Re: [OSL | CCIE_Voice] Equipment Config
What's BLS On Jul 14, 2012 8:51 PM, Chris Smolen csmo...@smolz.com wrote: I have the BLS and have access to a bunch of lab equipment. is there somewhere that shows the equipment with interface cards? ___ 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] dead air when calling CUE vmail gui doesn't work
Is it on cme / callmanager On Jul 5, 2012 8:59 AM, Krishna vinayak_...@yahoo.com wrote: hi folks, I couldn't able to understand why the CUE giving me the dead air though after the configuration is absolutely correct with the right codecs. when i pressed the vmail button on the phone, it connects to the vmail number but i cannot hear anything. And, also i couldn't access web gui for the cue even after providing all the right info such as ip http server, ip http path, ip http auth local.. the web browser sits there forever with no output... does anyone experienced the same problem as i am??? your advice on this matter is much appreciated. thank you krishna. ___ 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] BAT from 4.1.3 to 8.6.2
Use export /import in bat. But you will miss cdr records On Jul 2, 2012 7:16 PM, Emanuel Damasceno aedamasc...@gmail.com wrote: Does anybody know the fastest way to upgrade a system from 4.1.3 to 8.6.2a? There is a HUGE catch: DMA doesn't work, it gives me a lot of errors because of a corrupted Database. There is no way I can clean it up to use DMA. I opened a TAC with Cisco and they DIDN'T solve my problem in TWO MONTHS. Yes, I am really screwed, trying to work with this BAT file. Anyone can share any thoughts? I don't mean to be rude, but don't try to post anything DMA related here because I tried pretty much all versions with Cisco TAC and no versions worked. They always gave me errors. The errors were either extracting the DMA files or after successfully extracting the DMA files, the CUCM wouldn't install. And before anybody asks, YES, I was using the same DMA version as my CUCM's installation (if DMA was 7.0.2, CUCM was 7.0.2, if DMA was 7.1.5, CUCM was 7.1.5 and so on). I just want a help with the Excel file. There is no way to do anything with the DMA. Sorry... Thanks for anything... *Emanuel Damasceno* CCNP Voice ___ 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] transcoder not functioning in the cube section
Locally means you should have it on gk On Jun 21, 2012 11:57 AM, Krishna vinayak_...@yahoo.com wrote: i did enable the faststart but no use, and also this transcoder is locally available to cube as well... -- *From:* Lidiya Krunic lkru...@hotmail.com *To:* luv...@gmail.com; vinayak_...@yahoo.com *Cc:* ccie_voice@onlinestudylist.com *Sent:* Wednesday, June 20, 2012 10:40 PM *Subject:* RE: [OSL | CCIE_Voice] transcoder not functioning in the cube section Try to remove checkmark Wait for Far End TCS on CUM (or use faststart). -- Date: Thu, 21 Jun 2012 08:37:54 +0530 From: luv...@gmail.com To: vinayak_...@yahoo.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] transcoder not functioning in the cube section Transcoder should be available locally for cube On Jun 21, 2012 8:23 AM, Krishna vinayak_...@yahoo.com wrote: hi folks, I got stuck at vol2 lab1 in section 4.2, where the cube involves in routing the calls using gakekeeper. when i call from 1002 to 3002 the phone rings, and when i answer the 3002, the 1002 still makes the ringing sound, and after some time the call failing with the busy tone. I checked the show sccp connections, and surprisingly it is not showing any transcode sessions. when i call from 1002 or 5002 to sip phone 3006, it gives me immediately busy-tone/fail tone. can you guys advice me what to do in order to make this work. here is my config: HQ-RTR(config)#do sh sdspf unit mtp-1 Device:hqgk-xcode TCP socket:[1] REGISTERED in SCCP ver 65546/10 actual_stream:6 max_stream 6 IP:10.10.200.3 51291 MTP Dixieland keepalive 152 Supported codec: G711Ulaw G711Alaw G729 G729a G729b G729ab max-mtps:2, max-streams:6, alloc-streams:6, act-streams:0 HQ-RTR(Config)#dial-peer voice 3000 voip incoming called-number 3...$ dial-peer voice 3001 voip destination-pattern 3... session target ras codec g711ulaw gatekeeper zone local UCM ipexpert.com zone local UCME ipexpert.com outvia VIA zone local VIA ipexpert.com zone prefix UCM 1... gw-priority 10 gk-trunk_2 zone prefix UCM 1... gw-priority 9 gk-trunk_1 zone prefix UCME 3... zone prefix UCM 5... gw-priority 10 gk-trunk_2 zone prefix UCM 5... gw-priority 9 gk-trunk_1 gw-type-prefix 1#* default-technology no shutdown HQ-RTR(config)#do sh run | s gatew h323-gateway voip interface h323-gateway voip id VIA ipaddr 10.10.110.1 1719 h323-gateway voip h323-id HQ-RTR h323-gateway voip bind srcaddr 10.10.200.3 gateway Thank you krishna. ___ 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 http://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
Re: [OSL | CCIE_Voice] transcoder not functioning in the cube section
Transcoder should be available locally for cube On Jun 21, 2012 8:23 AM, Krishna vinayak_...@yahoo.com wrote: hi folks, I got stuck at vol2 lab1 in section 4.2, where the cube involves in routing the calls using gakekeeper. when i call from 1002 to 3002 the phone rings, and when i answer the 3002, the 1002 still makes the ringing sound, and after some time the call failing with the busy tone. I checked the show sccp connections, and surprisingly it is not showing any transcode sessions. when i call from 1002 or 5002 to sip phone 3006, it gives me immediately busy-tone/fail tone. can you guys advice me what to do in order to make this work. here is my config: HQ-RTR(config)#do sh sdspf unit mtp-1 Device:hqgk-xcode TCP socket:[1] REGISTERED in SCCP ver 65546/10 actual_stream:6 max_stream 6 IP:10.10.200.3 51291 MTP Dixieland keepalive 152 Supported codec: G711Ulaw G711Alaw G729 G729a G729b G729ab max-mtps:2, max-streams:6, alloc-streams:6, act-streams:0 HQ-RTR(Config)#dial-peer voice 3000 voip incoming called-number 3...$ dial-peer voice 3001 voip destination-pattern 3... session target ras codec g711ulaw gatekeeper zone local UCM ipexpert.com zone local UCME ipexpert.com outvia VIA zone local VIA ipexpert.com zone prefix UCM 1... gw-priority 10 gk-trunk_2 zone prefix UCM 1... gw-priority 9 gk-trunk_1 zone prefix UCME 3... zone prefix UCM 5... gw-priority 10 gk-trunk_2 zone prefix UCM 5... gw-priority 9 gk-trunk_1 gw-type-prefix 1#* default-technology no shutdown HQ-RTR(config)#do sh run | s gatew h323-gateway voip interface h323-gateway voip id VIA ipaddr 10.10.110.1 1719 h323-gateway voip h323-id HQ-RTR h323-gateway voip bind srcaddr 10.10.200.3 gateway Thank you krishna. ___ 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] IPexpert work book
