Re: [OSL | CCIE_Voice] Voice translation issue
That´s a good one! I've tried it out here and indeed does behaviour like that.. and I still can't figure out what on earth is happening that the Outpulsed digits are just 11 If you replace with /9911/, it sends only 1 !! It seems something is wrong with the 9's, because if you try to replace with /123/ for example, it works just fine. But when the replace pattern leads with a 9 something goes wrong.. Something misterious is happening in this translation-rule.. please let us know if you find out! I gave up on this one already On Sun, May 19, 2013 at 12:13 PM, Martin Sloan martinsloa...@gmail.comwrote: If I change the translation to this: voice translation-rule 8 rule 1 /^9911$/ /99911/ type any unknown plan any isdn rule 2 /^1...$/ /617394\0/ type any subscriber plan any isdn It sends '911' to the PSTN! The voice translation debug looks like this: *May 19 14:39:44.716: //-1/C76CA681803C/RXRULE/regxrule_dp_translate: calling_number=6173941002 calling_octet=0x41 called_number=99911 called_octet=0x1 I just don't understand the logic on this one. I know there's more than 1 way to skin this cat but it bugs the heck out of me to not understand this. I look at translations as a /MATCH/ /REPLACE/ setup with '/^9911$/' matching the dialed digits and then replacing the entire matched string with the replacement string '/911/'. I can't wrap my head around what's going on with this one. I tried it on the BR2 and HQ gateways and got the same results. On Sun, May 19, 2013 at 10:41 AM, Martin Sloan martinsloa...@gmail.comwrote: Thanks, Bill. Any thoughts on why the gw is only sending '11' to the PSTN? If the dial-peer is stripping explicitly matched digits it should strip all of the digits. It just doesn't make any sense to me that the voice translation debug and test shows that the digit manipulation happens correctly but the gw sends only '11'. I'm really confused about that! On Sun, May 19, 2013 at 10:33 AM, Bill whl...@gmail.com wrote: voice translation-rule 8 *rule 1 // // type any unknown plan any isdn* rule 2 /^1...$/ /617394\0/ type any subscriber plan any isdn Sent from my iPad On May 19, 2013, at 9:07 AM, Ravindra Lakpriya lakpr...@gmail.com wrote: In the dial peer configure no digit strip. :) On Sun, May 19, 2013 at 5:47 PM, Martin Sloan martinsloa...@gmail.comwrote: I have a voice translation rule in place for '9911' calls on BR1 during SRST. I'm running into some odd behavior (from my perspective) and I'm hoping it's a config issue I'm just not spotting. I have the translation profile applied to the dial peer and the only other translation that would be in the calling path is on the voice port but even that one is applied to inbound calls for stripping down to 4 digits. Here's the config related to this dial-peer: voice translation-rule 8 rule 1 /^9911$/ /911/ type any unknown plan any isdn rule 2 /^1...$/ /617394\0/ type any subscriber plan any isdn voice translation-profile 9911 translate calling 8 translate called 8 dial-peer voice 9911 pots translation-profile outgoing 9911 destination-pattern 9911$ port 0/0/0:23 BR1-RTR#test voice translation-rule 8 9911 Matched with rule 1 Original number: 9911 Translated number: 911 Original number type: none Translated number type: unknown Original number plan: none Translated number plan: isdn -Debug ISDN q931- Calling Party Number i = 0x4181, '6173941002' Plan:ISDN, Type:Subscriber(local) Called Party Number i = 0x81, '11' Plan:ISDN, Type:Unknown -Debug voice translation- *May 19 13:38:29.615: //-1/394E525380FD/RXRULE/regxrule_vp_translate: calling_number=6173941002 calling_octet=0x41 called_number=911 called_octet=0x1 From testing the voice translation and checking the translation debugs, it looks like everything works but the gw sends only '11' to the PSTN. Can someone please school me on this one? ___ 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 -- Ravindra Lakpriya ___ 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 -- *Bruno Takahashi da Silva* ___ 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
Re: [OSL | CCIE_Voice] Voice translation issue
flash0:c2900-universalk9-mz.SPA.152-4.M1.bin here On Sun, May 19, 2013 at 6:20 PM, Bill whl...@gmail.com wrote: What ios are you guys running? I don't see this happening in ios I am running flash:c2800nm-adventerprisek9_ivs_li-mz.124-20.T.bin Sent from my iPad On May 19, 2013, at 12:54 PM, Martin Sloan martinsloa...@gmail.com wrote: Bruno, thanks for giving it a try. I feel a little better about my sanity now :-) It is very strange and I also had the same results as you, with only the '9' exhibiting this behavior in the replace string. I was hoping to use translation rules for all my dial-peer digit manipulation but with this issue coming up, I think I'll use a forward digits command on the dial peer for 911/9911. I would love to know what's going on here though! On Sun, May 19, 2013 at 1:26 PM, Bruno Takahashi brun...