bK8095A4
From: "036482374326" <sip:22XXXXt7@217.10.68.150>;tag=4F38D18-219A
To: <sip:22XXXXt7@217.10.68.150>
Date: Mon, 25 May 2020 08:12:05 GMT
Call-ID: 41461754-9D9611EA-90D29AC8-546A44B2@192.168.83.254
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 1782251008-0000065536-0000000150-3360925888
User-Agent: Cisco-SIPGateway/IOS-15.7.3.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Timestamp: 1590394325
Contact: <sip:036482374326@192.168.83.254:5060>
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="22XXXXt7",realm="sipconnect.sipgate.de",uri="sip:+4917XXXX9872@217.10.68.150:5060",response="2f48fb412c4bfcee1e1177c2687fd50d",nonce="Xst/AV7LfdVcD+rDi7NMLoGm6bp5oxs3",algorithm=md5
Max-Forwards: 64
P-Preferred-Identity: "036482374326" <sip:4936482374326@192.168.83.254>
Diversion: "Mr. XXX"<sip:7811@192.168.83.200>;privacy=off;reason=follow-me;screen=yes
Session-ID: 5a25609346e650b1b89bec77e38e129f;remote=00000000000000000000000000000000
Session-Expires:  1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 322
v=0
o=CiscoSystemsSIP-GW-UserAgent 7662 6281 IN IP4 192.168.83.254
s=SIP Call
c=IN IP4 PUBLIC IP
t=0 0
m=audio 18434 RTP/AVP 0 8 18 101
c=IN IP4 PUBLIC IP
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20


Timestamp: 3673239125930
UTC Timestamp:3673239125930

May 25 08:12:06.947: //3581/6A3AFE000000/SIP/Msg/ccsipDisplayMsg:
Sent: 
INVITE sip:+4917XXXX9872@217.10.68.150:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.83.254:5060;branch=z9hG4bK8095A4
From: "036482374326" <sip:22XXXXt7@217.10.68.150>;tag=4F38D18-219A
To: <sip:22XXXXt7@217.10.68.150>
Date: Mon, 25 May 2020 08:12:06 GMT
Call-ID: 41461754-9D9611EA-90D29AC8-546A44B2@192.168.83.254
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 1782251008-0000065536-0000000150-3360925888
User-Agent: Cisco-SIPGateway/IOS-15.7.3.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Timestamp: 1590394326
Contact: <sip:036482374326@192.168.83.254:5060>
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="22XXXXt7",realm="sipconnect.sipgate.de",uri="sip:+4917XXXX9872@217.10.68.150:5060",response="2f48fb412c4bfcee1e1177c2687fd50d",nonce="Xst/AV7LfdVcD+rDi7NMLoGm6bp5oxs3",algorithm=md5
Max-Forwards: 64
P-Preferred-Identity: "036482374326" <sip:4936482374326@192.168.83.254>
Diversion: "Mr. XXX"<sip:7811@192.168.83.200>;privacy=off;reason=follow-me;screen=yes
Session-ID: 5a25609346e650b1b89bec77e38e129f;remote=00000000000000000000000000000000
Session-Expires:  1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 322
v=0
o=CiscoSystemsSIP-GW-UserAgent 7662 6281 IN IP4 192.168.83.254
s=SIP Call
c=IN IP4 PUBLIC IP
t=0 0
m=audio 18434 RTP/AVP 0 8 18 101
c=IN IP4 PUBLIC IP
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20


Timestamp: 3673239126947
UTC Timestamp:3673239126947

May 25 08:12:08.947: //3580/6A3AFE000000/SIP/Msg/ccsipDisplayMsg:
Sent: 
SIP/2.0 408 Request Timeout
Via: SIP/2.0/TCP 192.168.83.200:5060;branch=z9hG4bK2e2957760b30
From: "036482374326" <sip:036482374326@192.168.83.200>;tag=33761~badef395-9309-48d9-8bf8-030362ac44b2-19489563
To: <sip:+4917XXXX9872@192.168.83.254>;tag=4F39AF0-76A
Date: Mon, 25 May 2020 08:12:05 GMT
Call-ID: 6a3afe00-ecb17dd5-1fc2-c853a8c0@192.168.83.200
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.7.3.M5
Reason: Q.850;cause=102
Session-ID: cbe7beebbc5c55f7b773fcdd7c16604a;remote=5a25609346e650b1b89bec77e38e129f
Content-Length: 0


