[OSL | CCIE_Voice] no DTMF Relay to UC via PSTN during HA --
Hi All .. In Lab 5 Hand book -- 5th Lab, Not Unity Connection not recognizing the password (no DTMF) when the call is routed as following during a high availability situation. DTMF via SIP Trunk works fine. Call flow with DTMF Problem: SiteB PH2 --- MGCP T1 Port of SiteB GW My PSTN GW (use to switch call between all sites via pots dialpeers) - SiteA H323 GW - CUCM SUB Unity Connection. * The Unity Connection is playing Message -- Enter you PIN * Unity Connection recognizes SiteB PH2 is a registered user's number , so asks for password * When pressing password unity connection does not recognize that any key is pressed *debug mgcp packets *output at SITEB GW : (as shown in bold i pressed the PIN followed by # ---777#) BR1RTR is SiteB GW 10.131.150.11 is CUCM SUB Aug 9 09:28:14.407: MGCP Packet sent to 10.131.150.11:2427--- NTFY 230605050 *@BR1RTR MGCP 0.1 X: 0 O: --- Aug 9 09:28:14.443: MGCP Packet received from 10.131.150.11:2427--- 200 230605050 --- Aug 9 09:28:18.147: MGCP Packet received from 10.131.150.11:2427--- CRCX 404 S0/SU1/DS1-0/1@BR1RTR MGCP 0.1 C: D2080af600F50021 X: 1 L: p:20, a:G.729b, s:off, t:b8, fxr/fx:t38 M: recvonly R: D/[0-9ABCD*#] Q: process,loop --- Aug 9 09:28:18.171: MGCP Packet sent to 10.131.150.11:2427--- 200 404 OK I: 2C v=0 o=- 44 0 IN IP4 10.131.150.234 s=Cisco SDP 0 c=IN IP4 10.131.150.234 t=0 0 m=audio 19578 RTP/AVP 18 100 a=rtpmap:18 G.729/8000 a=fmtp:18 annexb=yes a=rtpmap:100 X-NSE/8000 a=fmtp:100 200-202 a=X-sqn:0 a=X-cap: 1 audio RTP/AVP 100 a=X-cpar: a=rtpmap:100 X-NSE/8000 a=X-cpar: a=fmtp:100 200-202 a=X-cap: 2 image udptl t38 --- Aug 9 09:28:18.555: MGCP Packet received from 10.131.150.11:2427--- MDCX 405 S0/SU1/DS1-0/1@BR1RTR MGCP 0.1 C: D2080af600F50021 I: 2C X: 1 L: p:20, a:G.729b, s:off, t:b8, fxr/fx:t38 M: sendrecv R: D/[0-9ABCD*#], FXR/t38 S: Q: process,loop v=0 o=- 44 0 IN EPN S0/SU1/DS1-0/1@BR1RTR s=Cisco SDP 0 t=0 0 m=audio 23676 RTP/AVP 18 c=IN IP4 10.131.150.164 a=X-sqn:0 a=X-cap:1 image udptl t38 --- Aug 9 09:28:18.567: MGCP Packet sent to 10.131.150.11:2427--- 200 405 OK --- Aug 9 09:28:20.799: MGCP Packet received from 10.131.150.11:2427--- RQNT 406 S0/SU1/DS1-0/1@BR1RTR MGCP 0.1 X: 1 R: D/[0-9ABCD*#], FXR/t38 *S: D/7* Q: process,loop --- Aug 9 09:28:20.807: MGCP Packet sent to 10.131.150.11:2427--- 200 406 OK --- Aug 9 09:28:21.223: MGCP Packet received from 10.131.150.11:2427--- RQNT 407 S0/SU1/DS1-0/1@BR1RTR MGCP 0.1 X: 1 R: D/[0-9ABCD*#], FXR/t38 *S: D/7* Q: process,loop --- Aug 9 09:28:21.227: MGCP Packet sent to 10.131.150.11:2427--- 200 407 OK --- Aug 9 09:28:21.619: MGCP Packet received from 10.131.150.11:2427--- RQNT 408 S0/SU1/DS1-0/1@BR1RTR MGCP 0.1 X: 1 R: D/[0-9ABCD*#], FXR/t38 *S: D/7* Q: process,loop --- Aug 9 09:28:21.623: MGCP Packet sent to 10.131.150.11:2427--- 200 408 OK --- Aug 9 09:28:22.331: MGCP Packet received from 10.131.150.11:2427--- RQNT 409 S0/SU1/DS1-0/1@BR1RTR MGCP 0.1 X: 1 R: D/[0-9ABCD*#], FXR/t38 *S: D/#* Q: process,loop --- Aug 9 09:28:22.335: MGCP Packet sent to 10.131.150.11:2427--- 200 409 OK --- Aug 9 09:28:25.235: MGCP Packet received from 10.131.150.11:2427--- MDCX 410 S0/SU1/DS1-0/1@BR1RTR MGCP 0.1 C: D2080af600F50021 I: 2C X: 1 M: recvonly R: D/[0-9ABCD*#] Q: process,loop --- Aug 9 09:28:25.243: MGCP Packet sent to 10.131.150.11:2427--- 200 410 OK v=0 o=- 44 1 IN IP4 10.131.150.234 s=Cisco SDP 0 c=IN IP4 10.131.150.234 t=0 0 m=audio 19578 RTP/AVP 18 a=rtpmap:18 G.729/8000 a=fmtp:18 annexb=yes a=X-sqn:0 a=X-cap: 1 audio RTP/AVP 100 a=X-cpar: a=rtpmap:100 X-NSE/8000 a=X-cpar: a=fmtp:100 200-202 a=X-cap: 2 image udptl t38 --- Aug 9 09:28:25.423: MGCP Packet received from 10.131.150.11:2427--- DLCX 411 S0/SU1/DS1-0/1@BR1RTR MGCP 0.1 C: D2080af600F50021 I: 2C X: 1 S: --- Aug 9 09:28:25.455: MGCP Packet sent to 10.131.150.11:2427--- 250 411 OK P: PS=141, OS=2282, PR=349, OR=6980, PL=0, JI=7, LA=0 --- Aug 9 09:28:44.407: MGCP Packet sent to 10.131.150.11:2427--- NTFY 230605051 *@BR1RTR MGCP 0.1 X: 0 O: --- - *debug voip dsm dsp* has no output on my PSTN GW ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] no DTMF Relay to UC via PSTN during HA --
DTMF Signaling Method[image: Required Field] OOB RFC 2833 The above configuration in SIP Trunk to Unity Connection solves the issue as the Call to unity connection was going via out of band DTMF in MGCP GW .. SiteB PH2 --- MGCP T1 Port of SiteB GW My PSTN GW (use to switch call between all sites via pots dialpeers) - SiteA H323 GW - CUCM SUB Unity Connection. Regards, Aman On Fri, Aug 9, 2013 at 3:34 PM, aman sinha aman.i...@gmail.com wrote: Hi All .. In Lab 5 Hand book -- 5th Lab, Not Unity Connection not recognizing the password (no DTMF) when the call is routed as following during a high availability situation. DTMF via SIP Trunk works fine. Call flow with DTMF Problem: SiteB PH2 --- MGCP T1 Port of SiteB GW My PSTN GW (use to switch call between all sites via pots dialpeers) - SiteA H323 GW - CUCM SUB Unity Connection. * The Unity Connection is playing Message -- Enter you PIN * Unity Connection recognizes SiteB PH2 is a registered user's number , so asks for password * When pressing password unity connection does not recognize that any key is pressed *debug mgcp packets *output at SITEB GW : (as shown in bold i pressed the PIN followed by # ---777#) BR1RTR is SiteB GW 10.131.150.11 is CUCM SUB Aug 9 09:28:14.407: MGCP Packet sent to 10.131.150.11:2427--- NTFY 230605050 *@BR1RTR MGCP 0.1 X: 0 O: --- Aug 9 09:28:14.443: MGCP Packet received from 10.131.150.11:2427--- 200 230605050 --- Aug 9 09:28:18.147: MGCP Packet received from 10.131.150.11:2427--- CRCX 404 S0/SU1/DS1-0/1@BR1RTR MGCP 0.1 C: D2080af600F50021 X: 1 L: p:20, a:G.729b, s:off, t:b8, fxr/fx:t38 