Can anyone please guide me to select a workbook / material from IPExpert for CCIE voice.I'm looking for a self study material with explanation for technology not a lab oriented stuff. Thanks in advance. San ___ 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] IPexpert work book
Thanks. I was specifically looking for the QoS part. The info SRND is not good enough to understand the variations On Thu, May 31, 2012 at 9:24 PM, A NN prince_karim...@yahoo.com wrote: IPExpert Voulume I lab and the Video Walkthrough is quite good. I wouldn't rely only on IP Expert to understand the technologies. The SRND guides and Feature and Services guides are a MUST. HTH -- *From:* san r luv...@gmail.com *To:* ccie_voice@onlinestudylist.com *Sent:* Thursday, 31 May 2012, 15:10 *Subject:* [OSL | CCIE_Voice] IPexpert work book Can anyone please guide me to select a workbook / material from IPExpert for CCIE voice.I'm looking for a self study material with explanation for technology not a lab oriented stuff. Thanks in advance. San ___ 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] IPexpert work book
Thanks ! On Thu, May 31, 2012 at 9:35 PM, A NN prince_karim...@yahoo.com wrote: There is a good Switch QoS blog articles from Vik (check it via google). For WAN, read the QoS SRND. -- *From:* san r luv...@gmail.com *To:* A NN prince_karim...@yahoo.com *Cc:* ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com *Sent:* Thursday, 31 May 2012, 16:56 *Subject:* Re: [OSL | CCIE_Voice] IPexpert work book Thanks. I was specifically looking for the QoS part. The info SRND is not good enough to understand the variations On Thu, May 31, 2012 at 9:24 PM, A NN prince_karim...@yahoo.com wrote: IPExpert Voulume I lab and the Video Walkthrough is quite good. I wouldn't rely only on IP Expert to understand the technologies. The SRND guides and Feature and Services guides are a MUST. HTH -- *From:* san r luv...@gmail.com *To:* ccie_voice@onlinestudylist.com *Sent:* Thursday, 31 May 2012, 15:10 *Subject:* [OSL | CCIE_Voice] IPexpert work book Can anyone please guide me to select a workbook / material from IPExpert for CCIE voice.I'm looking for a self study material with explanation for technology not a lab oriented stuff. Thanks in advance. San ___ 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] Gatekeeper trunk
/cch323_peer_caps_ind_common: Update the audio mask: old mask=0xC; new mask=0x0 May 24 06:25:16.057: //1548/8042FB5A1300/H323/cch323_peer_caps_ind_common: ExtendedCaps present May 24 06:25:16.057: //-1//H323/cch323_get_dp_pref_mask: cch323_get_dp_pref_mask:IPIPGW(1548):setting mask for 729ar8also as 729 is configured May 24 06:25:16.057: //1548/8042FB5A1300/H323/cch323_prepare_preferred_codec_list: Munging caps: 5:0:C May 24 06:25:16.057: //1548/8042FB5A1300/H323/cch323_prepare_preferred_codec_list: Munging caps: local audio_cap_mask: 1 pref_mask from dial-peer: C May 24 06:25:16.057: //1548/8042FB5A1300/H323/cch323_prepare_preferred_codec_list: Check the filter: Not a single match May 24 06:25:16.057: //1548/8042FB5A1300/H323/cch323_peer_caps_ind_common: No matching codec after filtering - use dial-peer codecs in TCS May 24 06:25:16.057: //1548/8042FB5A1300/H323/cch323_peer_caps_ind_common: need xcoder resource for codec mismatch May 24 06:25:16.061: //1548/8042FB5A1300/H323/cch323_peer_caps_ind_common: try to find transcoder May 24 06:25:16.061: //1548/8042FB5A1300/H323/cch323_peer_caps_ind_common: transcoder reservation failed, report H.245 failure May 24 06:25:16.069: //1548/8042FB5A1300/H323/cch323_h245_connection_sm: state=0, event=0, ccb=46011CF4, listen state=0 May 24 06:25:16.085: //1548/8042FB5A1300/H323/cch323_h245_connection_sm: state=1, event=2, ccb=46011CF4, listen state=1 May 24 06:25:16.089: //1548/8042FB5A1300/H323/cch323_h245_cap_ind: Masks au=0x100C data=0x4 uinp=0x32 May 24 06:25:16.089: //-1//H323/cch323_get_dp_pref_mask: cch323_get_dp_pref_mask:IPIPGW(1548):setting mask for 729ar8also as 729 is configured May 24 06:25:16.089: //1548/8042FB5A1300/H323/cch323_set_extended_caps: Copying codec list into extended caps structure May 24 06:25:16.089: //1548/8042FB5A1300/H323/cch323_set_extended_caps: G729IETF May 24 06:25:16.089: //1548/8042FB5A1300/H323/cch323_set_extended_caps: G729a May 24 06:25:16.089: //1548/8042FB5A1300/H323/cch323_iwf_cap_notify: Mask sent to other leg=C May 24 06:25:16.089: //1548/8042FB5A1300/H323/cch323_iwf_cap_notify: ../voip/cch323/gw/src/cch323_h245_iwf_util.c:cch323_iwf_cap_notify:1048 Post srtp info from tcs to other call leg May 24 06:25:16.097: //1548/8042FB5A1300/H323/cch323_peer_caps_ack: Sending caps resp event to CAP sm May 24 06:25:19.957: //-1//H323/cch323_post_caps_ind: callID=1548, ExtendedCaps not present May 24 06:25:19.961: //1548/8042FB5A1300/H323/cch323_peer_caps_ind_common: Update the audio mask: old mask=0xC; new mask=0x0 May 24 06:25:19.961: //1548/8042FB5A1300/H323/cch323_send_empty_cap_request: Sending NULL caps... May 24 06:25:19.969: //1548/8042FB5A1300/H323/h245_iwf_handle_send_caps_ack_to_peer: Sending caps ack to other leg May 24 06:25:19.969: //-1//H323/cch323_do_caps_ack: dstCallID=1549, srcCallID=1548 May 24 06:25:19.969: //1548/8042FB5A1300/H323/cch323_peer_caps_ind_common: Update the audio mask: old mask=0x0; new mask=0x1 May 24 06:25:19.969: //1548/8042FB5A1300/H323/cch323_peer_caps_ind_common: ExtendedCaps present May 24 06:25:19.969: //-1//H323/cch323_get_dp_pref_mask: cch323_get_dp_pref_mask:IPIPGW(1548):setting mask for 729ar8also as 729 is configured May 24 06:25:19.969: //1548/8042FB5A1300/H323/cch323_prepare_preferred_codec_list: Munging caps: 5:1:C May 24 06:25:19.969: //1548/8042FB5A1300/H323/cch323_prepare_preferred_codec_list: Munging caps: local audio_cap_mask: 1 pref_mask from dial-peer: C May 24 06:25:19.969: //1548/8042FB5A1300/H323/cch323_prepare_preferred_codec_list: Check the filter: Not a single match May 24 06:25:19.969: //1548/8042FB5A1300/H323/cch323_peer_caps_ind_common: No matching codec after filtering - use dial-peer codecs in TCS May 24 06:25:19.969: //1548/8042FB5A1300/H323/cch323_peer_caps_ind_common: need xcoder resource for codec mismatch May 24 06:25:19.969: //1548/8042FB5A1300/H323/cch323_peer_caps_ind_common: try to find transcoder May 24 06:25:19.969: //1548/8042FB5A1300/H323/cch323_peer_caps_ind_common: transcoder reservation failed, report H.245 failure When I pressed the hold button on cme phone: May 24 06:25:19.985: //1548/8042FB5A1300/H323/cch323_h245_connection_sm: state=0, event=4, ccb=46011CF4, listen state=2 May 24 06:25:19.989: //1548/8042FB5A1300/H323/cch323_call_generic_cleanup: De-allocating audioFastStartArray. On Wed, May 23, 2012 at 10:15 PM, Mohd Baqari baqari.voic...@gmail.comwrote: Hi, If the command emptycapability then it has to work assuming that u kept gk in media flow through mode. Plz share the output of debug ipipgw on cube Regards, Mohammed Al Baqari Sent from my iPhone On May 23, 2012, at 7:50 PM, san r luv...@gmail.com wrote: emptycapability ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP
Re: [OSL | CCIE_Voice] Gatekeeper trunk
Please paste your configs On May 23, 2012 11:23 PM, The Masterplan winmasterp...@gmail.com wrote: I configured that, but the result is the same. On Wed, May 23, 2012 at 6:50 PM, san r luv...@gmail.com wrote: Try the following commands in voice class or H.323 voice-service configuration mode - h323 end h225 connect-passthru emptycapability h245 passthru tcsnonstd-passthru On Wed, May 23, 2012 at 5:35 PM, The Masterplan winmasterp...@gmail.comwrote: Hi, I have the following setup: Sip IP phone --- CME ---h323 GK ---GK trunk CUCM-- Sip IP phone On the cme side I have a transcoder and sip ip phone configured with codec g711ulaw. On the cucm side I also have a transcoder on mrgl of gk trunk that is in a device pool configured to speak g729 only. I have unchecked on trunk Wait for Far End H.245 Terminal Capability Set and checked the following: - Media termination point required - Inbound Fast Start - Outbound Fast Start with G729 codec Given this facts, the call is established succesfully (transcoders are used) but supplementary services are not working from either side. Thank you ___ 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] Gatekeeper trunk
Do you have the same on GK as well? On May 24, 2012 12:05 AM, The Masterplan winmasterp...@gmail.com wrote: Hi, This is the cme side relevant config: voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip h323 emptycapability h225 connect-passthru h245 passthru tcsnonstd-passthru sip bind control source-interface FastEthernet0/0 bind media source-interface FastEthernet0/0 registrar server no update-callerid voice register global mode cme source-address 10.1.1.235 port 5060 max-dn 4 max-pool 3 authenticate register create profile sync 0054221011645706 voice register dn 2 number 1006 call-forward b2bua busy 2000 call-forward b2bua mailbox 2100 call-forward b2bua noan 2000 timeout 20 ! voice register pool 1 id mac .. number 1 dn 2 dtmf-relay rtp-nte username 1006 password cisco codec g711ulaw interface FastEthernet0/0 ip address 10.1.1.235 255.255.0.0 h323-gateway voip interface h323-gateway voip id UCM ipaddr 10.1.1.171 1719 h323-gateway voip h323-id cme h323-gateway voip tech-prefix 1# h323-gateway voip bind srcaddr 10.1.1.235 dial-peer voice 5000 voip translation-profile incoming stripgk destination-pattern 5...$ session target ras incoming called-number . tech-prefix 1# dtmf-relay h245-alphanumeric no vad On Wed, May 23, 2012 at 9:08 PM, san r luv...@gmail.com wrote: Please paste your configs On May 23, 2012 11:23 PM, The Masterplan winmasterp...@gmail.com wrote: I configured that, but the result is the same. On Wed, May 23, 2012 at 6:50 PM, san r luv...@gmail.com wrote: Try the following commands in voice class or H.323 voice-service configuration mode - h323 end h225 connect-passthru emptycapability h245 passthru tcsnonstd-passthru On Wed, May 23, 2012 at 5:35 PM, The Masterplan winmasterp...@gmail.com wrote: Hi, I have the following setup: Sip IP phone --- CME ---h323 GK ---GK trunk CUCM-- Sip IP phone On the cme side I have a transcoder and sip ip phone configured with codec g711ulaw. On the cucm side I also have a transcoder on mrgl of gk trunk that is in a device pool configured to speak g729 only. I have unchecked on trunk Wait for Far End H.245 Terminal Capability Set and checked the following: - Media termination point required - Inbound Fast Start - Outbound Fast Start with G729 codec Given this facts, the call is established succesfully (transcoders are used) but supplementary services are not working from either side. Thank you ___ 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] Gatekeeper trunk
I believe it's supposed to be there on GK. Do you have Mtp on trunk? On May 24, 2012 12:11 AM, The Masterplan winmasterp...@gmail.com wrote: No, only the gatekeeper configuration needed to route calls and the following: voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip On Wed, May 23, 2012 at 9:37 PM, san r luv...@gmail.com wrote: Do you have the same on GK as well? On May 24, 2012 12:05 AM, The Masterplan winmasterp...@gmail.com wrote: Hi, This is the cme side relevant config: voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip h323 emptycapability h225 connect-passthru h245 passthru tcsnonstd-passthru sip bind control source-interface FastEthernet0/0 bind media source-interface FastEthernet0/0 registrar server no update-callerid voice register global mode cme source-address 10.1.1.235 port 5060 max-dn 4 max-pool 3 authenticate register create profile sync 0054221011645706 voice register dn 2 number 1006 call-forward b2bua busy 2000 call-forward b2bua mailbox 2100 call-forward b2bua noan 2000 timeout 20 ! voice register pool 1 id mac .. number 1 dn 2 dtmf-relay rtp-nte username 1006 password cisco codec g711ulaw interface FastEthernet0/0 ip address 10.1.1.235 255.255.0.0 h323-gateway voip interface h323-gateway voip id UCM ipaddr 10.1.1.171 1719 h323-gateway voip h323-id cme h323-gateway voip tech-prefix 1# h323-gateway voip bind srcaddr 10.1.1.235 dial-peer voice 5000 voip translation-profile incoming stripgk destination-pattern 5...$ session target ras incoming called-number . tech-prefix 1# dtmf-relay h245-alphanumeric no vad On Wed, May 23, 2012 at 9:08 PM, san r luv...@gmail.com wrote: Please paste your configs On May 23, 2012 11:23 PM, The Masterplan winmasterp...@gmail.com wrote: I configured that, but the result is the same. On Wed, May 23, 2012 at 6:50 PM, san r luv...@gmail.com wrote: Try the following commands in voice class or H.323 voice-service configuration mode - h323 end h225 connect-passthru emptycapability h245 passthru tcsnonstd-passthru On Wed, May 23, 2012 at 5:35 PM, The Masterplan winmasterp...@gmail.com wrote: Hi, I have the following setup: Sip IP phone --- CME ---h323 GK ---GK trunk CUCM-- Sip IP phone On the cme side I have a transcoder and sip ip phone configured with codec g711ulaw. On the cucm side I also have a transcoder on mrgl of gk trunk that is in a device pool configured to speak g729 only. I have unchecked on trunk Wait for Far End H.245 Terminal Capability Set and checked the following: - Media termination point required - Inbound Fast Start - Outbound Fast Start with G729 codec Given this facts, the call is established succesfully (transcoders are used) but supplementary services are not working from either side. Thank you ___ 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] Gatekeeper trunk