Timestamp: 3673239128947
UTC Timestamp:3673239128947

May 25 08:12:08.947: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
ACK sip:+4917XXXX9872@192.168.83.254:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.83.200:5060;branch=z9hG4bK2e2957760b30
From: "036482374326" <sip:036482374326@192.168.83.200>;tag=33761~badef395-9309-48d9-8bf8-030362ac44b2-19489563
To: <sip:+4917XXXX9872@192.168.83.254>;tag=4F39AF0-76A
Date: Mon, 25 May 2020 08:12:05 GMT
Call-ID: 6a3afe00-ecb17dd5-1fc2-c853a8c0@192.168.83.200
User-Agent: Cisco-CUCM12.0
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Length: 0


Timestamp: 3673239128947
UTC Timestamp:3673239128947

May 25 08:12:09.779: //3579/3DF80E3390C4/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.83.254:5060;branch=z9hG4bK8075C0
From: "036482374326" <sip:036482374326@217.10.68.150>;tag=4F38520-21AA
To: <sip:7811@192.168.83.200>;tag=33755~badef395-9309-48d9-8bf8-030362ac44b2-19489562
Date: Mon, 25 May 2020 08:12:03 GMT
Call-ID: 400F6363-9D9611EA-90CC9AC8-546A44B2@192.168.83.254
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Allow-Events: presence, kpml
Supported: replaces
Server: Cisco-CUCM12.0
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires:  1800;refresher=uas
Require:  timer
Session-ID: 0000272500105000a00000059a3c7a00;remote=5a25609346e650b1b89bec77e38e129f
P-Preferred-Identity: "Mr. XXX" <sip:7811@192.168.83.200>
Contact: <sip:7811@192.168.83.200:5060>;+u.sip!devicename.ccm.cisco.com="CSFCO";video;bfcp
Content-Type: application/sdp
Content-Length: 329
v=0
o=CiscoSystemsCCM-SIP 33755 1 IN IP4 192.168.83.200
s=SIP Call
c=IN IP4 192.168.77.230
b=TIAS:64000
b=AS:64
t=0 0
a=cisco-mari:v1
a=cisco-mari-rate
m=audio 18582 RTP/AVP 0 101
a=extmap:14/sendrecv http://protocols.cisco.com/timestamp#100us
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15


Timestamp: 3673239129779
UTC Timestamp:3673239129779

May 25 08:12:09.783: //3579/3DF80E3390C4/SIP/Msg/ccsipDisplayMsg:
Sent: 
ACK sip:7811@192.168.83.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.83.254:5060;branch=z9hG4bK80A240B
From: "036482374326" <sip:036482374326@217.10.68.150>;tag=4F38520-21AA
To: <sip:7811@192.168.83.200>;tag=33755~badef395-9309-48d9-8bf8-030362ac44b2-19489562
Date: Mon, 25 May 2020 08:12:03 GMT
Call-ID: 400F6363-9D9611EA-90CC9AC8-546A44B2@192.168.83.254
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Session-ID: 5a25609346e650b1b89bec77e38e129f;remote=0000272500105000a00000059a3c7a00
Content-Length: 0


