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- About Frank Hohenadel
10-27-2017
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04-02-2013
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Created by
Frank Hohenadel
in IP Telephony and Phones
in IP Telephony and Phones
02-19-2015
02:00 AM
02-19-2015
02:00 AM
Unfortunately not. Seems that this is not working within CME environments Best, Frank
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11-18-2014
12:43 AM
Hi Neno, the radius server is located within an Synology NAS (as Synology packet). The NAS itself is located inside the network. My requirement is to limit the SSL VPN access to certain users located in the radius database. Thanks, Frank
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Created by
Frank Hohenadel
in Policy and Access
in Policy and Access
11-17-2014
11:40 AM
11-17-2014
11:40 AM
Hi, we are running a 2911 router with radius authentication for SSL as well as for VPN access. Is there a possibility to grant only specific users (out of the radius DB) access to the server ? Thanks, Frank
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Created by
Frank Hohenadel
in Switching
in Switching
11-16-2014
07:48 AM
11-16-2014
07:48 AM
Does anyone know if its possible to force a router only to assign an infinite DHCP address when a client connect to the network? In the configuration (2911 router) there is an DHCP server configured which assigns specific IP addresses to certain mac addresses (in this example to Client1 with mac address 00aa.00bb.00cc.ab). Due to security reasons I want to avoid that a user of a device can manually set a different ip address on his device and still be able to use/connect to the router (so far this is possible). DHCP config is: ip dhcp pool data network 192.168.2.0 255.255.255.0 default-router 192.168.2.1 dns-server 10.10.10.1 ! ip dhcp pool Client1 host 192.168.2.10 255.255.255.0 client-identifier 00aa.00bb.00cc.ab dns-server 10.10.10.1 default-router 192.168.2.1 Thanks, Frank
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Created by
Frank Hohenadel
in IP Telephony and Phones
in IP Telephony and Phones
03-01-2014
10:53 AM
03-01-2014
10:53 AM
where have you found the files on the website? Regards, Frank Sent from Cisco Technical Support iPad App
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Created by
Frank Hohenadel
in IP Telephony and Phones
in IP Telephony and Phones
01-18-2014
01:58 AM
01-18-2014
01:58 AM
Hi, I'm looking for a solution for inbound fax2mail on an 2900 router with an ISM module (CME/CUE). So far I managed to convert faxes from analog fax machines to get converted into an TiF image and send them by mail (using onramp.tcl script). For incoming SIP calls I havn't found anything yet. Any ideas? Regards, Frank
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Created by
Frank Hohenadel
in IP Telephony and Phones
in IP Telephony and Phones
12-26-2013
09:36 AM
12-26-2013
09:36 AM
That would be great ! Regards, Frank
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Created by
Frank Hohenadel
in IP Telephony and Phones
in IP Telephony and Phones
12-26-2013
02:15 AM
12-26-2013
02:15 AM
Could it be the case that this is only working with CUCM and not with CME ? Regards, Frank
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Created by
Frank Hohenadel
in IP Telephony and Phones
in IP Telephony and Phones
12-22-2013
09:14 AM
12-22-2013
09:14 AM
Hi, does anyone know if its possible to syncronize call-lists (placed calls, missed calls etc) between phones sharing the same line? Regards, Frank
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Created by
Frank Hohenadel
in IP Telephony and Phones
in IP Telephony and Phones
11-15-2013
01:43 PM
11-15-2013
01:43 PM
Hi, Rob, thanks for the reply / I found that information already in other posts - unfortunatelyit doesn't help. The phone loads the featurepolicy1.xml file but does not show the configured softkeys - really strange Regards, Frank
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Created by
Frank Hohenadel
in IP Telephony and Phones
in IP Telephony and Phones
11-15-2013
07:38 AM
11-15-2013
07:38 AM
Hi, I'm facing problems while trying to define softkeys (e.g. transfer or dnd) for the 9971 with cme 9.1. 1. trying to define softkeys by specifying them in the voice register template command doesnt work voice register template 1 softkeys idle cfwdall redial newcall voice register pool 1 type 9971 template 1 number 1 dn 1 number 2 dn 2 2. Another issue is the second dn - there I have different softkeys then on dn 1 (call transfer is missing) Most of the posts here say that its not possible to define softkeys with the template command. Whats the alternative? Any help would be highly appreciated Regards, Frank
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Created by
Frank Hohenadel
in IP Telephony and Phones
in IP Telephony and Phones
11-13-2013
09:16 AM
11-13-2013
09:16 AM
Hi, my CUE/CUCME MWI integration does not work as expected. After recording a voicemail the CUE system sends an SIP MWI NOTIFY to the CUCME and the red envelope symbol gets activated on the wrong line number (in my case dn 1 instead of dn 2) - phone is 9971. Resetting the CUE and configuring it form scratch didn't help. Somehow the whole 9971 integration seams a little bit unstable (call transfer does not work - no idea if there is anything in parallel). Configuration and debug is attached Thanks for any ideas. Regards, Frank ******************Configuration************************* version 15.2 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! boot-start-marker boot-end-marker ! ! no logging buffered ! aaa new-model ! ! aaa authentication login default local aaa authorization exec default local ! ! ! ! ! aaa session-id common network-clock-participate wic 1 mmi polling-interval 60 no mmi auto-configure no mmi pvc mmi snmp-timeout 180 ! ip dhcp relay information trust-all ip dhcp excluded-address 10.10.10.1 10.10.10.10 ! ip dhcp pool phone network 10.10.10.0 255.255.255.0 default-router 10.10.10.1 dns-server 217.0.43.97 217.0.43.113 option 150 ip 10.10.10.1 ! ! ip name-server 217.0.43.97 ip cef no ipv6 cef ! multilink bundle-name authenticated ! ! voice rtp send-recv ! voice service voip no ip address trusted authenticate callmonitor no callmonitor trace allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip no supplementary-service sip handle-replaces sip registrar server expires max 3600 min 180 asymmetric payload full video screening ! voice class codec 1 codec preference 1 g711alaw codec preference 2 g711ulaw codec preference 3 g723r53 codec preference 4 g729r8 codec preference 5 g729br8 codec