Re: [OSL | CCIE_Voice] DTMF from BR1 phone to CUC via SIP trunk notrecognized properly
Thanks Dan. I always had OOB RFC2833 option selected in SIP trunk configuration as there is no option to just select OOB. After reading your response, I started digging CUC side configuration and found that DTMF type can be set under ‘Port Group settings’ as well. I unchecked ‘Use DTMF RFC 2833’ option, while keeping oob method - ‘Use DTMF KPML’ option checked. After this change, DTMF was correctly recognized by CUC. Thanks for your input. -Tapan From: Dan Quinlan (daquinla) Sent: Monday, June 11, 2012 11:58 PM To: Tapan Gautam (tgautam) Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] DTMF from BR1 phone to CUC via SIP trunk notrecognized properly cRTP mangles in-band (audio) DTMF. If I understand correctly, you are SIP-integrated between CUC and UCM. You need to be OOB only for DTMF (not rfc2833). Rfc2833 is an in-band (audio channel) mechanism. You need OOB (signaling channel) for DTMF to function when cRTP is used. DQ d...@cisco.com Sent from my iPhone On Jun 11, 2012, at 11:33 PM, Tapan Gautam (tgautam) tgau...@cisco.com wrote: Hey Guys, When I call CUC pilot from BR1 phone, the dtmf tones are not recognized properly by CUC, i.e. BR1 phone cannot login to mailbox or select any other option via DTMF. If I remove crtp, everything works fine. Topology: SCCP phone(BR1 site) à g729r8 with crtp à CUCM à SIP trunk(with OOB and RFC2833 as dtmf options) à CUC Things I have tried so far, 1) All dtmf options in SIP trunk. 2) Enabled mtp option 3) In CUC, changed codec type to just g711u, just g729 and both(which is the default). I found other posts on this issue but none of them has the solution. Thanks, Tapan ___ 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] DTMF from BR1 phone to CUC via SIP trunk notrecognized properly
Bill, If I turn off crtp, everything works fine. Thanks, Tapan From: Bill Lake [mailto:whl...@gmail.com] Sent: Tuesday, June 12, 2012 11:29 AM To: Krishna Cc: Dan Quinlan (daquinla); Tapan Gautam (tgautam); ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] DTMF from BR1 phone to CUC via SIP trunk notrecognized properly Might want to read this as it is the authority on RFC 2833 and it appears to be out of band however it uses the RTP header. It also appears that the tones are actually carried in the RTP traffic just not the audio stream. http://www.ietf.org/rfc/rfc2833.txt Would it be interesting to see if cRTP was turned off if DTMF would work in this case? On Tue, Jun 12, 2012 at 7:58 AM, Krishna vinayak_...@yahoo.com wrote: Dan, A small correction to your statement..rfc2833 is out of band mechanism mostly, and moreover it doesn't use audio channel, infact it uses rtp header to relay the dtmf message with a payload identifier. thank you Krishna.. From: Dan Quinlan (daquinla) daqui...@cisco.com To: Tapan Gautam (tgautam) tgau...@cisco.com Cc: ccie_voice@onlinestudylist.com Sent: Monday, June 11, 2012 11:57 PM Subject: Re: [OSL | CCIE_Voice] DTMF from BR1 phone to CUC via SIP trunk notrecognized properly cRTP mangles in-band (audio) DTMF. If I understand correctly, you are SIP-integrated between CUC and UCM. You need to be OOB only for DTMF (not rfc2833). Rfc2833 is an in-band (audio channel) mechanism. You need OOB (signaling channel) for DTMF to function when cRTP is used. DQ d...@cisco.com Sent from my iPhone On Jun 11, 2012, at 11:33 PM, Tapan Gautam (tgautam) tgau...@cisco.com wrote: Hey Guys, When I call CUC pilot from BR1 phone, the dtmf tones are not recognized properly by CUC, i.e. BR1 phone cannot login to mailbox or select any other option via DTMF. If I remove crtp, everything works fine. Topology: SCCP phone(BR1 site) à g729r8 with crtp à CUCM à SIP trunk(with OOB and RFC2833 as dtmf options) à CUC Things I have tried so far, 1) All dtmf options in SIP trunk. 2) Enabled mtp option 3) In CUC, changed codec type to just g711u, just g729 and both(which is the default). I found other posts on this issue but none of them has the solution. Thanks, Tapan ___ 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 ___ 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] DTMF from BR1 phone to CUC via SIP trunk not recognized properly