Timestamp: 3673239129783
UTC Timestamp:3673239129783

May 25 08:12:09.787: //3577/3DF80E3390C4/SIP/Msg/ccsipDisplayMsg:
Sent: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.10.68.150;branch=z9hG4bK5dbb.3f5ed151baaa1ce19daa5e1aa4b8787a.0,SIP/2.0/UDP 172.20.40.8;branch=z9hG4bK5dbb.f2929beb70159fa048795107c9d9120e.0,SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK5dbb.5702be3bd00b028b10c11ddddb7b5559.0,SIP/2.0/UDP 212.9.44.6:5060;branch=z9hG4bK39b08e53
From: "036482374326" <sip:036482374326@sipconnect.sipgate.de>;tag=as09d979ac
To: <sip:0049366511397811@sipconnect.sipgate.de>;tag=4F38540-246D
Date: Mon, 25 May 2020 08:11:59 GMT
Call-ID: 259399cd0686c844702e00c51f7160b9@sipconnect.sipgate.de
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
P-Preferred-Identity: <sip:7811@192.168.83.254>
Contact: <sip:49366511397811@192.168.83.254:5060>
Record-Route: <sip:217.10.68.150;lr;ftag=as09d979ac>,<sip:172.20.40.8;lr>,<sip:217.10.68.137;lr;ftag=as09d979ac>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.7.3.M5
Session-ID: 0000272500105000a00000059a3c7a00;remote=5a25609346e650b1b89bec77e38e129f
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 250
v=0
o=CiscoSystemsSIP-GW-UserAgent 395 3009 IN IP4 192.168.83.254
s=SIP Call
c=IN IP4 PUBLIC IP
t=0 0
m=audio 18426 RTP/AVP 0 101
c=IN IP4 PUBLIC IP
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20


Timestamp: 3673239129787
UTC Timestamp:3673239129787

May 25 08:12:09.823: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
ACK sip:49366511397811@192.168.83.254:5060 SIP/2.0
Via: SIP/2.0/UDP 217.10.68.150;branch=z9hG4bK5dbb.3562a3b7603da2214b936945d53aa46e.0
Via: SIP/2.0/UDP 172.20.40.8;branch=z9hG4bK5dbb.3bd3946570ed3c536e07be9f5176842d.0
Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK5dbb.af475c19076ecb6b4094a7868b11ca87.0
Via: SIP/2.0/UDP 212.9.44.6:5060;branch=z9hG4bK012293ae
Max-Forwards: 67
From: "036482374326" <sip:036482374326@sipconnect.sipgate.de>;tag=as09d979ac
To: <sip:0049366511397811@sipconnect.sipgate.de>;tag=4F38540-246D
Contact: <sip:036482374326@212.9.44.6:5060>
Call-ID: 259399cd0686c844702e00c51f7160b9@sipconnect.sipgate.de
CSeq: 103 ACK
Content-Length: 0


Timestamp: 3673239129823
UTC Timestamp:3673239129823

May 25 08:12:11.719: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
BYE sip:036482374326@192.168.83.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.83.200:5060;branch=z9hG4bK2e436de144da
From: <sip:7811@192.168.83.200>;tag=33755~badef395-9309-48d9-8bf8-030362ac44b2-19489562
To: "036482374326" <sip:036482374326@217.10.68.150>;tag=4F38520-21AA
Date: Mon, 25 May 2020 08:12:09 GMT
Call-ID: 400F6363-9D9611EA-90CC9AC8-546A44B2@192.168.83.254
User-Agent: Cisco-CUCM12.0
Max-Forwards: 70
CSeq: 101 BYE
Reason: Q.850;cause=16
Session-ID: 0000272500105000a00000059a3c7a00;remote=5a25609346e650b1b89bec77e38e129f
Content-Length: 0


Timestamp: 3673239131719
UTC Timestamp:3673239131719

May 25 08:12:11.723: //3577/3DF80E3390C4/SIP/Msg/ccsipDisplayMsg:
Sent: 
BYE sip:036482374326@212.9.44.6:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.83.254:5060;branch=z9hG4bK80B6C4
From: <sip:22XXXXt7@sipconnect.sipgate.de>;tag=4F38540-246D
To: "036482374326" <sip:036482374326@sipconnect.sipgate.de>;tag=as09d979ac
Date: Mon, 25 May 2020 08:12:09 GMT
Call-ID: 259399cd0686c844702e00c51f7160b9@sipconnect.sipgate.de
User-Agent: Cisco-SIPGateway/IOS-15.7.3.M5
Max-Forwards: 70
Route: <sip:217.10.68.150;lr;ftag=as09d979ac>,<sip:172.20.40.8;lr>,<sip:217.10.68.137;lr;ftag=as09d979ac>
Timestamp: 1590394331
CSeq: 101 BYE
Reason: Q.850;cause=16
P-RTP-Stat: PS=87,OS=13920,PR=89,OR=14240,PL=0,JI=0,LA=0,DU=1
Session-ID: 0000272500105000a00000059a3c7a00;remote=5a25609346e650b1b89bec77e38e129f
Content-Length: 0