preference 6 g726r32 codec preference 7 g728 ! ! voice register global mode cme source-address 10.10.10.1 port 5060 max-dn 10 max-pool 4 load 9971 sip9971.9-4-1-9 load ATA-187 ATA187.9-2-3-1 authenticate register authenticate realm all timezone 21 time-format 24 date-format D/M/Y voicemail 999 tftp-path flash: file text create profile sync 0000420080667255 network-locale DE user-locale DE camera video overlap-signal ! voice register dn 1 number 91 allow watch no-reg mwi ! voice register dn 2 number 90 call-forward b2bua noan 999 timeout 20 allow watch no-reg mwi voice register pool 1 registration-timer max 300 min 60 id mac 0008.3030.6AAA feature-button 5 DnD session-transport tcp type 9971 number 1 dn 1 number 2 dn 2 dtmf-relay rtp-nte voice-class codec 1 no vad camera video ! ! ! ! no ip ftp passive ! ! ! interface FastEthernet0/0 description ***ISP Provider*** ip address dhcp no ip proxy-arp ip nat outside ip virtual-reassembly in duplex auto speed auto ! interface Integrated-Service-Engine0/0 ip unnumbered Vlan1 ip nat inside ip virtual-reassembly in service-module ip address 10.10.10.2 255.255.255.0 service-module ip default-gateway 10.10.10.1 ! interface Vlan1 ip address 10.10.10.1 255.255.255.0 ip nat inside ip virtual-reassembly in ! ip forward-protocol nd ip http server ip http authentication aaa ip http secure-server ip http timeout-policy idle 60 life 86400 requests 10000 ip http path flash: ! ! ip nat inside source list 1 interface FastEthernet0/0 overload ip route 0.0.0.0 0.0.0.0 87.186.225.92 254 ip route 10.10.10.2 255.255.255.255 Integrated-Service-Engine0/0 access-list 1 permit 10.10.10.0 0.0.0.255 ! ! control-plane ! ! ! mgcp profile default ! ! dial-peer voice 20 voip description **Mailbox** destination-pattern 99. session protocol sipv2 session target ipv4:10.10.10.2 voice-class codec 1 dtmf-relay rtp-nte no vad ! ! presence presence call-list ! sip-ua credentials username ***number*** password ***password*** realm **realm** authentication username ***user*** password ***password*** realm ***realm** nat symmetric role active nat symmetric check-media-src no remote-party-id retry invite 4 retry response 3 retry bye 2 retry cancel 2 retry notify 6 retry register 5 retry options 10 timers connect 100 timers notify 100 timers register 250 mwi-server ipv4:10.10.10.2 expires 3600 port 5060 transport udp registrar 1 dns:**registrar** expires 3600 sip-server dns:**registrar** connection-reuse host-registrar presence enable ! ! ! telephony-service no auto-reg-ephone max-ephones 12 max-dn 30 ip source-address 10.10.10.1 port 2000 max-redirect 5 timeouts interdigit 2 url services http://10.10.10.2/voiceview/common/login.do cnf-file location flash: user-locale U1 load CME-locale-de_DE-German-8.8.2.5.tar load 7925 CP7925G-1.4.4.3.LOADS time-zone 23 time-format 24 date-format dd-mm-yy voicemail 999 max-conferences 4 gain -6 dn-webedit time-webedit transfer-system full-consult create cnf-files version-stamp Jan 01 2002 00:00:00 ! ! ! line con 0 no modem enable line aux 0 line 2 no activation-character no exec transport preferred none transport input all transport output pad telnet rlogin lapb-ta mop udptn v120 ssh line vty 0 4 access-class 102 in privilege level 15 transport input all transport output all line vty 5 15 access-class 101 in exec-timeout 30 0 privilege level 15 transport input all transport output all ! ! end ********************************** debug ccsip messages ************************************** NOTIFY sip:90@10.10.10.1:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bKSLiy2tv6m.ZgKxUe8Y2eLQ~~144 Max-Forwards: 70 To: <sip:90@10.10.10.1>;tag=A7BEBF8-B74 From: <sip:90@10.10.10.2>;tag=daf14882-1160 Call-ID: BDAE6318-4BB311E3-8B318DCC-E4968F82@10.10.10.1 CSeq: 5 NOTIFY Content-Length: 109 Contact: sip:90@10.10.10.2 Event: message-summary Allow-Events: refer Allow-Events: message-summary Allow-Events: telephone-event Subscription-State: active Content-Type: application/simple-message-summary Messages-Waiting: yes Message-Account: sip:90@10.10.10.2 Voice-Message: 1/0 (0/0) Fax-Message: 0/0 (0/0) *Nov 13 18:08:29.743: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bKSLiy2tv6m.ZgKxUe8Y2eLQ~~144 From: <sip:90@10.10.10.2>;tag=daf14882-1160 To: <sip:90@10.10.10.1>;tag=A7BEBF8-B74 Date: Wed, 13 Nov 2013 18:08:29 GMT Call-ID: BDAE6318-4BB311E3-8B318DCC-E4968F82@10.10.10.1 CSeq: 5 NOTIFY Content-Length: 0 *Nov 13 18:08:29.747: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: NOTIFY sip:90@10.10.10.12:52895;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 10.10.10.1:5060;branch=z9hG4bK250B21A8 From: <sip:90@10.10.10.1>;tag=AEFE09C-119E To: <sip:90@10.10.10.12> Call-ID: 6F329427-4BC511E3-8C258DCC-E4968F82@10.10.10.1 CSeq: 101 NOTIFY Max-Forwards: 70 Date: Wed, 13 Nov 2013 18:08:29 GMT User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5 Event: message-summary Subscription-State: active Contact: <sip:90@10.10.10.1:5060;transport=tcp> Content-Type: application/simple-message-summary Content-Length: 23 Messages-Waiting: yes *Nov 13 18:08:29.751: //0/000000000000/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 OK Via: SIP/2.0/TCP 10.10.10.1:5060;branch=z9hG4bK250B21A8 From: <sip:90@10.10.10.1>;tag=AEFE09C-119E To: <sip:90@10.10.10.12> Call-ID: 6F329427-4BC511E3-8C258DCC-E4968F82@10.10.10.1 Date: Wed, 13 Nov 2013 18:08:31 GMT CSeq: 101 NOTIFY Content-Length: 0
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Created by
Frank Hohenadel
in IP Telephony and Phones
in IP Telephony and Phones
11-11-2013
10:53 AM
11-11-2013
10:53 AM
Hi, I'm still struggeling on dtmf tones within my router. While the SCCP Phones are working well, the SIP phones do not generate any DTMF tones. Problem occurs external and internal (e.g. when the SIP Phone calls the mailbox) Unfortunately the first on-hand solution (adding "pass-thru content sdp" in voice service voip / sip) didn't worked out. After adding this there is no more internal dial (SIP to SCCP) possible. Configuration and debug attached. Any ideas ? Regards, Frank ****************** Configuration ************************ version 15.2 service timestamps debug datetime msec service timestamps log datetime msec ! ! boot-start-marker boot-end-marker ! ! no logging buffered ! aaa new-model ! ! aaa authentication login default local aaa authorization exec default local ! ! ! aaa session-id common network-clock-participate wic 1 mmi polling-interval 60 no mmi auto-configure no mmi pvc mmi snmp-timeout 180 ! ! ! ip dhcp relay information trust-all ip dhcp excluded-address 10.10.10.1 10.10.10.10 ! ip dhcp pool phone network 10.10.10.0 255.255.255.0 default-router 10.10.10.1 dns-server 217.0.43.97 217.0.43.113 