Hey Guys, When I call CUC pilot from BR1 phone, the dtmf tones are not recognized properly by CUC, i.e. BR1 phone cannot login to mailbox or select any other option via DTMF. If I remove crtp, everything works fine. Topology: SCCP phone(BR1 site) à g729r8 with crtp à CUCM à SIP trunk(with OOB and RFC2833 as dtmf options) à CUC Things I have tried so far, 1) All dtmf options in SIP trunk. 2) Enabled mtp option 3) In CUC, changed codec type to just g711u, just g729 and both(which is the default). I found other posts on this issue but none of them has the solution. Thanks, Tapan ___ 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] NTP Master Dilemma
Based on my testing in the home lab, I suggest to not use 'ntp master' and ntp server at the same time. To sync to an external source, all you need is 'ntp server' command. Also, I could not find a cli command to setup ntp in CUCM, only way to do it is via GUI. Good luck!!! Hth, Tapan From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mann Chaddha Sent: Friday, June 08, 2012 12:19 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] NTP Master Dilemma Hi Experts I am due for my next Lab attempt in 5 days time. I so far have never been able to get marks for NTP section. So I ask this question to the forum that if someone has ever scored 100% in the Infra Section (without breaking the NDA of course) , can they lend some light on this innocuous looking command, ntp master. Here are some finding from my POD: [10.10.100.1 = PSTN RTR 10.10.110.1=HQ RTR] 1. With NTP Master Command R1: ntp source Loopback0 ntp master 10 ntp server 10.10.100.1 source Loopback0 ! clock timezone PST -8 clock summer-time PDT recurring R2: ntp server 10.10.110.1 Show commands: R1: HQ-R1#sh ntp status Clock is unsynchronized, stratum 16, no reference clock nominal freq is 250. Hz, actual freq is 250.0008 Hz, precision is 2**24 reference time is D37C1C18.AAB00668 (23:37:12.666 PDT Thu Jun 7 2012) clock offset is 0. msec, root delay is 0.00 msec root dispersion is 0.00 msec, peer dispersion is 0.00 msec loopfilter state is 'CTRL' (Normal Controlled Loop), drift is -0.03145 s/s system poll interval is 64, last update was 384 sec ago. HQ-R1# HQ-R1# HQ-R1#sh ntp ass HQ-R1#sh ntp associations address ref clock st when poll reach delay offset disp x~127.127.1.1 .LOCL. 9 16 16 377 0.000 0.000 0.240 x~10.10.100.1.LOCL. 1 50 64 377 0.000 13194.1 3.293 * sys.peer, # selected, + candidate, - outlyer, x falseticker, ~ configured R2: BR1-R2#sh ntp status Clock is unsynchronized, stratum 16, no reference clock nominal freq is 250. Hz, actual freq is 249.9990 Hz, precision is 2**24 reference time is . (16:00:00.000 PST Wed Dec 31 1899) clock offset is 0. msec, root delay is 0.00 msec root dispersion is 0.01 msec, peer dispersion is 0.00 msec loopfilter state is 'CTRL' (Normal Controlled Loop), drift is 0.03885 s/s system poll interval is 64, never updated. BR1-R2# BR1-R2#sh ntp ass address ref clock st when poll reach delay offset disp ~10.10.110.173.78.73.84 16 55 64 300 0.000 1325166 944.35 * sys.peer, # selected, + candidate, - outlyer, x falseticker, ~ configured CUCM: admin:utils ntp status ntpd (pid 6322) is running... remote refid st t when poll reach delay offset jitter == *127.127.1.0 LOCAL(0)10 l 63 64 3770.0000.000 0.001 10.10.110.1.INIT. 16 u 64 128 3701.897 -2.990 