Timestamp: 3673239131723
UTC Timestamp:3673239131723

May 25 08:12:11.723: //3579/3DF80E3390C4/SIP/Msg/ccsipDisplayMsg:
Sent: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.83.200:5060;branch=z9hG4bK2e436de144da
From: <sip:7811@192.168.83.200>;tag=33755~badef395-9309-48d9-8bf8-030362ac44b2-19489562
To: "036482374326" <sip:036482374326@217.10.68.150>;tag=4F38520-21AA
Date: Mon, 25 May 2020 08:12:11 GMT
Call-ID: 400F6363-9D9611EA-90CC9AC8-546A44B2@192.168.83.254
Server: Cisco-SIPGateway/IOS-15.7.3.M5
CSeq: 101 BYE
Reason: Q.850;cause=16
P-RTP-Stat: PS=89,OS=14240,PR=87,OR=13920,PL=0,JI=18,LA=0,DU=1
Session-ID: 5a25609346e650b1b89bec77e38e129f;remote=0000272500105000a00000059a3c7a00
Content-Length: 0


Timestamp: 3673239131723
UTC Timestamp:3673239131723

May 25 08:12:11.755: //3577/3DF80E3390C4/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.83.254:5060;rport=64340;received=PUBLIC IP;branch=z9hG4bK80B6C4
From: <sip:22XXXXt7@sipconnect.sipgate.de>;tag=4F38540-246D
To: "036482374326" <sip:036482374326@sipconnect.sipgate.de>;tag=as09d979ac
Call-ID: 259399cd0686c844702e00c51f7160b9@sipconnect.sipgate.de
CSeq: 101 BYE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0


Timestamp: 3673239131755
UTC Timestamp:3673239131755

Can someone here explain me please what i have to change or what i have done wrong?! 

FYI: All other phones/jabbers working fine.

 

Many thanks in advance.

 

Best regards 

Christoph

 

 

1 Accepted Solution

Accepted Solutions


@Chris9811 wrote:

Here are the debugs: 

 Nothing stands out as different between the working and non-working.  What strikes me is that you're getting no reply at all in he non working cases, if a header was wrong somewhere I'd expect an error response of some sort.  So this gets me thinking are their replies being sent but not getting to you for some reason. 

Do these calls all use the same outbound dial peer?

I can see some IP address inconsistencies, the media address is given in SDP as 24.134.12.225, I assume that's the outside interface of the CUBE.  But other fields show 192.168.83.254 which I presume is your internal address.  Could there be some reason that the provider is trying to reply to 192.168.83.254 which presumably isn't reachable from their side?

View solution in original post

16 Replies 16

TONY SMITH
Spotlight
Spotlight

 I assume this is the outgoing Invite to your cellphone ...

May 25 08:12:05.402: //3581/6A3AFE000000/SIP/Msg/ccsipDisplayMsg:
Sent: 
INVITE sip:+4917XXXX9872@217.10.68.150:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.83.254:5060;branch=z9hG4bK80825E7
From: "036482374326" <sip:22XXXXt7@217.10.68.150>;tag=4F38D18-219A
To: <sip:22XXXXt7@217.10.68.150>
Date: Mon, 25 May 2020 08:12:05 GMT
Call-ID: 41461754-9D9611EA-90D29AC8-546A44B2@192.168.83.254
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 1782251008-0000065536-0000000150-3360925888
User-Agent: Cisco-SIPGateway/IOS-15.7.3.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1590394325
Contact: <sip:036482374326@192.168.83.254:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 64
P-Preferred-Identity: "036482374326" <sip:4936482374326@192.168.83.254>
Diversion: "Mr. XXX"<sip:7811@192.168.83.200>;privacy=off;reason=follow-me;screen=yes
Session-ID: 5a25609346e650b1b89bec77e38e129f;remote=00000000000000000000000000000000
Session-Expires:  1800
Content-Type: application/sdp

I see that the carrier isn't replying with a Trying after you send with the proxy authentication.

Do normal outbound calls work correctly?   If so then I wonder if the carrier is not happy with your calling number in the redirected Invite (From and PPI headers).

Do you get the same behaviour if you set Call Forward All on an extension, to your mobile?