Can you configure the same commands on Gk? On May 24, 2012 12:21 AM, The Masterplan winmasterp...@gmail.com wrote: Yes, I have mtp on trunk. In the media resource group list assigned on trunk I have first a media resource group containing the hardware mtp and second a media resource group with the transcoder. On Wed, May 23, 2012 at 9:46 PM, san r luv...@gmail.com wrote: I believe it's supposed to be there on GK. Do you have Mtp on trunk? On May 24, 2012 12:11 AM, The Masterplan winmasterp...@gmail.com wrote: No, only the gatekeeper configuration needed to route calls and the following: voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip On Wed, May 23, 2012 at 9:37 PM, san r luv...@gmail.com wrote: Do you have the same on GK as well? On May 24, 2012 12:05 AM, The Masterplan winmasterp...@gmail.com wrote: Hi, This is the cme side relevant config: voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip h323 emptycapability h225 connect-passthru h245 passthru tcsnonstd-passthru sip bind control source-interface FastEthernet0/0 bind media source-interface FastEthernet0/0 registrar server no update-callerid voice register global mode cme source-address 10.1.1.235 port 5060 max-dn 4 max-pool 3 authenticate register create profile sync 0054221011645706 voice register dn 2 number 1006 call-forward b2bua busy 2000 call-forward b2bua mailbox 2100 call-forward b2bua noan 2000 timeout 20 ! voice register pool 1 id mac .. number 1 dn 2 dtmf-relay rtp-nte username 1006 password cisco codec g711ulaw interface FastEthernet0/0 ip address 10.1.1.235 255.255.0.0 h323-gateway voip interface h323-gateway voip id UCM ipaddr 10.1.1.171 1719 h323-gateway voip h323-id cme h323-gateway voip tech-prefix 1# h323-gateway voip bind srcaddr 10.1.1.235 dial-peer voice 5000 voip translation-profile incoming stripgk destination-pattern 5...$ session target ras incoming called-number . tech-prefix 1# dtmf-relay h245-alphanumeric no vad On Wed, May 23, 2012 at 9:08 PM, san r luv...@gmail.com wrote: Please paste your configs On May 23, 2012 11:23 PM, The Masterplan winmasterp...@gmail.com wrote: I configured that, but the result is the same. On Wed, May 23, 2012 at 6:50 PM, san r luv...@gmail.com wrote: Try the following commands in voice class or H.323 voice-service configuration mode - h323 end h225 connect-passthru emptycapability h245 passthru tcsnonstd-passthru On Wed, May 23, 2012 at 5:35 PM, The Masterplan winmasterp...@gmail.com wrote: Hi, I have the following setup: Sip IP phone --- CME ---h323 GK ---GK trunk CUCM-- Sip IP phone On the cme side I have a transcoder and sip ip phone configured with codec g711ulaw. On the cucm side I also have a transcoder on mrgl of gk trunk that is in a device pool configured to speak g729 only. I have unchecked on trunk Wait for Far End H.245 Terminal Capability Set and checked the following: - Media termination point required - Inbound Fast Start - Outbound Fast Start with G729 codec Given this facts, the call is established succesfully (transcoders are used) but supplementary services are not working from either side. Thank you ___ 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] Vol 2 Lab 2 Question 9.3 SRST Call Forwarding
Do you have ephone - DN configured as dual line or octo line? Also add Transfer pattern .T command. Assuming 5234 is an external number On May 19, 2012 10:37 PM, Jason Murray murr...@usa.com wrote: ___ 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] Vol 2 Lab 2 Question 9.3 SRST Call Forwarding
Do you have ephone - dn configured as dual line or octo line? Also add Transfer pattern .T command. Assuming 5234 is an external number configured on a dial-peer On Sat, May 19, 2012 at 9:17 PM, Jason Murray murr...@usa.com wrote: Hello everyone, I searched the forums and found a question that I have but there is no answer to it. Was wondering if anyone had any solution to this. I am going to post the original question since it is the same as I have. Thanks I have been working on Vol2 Lab 2 Question 9.3 SRST section (as below), but I could not find any solution for this. The IPX PG PDF file solution does not work, since it configured with an 'old fashion' alias in call-manager-fallback (as below), which redirects the incoming call to 1001 VM box, not the UC call-handler 5234 as supposed, which is configured as an announcement 'the number you call is not available' and then drop the call. Question - ensure that during SRST mode, incoming calls to ext 1001 will ring on ext 1002. If 1002 does not answer the call, the caller must hear an announcement 'this number is not available' and then drop the call. Solution provided in IPX PG PDF file: call-manager-fallback alias 1 1001 to 1003 cfw 5234 timeout 12 The VOL2 VOD is missing this section as well; and I did a search but not see any solution or suggestion on this OSL neither. I have tried a couple ways which configured ephone-dn call-forwarding or voice hunt-group and then finally sending caller to UC, but it always redirect to 1001 VM. Is this question is still a valid for Vol2 Lab2 SRST, and is anyone has been able to come up with a working solution for it? Thanks, TN. ___ 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] Lab 7 Lan QOS