option 150 ip 10.10.10.1 ! ! ip name-server 217.0.43.97 ip cef no ipv6 cef ! multilink bundle-name authenticated ! ! voice rtp send-recv ! voice service voip ip address trusted list no ip address trusted authenticate callmonitor no callmonitor trace allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip no supplementary-service sip handle-replaces sip registrar server expires max 3600 min 180 asymmetric payload full video screening ! voice class codec 1 codec preference 1 g711alaw codec preference 2 g711ulaw codec preference 3 g723r53 codec preference 4 g729r8 codec preference 5 g729br8 codec preference 6 g726r32 codec preference 7 g728 ! ! voice register global mode cme source-address 10.10.10.1 port 5060 max-dn 10 max-pool 4 load 9971 sip9971.9-3-2SR1-1 authenticate register authenticate realm all timezone 21 time-format 24 date-format D/M/Y voicemail 999 tftp-path flash: file text create profile sync 0000420080667251 network-locale DE user-locale DE camera video overlap-signal ! voice register dn 1 number 91 allow watch no-reg mwi ! voice register pool 1 registration-timer max 300 min 60 id mac 0008.3030.6AAA session-transport tcp type 9971 number 1 dn 1 template 1 voice-class codec 1 username cisco password cisco no vad camera video ! ! ! voice translation-rule 1 rule 2 /^\(\+..\)\(.*\)/ /0\2/ ! voice translation-rule 2 rule 1 /123456/ /91/ ! voice translation-rule 3 rule 1 /^.*/ /123456/ ! ! voice translation-profile IN translate calling 1 translate called 2 ! voice translation-profile OUT translate calling 3 ! ! voice-card 0 ! ! license udi pid C1861-UC-B/K9-MS sn FHK144871U3 file privilege 0 ! ! no ip ftp passive ! ! interface FastEthernet0/0 description ***ISP Provider*** ip address dhcp no ip proxy-arp ip nat outside ip virtual-reassembly in duplex auto speed auto ! interface Integrated-Service-Engine0/0 ip unnumbered Vlan1 ip nat inside ip virtual-reassembly in service-module ip address 10.10.10.2 255.255.255.0 service-module ip default-gateway 10.10.10.1 ! ! interface Vlan1 ip address 10.10.10.1 255.255.255.0 ip nat inside ip virtual-reassembly in ! ip forward-protocol nd ip http server ip http authentication aaa ip http secure-server ip http timeout-policy idle 60 life 86400 requests 10000 ip http path flash: ! ! ip nat inside source list 1 interface FastEthernet0/0 overload ip route 0.0.0.0 0.0.0.0 87.186.225.92 254 ip route 10.10.10.2 255.255.255.255 Integrated-Service-Engine0/0 ! access-list 1 permit 10.10.10.0 0.0.0.255 access-list 1 permit 192.168.2.0 0.0.0.255 ! ! ! control-plane ! ! voice-port 0/0/0 cptone DE station-id number 94 caller-id enable ! voice-port 0/0/1 ! voice-port 0/0/2 ! voice-port 0/0/3 ! voice-port 0/1/0 ! voice-port 0/1/1 ! voice-port 0/4/0 auto-cut-through signal immediate input gain auto-control description Music On Hold Port ! ! mgcp profile default ! ! dial-peer voice 4 voip description **Outgoing VOIP** translation-profile outgoing OUT destination-pattern ...T session protocol sipv2 session target dns:tel.t-online.de voice-class sip dtmf-relay force rtp-nte dtmf-relay rtp-nte codec g711alaw no vad ! dial-peer voice 10 voip description **Incoming VOIP** translation-profile incoming IN session protocol sipv2 session target ipv4:10.10.10.1 incoming called-number 123456 voice-class sip dtmf-relay force rtp-nte dtmf-relay rtp-nte no vad ! ! dial-peer voice 20 voip description **Mailbox** destination-pattern 99. session protocol sipv2 session target ipv4:10.10.10.2 voice-class codec 1 dtmf-relay rtp-nte no vad ! dial-peer voice 19 pots destination-pattern 94 port 0/0/0 no sip-register ! ! presence presence call-list ! sip-ua credentials username 123456 password ***password*** realm tel.t-online.de authentication username ***user*** password ***password*** realm tel.t-online.de nat symmetric role active nat symmetric check-media-src no remote-party-id retry invite 4 retry response 3 retry bye 2 retry cancel 2 retry notify 6 retry register 5 retry options 10 timers connect 100 timers notify 100 timers register 250 mwi-server ipv4:10.10.10.2 expires 3600 port 5060 transport udp registrar 1 dns:tel.t-online.de expires 3600 sip-server dns:tel.t-online.de connection-reuse host-registrar presence enable ! ! ! telephony-service no auto-reg-ephone max-ephones 12 max-dn 30 ip source-address 10.10.10.1 port 2000 max-redirect 5 timeouts interdigit 2 url services http://10.10.10.2/voiceview/common/login.do cnf-file location flash: user-locale U1 load CME-locale-de_DE-German-8.8.2.5.tar load 7925 CP7925G-1.4.4.3.LOADS time-zone 23 time-format 24 date-format dd-mm-yy voicemail 999 max-conferences 4 gain -6 dn-webedit time-webedit transfer-system full-consult create cnf-files version-stamp 7960 Nov 11 2013 14:07:24 ! ! ! ephone-dn 1 dual-line number 92 no-reg primary label IP-Mobil-1 description IP-Mobil-1 name IP-Mobil-1 allow watch ! ! ephone-dn 2 dual-line number 93 no-reg primary label IP-Mobil-2 description IP-Mobil-2 name IP-Mobil-2 allow watch ! ! ! ephone 1 device-security-mode none description IP-Mobil-1 mac-address E05F.B9BC.74CE max-calls-per-button 4 type 7925 button 1:1 ! ! ! ephone 2 device-security-mode none description IP-Mobil-2 mac-address E05F.B9BB.F4D9 max-calls-per-button 4 type 7925 button 1:2 ! ! ! line con 0 no modem enable line aux 0 line 2 no activation-character no exec transport preferred none transport input all transport output pad telnet rlogin lapb-ta mop udptn v120 ssh line vty 0 4 access-class 102 in privilege level 15 transport input all transport output all line vty 5 15 access-class 101 in exec-timeout 30 0 privilege level 15 transport input all transport output all ! ! end *************************************************************************************** **********debug (SIP Phone calls Unity Express Mailbox System) *********** *Nov 11 19:29:14.642: //-1/6226DD5D8211/SIP/Media/sipSPICopyStunConfigFromPeerToCCB: Firewall traversal is not enabledimageattr parse payload numtok not foundimageattr payload found, specific is 255 *Nov 11 19:29:14.642: //556/6226DD5D8211/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1 *Nov 11 19:29:14.642: //556/6226DD5D8211/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 2 = 10.10.10.1 *Nov 11 19:29:14.642: //556/6226DD5D8211/SIP/Media/sipSPIDoMediaNegotiation: DO_Media_Negotiation: video m-line detected. *Nov 11 19:29:14.642: //556/6226DD5D8211/SIP/Media/sipSPIUpdCallWithSdpInfo: Preferred Codec : g711alaw, bytes :160 Preferred DTMF relay : inband-voice Preferred NTE payload : 101 Early Media : No Delayed Media : No Bridge Done : No New Media : No DSP DNLD Reqd : No *Nov 11 19:29:14.646: //556/6226DD5D8211/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1 *Nov 