@gmail.comwrote: That´s a good one! I've tried it out here and indeed does behaviour like that.. and I still can't figure out what on earth is happening that the Outpulsed digits are just 11 If you replace with /9911/, it sends only 1 !! It seems something is wrong with the 9's, because if you try to replace with /123/ for example, it works just fine. But when the replace pattern leads with a 9 something goes wrong.. Something misterious is happening in this translation-rule.. please let us know if you find out! I gave up on this one already On Sun, May 19, 2013 at 12:13 PM, Martin Sloan martinsloa...@gmail.comwrote: If I change the translation to this: voice translation-rule 8 rule 1 /^9911$/ /99911/ type any unknown plan any isdn rule 2 /^1...$/ /617394\0/ type any subscriber plan any isdn It sends '911' to the PSTN! The voice translation debug looks like this: *May 19 14:39:44.716: //-1/C76CA681803C/RXRULE/regxrule_dp_translate: calling_number=6173941002 calling_octet=0x41 called_number=99911 called_octet=0x1 I just don't understand the logic on this one. I know there's more than 1 way to skin this cat but it bugs the heck out of me to not understand this. I look at translations as a /MATCH/ /REPLACE/ setup with '/^9911$/' matching the dialed digits and then replacing the entire matched string with the replacement string '/911/'. I can't wrap my head around what's going on with this one. I tried it on the BR2 and HQ gateways and got the same results. On Sun, May 19, 2013 at 10:41 AM, Martin Sloan martinsloa...@gmail.comwrote: Thanks, Bill. Any thoughts on why the gw is only sending '11' to the PSTN? If the dial-peer is stripping explicitly matched digits it should strip all of the digits. It just doesn't make any sense to me that the voice translation debug and test shows that the digit manipulation happens correctly but the gw sends only '11'. I'm really confused about that! On Sun, May 19, 2013 at 10:33 AM, Bill whl...@gmail.com wrote: voice translation-rule 8 *rule 1 // // type any unknown plan any isdn* rule 2 /^1...$/ /617394\0/ type any subscriber plan any isdn Sent from my iPad On May 19, 2013, at 9:07 AM, Ravindra Lakpriya lakpr...@gmail.com wrote: In the dial peer configure no digit strip. :) On Sun, May 19, 2013 at 5:47 PM, Martin Sloan martinsloa...@gmail.com wrote: I have a voice translation rule in place for '9911' calls on BR1 during SRST. I'm running into some odd behavior (from my perspective) and I'm hoping it's a config issue I'm just not spotting. I have the translation profile applied to the dial peer and the only other translation that would be in the calling path is on the voice port but even that one is applied to inbound calls for stripping down to 4 digits. Here's the config related to this dial-peer: voice translation-rule 8 rule 1 /^9911$/ /911/ type any unknown plan any isdn rule 2 /^1...$/ /617394\0/ type any subscriber plan any isdn voice translation-profile 9911 translate calling 8 translate called 8 dial-peer voice 9911 pots translation-profile outgoing 9911 destination-pattern 9911$ port 0/0/0:23 BR1-RTR#test voice translation-rule 8 9911 Matched with rule 1 Original number: 9911 Translated number: 911 Original number type: none Translated number type: unknown Original number plan: none Translated number plan: isdn -Debug ISDN q931- Calling Party Number i = 0x4181, '6173941002' Plan:ISDN, Type:Subscriber(local) Called Party Number i = 0x81, '11' Plan:ISDN, Type:Unknown -Debug voice translation- *May 19 13:38:29.615: //-1/394E525380FD/RXRULE/regxrule_vp_translate: calling_number=6173941002 calling_octet=0x41 called_number=911 called_octet=0x1 From testing the voice translation and checking the translation debugs, it looks like everything works but the gw sends only '11' to the PSTN. Can someone please school me on this one? ___ For more information regarding
Re: [OSL | CCIE_Voice] Removing B-ACD from running-config IOS 12.4.20?
Maybe a : ROUTER(config)#application ROUTER(config-app)# service default To return it to default config? On Sat, Jan 26, 2013 at 11:52 AM, Jason Aarons scubajas...@gmail.comwrote: Is there any way to remove B-ACD from the running config, short of doing a write erase? Trying to default my router back after practing B-ACD. ** ** ** ** With these commands B-ACD is still present in the running-config. ** ** ROUTER(config)#application ROUTER(config-app)#no service app-b-acd-aa ROUTER(config-app)#no service app-b-acd ROUTER(config)#no application ^ % Invalid input detected at '^' marker. ___ 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 -- *Bruno Takahashi da Silva* ___ 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] CME Call List Presence