The carrier may have some specific requirements for calling number presentation for redirected calls, or some specific header needing to be modified or added.

Normal outbound calls working fine from deskphones/jabber for Windows/Jabber for mac. 

From Jabber for iPhone or Android it is not working. I allways get a occupied tone.

From the CISCO 8821 i also get an occupied tone after the T302 timer is passed. 

That is what i dont understand - because i think if it is a problem with the provider (Headers - ppi-pai) nothing should work?!

 

But on all of the devices named above the inbound calls working fine.

 

I also have the same behaviour if I set "Call Forward All" on an extension to my mobile.

My mobile number is +49172XXXX872.

 

Or has this something to do with the dialpeers on the CUBE and the route Pattern on my CUCM?

 

Thanks for your quick response.

 

 

 


@Chris9811 wrote:

That is what i dont understand - because i think if it is a problem with the provider (Headers - ppi-pai) nothing should work?!


The reason I mention these headers is that for a normal outbound call you will be presenting your own number in these headers, a number which the carrier may recognise as valid.  Whereas these redirected calls either via Mobility or by CFA are presenting the original caller's number, not one of your own.

Here's where we get into the non-standard nature of SIP because different carriers have different requirements for redirected calls.

Ok so i need to talk again with my Provider. In this Thread we already had done this and nothing has changed on my site since we solved the issue.

https://community.cisco.com/t5/ip-telephony-and-phones/cisco2951-as-cube-setup-clip-no-screening-diversion-header/m-p/4023207#M386097

 

But all this doesnt explain why i can not call outbound with the 8821 wireless Phone and the Jabber for iPhone and Android?

 

Thanks for your help so far.

 


@Chris9811 wrote:

Ok so i need to talk again with my Provider. In this Thread we already had done this and nothing has changed on my site since we solved the issue.

https://community.cisco.com/t5/ip-telephony-and-phones/cisco2951-as-cube-setup-clip-no-screening-diversion-header/m-p/4023207#M386097

OK, in that previous thread you had to tweak the Diversion header and from what I can tell it looks like your final working format was ..

Diversion: "Christoph"<sip:2XXXX146t7@192.168.83.200>;privacy=off;reason=follow-me;screen=yes

The Invite that I quoted showed this, which is not the same.  

Diversion: "Mr. XXX"<sip:7811@192.168.83.200>;privacy=off;reason=follow-me;screen=yes

In fact it look like what you were sending before you made your fix in February


@Chris9811 wrote:

But all this doesnt explain why i can not call outbound with the 8821 wireless Phone and the Jabber for iPhone and Android?

 


I think we probably need to see SIP debugs for call attempts from each of these.  Did they ever work, and if so then did they stop working at the same time as your diverted calls stopped working?

Yes they did stop working at the same time as the diverted calls stopped working.

Please see attached a picture where we can see the 408 Timeout and i think that timeout is comeing from my CUCM and not from the ITSP.

 

Diverted call:

2020-05-26 12_23_24-Cisco Collaboration Trace Translator.png

Call from Jabber on iPhone

2020-05-26 15_00_45-.png

 


@Chris9811 wrote:

Yes they did stop working at the same time as the diverted calls stopped working.

Please see attached a picture where we can see the 408 Timeout and i think that timeout is comeing from my CUCM and not from the ITSP.

Based on your labelling, 408 Timeout is sent from CUBE to CUCM.  It's being sent two seconds after the last Invite sent to 217.10.68.150 with no reply.  I guess you have it set to two retries and 500 ms timeout.

Your trace for the redirected call now doesn't show a Diversion header at all.

Would need to see the actual debugs for the Jabber call to be able to comment, but I think it would be worth comparing Invites between a working and non-working normal outbound call.  Maybe go so far as to configure the same DN on both Jabber and on a working phone.

How's your Jabber connecting to CUCM, is it via Expressways or is it on your network?

Where can i set these timers? Under voice service voip on the CUBE or on the CUCM?

 

Yes thats true because my ITSP told me that they only need to get the PPI and it has to work.

 

My Jabber connects to the CUCM over WiFi in my network. 

Expressway is also not working outbound but for these test i always connect the iPhone over WiFi, because i think the Expressway problem is related to this "normal outbound problem"?!