Is it something like SIP/SIMPLE SOAP towards Presence server and SIP/SIMPLE CTI/QBE towards CUCM ? 2 ACL - one to CUPS and other to CUCM. Together can we call it as 'signalling' from presence communicator? http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/presence.html -- Figure 22-1 Cisco Unified Presence Components http://docwiki.cisco.com/wiki/Cisco_Unified_Presence,_Release_7.x_--_About_the_Interfaces_with_Cisco_Unified_Communications_Manager#About_the_Interfaces_with_Cisco_Unified_Communications_Manager Figure: Cisco Unified Presence Basic Deployment On Mon, May 14, 2012 at 4:19 AM, Brian Turner brianstur...@gmail.comwrote: I think you guys are over thinking it maybe. Isn't all traffic from the CUPC server Signaling traffic? The CUPC server doesn't really do anything but Signaling Media traffic goes from enpoint to endpoint, or endpoint to MTP etc. All other voice traffic is typically considered signaling. So an ACL that just matched all traffic to / from the CUPC server IP address would include signaling traffic for the CUPC server and little else. I didn't see the earlier email concerning the wording of the question, so this is a stab in the dark. Brian S Turner CCIE 6145 On Sun, May 13, 2012 at 11:39 AM, san r luv...@gmail.com wrote: Release Notes for Cisco Unified Personal Communicator Release 7.0 https://mail.google.com/mail/html/compose/static_files/blank_quirks.html#wp85054has got all inbound outbound port usage for CUPC http://www.cisco.com/en/US/docs/voice_ip_comm/cupc/7_0/english/release/notes/ol15710.html On Sun, May 13, 2012 at 5:27 PM, san r luv...@gmail.com wrote: I think we will need to consider the citi port number too On May 13, 2012 2:57 PM, steven moran smoran...@gmail.com wrote: I've been doing some work on this topic and have moved away from my original thoughts which were along the same lines as originally posted by Nazeer and a few others . Here are some recent thoughts on the subject the question is not particularly well worded, we can't really influence the traffic from the CUPC client to the server by applying policy on the server port. What we can do is control what the switch port does with packets from the CUPS server to the clients by classifying and applying an input ploicy on the server port. I went back to the presence section in the CUCM srnd - the only traffic between CUPC and CUPS seems be 5060 for sip/simple and 443 for SOAP, this was sort of backed up by the information in the CUPC 7.1 release. The release notes also mention that ports 16384-16424 are used for TFTP - but this would not be classed as signalling. Since there are other servers on the port any access list would need to specifiy the server IP address. If you wanted to make it really specific you could also specify the destination address of the CUPC clients as well. Finally by putting bidirectional access lists (any any eq and any eq any) you may loose the points as it might appear you are hedging your bets and do not know how directional QoS works. Also we are asked to guarantee 32k for signalling traffic if we put non signalling traffic into this policy we are not achieving the requirements. These statements are not facts, just my current opinion based on the last bits of documentation read and I am happy to discuss. Best regards. Steve ___ 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 http://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
Re: [OSL | CCIE_Voice] Lab 7 Lan QOS
If Cisco Unified Personal Communicator is configured for Desk Phone mode, a connection is established with the CTI Manager of Unified CM for phone control. Looks like CTI port needs to be consider only if the CUPC configured in deskphone mode. On Mon, May 21, 2012 at 8:05 PM, san r luv...@gmail.com wrote: Is it something like SIP/SIMPLE SOAP towards Presence server and SIP/SIMPLE CTI/QBE towards CUCM ? 2 ACL - one to CUPS and other to CUCM. Together can we call it as 'signalling' from presence communicator? http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/presence.html -- Figure 22-1 Cisco Unified Presence Components http://docwiki.cisco.com/wiki/Cisco_Unified_Presence,_Release_7.x_--_About_the_Interfaces_with_Cisco_Unified_Communications_Manager#About_the_Interfaces_with_Cisco_Unified_Communications_Manager Figure: Cisco Unified Presence Basic Deployment On Mon, May 14, 2012 at 4:19 AM, Brian Turner brianstur...@gmail.comwrote: I think you guys are over thinking it maybe. Isn't all traffic from the CUPC server Signaling traffic? The CUPC server doesn't really do anything but Signaling Media traffic goes from enpoint to endpoint, or endpoint to MTP etc. All other voice traffic is typically considered signaling. So an ACL that just matched all traffic to / from the CUPC server IP address would include signaling traffic for the CUPC server and little else. I didn't see the earlier email concerning the wording of the question, so this is a stab in the dark. Brian S Turner CCIE 6145 On Sun, May 13, 2012 at 11:39 AM, san r luv...@gmail.com wrote: Release Notes for Cisco Unified Personal Communicator Release 7.0 https://mail.google.com/mail/html/compose/static_files/blank_quirks.html#wp85054has got all inbound outbound port usage for CUPC http://www.cisco.com/en/US/docs/voice_ip_comm/cupc/7_0/english/release/notes/ol15710.html On Sun, May 13, 2012 at 5:27 PM, san r luv...@gmail.com wrote: I think we will need to consider the citi port number too On May 13, 2012 2:57 PM, steven moran smoran...@gmail.com wrote: I've been doing some work on this topic and have moved away from my original thoughts which were along the same lines as originally posted by Nazeer and a few others . Here are some recent thoughts on the subject the question is not particularly well worded, we can't really influence the traffic from the CUPC client to the server by applying policy on the server port. What we can do is control what the switch port does with packets from the CUPS server to the clients by classifying and applying an input ploicy on the server port. I went back to the presence section in the CUCM srnd - the only traffic between CUPC and CUPS seems be 5060 for sip/simple and 443 for SOAP, this was sort of backed up by the information in the CUPC 7.1 release. The release notes also mention that ports 16384-16424 are used for TFTP - but this would not be classed as signalling. Since there are other servers on the port any access list would need to specifiy the server IP address. If you wanted to make it really specific you could also specify the destination address of the CUPC clients as well. Finally by putting bidirectional access lists (any any eq and any eq any) you may loose the points as it might appear you are hedging your bets and do not know how directional QoS works. Also we are asked to guarantee 32k for signalling traffic if we put non signalling traffic into this policy we are not achieving the requirements. These statements are not facts, just my current opinion based on the last bits of documentation read and I am happy to discuss. Best regards. Steve ___ 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 http://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