Thanks Kevin! By doing a create cnf-file, I worked without a reboot! :) On Fri, Oct 5, 2012 at 9:15 PM, Edgar Feliz ejzi...@gmail.com wrote: Happens to me every time. It will not work unless I reboot and I have the the create cnf and reset the phones too. Edgar On Fri, Oct 5, 2012 at 12:52 PM, Bruno Nonogaki brun...@gmail.com wrote: Hello, When I configure CME Call List presence, I always have to reboot the router to make it work. Does this also happen with you guys? Or is this a Proctor Lab's bug? Even after an ephone reset, I cannot see the presence indication in call lists. But right after a reboot, it works perfectly. The configuration I apply is as follow: presence presence call-list sip-ua presence enable ephone-dn X allow watch Thank you, Bruno ___ 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] CME Call List Presence
Hello, When I configure CME Call List presence, I always have to reboot the router to make it work. Does this also happen with you guys? Or is this a Proctor Lab's bug? Even after an ephone reset, I cannot see the presence indication in call lists. But right after a reboot, it works perfectly. The configuration I apply is as follow: presence presence call-list sip-ua presence enable ephone-dn X allow watch Thank you, Bruno ___ 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] Plan and Type fields
Hey guys, If the question asks me to mark a Called Party Number *Type* as Subscriber/Local/International, but does not say anything about Called Party Numbering *Plan*, am I supposed to mark it with ISDN also? Thank you, Bruno ___ 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] Voice Mail Pilot 2220 Issue
I think the first question is, can you dial to 2220? I assume not since you already get a busy tone when pressing the Message button. 2220 is configured as the VoicemailPilot right? And did you already configured the LineGroup/HuntList where 2220 will hunt for the voicemail ports? Maybe that is the problem. On Mon, Sep 3, 2012 at 8:25 AM, Tejveer singh Panwar tejvee...@yahoo.inwrote: Hi All, I am doing Lab5. When i am press message button on phone so call should go on voice mail 2220 but i am getting busy tone. when i direct dial voice mail port 2111 from phone, it is going on voice mail but my voice mail pilot is not working. Call Manager users successfully integrated with unity. All user is successfully downloaded. Please help my. ___ 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] MGCP Calls to DN not in use and debug mgcp packet
I think you are trying to see messages which are backhauled to CUCM (q931). Gateway doesn't care about numbers, dial plans and stuffs like that, because it is all inside the q931 messages, which you can see with debug ccm-manager backhaul packets. So the gateway doesn't ask the call-agent about the number. It does't have this intelligence... On Thu, Aug 30, 2012 at 2:03 PM, Jason Aarons (AM) jason.aar...@dimensiondata.com wrote: Playing around with debug mgcp packet on gateway. ** ** CallManager doesn’t have any DNs matching the number. How come I don’t see the gateway asking the call-agent about the number via MGCP messages? How does the gateway know it’s unassigned without asking the call-agent? ** ** I suspect a RQNT R: D/[0-9ABCD*#] is in a loop watching the digits, but the number should be sent to callmanager to check the dialplan right? Or does gateway have dial-pan in memory, etc? ** ** ** ** Debug mgcp packet Debug isdn q931 ** ** Aug 30 16:52:41.620: ISDN Se0/0/0:23 Q931: RX - SETUP pd = 8 callref = 0x008E Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Progress Ind i = 0x8583 - Origination address is non-ISDN Display i = 'Emergency Services' Calling Party Number i = 0x0080, '911' Plan:Unknown, Type:Unknown Called Party Number i = 0xC1, '2522009' Plan:ISDN, Type:Subscriber(local) Aug 30 16:52:41.628: ISDN Se0/0/0:23 Q931: TX - RELEASE_COMP pd = 8 callref = 0x808E Cause i = 0x8081 - Unallocated/unassigned number ** ** No mgcp packets in debug ___ 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] MGCP Calls to DN not in use and debug mgcp packet
debug ccm-manager backhaul packets But it is not so clear text as debug isdn q931... Your output will be something like that: cmbrl_send_pak: -- Sending backhauled msg for Se0/0/0:15 : | bk_msg_type = DATA_IND | bk_chan_id (slot:port) = 0:0 | Q.931 length = 62 | Q.931 message type: SETUP | Q.931 message = 080200830504038090A31803A983811E028183280F5053544E2D53495445422D50484F4E6C0D118031393732353235323232327009813234303434303032 Aug 30 22:44:37.120: cmbh_rcv_callback: -- Receiving backhaul msg for Se0/0/0:15 : | bk_msg_type = DATA_REQ | bk_chan_id (slot:port) = 0:0 | Q.931 length = 10 | Q.931 message type: CALL PROCEEDING | Q.931 message = 08028083021803A98381 SiteC-RTR# Aug 30 22:44:37.120: cmbh_rcv_callback: -- Receiving backhaul msg for Se0/0/0:15 : | bk_msg_type = DATA_REQ | bk_chan_id (slot:port) = 0:0 | Q.931 length = 9 | Q.931 message type: ALERTING | Q.931 message = 08028083011E028088 On Thu, Aug 30, 2012 at 3:42 PM, Jason Aarons (AM) jason.aar...@dimensiondata.com wrote: Is there a debug command to view the Q931 backhaul messages? ** ** About to Wireshark the Q931 backhaul.. ** ** *From:* Ovidiu Popa [mailto:ovi.p...