@Chris9811 wrote:

Where can i set these timers? Under voice service voip on the CUBE or on the CUCM?

 


These specific timers are set under "sip-ua" on the CUBE.  Remember that the timeout doubles with each retry.  For example ..

sip-ua
 retry invite 2
 timers trying 350

This will send an Invite, if it doesn't receive Trying within 350 ms it sends another Invite, then waits 700 ms before the second retry, and if there's no answer to that in 1400 ms it gives up.  It's the doubling of the timer at each retry that makes the default so absurdly long, something like 32 seconds before it gives up, which makes any sort of fail over to alternate route pretty much impossible.


@Chris9811 wrote:

Here are the debugs: 

 Nothing stands out as different between the working and non-working.  What strikes me is that you're getting no reply at all in he non working cases, if a header was wrong somewhere I'd expect an error response of some sort.  So this gets me thinking are their replies being sent but not getting to you for some reason. 

Do these calls all use the same outbound dial peer?

I can see some IP address inconsistencies, the media address is given in SDP as 24.134.12.225, I assume that's the outside interface of the CUBE.  But other fields show 192.168.83.254 which I presume is your internal address.  Could there be some reason that the provider is trying to reply to 192.168.83.254 which presumably isn't reachable from their side?

By the way, it might be better to put long debugs in as file attachments.

Ayodeji Okanlawon
VIP Alumni
VIP Alumni

First of all great job to @TONY SMITH  for the help he has given you so far. Having read some part of the thread and looking at the logs, the issue in scenarios like this is down to the diversion header.

The Diversion header though is not used for CLI/ANI presentation is rather very critical and when presents supersedes the other CLI headers ie RPID, PPI, PAI etc. This is used in most times to authenticate the caller and if the number in the diversion header is not part of your DDI range your provider will not allow the call to go through.

As you can see the number presented in the diversion header is not part of your DDI range...

 

Sent: 
INVITE sip:+4917XXXX9872@217.10.68.150:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.83.254:5060;branch=z9hG4bK8095A4
From: "036482374326" <sip:22XXXXt7@217.10.68.150>;tag=4F38D18-219A
To: <sip:22XXXXt7@217.10.68.150>
Date: Mon, 25 May 2020 08:12:05 GMT
Call-ID: 41461754-9D9611EA-90D29AC8-546A44B2@192.168.83.254
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 1782251008-0000065536-0000000150-3360925888
User-Agent: Cisco-SIPGateway/IOS-15.7.3.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Timestamp: 1590394325
Contact: <sip:036482374326@192.168.83.254:5060>
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="22XXXXt7",realm="sipconnect.sipgate.de",uri="sip:+4917XXXX9872@217.10.68.150:5060",response="2f48fb412c4bfcee1e1177c2687fd50d",nonce="Xst/AV7LfdVcD+rDi7NMLoGm6bp5oxs3",algorithm=md5
Max-Forwards: 64
P-Preferred-Identity: "036482374326" <sip:4936482374326@192.168.83.254>
Diversion: "Mr. XXX"<sip:7811@192.168.83.200>;privacy=off;reason=follow-me;screen=yes
Session-ID: 5a25609346e650b1b89bec77e38e129f;remote=00000000000000000000000000000000 

 You will need to configure a SIP profile and use the sip profile to massage the diversion header to present a valid DDI.

 

voice class sip-profiles 200

request INVITE sip-header Diversion modify "<sip:.*@.*>" "<sip:4936482374326@192.168.83.254>"

 

NB: I just put the 49XXXX number there, you can change this to any of your valid DDI

Then apply the sip-profile to your outbound ITSP dial-peer

Please rate all useful posts


@Ayodeji Okanlawon wrote:

 


If you look at the OP's previous thread you'll see he went through all that and got it working.  Following your suggestion in fact!

https://community.cisco.com/t5/ip-telephony-and-phones/cisco2951-as-cube-setup-clip-no-screening-diversion-header/m-p/4023207#M386097

So now it looks like bits are falling off the previously working configuration, as early in this current thread the Diversion headers were back to being incorrect, and more recently missing altogether.

I think we need to see a full copy of the CUBE configuration.

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