1087693 synchronised to local net at stratum 11 time correct to within 12 ms polling server every 512 s Current time in UTC is : Fri Jun 8 06:22:22 UTC 2012 Current time in America/Los_Angeles is : Thu Jun 7 23:22:22 PDT 2012 -- Scenario 2: Without NTP Master Command. -- R1: ntp source Loopback0 ntp server 10.10.100.1 source Loopback0 ! clock timezone PST -8 clock summer-time PDT recurring R2: BR1-R2#r | s ntp ntp server 10.10.110.1 Show Commands: HQ-R1#sh ntp status Clock is synchronized, stratum 2, reference is 10.10.100.1 nominal freq is 250. Hz, actual freq is 250.0008 Hz, precision is 2**24 reference time is D37C21AB.D80D6753 (00:00:59.843 PDT Fri Jun 8 2012) clock offset is -0.0078 msec, root delay is 0.00 msec root dispersion is 0.01 msec, peer dispersion is 0.00 msec loopfilter state is 'CTRL' (Normal Controlled Loop), drift is -0.03182 s/s system poll interval is 64, last update was 264 sec ago. HQ-R1# HQ-R1#sh ntp ass address ref clock st when poll reach delay offset disp *~10.10.100.1.LOCL. 1 11 64 377 0.000 -7.866 2.813 * sys.peer, # selected, + candidate, - outlyer, x falseticker, ~ configured HQ-R1#sh ntp ass d 10.10.100.1 configured, our_master, sane, valid, stratum 1 ref ID .LOCL., time D37C229E.73B36D17 (00:05:02.451 PDT Fri Jun 8 2012) our mode client, peer mode server, our poll intvl 64, peer poll intvl 64 root delay 0.00 msec, root disp 0.47, reach 377, sync dist 0.00 delay 0.00 msec, offset -7.8662 msec, dispersion 2.81 precision 2**24, version 4 org time D37C22AE.D47DF91B (00:05:18.830 PDT Fri Jun 8 2012) rec time D37C22AE.D7653258 (00:05:18.841 PDT Fri Jun 8 2012) xmt time D37C22AE.D6E2248F (00:05:18.839 PDT Fri Jun 8 2012)
Re: [OSL | CCIE_Voice] VPIM between CUE and CUC
Try this http://pushkarbhatkoti.wordpress.com/category/cue-voicemail-vpim-network ing-cue-to-unity-in-10-minutes/ From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Joaquim Fernandes Sent: Friday, August 06, 2010 8:23 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] VPIM between CUE and CUC Hi Team, Is there any document where i can configure VPIM between CUE and CUC. I am hunting over the google but havent found anything great. I need a document giving step by step procedure for configuring vpim between CUC AND CUE. Thanx in advance. Regards, JF ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Switch QoS
I think, if you read the explanation as, it configures the drop thresholds for queue set 2 to 40 and 60 percent of the allocated memory, guarantees 100% of the allocated memory, and configures 200% as the maximum memory that this queue set can have before packets are dropped. , it would make more sense. Better explanation of the command is given here, http://www.cisco.com/en/US/products/hw/switches/ps5023/products_tech_not e09186a0080883f9e.shtml hope that helps, Tapan From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of CCIE Voice Sent: Friday, July 30, 2010 3:17 PM To: 'ccie_voice@onlinestudylist.com' Subject: [OSL | CCIE_Voice] Switch QoS OK folks, I REALLY do not understand the following command. Cisco's explanation states that it configures the drop thresholds for queue 2 to 40 and 60 percent of the allocated memory, guarantees 100% of the allocated memory, and configures 200% as the maximum memory that this queue can have before packets are dropped. My question is: what the hell does that even mean? Does it mean that queue 2 threshold 1 is set to drop at 40%, queue 2 threshold 2 is set to drop at 60%, queue 2 threshold 3 can have 100%? That's how I understood it, but apparently it is NOT the correct way to interpret it. Can anyone explain this to me in plain English? Switch(config)# mls qos queue-set output 2 threshold 2 40 60 100 200 Link to Cisco's explanation: http://www.cisco.com/en/US/docs/switches/lan/catalyst3750/software/relea se/12.2_25_see/configuration/guide/swqos.html#wp1179728 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SRST dial-peer behaviour