@gmail.com] *Sent:* Thursday, August 30, 2012 2:41 PM *To:* Jason Aarons (AM) *Cc:* ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] MGCP Calls to DN not in use and debug mgcp packet ** ** ** ** Hello Jason ** ** That's because the q931 is terminated on the cucm and it (the cucm) knows the dn is unavailable. Basically the mgcp gw is used only to open rtp streams (connections) as the call setup is handled by the cucm. ** ** HTH ** ** Regards, Ovidiu ** ** On Aug 30, 2012, at 19:03, Jason Aarons (AM) jason.aar...@dimensiondata.com wrote: Playing around with debug mgcp packet on gateway. CallManager doesn’t have any DNs matching the number. How come I don’t see the gateway asking the call-agent about the number via MGCP messages? How does the gateway know it’s unassigned without asking the call-agent? I suspect a RQNT R: D/[0-9ABCD*#] is in a loop watching the digits, but the number should be sent to callmanager to check the dialplan right? Or does gateway have dial-pan in memory, etc? Debug mgcp packet Debug isdn q931 Aug 30 16:52:41.620: ISDN Se0/0/0:23 Q931: RX - SETUP pd = 8 callref = 0x008E Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Progress Ind i = 0x8583 - Origination address is non-ISDN Display i = 'Emergency Services' Calling Party Number i = 0x0080, '911' Plan:Unknown, Type:Unknown Called Party Number i = 0xC1, '2522009' Plan:ISDN, Type:Subscriber(local) Aug 30 16:52:41.628: ISDN Se0/0/0:23 Q931: TX - RELEASE_COMP pd = 8 callref = 0x808E Cause i = 0x8081 - Unallocated/unassigned number No mgcp packets in debug ___ 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 itevomcid ___ 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] MGCP Calls to DN not in use and debug mgcp packet
Oh, by the way... You can extract the dialed number in the Q.931 setup message by erasing the 3 at the final. So for example, this setup message is: 080200830504038090A31803A983811E028183280F5053544E2D53495445422D50484F4E6C0D118031393732353235323232327009813234303434303032 If you get the final part of this string, we have: 3234303434303032 If you delete the 3: 24044002 This is the dialed number. On Thu, Aug 30, 2012 at 3:47 PM, Bruno Nonogaki brun...@gmail.com wrote: debug ccm-manager backhaul packets But it is not so clear text as debug isdn q931... Your output will be something like that: cmbrl_send_pak: -- Sending backhauled msg for Se0/0/0:15 : | bk_msg_type = DATA_IND | bk_chan_id (slot:port) = 0:0 | Q.931 length = 62 | Q.931 message type: SETUP | Q.931 message = 080200830504038090A31803A983811E028183280F5053544E2D53495445422D50484F4E6C0D118031393732353235323232327009813234303434303032 Aug 30 22:44:37.120: cmbh_rcv_callback: -- Receiving backhaul msg for Se0/0/0:15 : | bk_msg_type = DATA_REQ | bk_chan_id (slot:port) = 0:0 | Q.931 length = 10 | Q.931 message type: CALL PROCEEDING | Q.931 message = 08028083021803A98381 SiteC-RTR# Aug 30 22:44:37.120: cmbh_rcv_callback: -- Receiving backhaul msg for Se0/0/0:15 : | bk_msg_type = DATA_REQ | bk_chan_id (slot:port) = 0:0 | Q.931 length = 9 | Q.931 message type: ALERTING | Q.931 message = 08028083011E028088 On Thu, Aug 30, 2012 at 3:42 PM, Jason Aarons (AM) jason.aar...@dimensiondata.com wrote: Is there a debug command to view the Q931 backhaul messages? ** ** About to Wireshark the Q931 backhaul.. ** ** *From:* Ovidiu Popa [mailto:ovi.p...@gmail.com] *Sent:* Thursday, August 30, 2012 2:41 PM *To:* Jason Aarons (AM) *Cc:* ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] MGCP Calls to DN not in use and debug mgcp packet ** ** ** ** Hello Jason ** ** That's because the q931 is terminated on the cucm and it (the cucm) knows the dn is unavailable. Basically the mgcp gw is used only to open rtp streams (connections) as the call setup is handled by the cucm. ** ** HTH ** ** Regards, Ovidiu ** ** On Aug 30, 2012, at 19:03, Jason Aarons (AM) jason.aar...@dimensiondata.com wrote: Playing around with debug mgcp packet on gateway. CallManager doesn’t have any DNs matching the number. How come I don’t see the gateway asking the call-agent about the number via MGCP messages? How does the gateway know it’s unassigned without asking the call-agent?* *** I suspect a RQNT R: D/[0-9ABCD*#] is in a loop watching the digits, but the number should be sent to callmanager to check the dialplan right? Or does gateway have dial-pan in memory, etc? Debug mgcp packet Debug isdn q931 Aug 30 16:52:41.620: ISDN Se0/0/0:23 Q931: RX - SETUP pd = 8 callref = 0x008E Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Progress Ind i = 0x8583 - Origination address is non-ISDN Display i = 'Emergency Services' Calling Party Number i = 0x0080, '911' Plan:Unknown, Type:Unknown Called Party Number i = 0xC1, '2522009' Plan:ISDN, Type:Subscriber(local) Aug 30 16:52:41.628: ISDN Se0/0/0:23 Q931: TX - RELEASE_COMP pd = 8 callref = 0x808E Cause i = 0x8081 - Unallocated/unassigned number No mgcp packets in debug ___ 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 itevomcid ___ 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] predot trailing # is required to strip for international numbers for sccp phones??