M: recvonly R: D/[0-9ABCD*#] Q: process,loop --- Aug 9 09:28:18.171: MGCP Packet sent to 10.131.150.11:2427--- 200 404 OK I: 2C v=0 o=- 44 0 IN IP4 10.131.150.234 s=Cisco SDP 0 c=IN IP4 10.131.150.234 t=0 0 m=audio 19578 RTP/AVP 18 100 a=rtpmap:18 G.729/8000 a=fmtp:18 annexb=yes a=rtpmap:100 X-NSE/8000 a=fmtp:100 200-202 a=X-sqn:0 a=X-cap: 1 audio RTP/AVP 100 a=X-cpar: a=rtpmap:100 X-NSE/8000 a=X-cpar: a=fmtp:100 200-202 a=X-cap: 2 image udptl t38 --- Aug 9 09:28:18.555: MGCP Packet received from 10.131.150.11:2427--- MDCX 405 S0/SU1/DS1-0/1@BR1RTR MGCP 0.1 C: D2080af600F50021 I: 2C X: 1 L: p:20, a:G.729b, s:off, t:b8, fxr/fx:t38 M: sendrecv R: D/[0-9ABCD*#], FXR/t38 S: Q: process,loop v=0 o=- 44 0 IN EPN S0/SU1/DS1-0/1@BR1RTR s=Cisco SDP 0 t=0 0 m=audio 23676 RTP/AVP 18 c=IN IP4 10.131.150.164 a=X-sqn:0 a=X-cap:1 image udptl t38 --- Aug 9 09:28:18.567: MGCP Packet sent to 10.131.150.11:2427--- 200 405 OK --- Aug 9 09:28:20.799: MGCP Packet received from 10.131.150.11:2427--- RQNT 406 S0/SU1/DS1-0/1@BR1RTR MGCP 0.1 X: 1 R: D/[0-9ABCD*#], FXR/t38 *S: D/7* Q: process,loop --- Aug 9 09:28:20.807: MGCP Packet sent to 10.131.150.11:2427--- 200 406 OK --- Aug 9 09:28:21.223: MGCP Packet received from 10.131.150.11:2427--- RQNT 407 S0/SU1/DS1-0/1@BR1RTR MGCP 0.1 X: 1 R: D/[0-9ABCD*#], FXR/t38 *S: D/7* Q: process,loop --- Aug 9 09:28:21.227: MGCP Packet sent to 10.131.150.11:2427--- 200 407 OK --- Aug 9 09:28:21.619: MGCP Packet received from 10.131.150.11:2427--- RQNT 408 S0/SU1/DS1-0/1@BR1RTR MGCP 0.1 X: 1 R: D/[0-9ABCD*#], FXR/t38 *S: D/7* Q: process,loop --- Aug 9 09:28:21.623: MGCP Packet sent to 10.131.150.11:2427--- 200 408 OK --- Aug 9 09:28:22.331: MGCP Packet received from 10.131.150.11:2427--- RQNT 409 S0/SU1/DS1-0/1@BR1RTR MGCP 0.1 X: 1 R: D/[0-9ABCD*#], FXR/t38 *S: D/#* Q: process,loop --- Aug 9 09:28:22.335: MGCP Packet sent to 10.131.150.11:2427--- 200 409 OK --- Aug 9 09:28:25.235: MGCP Packet received from 10.131.150.11:2427--- MDCX 410 S0/SU1/DS1-0/1@BR1RTR MGCP 0.1 C: D2080af600F50021 I: 2C X: 1 M: recvonly R: D/[0-9ABCD*#] Q: process,loop --- Aug 9 09:28:25.243: MGCP Packet sent to 10.131.150.11:2427--- 200 410 OK v=0 o=- 44 1 IN IP4 10.131.150.234 s=Cisco SDP 0 c=IN IP4 10.131.150.234 t=0 0 m=audio 19578 RTP/AVP 18 a=rtpmap:18 G.729/8000 a=fmtp:18 annexb=yes a=X-sqn:0 a=X-cap: 1 audio RTP/AVP 100 a=X-cpar: a=rtpmap:100 X-NSE/8000 a=X-cpar: a=fmtp:100 200-202 a=X-cap: 2 image udptl t38 --- Aug 9 09:28:25.423: MGCP Packet received from 10.131.150.11:2427--- DLCX 411 S0/SU1/DS1-0/1@BR1RTR MGCP 0.1 C: D2080af600F50021 I: 2C X: 1 S: --- Aug 9 09:28:25.455: MGCP Packet sent to 10.131.150.11:2427--- 250 411 OK P: PS=141, OS=2282, PR=349, OR=6980, PL=0, JI=7, LA=0 --- Aug 9 09:28:44.407: MGCP Packet sent to 10.131.150.11:2427--- NTFY 230605051 *@BR1RTR MGCP 0.1 X: 0 O: --- - *debug voip dsm dsp* has no output on my PSTN GW ___ For more information regarding