Re: [OSL | CCIE_Voice] Lab 7 Lan QOS
If NDA is strict. Why we have lab work books. Most of them are claiming it's 'exactly ' as in lab. Even everyone is using the name CCIE - I do believe its 'Cisco ' certified internetwork expert On May 15, 2012 4:07 PM, Ken Wyan kew...@gmail.com wrote: Hi, Didn't you forget cisco NDA discuss exam questions like this? (Cisco seems given a never-ending exercise to NDA violators) You can't conclude answers this way don't hope to pass first attempt or if cisco gives such questions. Just try again youll pass next time or a in a subsequent attempt if you prepared very well using IPExpert material. Thats only I can say. Thanks On Tue, May 15, 2012 at 12:21 PM, Kevin Spicer ke...@kevinspicer.co.ukwrote: But cucm and cup both run on the same VMWare server so will use the internal vswitch to communicate only traffic to the clients will traverse the switch port. On 15 May 2012 03:41, steven moran smoran...@gmail.com wrote: Whilst in some aspects you are right in that the CUPS server is really only involved in signalling - the question requires a guarantee of 32k for signal traffic between CUPC and CUPS (that's how I read it) as we are only instructed to put a policy of the CUPS server port, then we have to be careful not to put traffic between the CUPS and CUCM into the same policy as above as this would impact of the bandwidth allocated. At the end of the day it is a badly worded question. Steve ___ 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 http://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 http://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] GOT LAB 5 in Dubai but failed
I finished my first attempt in dubai on may 14. This lab is really lengthy . You should enough luck to finish in time...started preparation for the next attempt ___ 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] GOT LAB 5 in Dubai but failed
I finished my first attempt in dubai on may 14. This lab is really lengthy . You should have enough luck to finish it in time...started preparation for the next attempt. ___ 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] Lab 7 Lan QOS
Since they specifically asked for 'CUPC Signalling ' I don't think you need to mark the RTP traffic On May 13, 2012 11:53 AM, Nazeer rahiman nazs...@yahoo.com wrote: For LAN QoS I got below question All servers are connected (running on vmware) to SW int G 1/0/4. CUPC is running in UCCX and test pc. They asked to configure one in softphone mode other one is desktop mode. also configre voice mail on both clients QoS question was - In Gig 1/0/4 , make sure all incoming CUPC signaling traffic to mark CS3 and gurantee 32k BW. anythung exess should be mark down to DSCP 8 and retransmit. My ans was - mls qos mls qos map cos-dscp 0 8 16 24 32 46 48 56 mls qos map policed-dscp 24 26 to 8 ip access-list extended voice-rtp permit udp any any range 16384 32767 ip access-list extended cupc-sig permit tcp any any eq 5060 permit tcp any any eq 5060 permit tcp any eq 5060 any permit udp any any eq 5060 permit udp any eq 5060 any permit tcp any any eq 143 permit tcp any eq 143 any permit tcp any any eq 80 permit tcp any eq 80 any permit tcp any any eq 443 permit tcp any eq 443 any permit tcp any any eq 993 permit tcp any eq 993 any permit tcp any any eq 7993 permit tcp any eq 7993 any permit tcp any any eq 389 permit tcp any eq 389 any permit tcp any any eq 2748 permit tcp any eq 2748 any config)#class-map voice-rtp (config-cmap)#match access-group name voice-rtp (config)#class-map match any cupc-sig (config-cmap)#match access-group name cupc-sig (config-cmap)#policy-map cupc (config-pmap)#class voice-rtp (config-pmap-c)#set dscp ef (config-pmap)#class cupc-sig (config-pmap-c)#police 32000 8000 exceed-action policed-dscp-transmit (config-pmap-c)#set dscp cs3 (config)#interface GigabitEthernet1/0/4 config-if)#service-policy input cupc Phone ports mls qos trust cos mls qos trust device cisco phone Server ports mls qos trust dscp I got 0 marks for this question - any body can clarify where it's wrong ? ___ 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] Lab 7 Lan QOS
I think we will need to consider the citi port number too On May 13, 2012 2:57 PM, steven moran smoran...@gmail.com wrote: I've been doing some work on this topic and have moved away from my original thoughts which were along the same lines as originally posted by Nazeer and a few others . Here are some recent thoughts on the subject the question is not particularly well worded, we can't really influence the traffic from the CUPC client to the server by applying policy on the server port. What we can do is control what the switch port does with packets from the CUPS server to the clients by classifying and applying an input ploicy on the server port. I went back to the presence section in the CUCM srnd - the only traffic between CUPC and CUPS seems be 5060 for sip/simple and 443 for SOAP, this was sort of backed up by the information in the CUPC 7.1 release. The release notes also mention that ports 16384-16424 are used for TFTP - but this would not be classed as signalling. Since there are other servers on the port any access list would need to specifiy the server IP address. If you wanted to make it really specific you could also specify the destination address of the CUPC clients as well. Finally by putting bidirectional access lists (any any eq and any eq any) you may loose the points as it might appear you are hedging your bets and do not know how directional QoS works. Also we are asked to guarantee 32k for signalling traffic if we put non signalling traffic into this policy we are not achieving the requirements. These statements are not facts, just my current opinion based on the last bits of documentation read and I am happy to discuss. Best regards. Steve ___ 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] No audio on H323 Trunk