Hi Krishna, You don't need trailing # for Route Patterns. But you need when you have a Translation Pattern that sends the call to a Route Pattern. For example, if you have a TP 9011.!#, which Discards Digits pre-dot and prefix a + sign. And you have a Route Pattern \+.! In this case, you will need a trailing # in the Translation Pattern. HTH Bruno On Thu, Aug 30, 2012 at 1:55 PM, Krishna vinayak_...@yahoo.com wrote: hi guys... i saw kevin's video about call routing section where he discussed about predot trailing # i.e. for a pattern 9.011!#, he said to put the digit discard as predot trailing #... my question is that terminating character has to stripped off while sending the call to gateway??? is it mandatory to do this??? i know for sip phones it is required since rfc 3261 cites the # symbol is no more recognized as terminating character..in sip terminology the # is represented as '%23'... in short, do i have to do predot trailing# for sccp phones if the dial pattern requires that there shouldn't be no interdigit timeout... please advice me on this matter... 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] H.323 Troubleshooting
Hello guys, I was practicing some troubleshooting today, making some changes in the PSTN router and analysing the SDI traces. And I found out in a codec mismatch situation, the trace is different if I configure the trunk as an H.323 gateway or as an Inter Cluster Trunk (non GK controlled). If I configure an ICT, I can clearly see the incoming TCS from the PSTN supporting g711u, then the outgoing TCS from CUCM saying it supports g729 only. And finally the PSTN sends a terminalCapabilitySetReject, and the call is released with cause code C1 (which is Bearer capability not implemented). But if I configure an H.323 gateway, I can only see the incoming TCS from the PSTN saying it supports g711u only, and after that, the call is released. CUCM does not send its TCS, and I cannot see any terminalCapabilitySetReject message. The cause code now is AF (which is Resources unavailable, unspecified: The channel or service that the user requests is unavailable for an unknown reason. This problem is usually temporary.) Is this the correct behavior? So how can I explain a codec mismatch situation in a H.323 gateway, since no message clearly says that? Thanks, Bruno ___ 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] CUPC - Login (DNS)
Salman, What I do is to change the hostname to IP Address in CUPS: System Topology Do this before starting the services, otherwise you will need to reboot the server. Regards, Bruno On Mon, Aug 20, 2012 at 6:10 PM, Salman Shah s.s...@site-technology.comwrote: ** ** Hi, ** ** Just a quick question, for CUPC is it mandatory to have DNS on local system to resolve FQDN of CUPS or host file entry will be enough. OR even that is not required. ** ** Thank you. ** ** Regards, Salman Shah. ** ** ___ 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] CUE - DEAD air
Have you tried no sccp / sccp? I have already gotten some issues with audio, and it was a bug in the media resources. On Wed, Aug 15, 2012 at 2:37 PM, Randall Crumm rrcr...@yahoo.com wrote: Hello, When I call sc ph2 from another site I am getting dead air once it drops to VM. If I press the messages button from sc ph2 I hear the correct greeting. CUCM -- CUE intergration I see the CTI port on the screen of the calling phone. Any thoughts? Cheers, Randall ___ 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] CUE - New Labs Lab 2
If you configured MTP with g729r8, don't forget to add this codec in the XCD, since it is not the default. Or change the MTP to g729ar8. On Sun, Aug 12, 2012 at 6:54 PM, Randall Crumm rrcr...@yahoo.com wrote: The difference I can see between the sights is RSVP CAC between SA and SC. The is no CAC between SB and SC. Probably not the issue, but I am not sure while I am scratching my head. Cheers, Randall -- *From:* Randall Crumm rrcr...@yahoo.com *To:* Online Study ccie_voice@onlinestudylist.com *Sent:* Sunday, August 12, 2012 2:18 PM *Subject:* [OSL | CCIE_Voice] CUE - New Labs Lab 2 I set up cue at SC it works from SB and SC , but when I call from SA I get a fast busy after 10 seconds, which is the fwd noan setting. Any thoughts? Cheers, Randall ___ 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
Re: [OSL | CCIE_Voice] no prompts with bacd
Hi Krishna, If you are asked to hear Thank you for calling, you can use en_bacd_welcome.au as drop-through audio. So, use this parameter: param drop-through-prompt _bacd_welcome.au And make sure you have the file en_bacd_welcome.au in flash: (root directory), since your location is paramspace english location flash:, and your language is paramspace english language en Bruno On Sat, Aug 11, 2012 at 1:22 AM, Krishna vinayak_...@yahoo.com wrote: hi folks, i configured on site C cme with the bacd script using the cisco configuration example guide... but unfortunately it didnt work for me while i try to establish the requirements.. the requirement is 1.) thank you or welcome prompt 2.) call route to ephone-hunt group 4000 3.) if both phones are busy then it should play busy prompt here is my configuration: application service app-b-acd-aa param voice-mail 4110 paramspace english index 0 param max-time-call-retry 700 param service-name app-b-acd param number-of-hunt-grps 1 param drop-through-option 1 paramspace english language en param handoff-string app-b-acd-aa param max-time-vm-retry 2 paramspace english location flash: param aa-pilot 4000 param second-greeting-time 60 param drop-through-prompt _dt_prompt.au param busy _bacd_allagentsbusy.au param welcome _bacd_welcome.au param call-retry-timer 15 ! service app-b-acd param queue-len 10 param aa-hunt1 4000 param queue-manager-debugs 1 param number-of-hunt-grps 1 can anyone tell me what _dt_prompt.au stands for... i dont this .au file anywhere on the cme router flash... and moreover param weclome and param busy are the one created by me in the application but no use since it didnt work.. any help 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