11 19:29:14.646: //556/6226DD5D8211/SIP/Media/sipSPISelectCodecVersion: g729r8 flavor of g729 codec will be used *Nov 11 19:29:14.646: //556/6226DD5D8211/SIP/Media/sipSPISelectCodecVersion: g722-64 flavor of G722 codec will be used *Nov 11 19:29:14.646: //556/6226DD5D8211/SIP/Media/sipSPIDisplayStreamInfo: Stream type : voice-only Media line : 1 State : STREAM_ADDING (2) Stream address type : 1 Callid : 556 Peer Callid : -1 RTP/SRTP Negotiated : 8 Negotiated Codec : g711alaw, bytes :160 Nego. Codec payload : 8 (tx), 8 (rx) Negotiated DTMF relay : inband-voice Negotiated NTE payload : 0 (tx), 0 (rx) Negotiated CN payload : 0 Media Srce Addr/Port : [10.10.10.1]:0 Media Dest Addr/Port : [10.10.10.12]:17166 *Nov 11 19:29:14.646: //556/6226DD5D8211/SIP/Media/sipSPIDisplayStreamInfo: Stream type : video Media line : 2 State : STREAM_ADDING (2) Stream address type : 1 Callid : 557 Peer Callid : -1 RTP/SRTP Negotiated : 8 Negotiated Codec : h264, bytes Nego. Codec payload : 97 (tx), 119 (rx) Negotiated DTMF relay : inband-voice Negotiated NTE payload : 0 (tx), 0 (rx) Negotiated CN payload : 0 Media Srce Addr/Port : [10.10.10.1]:0 Media Dest Addr/Port : [10.10.10.12]:28736 *Nov 11 19:29:14.646: //556/6226DD5D8211/SIP/Media/sipSPIUpdCallWithSdpInfo: Stream type : voice-only Media line : 1 State : STREAM_ADDING (2) Stream address type : 1 Callid : 556 Negotiated Codec : g711alaw, bytes :160 Nego. Codec payload : 8 (tx), 8 (rx) Negotiated DTMF relay : inband-voice Negotiated NTE payload : 0 (tx), 0 (rx) Negotiated CN payload : 0 Media Srce Addr/Port : [10.10.10.1]:0 Media Dest Addr/Port : [10.10.10.12]:17166 *Nov 11 19:29:14.646: //556/6226DD5D8211/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 2 = 10.10.10.1 *Nov 11 19:29:14.646: //556/6226DD5D8211/SIP/Media/sipSPIUpdCallWithSdpInfo: Stream type : video Media line : 2 State : STREAM_ADDING (2) Stream address type : 1 Callid : 557 Negotiated Codec : h264, bytes Nego. Codec payload : 97 (tx), 119 (rx) Negotiated DTMF relay : inband-voice Negotiated NTE payload : 0 (tx), 0 (rx) Negotiated CN payload : 0 Media Srce Addr/Port : [10.10.10.1]:0 Media Dest Addr/Port : [10.10.10.12]:28736 *Nov 11 19:29:14.646: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 16662 for stream 1 *Nov 11 19:29:14.646: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 16664 for stream 2 *Nov 11 19:29:14.650: //556/6226DD5D8211/SIP/Media/sipSPIDisplayStreamInfo: Stream type : voice-only Media line : 1 State : STREAM_ADDING (2) Stream address type : 1 Callid : 556 Peer Callid : -1 RTP/SRTP Negotiated : 8 Negotiated Codec : g711alaw, bytes :160 Nego. Codec payload : 8 (tx), 8 (rx) Negotiated DTMF relay : inband-voice Negotiated NTE payload : 0 (tx), 0 (rx) Negotiated CN payload : 0 Media Srce Addr/Port : [10.10.10.1]:16662 Media Dest Addr/Port : [10.10.10.12]:17166 *Nov 11 19:29:14.650: //556/6226DD5D8211/SIP/Media/sipSPIDisplayStreamInfo: Stream type : video Media line : 2 State : STREAM_ADDING (2) Stream address type : 1 Callid : 557 Peer Callid : -1 RTP/SRTP Negotiated : 8 Negotiated Codec : h264, bytes Nego. Codec payload : 97 (tx), 119 (rx) Negotiated DTMF relay : inband-voice Negotiated NTE payload : 0 (tx), 0 (rx) Negotiated CN payload : 0 Media Srce Addr/Port : [10.10.10.1]:16664 Media Dest Addr/Port : [10.10.10.12]:28736 *Nov 11 19:29:14.682: //558/6226DD5D8211/SIP/Media/sipSPICopyStunConfigFromPeerToCCB: Firewall traversal is not enabled *Nov 11 19:29:14.682: //558/6226DD5D8211/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1 *Nov 11 19:29:14.682: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 16666 for stream 1 *Nov 11 19:29:14.682: //558/6226DD5D8211/SIP/Media/sipSPIAddSDPMediaPayload: Preferred method of dtmf relay is: 6, with payload: 101 *Nov 11 19:29:14.682: //558/6226DD5D8211/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 2 = 10.10.10.1 *Nov 11 19:29:14.682: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 16668 for stream 2 *Nov 11 19:29:14.682: //558/6226DD5D8211/SIP/Media/sipSPI_ipip_UpdateSDPVideoMediaPayload: update ccb->pld.negotiated_bandwidth with current_max_bit_rate (100bps):10000 *Nov 11 19:29:14.682: //558/6226DD5D8211/SIP/Media/sipSPI_ipip_UpdateSDPVideoMediaPayload: VIDEO m-line 2, set ccb negotiated_bandwidth (kbps):1000; current_max_bit_rate:10000, local_bandwidth_req (kbps):0, remote_bandwidth_req (kbps):0 *Nov 11 19:29:14.682: //558/6226DD5D8211/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_IDLE *Nov 11 19:29:14.682: //558/6226DD5D8211/SIP/Media/sipSPIProcessRtpSessions: No active streams. *Nov 11 19:29:14.686: //558/6226DD5D8211/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_ADDING *Nov 11 19:29:14.686: //558/6226DD5D8211/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice+dtmf (callid 558) to the VOIP RTP library *Nov 11 19:29:14.686: //558/6226DD5D8211/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1 *Nov 11 19:29:14.686: //558/6226DD5D8211/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1 *Nov 11 19:29:14.690: //558/6226DD5D8211/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info laddr = 10.10.10.1, lport = 16666, raddr = 0.0.0.0, rport=0, do_rtcp=FALSE src_callid = 558, dest_callid = -1, stream type = voice+dtmf, stream direction = RECVONLY media_ip_addr = - , vrf tableid = 0 media_addr_type = 1 negotiated_bandwidth (kbps) = 0 *Nov 11 19:29:14.690: //558/6226DD5D8211/SIP/Media/sipSPIUpdateRtcpSession: No rtp session, creating a new one *Nov 11 19:29:14.690: //558/6226DD5D8211/SIP/Media/sipSPICreateRtpSession: stun is disabled *Nov 11 19:29:14.690: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: Voice quality monitoring is not enabled for this RTP session due to sdp passthru enabled *Nov 11 19:29:14.690: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: VQM: gccb=0x0, gccb->callId=0, ccb->ccCallID=558 *Nov 11 19:29:14.806: //558/6226DD5D8211/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1 *Nov 11 19:29:14.810: //558/6226DD5D8211/SIP/Media/sipSPIReplaceSDP: Main stream got changed & it's Flow Around *Nov 11 19:29:14.810: //558/6226DD5D8211/SIP/Media/sipSPIUpdCallWithSdpInfo: Preferred Codec : g711alaw, bytes :160 Preferred DTMF relay : rtp-nte Preferred NTE payload : 101 Early Media : No Delayed Media : No Bridge Done : No New Media : No DSP DNLD Reqd : No *Nov 11 19:29:14.810: //558/6226DD5D8211/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1 *Nov 11 19:29:14.810: //558/6226DD5D8211/SIP/Media/sipSPIDisplayStreamInfo: Stream type : voice+dtmf Media line : 1 State : STREAM_ADDING (2) Stream address type : 1 Callid : 558 Peer Callid : -1 RTP/SRTP Negotiated : 8 Negotiated Codec : g711ulaw, bytes :160 Nego. Codec payload : 0 (tx), 0 (rx) Negotiated DTMF relay : rtp-nte Negotiated NTE payload : 101 (tx), 101 (rx) Negotiated CN payload : 0 Media Srce Addr/Port : [10.10.10.1]:16666 Media Dest Addr/Port : [10.10.10.2]:21046 *Nov 11 19:29:14.810: //556/6226DD5D8211/SIP/Media/sipSPISetStreamInfo: 0 Active Streams *Nov 11 19:29:14.810: //556/6226DD5D8211/SIP/Media/sipSPISetStreamInfo: Number of active streams is zero (0)! *Nov 11 19:29:14.810: //556/6226DD5D8211/SIP/Media/sipSPISetStreamInfo: caps.stream_count=0,caps.stream[0].stream_type=0xFFFF, caps.stream_list.xmitFunc= *Nov 11 19:29:14.810: //556/6226DD5D8211/SIP/Media/sipSPISetStreamInfo: ??unknown??, caps.stream_list.context= *Nov 11 19:29:14.810: //556/6226DD5D8211/SIP/Media/sipSPISetStreamInfo: 0x0 (gccb) *Nov 11 19:29:14.810: //558/6226DD5D8211/SIP/Media/sipSPIUpdCallWithSdpInfo: Stream type : voice+dtmf Media line : 1 State : STREAM_ADDING (2) Stream address type : 1 Callid : 558 Negotiated Codec : g711ulaw, bytes :160 Nego. Codec payload : 0 (tx), 0 (rx) Negotiated DTMF relay : rtp-nte Negotiated NTE payload : 101 (tx), 101 (rx) Negotiated CN payload : 0 Media Srce Addr/Port : [10.10.10.1]:16666 Media Dest Addr/Port : [10.10.10.2]:21046 *Nov 11 19:29:14.814: //558/6226DD5D8211/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8B7786D0 *Nov 11 19:29:14.818: //556/6226DD5D8211/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_ADDING *Nov 11 19:29:14.818: //556/6226DD5D8211/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice-only (callid 556) to the VOIP RTP library *Nov 11 19:29:14.818: //556/6226DD5D8211/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1 *Nov 11 19:29:14.818: //556/6226DD5D8211/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1 *Nov 11 19:29:14.818: //556/6226DD5D8211/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info laddr = 10.10.10.1, lport = 16662, raddr = 10.10.10.12, rport=17166, do_rtcp=TRUE src_callid = 556, dest_callid = 558, stream type = voice-only, stream direction = SENDRECV media_ip_addr = 10.10.10.12, vrf tableid = 0 media_addr_type = 1 negotiated_bandwidth (kbps) = 0 *Nov 11 19:29:14.818: //556/6226DD5D8211/SIP/Media/sipSPIUpdateRtcpSession: No rtp session, creating a new one *Nov 11 19:29:14.818: //556/6226DD5D8211/SIP/Media/sipSPICreateRtpSession: stun is disabled *Nov 11 19:29:14.818: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: Voice quality monitoring is not enabled for this RTP session due to sdp passthru enabled *Nov 11 19:29:14.818: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: VQM: gccb=0x0, gccb->callId=0, ccb->ccCallID=556 *Nov 11 19:29:14.818: //556/6226DD5D8211/SIP/Media/sipSPIGetNewLocalMediaDirection: New Remote Media Direction = SENDRECV Present Local Media Direction = SENDRECV New Local Media Direction = SENDRECV retVal = 0 *Nov 11 19:29:14.822: //558/6226DD5D8211/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_ADDING *Nov 11 19:29:14.822: //558/6226DD5D8211/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice+dtmf (callid 558) to the VOIP RTP library *Nov 11 19:29:14.822: //558/6226DD5D8211/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1 *Nov 11 19:29:14.822: //558/6226DD5D8211/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1 *Nov 11 19:29:14.822: //558/6226DD5D8211/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info laddr = 10.10.10.1, lport = 16666, raddr = 10.10.10.2, rport=21046, do_rtcp=TRUE src_callid = 558, dest_callid = 556, stream type = voice+dtmf, stream direction = SENDRECV media_ip_addr = 10.10.10.2, vrf tableid = 0 media_addr_type = 1 negotiated_bandwidth (kbps) = 0 *Nov 11 19:29:14.822: //558/6226DD5D8211/SIP/Media/sipSPIUpdateRtcpSession: RTP session already created - update *Nov 11 19:29:14.822: //558/6226DD5D8211/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8B7786D0 *Nov 11 19:29:14.822: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: VQM: gccb=0x0, gccb->callId=0, ccb->ccCallID=558 *Nov 11 19:29:14.822: //558/6226DD5D8211/SIP/Media/sipSPIGetNewLocalMediaDirection: New Remote Media Direction = SENDRECV Present Local Media Direction = SENDRECV New Local Media Direction = SENDRECV retVal = 0 *Nov 11 19:29:14.850: //558/6226DD5D8211/SIP/Media/act_update_nat_media: NEW: rtp session to be updated with the new src info, old addr:port [10.10.10.2]:21046 , newaddr:port [10.10.10.2]:21046 *Nov 11 19:29:14.850: //558/6226DD5D8211/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8B7786D0 *Nov 11 19:29:14.898: //556/6226DD5D8211/SIP/Media/act_update_nat_media: NEW: rtp session to be updated with the new src info, old addr:port [10.10.10.12]:17166 , newaddr:port [10.10.10.12]:17166 *Nov 11 19:29:14.902: //556/6226DD5D8211/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8D6F25F8 *Nov 11 19:29:34.322: //556/6226DD5D8211/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8D6F25F8 *Nov 11 19:29:34.322: //558/6226DD5D8211/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8B7786D0 *Nov 11 19:29:34.326: //556/6226DD5D8211/SIP/Media/sipSPIHandleDestroyRtpSession: stream:8D6F25F8 *Nov 11 19:29:34.338: //558/6226DD5D8211/SIP/Media/sipSPIHandleDestroyRtpSession: stream:8B7786D0 *Nov 11 19:31:36.042: //565/B61D0194821B/SIP/Media/sipSPICopyStunConfigFromPeerToCCB: Firewall traversal is not enabled *Nov 11 19:31:36.042: //565/B61D0194821B/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1 *Nov 11 19:31:36.042: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 16670 for stream 1 *Nov 11 19:31:36.042: //565/B61D0194821B/SIP/Media/sipSPIAddSDPMediaPayload: Preferred method of dtmf relay is: 6, with payload: 101 *Nov 11 19:31:36.042: //565/B61D0194821B/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_IDLE *Nov 11 19:31:36.042: //565/B61D0194821B/SIP/Media/sipSPIProcessRtpSessions: No active streams. *Nov 11 19:31:36.046: //565/B61D0194821B/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_ADDING *Nov 11 19:31:36.046: //565/B61D0194821B/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice+dtmf (callid 565) to the VOIP RTP library *Nov 11 19:31:36.046: //565/B61D0194821B/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1 *Nov 11 19:31:36.046: //565/B61D0194821B/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1 *Nov 11 19:31:36.046: //565/B61D0194821B/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info laddr = 10.10.10.1, lport = 16670, raddr = 0.0.0.0, rport=0, do_rtcp=FALSE src_callid = 565, dest_callid = -1, stream type = voice+dtmf, stream direction = RECVONLY media_ip_addr = - , vrf tableid = 0 media_addr_type = 1 negotiated_bandwidth (kbps) = 0 *Nov 11 19:31:36.046: //565/B61D0194821B/SIP/Media/sipSPIUpdateRtcpSession: No rtp session, creating a new one *Nov 11 19:31:36.046: //565/B61D0194821B/SIP/Media/sipSPICreateRtpSession: stun is disabled *Nov 11 19:31:36.046: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: Voice quality monitoring is not enabled for this RTP session due to sdp passthru enabled *Nov 11 19:31:36.046: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: VQM: gccb=0x0, gccb->callId=0, ccb->ccCallID=565 *Nov 11 19:31:36.170: //565/B61D0194821B/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1 *Nov 11 19:31:36.170: //565/B61D0194821B/SIP/Media/sipSPIReplaceSDP: Main stream got changed & it's Flow Around *Nov 11 19:31:36.170: //565/B61D0194821B/SIP/Media/sipSPIUpdCallWithSdpInfo: Preferred Codec : g711alaw, bytes :160 Preferred DTMF relay : rtp-nte Preferred NTE payload : 101 Early Media : No Delayed Media : No Bridge Done : No New Media : No DSP DNLD Reqd : No *Nov 11 19:31:36.174: //565/B61D0194821B/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1 *Nov 11 19:31:36.174: //565/B61D0194821B/SIP/Media/sipSPIUpdCallWithSdpInfo: Stream type : voice+dtmf Media line : 1 State : STREAM_ADDING (2) Stream address type : 1 Callid : 565 Negotiated Codec : g711ulaw, bytes :160 Nego. Codec payload : 0 (tx), 0 (rx) Negotiated DTMF relay : rtp-nte Negotiated NTE payload : 101 (tx), 101 (rx) Negotiated CN payload : 0 Media Srce Addr/Port : [10.10.10.1]:16670 Media Dest Addr/Port : [10.10.10.2]:21114 *Nov 11 19:31:36.174: //565/B61D0194821B/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_ADDING *Nov 11 19:31:36.178: //565/B61D0194821B/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice+dtmf (callid 565) to the VOIP RTP library *Nov 11 19:31:36.178: //565/B61D0194821B/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1 *Nov 11 19:31:36.178: //565/B61D0194821B/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1 *Nov 11 19:31:36.178: //565/B61D0194821B/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info laddr = 10.10.10.1, lport = 16670, raddr = 10.10.10.2, rport=21114, do_rtcp=TRUE src_callid = 565, dest_callid = 564, stream type = voice+dtmf, stream direction = SENDRECV media_ip_addr = 10.10.10.2, vrf tableid = 0 media_addr_type = 1 negotiated_bandwidth (kbps) = 0 *Nov 11 19:31:36.178: //565/B61D0194821B/SIP/Media/sipSPIUpdateRtcpSession: RTP session already created - update *Nov 11 19:31:36.178: //565/B61D0194821B/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8D6C478C *Nov 11 19:31:36.178: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: VQM: gccb=0x0, gccb->callId=0, ccb->ccCallID=565 *Nov 11 19:31:36.178: //565/B61D0194821B/SIP/Media/sipSPIGetNewLocalMediaDirection: New Remote Media Direction = SENDRECV Present Local Media Direction = SENDRECV New Local Media Direction = SENDRECV retVal = 0 *Nov 11 19:31:36.178: //565/B61D0194821B/SIP/Media/sipSPISetStreamInfo: 1 Active Streams *Nov 11 19:31:36.178: //565/B61D0194821B/SIP/Media/sipSPISetStreamInfo: Adding stream type (voice+dtmf) from media line 1 codec g711ulaw *Nov 11 19:31:36.178: //565/B61D0194821B/SIP/Media/sipSPISetStreamInfo: caps.stream_count=1,caps.stream[0].stream_type=0x3, caps.stream_list.xmitFunc= *Nov 11 19:31:36.178: //565/B61D0194821B/SIP/Media/sipSPISetStreamInfo: voip_rtp_xmit, caps.stream_list.context= *Nov 11 19:31:36.178: //565/B61D0194821B/SIP/Media/sipSPISetStreamInfo: 0x8D6F1880 (gccb) *Nov 11 19:31:36.218: //565/B61D0194821B/SIP/Media/act_update_nat_media: NEW: rtp session to be updated with the new src info, old addr:port [10.10.10.2]:21114 , newaddr:port [10.10.10.2]:21114 *Nov 11 19:31:36.218: //565/B61D0194821B/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8D6C478C *Nov 11 19:31:43.474: //565/B61D0194821B/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8D6C478C *Nov 11 19:31:43.490: //565/B61D0194821B/SIP/Media/sipSPIHandleDestroyRtpSession: stream:8D6C478C *Nov 11 19:45:41.206: //-1/AE3093EE8227/SIP/Media/sipSPICopyStunConfigFromPeerToCCB: Firewall traversal is not enabledimageattr parse payload numtok not foundimageattr payload found, specific is 255 *Nov 11 19:45:41.206: //589/AE3093EE8227/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1 *Nov 11 19:45:41.210: //589/AE3093EE8227/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 2 = 10.10.10.1 *Nov 11 19:45:41.210: //589/AE3093EE8227/SIP/Media/sipSPIDoMediaNegotiation: DO_Media_Negotiation: video m-line detected. *Nov 11 19:45:41.210: //589/AE3093EE8227/SIP/Media/sipSPIUpdCallWithSdpInfo: Preferred Codec : g711alaw, bytes :160 Preferred DTMF relay : inband-voice Preferred NTE payload : 101 Early Media : No Delayed Media : No Bridge Done : No New Media : No DSP DNLD Reqd : No *Nov 11 19:45:41.210: //589/AE3093EE8227/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1 *Nov 11 19:45:41.210: //589/AE3093EE8227/SIP/Media/sipSPISelectCodecVersion: g729r8 flavor of g729 codec will be used *Nov 11 19:45:41.210: //589/AE3093EE8227/SIP/Media/sipSPISelectCodecVersion: g722-64 flavor of G722 codec will be used *Nov 11 19:45:41.210: //589/AE3093EE8227/SIP/Media/sipSPIDisplayStreamInfo: Stream type : voice-only Media line : 1 State : STREAM_ADDING (2) Stream address type : 1 Callid : 589 Peer Callid : -1 RTP/SRTP Negotiated : 8 Negotiated Codec : g711alaw, bytes :160 Nego. Codec payload : 8 (tx), 8 (rx) Negotiated DTMF relay : inband-voice Negotiated NTE payload : 0 (tx), 0 (rx) Negotiated CN payload : 0 Media Srce Addr/Port : [10.10.10.1]:0 Media Dest Addr/Port : [10.10.10.12]:20200 *Nov 11 19:45:41.210: //589/AE3093EE8227/SIP/Media/sipSPIDisplayStreamInfo: Stream type : video Media line : 2 State : STREAM_ADDING (2) Stream address type : 1 Callid : 590 Peer Callid : -1 RTP/SRTP Negotiated : 8 Negotiated Codec : h264, bytes Nego. Codec payload : 97 (tx), 119 (rx) Negotiated DTMF relay : inband-voice Negotiated NTE payload : 0 (tx), 0 (rx) Negotiated CN payload : 0 Media Srce Addr/Port : [10.10.10.1]:0 Media Dest Addr/Port : [10.10.10.12]:22496 *Nov 11 19:45:41.210: //589/AE3093EE8227/SIP/Media/sipSPIUpdCallWithSdpInfo: Stream type : voice-only Media line : 1 State : STREAM_ADDING (2) Stream address type : 1 Callid : 589 Negotiated Codec : g711alaw, bytes :160 Nego. Codec payload : 8 (tx), 8 (rx) Negotiated DTMF relay : inband-voice Negotiated NTE payload : 0 (tx), 0 (rx) Negotiated CN payload : 0 Media Srce Addr/Port : [10.10.10.1]:0 Media Dest Addr/Port : [10.10.10.12]:20200 *Nov 11 19:45:41.210: //589/AE3093EE8227/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 2 = 10.10.10.1 *Nov 11 19:45:41.210: //589/AE3093EE8227/SIP/Media/sipSPIUpdCallWithSdpInfo: Stream type : video Media line : 2 State : STREAM_ADDING (2) Stream address type : 1 Callid : 590 Negotiated Codec : h264, bytes Nego. Codec payload : 97 (tx), 119 (rx) Negotiated DTMF relay : inband-voice Negotiated NTE payload : 0 (tx), 0 (rx) Negotiated CN payload : 0 Media Srce Addr/Port : [10.10.10.1]:0 Media Dest Addr/Port : [10.10.10.12]:22496 *Nov 11 19:45:41.210: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 16672 for stream 1 *Nov 11 19:45:41.210: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 16674 for stream 2 *Nov 11 19:45:41.214: //589/AE3093EE8227/SIP/Media/sipSPIDisplayStreamInfo: Stream type : voice-only Media line : 1 State : STREAM_ADDING (2) Stream address type : 1 Callid : 589 Peer Callid : -1 RTP/SRTP Negotiated : 8 Negotiated Codec : g711alaw, bytes :160 Nego. Codec payload : 8 (tx), 8 (rx) Negotiated DTMF relay : inband-voice Negotiated NTE payload : 0 (tx), 0 (rx) Negotiated CN payload : 0 Media Srce Addr/Port : [10.10.10.1]:16672 Media Dest Addr/Port : [10.10.10.12]:20200 *Nov 11 19:45:41.214: //589/AE3093EE8227/SIP/Media/sipSPIDisplayStreamInfo: Stream type : video Media line : 2 State : STREAM_ADDING (2) Stream address type : 1 Callid : 590 Peer Callid : -1 RTP/SRTP Negotiated : 8 Negotiated Codec : h264, bytes Nego. Codec payload : 97 (tx), 119 (rx) Negotiated DTMF relay : inband-voice Negotiated NTE payload : 0 (tx), 0 (rx) Negotiated CN payload : 0 Media Srce Addr/Port : [10.10.10.1]:16674 Media Dest Addr/Port : [10.10.10.12]:22496 *Nov 11 19:45:41.230: //591/AE3093EE8227/SIP/Media/sipSPICopyStunConfigFromPeerToCCB: Firewall traversal is not enabled *Nov 11 19:45:41.230: //591/AE3093EE8227/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1 *Nov 11 19:45:41.230: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 16676 for stream 1 *Nov 11 19:45:41.230: //591/AE3093EE8227/SIP/Media/sipSPIAddSDPMediaPayload: Preferred method of dtmf relay is: 6, with payload: 101 *Nov 11 19:45:41.234: //591/AE3093EE8227/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 2 = 10.10.10.1 *Nov 11 19:45:41.234: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 16678 for stream 2 *Nov 11 19:45:41.234: //591/AE3093EE8227/SIP/Media/sipSPI_ipip_UpdateSDPVideoMediaPayload: update ccb->pld.negotiated_bandwidth with current_max_bit_rate (100bps):10000 *Nov 11 19:45:41.234: //591/AE3093EE8227/SIP/Media/sipSPI_ipip_UpdateSDPVideoMediaPayload: VIDEO m-line 2, set ccb negotiated_bandwidth (kbps):1000; current_max_bit_rate:10000, local_bandwidth_req (kbps):0, remote_bandwidth_req (kbps):0 *Nov 11 19:45:41.234: //591/AE3093EE8227/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_IDLE *Nov 11 19:45:41.234: //591/AE3093EE8227/SIP/Media/sipSPIProcessRtpSessions: No active streams. *Nov 11 19:45:41.238: //591/AE3093EE8227/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_ADDING *Nov 11 19:45:41.238: //591/AE3093EE8227/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice+dtmf (callid 591) to the VOIP RTP library *Nov 11 19:45:41.238: //591/AE3093EE8227/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1 *Nov 11 19:45:41.238: //591/AE3093EE8227/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1 *Nov 11 19:45:41.238: //591/AE3093EE8227/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info laddr = 10.10.10.1, lport = 16676, raddr = 0.0.0.0, rport=0, do_rtcp=FALSE src_callid = 591, dest_callid = -1, stream type = voice+dtmf, stream direction = RECVONLY media_ip_addr = - , vrf tableid = 0 media_addr_type = 1 negotiated_bandwidth (kbps) = 0 *Nov 11 19:45:41.238: //591/AE3093EE8227/SIP/Media/sipSPIUpdateRtcpSession: No rtp session, creating a new one *Nov 11 19:45:41.238: //591/AE3093EE8227/SIP/Media/sipSPICreateRtpSession: stun is disabled *Nov 11 19:45:41.238: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: Voice quality monitoring is not enabled for this RTP session due to sdp passthru enabled *Nov 11 19:45:41.238: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: VQM: gccb=0x0, gccb->callId=0, ccb->ccCallID=591 *Nov 11 19:45:41.350: //591/AE3093EE8227/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1 *Nov 11 19:45:41.350: //591/AE3093EE8227/SIP/Media/sipSPIReplaceSDP: Main stream got changed & it's Flow Around *Nov 11 19:45:41.350: //591/AE3093EE8227/SIP/Media/sipSPIUpdCallWithSdpInfo: Preferred Codec : g711alaw, bytes :160 Preferred DTMF relay : rtp-nte Preferred NTE payload : 101 Early Media : No Delayed Media : No Bridge Done : No New Media : No DSP DNLD Reqd : No *Nov 11 19:45:41.350: //591/AE3093EE8227/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1 *Nov 11 19:45:41.350: //591/AE3093EE8227/SIP/Media/sipSPIDisplayStreamInfo: Stream type : voice+dtmf Media line : 1 State : STREAM_ADDING (2) Stream address type : 1 Callid : 591 Peer Callid : -1 RTP/SRTP Negotiated : 8 Negotiated Codec : g711ulaw, bytes :160 Nego. Codec payload : 0 (tx), 0 (rx) Negotiated DTMF relay : rtp-nte Negotiated NTE payload : 101 (tx), 101 (rx) Negotiated CN payload : 0 Media Srce Addr/Port : [10.10.10.1]:16676 Media Dest Addr/Port : [10.10.10.2]:20670 *Nov 11 19:45:41.354: //589/AE3093EE8227/SIP/Media/sipSPISetStreamInfo: 0 Active Streams *Nov 11 19:45:41.354: //589/AE3093EE8227/SIP/Media/sipSPISetStreamInfo: Number of active streams is zero (0)! *Nov 11 19:45:41.354: //589/AE3093EE8227/SIP/Media/sipSPISetStreamInfo: caps.stream_count=0,caps.stream[0].stream_type=0xFFFF, caps.stream_list.xmitFunc= *Nov 11 19:45:41.354: //589/AE3093EE8227/SIP/Media/sipSPISetStreamInfo: ??unknown??, caps.stream_list.context= *Nov 11 19:45:41.354: //589/AE3093EE8227/SIP/Media/sipSPISetStreamInfo: 0x0 (gccb) *Nov 11 19:45:41.354: //591/AE3093EE8227/SIP/Media/sipSPIUpdCallWithSdpInfo: Stream type : voice+dtmf Media line : 1 State : STREAM_ADDING (2) Stream address type : 1 Callid : 591 Negotiated Codec : g711ulaw, bytes :160 Nego. Codec payload : 0 (tx), 0 (rx) Negotiated DTMF relay : rtp-nte Negotiated NTE payload : 101 (tx), 101 (rx) Negotiated CN payload : 0 Media Srce Addr/Port : [10.10.10.1]:16676 Media Dest Addr/Port : [10.10.10.2]:20670 *Nov 11 19:45:41.354: //591/AE3093EE8227/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8D6E4A28 *Nov 11 19:45:41.362: //589/AE3093EE8227/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_ADDING *Nov 11 19:45:41.362: //589/AE3093EE8227/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice-only (callid 589) to the VOIP RTP library *Nov 11 19:45:41.362: //589/AE3093EE8227/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1 *Nov 11 19:45:41.362: //589/AE3093EE8227/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1 *Nov 11 19:45:41.362: //589/AE3093EE8227/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info laddr = 10.10.10.1, lport = 16672, raddr = 10.10.10.12, rport=20200, do_rtcp=TRUE src_callid = 589, dest_callid = 591, stream type = voice-only, stream direction = SENDRECV media_ip_addr = 