Let's say I'm asked to setup an H323 trunk to a backbone router (address 1.1.1.1) make a call to a number hosted on that (011321234567) and then troubleshoot why the remote side has no audio and the originating site drops the call. This seems to be a H245 Negotiation issue. May be MSD failed and endpoints migt have started the communication with an incompatable media type resulting no audio. I found something interesting on Wikipedia link below http://en.wikipedia.org/wiki/H.323#H.245_Call_Control Once a call has initiated (but not necessarily fully connected) endpoints may initiate H.245 call control signaling in order to provide more extensive control over the conference. H.245 is a rather voluminous specification with many procedures that fully enable multipoint communication, though in practice most implementations only implement the minimum necessary in order to enable point-to-point voice and video communication. H.245 provides capabilities such as capability negotiation, master/slave determination, opening and closing of logical channels (i.e., audio and video flows), flow control, and conference control. It has support for both unicast and multicast communication, allowing the size of a conference to theoretically grow without bound When an H.323 device initiates communication with a remote H.323 device and when H.245 communication is established between the two entities, the Terminal Capability Set (TCS) message is the first message transmitted to the other side. After sending a TCS message, H.323 entities (through H.245 exchanges) will attempt to determine which device is the master and which is the slave. This process, referred to as Master/Slave Determination (MSD), is important, as the master in a call settles all disputes between the two devices. For example, if both endpoints attempt to open incompatible media flows, it is the master who takes the action to reject the incompatible flow ___ 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] H225 or ICT Trunk conundrum
Steve, This seems to be a H245 Negotiation issue. May be MSD failed and endpoints migt have started the communication with an incompatable media type resulting no audio. Let me know if you have any different thoughts/ comments I found something interesting on Wikipedia link below http://en.wikipedia.org/wiki/H.323#H.245_Call_Control Once a call has initiated (but not necessarily fully connected) endpoints may initiate H.245 call control signaling in order to provide more extensive control over the conference. H.245 is a rather voluminous specification with many procedures that fully enable multipoint communication, though in practice most implementations only implement the minimum necessary in order to enable point-to-point voice and video communication. H.245 provides capabilities such as capability negotiation, master/slave determination, opening and closing of logical channels (i.e., audio and video flows), flow control, and conference control. It has support for both unicast and multicast communication, allowing the size of a conference to theoretically grow without bound When an H.323 device initiates communication with a remote H.323 device and when H.245 communication is established between the two entities, the Terminal Capability Set (TCS) message is the first message transmitted to the other side. After sending a TCS message, H.323 entities (through H.245 exchanges) will attempt to determine which device is the master and which is the slave. This process, referred to as Master/Slave Determination (MSD), is important, as the master in a call settles all disputes between the two devices. For example, if both endpoints attempt to open incompatible media flows, it is the master who takes the action to reject the incompatible flow On Fri, May 11, 2012 at 1:07 AM, steven moran smoran...@gmail.com wrote: The question - set up a POC H.323 trunk to your service provider ip address x.x.x.x and route international calls, these calls should have audio issues, It is not clear whether to set up a H.225 gatekeeerp controlled or a non gatekeeer controlled ICT. I would take it that since it does not mention that the service provider is using CUCM that you cannot use an ICT (GK or non GK controlled) -although in practice this works just fine to the PSTN-WAN router and without audio issues. So I take it then the correct way is to set up a gatekeeper and a gatekeeper controlled H225 trunk. But I still do not have any audio connection issues, it would seem the only way to ensure these is to set up CUBE which breaks the H.245 negotiation, unless of course the far end PSTN-WAN has CUBE installed, which of course would result in the same H245 issues. Has anyone had audio connection issues on a correctly set up GK controlled H.225 trunk without CUBE? ___ 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 http://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] Lab 7 Lan QOS
Release Notes for Cisco Unified Personal Communicator Release 7.0 https://mail.google.com/mail/html/compose/static_files/blank_quirks.html#wp85054has got all inbound outbound port usage for CUPC http://www.cisco.com/en/US/docs/voice_ip_comm/cupc/7_0/english/release/notes/ol15710.html On Sun, May 13, 2012 at 5:27 PM, san r luv...@gmail.com wrote: I think we will need to consider the citi port number too On May 13, 2012 2:57 PM, steven moran smoran...@gmail.com wrote: I've been doing some work on this topic and have moved away from my original thoughts which were along the same lines as originally posted by Nazeer and a few others . Here are some recent thoughts on the subject the question is not particularly well worded, we can't really influence the traffic from the CUPC client to the server by applying policy on the server port. What we can do is control what the switch port does with packets from the CUPS server to the clients by classifying and applying an input ploicy on the server port. I went back to the presence section in the CUCM srnd - the only traffic between CUPC and CUPS seems be 5060 for sip/simple and 443 for SOAP, this was sort of backed up by the information in the CUPC 7.1 release. The release notes also mention that ports 16384-16424 are used for TFTP - but this would not be classed as signalling. Since there are other servers on the port any access list would need to specifiy the server IP address. If you wanted to make it really specific you could also specify the destination address of the CUPC clients as well. Finally by putting bidirectional access lists (any any eq and any eq any) you may loose the points as it might appear you are hedging your bets and do not know how directional QoS works. Also we are asked to guarantee 32k for signalling traffic if we put non signalling traffic into this policy we are not achieving the requirements. These statements are not facts, just my current opinion based on the last bits of documentation read and I am happy to discuss. Best regards. Steve ___ 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 http://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] Attempt my first exam in Dubai but I fail