Remove $ from dial-peer 7. It'll wait for inter-digit timeout and match dial-peer 9 after you press the last digit. Let me know if that works. Hth, Tapan From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Erwan Erwan Sent: Wednesday, July 28, 2010 11:40 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] SRST dial-peer behaviour Hi Experts, I am trying to configure so that calling phone will show 7 digit To 8884343 in SRST and Normal mode. I use dial-peer 7 pots in normal mode (send 8884343 from Route Pattern to BR-1 GW) RP Local : 9.[2-9]xx , predot , send to BR-1 (H323) , hit dial-peer 7 pots And it did show 7 digit 8884343 in my Calling phone BR-1 dialpeer -- dial-peer voice 7 pots destination-pattern [2-9]..$ port 0/0/0:23 forward-digits all However when I dial 98884343 in SRST mode,I expect it will use dial-peer 9 pots (because I have to dial 9 for local call) dial-peer voice 9 pots destination-pattern 9[2-9]..$ port 0/0/0:23 forward-digits 7 But the call from phone always hit dial-peer 7. And if I shut down dial peer 7, local call will work fine in SRST. But why it hit dial-peer 7 in SRST for 98884343 , which I think dial-peer 9 is more precise match ?? And if I tested using csim start 98884343 in SRST , it will hit dial-peer 9 (which is right for this case). But if from IP phone it will use dial-peer 7 when I dial 98884343 in SRST mode. Anybody know why and shade light on this ? Thks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] QoS question about uplink port
Hi Matthew, My understanding is that you trust dscp on uplink ports. There is a cos-dscp mapping table and you enable mapping using mls qos map cos-dscp command. More details here, https://www.cisco.com/en/US/products/hw/switches/ps5023/products_tech_no te09186a0080883f9e.shtml Hope that helps, Tapan From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Matthew Berry Sent: Monday, July 26, 2010 8:23 AM To: OSL Group Subject: [OSL | CCIE_Voice] QoS question about uplink port Guys - When configuring QoS on an uplink port, how do I determine whether to trust CoS or DSCP markings? I always thought that you would trust CoS markings on access ports with IP phones on the other end since the phone will mark packets as CoS3 (signaling) or CoS 5 (media). The access ports connected to servers would be configured to trust DSCP since CUCM marks according to DSCP. My understanding is that the mls qos trust cos or mls qos trust dscp applies only for inbound packets. Ideas? Thanks! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Gatekeeper Issue
Hi Jeff, Have you tried following configuration. If not, give it a shot. ! gatekeeper zone local ZONE_1 asccie.com 10.5.200.1 zone prefix ZONE_1 1775... gw-priority 10 CUCM_GK_TRUNK_2 zone prefix ZONE_1 1775... gw-priority 9 CUCM_GK_TRUNK_1 zone prefix ZONE_1 1775... gw-priority 0 BR2_R3_GW BR1_R2_GW zone prefix ZONE_1 44* gw-priority 10 BR2_R3_GW zone prefix ZONE_1 44* gw-priority 0 BR1_R2_GW CUCM_GK_TRUNK_2 CUCM_GK_TRUNK_1 gw-type-prefix 1# default-technology no shutdown ! Hope that helps, Tapan From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jeff Price (jeffpric) Sent: Tuesday, May 25, 2010 5:54 PM To: CCIE Voice Maillist Subject: [OSL | CCIE_Voice] Gatekeeper Issue Hi everyone, I am having trouble with my GK. I have made Bold what is the problem, but I can't seem to understand why I'm having this issue. I configured a tech-prefix of 1# under the Trunk configuration page. Here is the config - gatekeeper zone local ZONE_1 asccie.com 10.5.200.1 zone prefix ZONE_1 1* gw-priority 10 CUCM_GK_TRUNK_2 zone prefix ZONE_1 1* gw-priority 9 