[OSL | CCIE_Voice] UCCX DSCP marking
Hello guys, Does anybody know if UCCX marks its packets with DSCP by default? If we are told on router to trust the EF/CS3 signalling comming from the switches, can I set mls qos trust dscp on the CCX interface, and assume it is marking RTP with EF and JTAPI with CS3? Thanks, Bruno ___ 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] UCCX DSCP marking
Actually, CS3 is not a problem since we are not crossing CCX signalling traffic over the WAN. But what about RTP from the Prompts? On Fri, Aug 10, 2012 at 1:21 PM, Bruno Nonogaki brun...@gmail.com wrote: Hello guys, Does anybody know if UCCX marks its packets with DSCP by default? If we are told on router to trust the EF/CS3 signalling comming from the switches, can I set mls qos trust dscp on the CCX interface, and assume it is marking RTP with EF and JTAPI with CS3? Thanks, Bruno ___ 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- is it necessary
Hello Krishna, Yes, you are right. ntp master is not required. If you do ntp master, it may synchronize with its internal clock. It is a big mistake a lot of people do, including me before the OWLE Bootcamp, which I really recommend. Regards, Bruno On Sun, Aug 5, 2012 at 2:37 AM, Krishna vinayak_...@yahoo.com wrote: hi folks, i see some guys posts on ntp master command on the hq router ... i was wondering why one would be needing ntp master command when it is already being synchronized with external ntp server ntp master will infact mess up the time if not configured correctly since ntp master takes the stratum from the hardware(device) and be careful when putting the command ntp master .. if it is required then it is advised to keep the stratum number high compared to the extrenal ntp server... please correct me guys if i m wrong precisely, i felt that ntp master command is not required if that device is synchronized with external ntp server.. any comments on my advice 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] Switch QOS query
Hi Vir, I agree with Justin regarding the issue with the requirements. And I also recommend you the Kevin Wallace's video: http://www.youtube.com/watch?v=IA4iOrn2eiU Regards, Bruno On Sun, Aug 5, 2012 at 10:10 AM, Justin McIntyre justin.mcint...@blackbox.com wrote: So I believe your on the right track with your QOS config but there are a few things that need to be modified. 1. I see an issue with your requirements. Have the priority-queue enabled but then also give queue 1 30% bandwidth. If priority-queue out is enabled then this over-rides the bandwidth command for that queue. I know you had some other questions as well specifically about how to drop certain traffic if a queue were 80% full. My suggestion to you would be to review Vik Mahlis QOS blog on the IPEXPERT website. Go to blog.ipexpert.com and select the voice blog on the left. Then look for the QOS section. I think this will clear up most of your questions and get you on your way. Thanks, Justin McIntyre This email and any files transmitted with it are confidential and are intended for the sole use of the individual to whom they are addressed. Black Box Corporation reserves the right to scan all e-mail traffic for restricted content and to monitor all e-mail in general. If you are not the intended recipient or you have received this email in error, any use, dissemination or forwarding of this email is strictly prohibited. If you have received this email in error, please notify the sender by replying to this email. ___ 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] Download a .wav file from CUCM
Hi, The MOH files are not stored in TFTP, but under the mohprep folder. You can view them with the command file list activelog mohprep So if you want to download them, you should do a file get activelog mohprep/file.wav But I think you can only transfer the file via SFTP, and not FTP or TFTP. Regards, Bruno On Sat, Aug 4, 2012 at 3:28 PM, Randall Crumm rrcr...@yahoo.com wrote: Hello, How do I download a .wav file (userd for MOH) from CUCM? I am trying to tftp the file down using tftpd32 but it always says file and hostname firleds are required. I did fill out the fields Something is wrong. Any help on tftpd32 or something else? Thanks, Thanks, Cheers, Randall ___ 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] srst - voicemail for cue intergrated with cucm
Hi Krishna, Yes, just create a sip dial-peer to CUE... And for MWI to work, enable unsolicited notify: sip-ua mwi server CUE-IP unsolicited Regards, Bruno On Fri, Aug 3, 2012 at 3:43 PM, Krishna vinayak_...@yahoo.com wrote: hi folks, i have a site C with cue integrated with cucm, and the site C phones are registered to call manager as well. when site operates in SRST, how can i able to make voicemail to work, since CUE is integrated with cucm... Does CUE supports both ccm and cue features or the voicemail doesn't work in srst mode?? any advice or suggestion on this query 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] srst - voicemail for cue intergrated with cucm
Sorry, mwi-server ipv4:x.x.x.x unsolicited Rgds, Bruno On Fri, Aug 3, 2012 at 4:31 PM, Bruno Nonogaki brun...@gmail.com wrote: Hi Krishna, Yes, just create a sip dial-peer to CUE... And for MWI to work, enable unsolicited notify: sip-ua mwi server CUE-IP unsolicited Regards, Bruno On Fri, Aug 3, 2012 at 3:43 PM, Krishna vinayak_...@yahoo.com wrote: hi folks, i have a site C with cue integrated with cucm, and the site C phones are registered to call manager as well. when site operates in SRST, how can i able to make voicemail to work, since CUE is integrated with cucm... Does CUE supports both ccm and cue features or the voicemail doesn't work in srst mode?? any advice or suggestion on this query 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] SIP/SCCP Phones ???