Did you disable g722 from the service parameter? On May 9, 2012 2:58 PM, amr mustafa amr.m.a...@gmail.com wrote: plz if anyone can answer me did I have to do something especial in codec section than the call should work between phones with G711 and between branches with G729 to get full score instead of ((0)) and if I didn’t requested in HA to make connection with Site C by dialing 4 digits so I have to make it to get HA full score??? I did everything except if GK go down I wasn’t able to send the call to site C with + while the call go out from HQ with +E164 and I did translation in site C but didn’t appear at the screen ___ 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] Attempt my first exam in Dubai but I fail
If you set the interregion intraregion codecs then I think you're good. Also for + dialing did you check what exactly displayed on just above the softkeys while it was ringing? 'Your current options ' text will be displayed if phone is idle On May 9, 2012 2:58 PM, amr mustafa amr.m.a...@gmail.com wrote: plz if anyone can answer me did I have to do something especial in codec section than the call should work between phones with G711 and between branches with G729 to get full score instead of ((0)) and if I didn’t requested in HA to make connection with Site C by dialing 4 digits so I have to make it to get HA full score??? I did everything except if GK go down I wasn’t able to send the call to site C with + while the call go out from HQ with +E164 and I did translation in site C but didn’t appear at the screen ___ 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] Presence Client QoS marking
Hello All, Can anyone please guide me to answer the below question. Configure marking on port where the lab PC is connected. Guarantee 128k bw for the presence client signalling traffic. excees traffic may be marked to DSCP 8 and trasmit. San. ___ 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 Hangs Up
Do you have the same number (no. Of digits) configured ccm and dial-peer? On May 6, 2012 7:27 PM, Joe Fearday feard...@trinity-health.org wrote: Any ideas? The MVA application receives the call on the H323 gateway and sends to IVRMainpage.vxml. The call is prompted for PIN and Press 1 to make a call. When the desired number is entered, followed by a # there is an immediate hang up. The Remote Destination Profile CSS has visibility to the number being dialed. What might be the problem or how do I troubleshoot? Thanks, Joe ___ 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] the directories button display
Does anyone know what exact directory order required for the lab 7 ? Is it same as below? Missed Calls Placed Calls Received Calls Corporate Directory On Tue, Apr 3, 2012 at 7:44 AM, Rick Long rick.l...@ensi.com wrote: Search for restoring in Cisco Call Manager 7.01 release notes It will give you the sql statements that you need to reapply to CUCM after removing the services. HTH Rick Long -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Monday, April 02, 2012 9:55 PM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 74, Issue 8 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Directories Question (Pablo Meneses) 2. Re: UCCX failure updating rmcm user on Call manager (Baktha Muralidharan) 3. Re: UCCX failure updating rmcm user on Call manager (Gurpreet Singh Kukreja) 4. Re: switch-QoS-Quick Question (steven moran) 5. Re: switch-QoS-Quick Question (Jurassic Labs) -- Message: 1 Date: Mon, 2 Apr 2012 16:45:45 -0600 From: Pablo Meneses pmenese...@gmail.com To: ccie voice ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Directories Question Message-ID: CABz8te5zz15QHDkj3D8gRaqZz-T-7jzjKYGtpB=r6+nUH=+x...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi Experts, Does anybody know how to make sure that the directories button display the services in the correct order after you have disabled Enterprise Subscription by deleting and adding the services back? Thanks in advance. -Pablo Meneses. -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20120402/92f7c187/attachment-0001.html -- Message: 2 Date: Mon, 2 Apr 2012 18:47:27 -0400 From: Baktha Muralidharan muralic...@gmail.com To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] UCCX failure updating rmcm user on Callmanager Message-ID: ca+ne39upbj1k_n-0zmfccwkuhto+tbf9vianfz164hojasg...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Thanks to all for the tips/suggestions- Does uccx automatically create the rmcm application user or do we need to manually create it? I don't remember having to create it in the past. whenever, the uccx was partially integrated, I used to be able to do the following- - go into uccx and create/complete telephony-group and rmcm (resources, skills etc.) - on UCM, associate users to rmcm application user - configure application/trigger etc. thanks, /Baktha -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20120402/14032129/attachment-0001.html -- Message: 3 Date: Mon, 2 Apr 2012 20:00:52 -0400 From: Gurpreet Singh Kukreja tycoononway1...@gmail.com To: Baktha Muralidharan muralic...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] UCCX failure updating rmcm user on Callmanager Message-ID: CABmpSZQALgAH4b4uL+auSZP4C=y9go8ystix-q5eufpc6g1...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hey Baktha, When you run the Fresh Install setup from UCCX the first time, those users are created automatically and the CTI Enabled role is also assigned to them at the same time; no need for manual intervention. However, the steps i suggested was to bring back the RM Subsystem back in service on the CCX if it does not automatically. Let me know if you have more questions. Regards Gurpreet On Mon, Apr 2, 2012 at 6:47 PM, Baktha Muralidharan muralic...@gmail.com wrote: Thanks to all for the tips/suggestions- Does uccx automatically create the rmcm application user or do we need to manually create it? I don't remember having to create it in the past. whenever, the uccx was partially integrated, I used to be able to do the following- - go into uccx and create/complete telephony-group and rmcm (resources, skills etc.) - on UCM, associate users to rmcm application user - configure application/trigger etc. thanks, /Baktha ___ For more information regarding industry leading CCIE Lab training, please visit