CUCM_GK_TRUNK_1 zone prefix ZONE_1 1* gw-priority 0 BR2_R3_GW BR1_R2_GW zone prefix ZONE_1 44* gw-priority 10 BR2_R3_GW zone prefix ZONE_1 44* gw-priority 0 BR1_R2_GW CUCM_GK_TRUNK_2 CUCM_GK_TRUNK_1 gw-type-prefix 1#* default-technology no shutdown Here is the debug gatekeeper main 10 output: May 25 23:55:58.011: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup May 25 23:55:58.187: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup R1(config-gk)# May 25 23:56:00.115: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup May 25 23:56:00.115: ////GK/gk_rassrv_arq: arqp=0x4AE0FB04,crv=0x19, answerCall=0 May 25 23:56:00.115: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC May 25 23:56:00.115: //E69BDC8380B8/E69BDC8380BA/GK/gk_dns_query: No Name servers May 25 23:56:00.115: //E69BDC8380B8/E69BDC8380BA/GK/rassrv_get_addrinfo: (1#17752011001) Matched tech-prefix 1# May 25 23:56:00.115: //E69BDC8380B8/E69BDC8380BA/GK/rassrv_get_addrinfo: (1#17752011001) Matched zone prefix 1 and remainder 7752011001 May 25 23:56:00.115: ////GK/gk_rassrv_get_ingress_network: ARQ non-std ingress network = 1 May 25 23:56:00.115: //E69BDC8380B8/E69BDC8380BA/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x4AE06200 May 25 23:56:00.115: //E69BDC8380B8/E69BDC8380BA/GK/rassrv_arq_select_viazone: matched zone is ZONE_1, and z_invian R1(config-gk)#amelen=0 May 25 23:56:00.115: //E69BDC8380B8/E69BDC8380BA/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x4AE06200 May 25 23:56:00.115: //E69BDC8380B8/E69BDC8380BA/GK/rassrv_arq_select_viazone: matched zone is ZONE_1, and z_outvianamelen=0 May 25 23:56:00.115: ////GK/gk_rassrv_get_ingress_network: ARQ non-std ingress network = 1 May 25 23:56:00.115: //E69BDC8380B8/E69BDC8380BA/GK/rassrv_get_addrinfo: (1#17752011001) tech-prefix gateway selection failed. May 25 23:56:00.115: //E69BDC8380B8/E69BDC8380BA/GK/gk_rassrv_sep_arq: rassrv_get_addrinfo() failed (return code = 0x103) Here is the show gatekeeper call 10 output: May 26 00:02:15.899: ////GK/gk_call_new: src_endptp=0x4AE0F9F0, dst_endptp=0x0, src_pxp=0x0, dst_pxp=0x0, bw=160, crv=31, whichcrv=0x1, circuit=0x0, capacity=0x0, ret_callpp=0x4925F3F8 May 26 00:02:15.899: ////GK/gk_call_find_endpts: NOT_FOUND May 26 00:02:15.899: ////GK/gk_call_new: checking for default (CLI) carrier for sep endpt 0x4AE0F9F0 May 26 00:02:15.899: //C6CEF7C380D2/C6CEF7C380D4/GK/gk_call_delete: callp=4AB57F54 May 26 00:02:15.899: //C6CEF7C380D2/C6CEF7C380D4/GK/gk_call_delete: c_callstate 0x0, c_resbw1 0, resbw2 0, c_reszp1 0x0, c_reszp2 0x0 Here is the show gatekeeper endpoints output: R1(config-gk)#do show gatekeeper end GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags --- - --- - - - 10.5.201.1 1720 10.5.201.1 61751 ZONE_1VOIP-GW H323-ID: BR1_R2_GW Voice Capacity Max.= Avail.= Current.= 0 10.5.202.1 1720 10.5.202.1 52635 ZONE_1VOIP-GW H323-ID: BR2_R3_GW E164-ID: 3001 E164-ID: 3002 Voice Capacity Max.= Avail.= Current.= 0 172.21.51.204 37257 172.21.51.204 32858 ZONE_1TERM H323-ID: CUCM_GK_TRUNK_1 172.21.51.205 34279 172.21.51.205 32814 ZONE_1TERM H323-ID: CUCM_GK_TRUNK_2 Total number of active registrations = 4
Re: [OSL | CCIE_Voice] verify QoS on 3750
Clear counters. Place a call and check interface stats using 'show mls qos interface port# statistics'. Could not locate any other command. Tapan From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Erwan Erwan Sent: Monday, May 24, 2010 1:45 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] verify QoS on 3750 hi, any body know what command to see if QOS configure in 3750 is actually working, when we place call. ie: we cofigure auto qos voip trust in switch port As I did not see in sh mls qos interface . Maybe there is other command or tool ? tks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
