05-25-2020 01:57 AM
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
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
Solved! Go to Solution.
05-27-2020 02:36 AM
@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?
05-25-2020 08:43 AM
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.
05-25-2020 12:16 PM - edited 05-25-2020 12:19 PM
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.
05-26-2020 12:37 AM
@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.
05-26-2020 01:13 AM
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.
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.
05-26-2020 02:00 AM
@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.
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
05-26-2020 02:26 AM
@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?
05-26-2020 06:10 AM - edited 05-27-2020 12:30 PM
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:
Call from Jabber on iPhone
05-26-2020 07:05 AM
@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?
05-26-2020 10:52 PM - edited 05-27-2020 12:31 PM
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"?!
05-27-2020 01:17 AM
@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.
05-27-2020 02:36 AM
@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?
05-27-2020 06:54 AM
By the way, it might be better to put long debugs in as file attachments.
05-27-2020 09:09 AM
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
05-27-2020 09:22 AM
@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!
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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