[OSL | CCIE_Voice] Ip expert video
Hi Can anyone tell me where can i find the VIK MALHI Ip Expert VOice Troubleshooting Video.. -- Regards, Dharambir Kumar ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Ip expert video
http://www.ipexpert.com/ They accept all major Credit cards On Fri, Aug 9, 2013 at 4:56 AM, Dharambir kumar varma dharambi...@gmail.com wrote: Hi Can anyone tell me where can i find the VIK MALHI Ip Expert VOice Troubleshooting Video.. -- Regards, Dharambir Kumar ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Extension Mobility on CUCME
Hello, I am trying to setup the Extension Mobility on CME, but when I press the Mobility key, it shows key is not active here is my config *telephony-service* * no auto-reg-ephone* * authentication credential username password* * em keep-history* * max-ephones 1* * max-dn 2 no-reg both* * ip source-address 10.0.38.254 port 2000* * service phone webAccess 0* * system message ITASSISTANT* * url authentication http://10.0.38.254/CCMCIP/authenticate.asp username password* * load 7945 flash0:term45.default.loads* * time-format 24* * date-format dd-mm-yy* * max-conferences 8 gain -6* * dn-webedit * * transfer-system full-consult* * create cnf-files* *!* *voice logout-profile 400* * pin 2400* * user 2400 password cisco* * number 250032400 type normal* *!* *voice user-profile 2400* * pin 2400* * user 250032400 password 2400* * number 250032400 type normal* *!* *ephone 1* * device-security-mode none* * mac-address 0021.55D6.05AE* * ephone-template 1* * type 7945* * no auto-line* * logout-profile 400* Please let me know what I am missing. Thanks, Viki ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Guide Me
Hi All, I have one branch site At UK and on HQ site at Mumbai. when i call from India to UK , two way audio is perfect. but whe the call comes from UK to India, Audio is intermittent,Uk user can not hear but india user is hearing. There is One firewall at UK and one firewall at India through IPSEC. Regards, Dharambir Kumar ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Guide Me
https://supportforums.cisco.com On Fri, Aug 9, 2013 at 8:31 AM, Dharambir kumar varma dharambi...@gmail.com wrote: Hi All, I have one branch site At UK and on HQ site at Mumbai. when i call from India to UK , two way audio is perfect. but whe the call comes from UK to India, Audio is intermittent,Uk user can not hear but india user is hearing. There is One firewall at UK and one firewall at India through IPSEC. Regards, Dharambir Kumar ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] LiveRecord
hi folks. After I press Live Record and press disconnected to end conversation , why the Live Record session still stay? is this expected or any configuration we need? K ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Live Record.