10.10.10.12, vrf tableid = 0 media_addr_type = 1 negotiated_bandwidth (kbps) = 0 *Nov 11 19:45:41.362: //589/AE3093EE8227/SIP/Media/sipSPIUpdateRtcpSession: No rtp session, creating a new one *Nov 11 19:45:41.362: //589/AE3093EE8227/SIP/Media/sipSPICreateRtpSession: stun is disabled *Nov 11 19:45:41.362: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: Voice quality monitoring is not enabled for this RTP session due to sdp passthru enabled *Nov 11 19:45:41.362: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: VQM: gccb=0x0, gccb->callId=0, ccb->ccCallID=589 *Nov 11 19:45:41.362: //589/AE3093EE8227/SIP/Media/sipSPIGetNewLocalMediaDirection: New Remote Media Direction = SENDRECV Present Local Media Direction = SENDRECV New Local Media Direction = SENDRECV retVal = 0 *Nov 11 19:45:41.362: //591/AE3093EE8227/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_ADDING *Nov 11 19:45:41.362: //591/AE3093EE8227/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice+dtmf (callid 591) to the VOIP RTP library *Nov 11 19:45:41.362: //591/AE3093EE8227/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1 *Nov 11 19:45:41.362: //591/AE3093EE8227/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1 *Nov 11 19:45:41.362: //591/AE3093EE8227/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info laddr = 10.10.10.1, lport = 16676, raddr = 10.10.10.2, rport=20670, do_rtcp=TRUE src_callid = 591, dest_callid = 589, stream type = voice+dtmf, stream direction = SENDRECV media_ip_addr = 10.10.10.2, vrf tableid = 0 media_addr_type = 1 negotiated_bandwidth (kbps) = 0 *Nov 11 19:45:41.362: //591/AE3093EE8227/SIP/Media/sipSPIUpdateRtcpSession: RTP session already created - update *Nov 11 19:45:41.362: //591/AE3093EE8227/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8D6E4A28 *Nov 11 19:45:41.362: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: VQM: gccb=0x0, gccb->callId=0, ccb->ccCallID=591 *Nov 11 19:45:41.362: //591/AE3093EE8227/SIP/Media/sipSPIGetNewLocalMediaDirection: New Remote Media Direction = SENDRECV Present Local Media Direction = SENDRECV New Local Media Direction = SENDRECV retVal = 0 *Nov 11 19:45:41.386: //591/AE3093EE8227/SIP/Media/act_update_nat_media: NEW: rtp session to be updated with the new src info, old addr:port [10.10.10.2]:20670 , newaddr:port [10.10.10.2]:20670 *Nov 11 19:45:41.386: //591/AE3093EE8227/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8D6E4A28 *Nov 11 19:45:41.438: //589/AE3093EE8227/SIP/Media/act_update_nat_media: NEW: rtp session to be updated with the new src info, old addr:port [10.10.10.12]:20200 , newaddr:port [10.10.10.12]:20200 *Nov 11 19:45:41.438: //589/AE3093EE8227/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8EE45620 *Nov 11 19:45:47.322: //589/AE3093EE8227/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8EE45620 *Nov 11 19:45:47.322: //591/AE3093EE8227/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8D6E4A28 *Nov 11 19:45:47.326: //589/AE3093EE8227/SIP/Media/sipSPIHandleDestroyRtpSession: stream:8EE45620 *Nov 11 19:45:47.338: //591/AE3093EE8227/SIP/Media/sipSPIHandleDestroyRtpSession: stream:8D6E4A28
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Created by
Frank Hohenadel
in IP Telephony and Phones
in IP Telephony and Phones
10-20-2013
09:44 AM
10-20-2013
09:44 AM
Hi, I'm looking for a solution to use my different SIP accounts for the different line configurations (dn 1 and dn 2) on the 9971. With dn 1 the system should use voice translation-rule 3 (phone number 06209123456) where for dn 2 the system should use translation-rule 4 (phone number 05209789012) for all outbound calls. So far the system is always using the dial-peer voice 1 voip (where translation profile OUT is assigned). Even removing the translation profile assignment in dial-peer voice 1 voip doesn't help. Configuration is attached. Any ideas? Regards, Frank
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Labels:
10-20-2013
09:30 AM
Finally I made it by modifying the INVITE string by: voice class sip-profiles 10 request INVITE sip-header From modify " phone-number@(.*)>" " phone-number@tel.t-online.de>" request ANY sip-header From modify "Frank" "phone-number" AND voice service voip sip sip-profiles 10 Instead of "phone-number" you have to insert your sip phone number given by T-Online T-Online expects the realm name in the From INVITE string. In addition the callers name needs to be replaced into the sip phone numer Regards, Frank
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11-18-2014
Hi Neno,the radius server is located within an Synology NAS (as Synology
packet). The NAS itself is ...
11-17-2014
Hi,we are running a 2911 router with radius authentication for SSL as
well as for VPN access. Is the...
Created by
Frank Hohenadel
in Switching
11-16-2014
11-16-2014
Does anyone know if its possible to force a router only to assign an
infinite DHCP address when a cl...
Created by
Frank Hohenadel
in IP Telephony and Phones
03-01-2014
03-01-2014
where have you found the files on the website?Regards, FrankSent from
Cisco Technical Support iPad A...
01-18-2014
Hi,I'm looking for a solution for inbound fax2mail on an 2900 router
with an ISM module (CME/CUE).So...
Created by
Frank Hohenadel
in IP Telephony and Phones
12-26-2013
12-26-2013
That would be great !Regards, Frank
Created by
Frank Hohenadel
in IP Telephony and Phones
12-26-2013
12-26-2013
Could it be the case that this is only working with CUCM and not with
CME ?Regards, Frank
Created by
Frank Hohenadel
in IP Telephony and Phones
12-22-2013
12-22-2013
Hi,does anyone know if its possible to syncronize call-lists (placed
calls, missed calls etc) betwee...
Created by
Frank Hohenadel
in IP Telephony and Phones
11-15-2013
11-15-2013
Hi, Rob,thanks for the reply / I found that information already in other
posts - unfortunatelyit doe...
Created by
Frank Hohenadel
in IP Telephony and Phones
11-15-2013
11-15-2013
Hi,I'm facing problems while trying to define softkeys (e.g. transfer or
dnd) for the 9971 with cme ...
Created by
Frank Hohenadel
in IP Telephony and Phones
11-13-2013
11-13-2013
Hi,my CUE/CUCME MWI integration does not work as expected. After
recording a voicemail the CUE syste...
Created by
Frank Hohenadel
in IP Telephony and Phones
11-11-2013
11-11-2013
Hi, I'm still struggeling on dtmf tones within my router. While the SCCP
Phones are working well, th...
Created by
Frank Hohenadel
in IP Telephony and Phones
10-20-2013
10-20-2013
Hi,I'm looking for a solution to use my different SIP accounts for the
different line configurations...
Public Statistics
| Date Registered | 04-02-2013 09:55 AM |
| Date Last Visited | 10-27-2017 03:43 PM |
| Total Messages Posted | 19 |