Yes, I agree with Dimuthu. Do your firmware upgrade on CUCM... But to save you time on proctorlabs, what I use to do is to take a look at the lab before I start the session, and upgrade the firmwares locally in my home lab. Upgrading it during the session is a pain in the ass. Regards, Bruno On Wed, Aug 1, 2012 at 8:21 AM, Dimuthu dim...@yahoo.com wrote: Firmware upgrades via WAN links takes few hours (at times) whether it's Proctorlabs or actual Cisco Lab facility. SIP -- SCCP conversion procedure varies depending on phone model. If you are using 7965 phones , best way is to first register to CUCM using existing phone type (SIP or SCCP). In phone configuration page you can specify individual phone load file. You can use new phone-load file name here (to be converted). After registering to CUCM , phone will then upgrade to new phone load. Once upgrade is done delete the phone from CUCM re-add with other type (SIP/SCCP). Also I think it's better to use CUCM rather than CME for firmware upgrades. Thanks *From:* Nicolas MICHEL mcl.nico...@gmail.com *To:* Online Study ccie_voice@onlinestudylist.com *Sent:* Wednesday, August 1, 2012 4:11 PM *Subject:* [OSL | CCIE_Voice] SIP/SCCP Phones ??? Hey Guys. Just did my first session this morning and noticed that in a few labs that some phones were using SIP. One of my phone was luckily using SIP but upgraded through the WAN and I lost so much time (upgrade took 3 hours ... blame the TFTP latency) Can you guys tell me if I can revert the phone to SCCP and continue to use it throughout my studies ? Also I should modify the XMLdefault.cnf.xml on a local TFTP server to modify it to SCCP right ? Thanks for your help :) Nic -- Nicolas MICHEL Ingenieur Réseaux CCIE #29410 ___ 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
Re: [OSL | CCIE_Voice] [OSL|CCIE Voice] no notification sound on ip phone for messages
And also check the Authenticate URL on Enterprise Parameters... On Mon, Jul 23, 2012 at 5:44 PM, Justin McIntyre justin.mcint...@blackbox.com wrote: this means you do not have all configuration completed. You need to check these few places: 1. user licensed for CUP in UCM 2. You have created the Application user for IPPM(PhoneMessenger) in UCM and the phone you are using the IPPM service on is associated with this Application user.. Also make sure this user is CTI enabled and that the passwords in UCM and Application IPPM are the same, also make sure the IPPM status is set to on. Additionally if you want to see presence updates make sure you have your SIP trunk from UCM to CUPS set properly and that the user that you want to see presence updates from has been associated with the line/DN. This email and any files transmitted with it are confidential and are intended for the sole use of the individual to whom they are addressed. Black Box Corporation reserves the right to scan all e-mail traffic for restricted content and to monitor all e-mail in general. If you are not the intended recipient or you have received this email in error, any use, dissemination or forwarding of this email is strictly prohibited. If you have received this email in error, please notify the sender by replying to this email. ___ 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] LAB8 - Question 7.1
Hey guys, Yeah, maybe num-exp and dialplan-pattern could be used for inbound calls. To translate the number received from PSTN to 4 digits extension. But it can't manipulate the ANI to 10 Digits on outgoing calls. I would still need voice translation-rules. And what do you mean by a number expansion on Serial or voice-port? Do you mean the regular translation-rule X instead of voice translation-rules / voice translation-profile? If so, the question says you can't use both... :( Thank you, Bruno On Sun, Jun 17, 2012 at 9:35 AM, Justin McIntyre justin.mcint...@blackbox.com wrote: You could try using number expansion. Additionally, you can apply number expansion via a regular expression to either the Serial Interface or the Voice-Port I can't remember which. Technically this is not a voice translation-rule. Thanks, Justin McIntyre This email and any files transmitted with it are confidential and are intended for the sole use of the individual to whom they are addressed. Black Box Corporation reserves the right to scan all e-mail traffic for restricted content and to monitor all e-mail in general. If you are not the intended recipient or you have received this email in error, any use, dissemination or forwarding of this email is strictly prohibited. If you have received this email in error, please notify the sender by replying to this email. ___ 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] LAB8 - Question 7.1
Hi everybody, LAB8 Q7.1 asks us to setup SRST for BR1 (which has SIP Phones). And when registered on SRST gateway, the BR1 phone should be able to dial local, emergency and receive calls from PSTN. But for this task we are not allowed to create voice translation-rules nor translation-rules. DSG solution is completely based on voice-translation rules. And indeed I cannot see another way to accomplish the task. Does anyone have an idea? Thanks, Bruno ___ 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] does MLP LFI break your wan?