Karen, To stop the live record session you should press the live record softkey again and it will end the recording and send to voicemail. If you just disconnect the recording will continue. --Michael Message: 6 Date: Fri, 9 Aug 2013 07:27:47 -0700 (PDT) From: Karen Johnson karen.johnson...@yahoo.ca To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] LiveRecord Message-ID: 1376058467.46146.yahoomail...@web163906.mail.gq1.yahoo.com Content-Type: text/plain; charset=us-ascii hi folks. After I press Live Record and press disconnected to end conversation , why the Live Record session still stay? is this expected or any configuration we need? K ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] lab 5 sip + call
i am not able to connect call to pstn... is it IOS issue or my config issue/ please help me...when trying without + call working... but when aplly + called number call failes.. what could be issue? -- Thanks Regard's Amit Sharma ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] BACD limit max 2 call
all, is there a way to limit so BACD can only accept 2 call ? i have used -max-conn under dial-peer -param queue-len under sript app-b-acd however it still play Thanks for calling then reject the call. Can we achieve rejecting call right away, without play Thanks for calling ? K ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] BACD limit max 2 call
Hi, Thats the way you do it to fulfil your requirement ccm-manager music-on-hold ephone-hunt 1 longest-idle pilot 4500 list 4101,4102 timeout 10,10 auto logout 2 dynamic application service app-b-acd param number-of-hunt-grps 1 param second-greeting-time 40 param aa-hunt1 4500 param queue-len 2 param queue-manager-debugs 1 ! service app-b-acd-aa paramspace english index 1 paramspace english language en paramspace english location flash: param service-name app-b-acd param handoff-string app-b-acd-aa param aa-pilot 4000 param number-of-hunt-grps 1 param dial-by-extension-option 1 param second-greeting-time 32 param call-retry-timer 10 param max-time-call-retry 60 param max-time-vm-retry 2 param voice-mail *4001 param drop-through-option 1 param drop-through-prompt _bacd_welcome.au ! dial-peer voice 4000 voip service app-b-acd-aa destination-pattern 4000 session target ipv4:142.102.66.254 incoming called-number 4000 dtmf-relay h245-alphanumeric codec g711ulaw On 9 August 2013 16:43, Karen Johnson karen.johnson...@yahoo.ca wrote: all, is there a way to limit so BACD can only accept 2 call ? i have used -max-conn under dial-peer -param queue-len under sript app-b-acd however it still play Thanks for calling then reject the call. Can we achieve rejecting call right away, without play Thanks for calling ? K ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] mva
I know there is an issue that can be created with single number reach using the standard local route group, so you might (somehow) be hitting a related issue. Just a guess for you to try. On Aug 9, 2013 12:51 AM, Somphol Boonjing somp...@gmail.com wrote: On Fri, Aug 9, 2013 at 1:15 PM, Alex Mendoza aa.mend...@icloud.comwrote: Calling from my SNR to MVA is working, MVA asks for my pin number, then press 1, after that I dialed internal 4 digit extension but this internal phone only shows the caller number and not the caller name. I think is normal behavior, but when a calling from my SNR directly to a internal extension, it shows the caller number and the caller id. I am seeing the same thing for my MVA setup.I also presume this is expected behavior, but I'm not able to find any bug report or any concrete Cisco document to back it up though. Some people seems to report that the name will display after the call is connected. I can't reproduce that behavior, just the caller number for me during ringing and connected state of the call --- when made via MVA pilot number. Call directly from SNR number, i.e. Enterprise Feature Access, seems to be no problem with both Calling Name and Number. I'll be interested to see whether anyone else have different outcome. Regards, --Somphol. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] BACD limit max 2 call
thanks, yes this work From: Hesham Abdelkereem heshamcentr...@gmail.com To: Karen Johnson karen.johnson...@yahoo.ca Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Friday, August 9, 2013 6:22:38 PM Subject: Re: [OSL | CCIE_Voice] BACD limit max 2 call Hi, Thats the way you do it to fulfil your requirement ccm-manager music-on-hold ephone-hunt 1 longest-idle pilot 4500 list 4101,4102 timeout 10,10 auto logout 2 dynamic application service app-b-acd param number-of-hunt-grps 1 param second-greeting-time 40 param aa-hunt1 4500 param queue-len 2 param queue-manager-debugs 1 ! service app-b-acd-aa paramspace english index 1 paramspace english language en paramspace english location flash: param service-name app-b-acd param handoff-string app-b-acd-aa param aa-pilot 4000 param number-of-hunt-grps 1 param dial-by-extension-option 1 param second-greeting-time 32 param call-retry-timer 10 param max-time-call-retry 60 param max-time-vm-retry 2 param voice-mail *4001 param drop-through-option 1 param drop-through-prompt _bacd_welcome.au ! dial-peer voice 4000 voip service app-b-acd-aa destination-pattern 4000 session target ipv4:142.102.66.254 incoming called-number 4000 dtmf-relay h245-alphanumeric codec g711ulaw On 9 August 2013 16:43, Karen Johnson karen.johnson...@yahoo.ca wrote: all, is there a way to limit so BACD can only accept 2 call ? i have used -max-conn under dial-peer -param queue-len under sript app-b-acd however it still play Thanks for calling then reject the call. Can we achieve rejecting call right away, without play Thanks for calling ? K ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