Hey guys, Has anyone discovered what goes wrong with the interfaces after applying MLP LFI? I am running the same problem on my lab3. When I configure MLP LFI on both sides, the virtual interface stays up/down: Virtual-Access110.10.111.2 YES TFTP up down I have already rebooted the router several times, but it does not come up. The auto qos is applied on both sides: HQ: interface Serial0/0/1:0.1 point-to-point bandwidth 384 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 201 ppp Virtual-Template200 class AutoQoS-FR-Se0/0/1:0-201 auto qos voip trust fr-atm ip rsvp bandwidth 112 BR1: interface Serial0/0/1:0.1 point-to-point bandwidth 384 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 101 ppp Virtual-Template200 class AutoQoS-FR-Se0/0/1:0-101 auto qos voip trust fr-atm ip rsvp bandwidth 112 Has anyone run into this issue? Thank you, Bruno On Sat, Mar 5, 2011 at 6:51 PM, adam compton com...@gmail.com wrote: Yeah I've had this problem before. First off, you have to configure it on both sides for it to come back up. It usually comes right back up once you run auto qos voip fr-atm on both sides. If it doesn't, wr mem and reboot both routers should fix it. In proctorlabs, I haven't had to reboot the routers in this situation. In GNS3, I have to reboot them every single time. On Fri, Mar 4, 2011 at 11:01 PM, CCIE Voice cc...@corb.net wrote: I am having the exact same problem. virtual-access1, virtual-access2, and virtual-template200 are all in a down/down or up/down state. Not sure how to rectify it. Anyone else experienced this and figure out what was wrong? On Tue, Nov 23, 2010 at 12:33 PM, Romain Mullier romain.mull...@gmail.com wrote: Hi guys, was working on lab3, RSVP configuration worked well but after applying MLP LFI between HQ and BR1, I cannot bring the virtual interfaces up. Has anyone seen this before? (Routers have been reloaded) Thanks for your help. HQ class-map match-any AutoQoS-VoIP-RTP-Trust match ip dscp ef class-map match-any AutoQoS-VoIP-Control-Trust match ip dscp cs3 match ip dscp af31 ! ! policy-map AutoQoS-Policy-Trust class AutoQoS-VoIP-RTP-Trust priority 56 compress header ip rtp class AutoQoS-VoIP-Control-Trust bandwidth 17 class class-default fair-queue ! interface Serial0/0/1:0 no ip address encapsulation frame-relay no fair-queue frame-relay traffic-shaping frame-relay lmi-type ansi ip rsvp bandwidth ! interface Serial0/0/1:0.1 point-to-point bandwidth 384 ip pim sparse-dense-mode ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 201 ppp Virtual-Template200 class AutoQoS-FR-Se0/0/1:0-101 ip rsvp bandwidth 112 ! ! interface Virtual-Template200 bandwidth 384 ip address 10.10.111.1 255.255.255.0 ip ospf mtu-ignore ppp multilink ppp multilink interleave ppp multilink fragment delay 10 service-policy output AutoQoS-Policy-Trust ip rsvp bandwidth 112 ! map-class frame-relay AutoQoS-FR-Se0/0/1:0-101 frame-relay cir 384000 frame-relay bc 3840 frame-relay be 0 frame-relay mincir 384000 ! Virtual-Access110.10.111.1 YES TFTP up*down* On BR1 ! class-map match-any AutoQoS-VoIP-RTP-Trust match ip dscp ef class-map match-any AutoQoS-VoIP-Control-Trust match ip dscp cs3 match ip dscp af31 ! ! policy-map AutoQoS-Policy-Trust class AutoQoS-VoIP-RTP-Trust priority 56 compress header ip rtp class AutoQoS-VoIP-Control-Trust bandwidth 17 class class-default fair-queue ! ! interface Serial0/0/1:0 no ip address encapsulation frame-relay IETF no fair-queue frame-relay traffic-shaping frame-relay lmi-type ansi ip rsvp bandwidth ! interface Serial0/0/1:0.1 point-to-point bandwidth 384 ip pim sparse-dense-mode ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 101 ppp Virtual-Template200 class AutoQoS-FR-Se0/0/1:0-101 auto qos voip trust fr-atm ip rsvp bandwidth 112 ! interface Virtual-Template200 bandwidth 384 ip address 10.10.111.2 255.255.255.0 ip ospf mtu-ignore ppp multilink ppp multilink interleave ppp multilink fragment delay 10 service-policy output AutoQoS-Policy-Trust ip rsvp bandwidth 112 ! ! ! map-class frame-relay AutoQoS-FR-Se0/0/1:0-101 frame-relay cir 384000 frame-relay bc 3840 frame-relay be 0 frame-relay mincir 384000 ! Virtual-Access110.10.111.2 YES TFTP up * down* ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab
Re: [OSL | CCIE_Voice] does MLP LFI break your wan?
Once I removed the auto-qos command, the router crashed... no lucky today with that! Well, I was searching for older topics related to that on the list, and I saw one from Cristobal Priego. Basically he suggests to do the following workaround after configuring auto qos fr-atm: remove the frame-relay interface-dlci 201 ppp virtual-template 1 command re-add the command re-add the class-map to the frame-relay interface connectivity came back up Reload the HQ router. And once it comes back online again, you can reload the Branch router the number of times you want, and the virtual interface does not get down anymore. I will test that on my next remote lab. Thank you all, Bruno On Sat, May 19, 2012 at 3:17 PM, khaled Saholy khaled_sah...@hotmail.comwrote: Bruno, save the generated config of auto-qos into a text file (class-map , policy-map, interface, ..etc) , the remove the auto-qos command under the interface. paste the config and if not worked, reboot the two routers. Khaled -- From: cm3_...@hotmail.com To: brun...@gmail.com; ccie_voice@onlinestudylist.com Date: Sat, 19 May 2012 12:08:12 -0500 CC: com...@gmail.com; cc...@corb.net Subject: Re: [OSL | CCIE_Voice] does MLP LFI break your wan? I've run into it... Blow auto qos away and reboot, sometimes it takes 2 or 3 times, assuming the router doesn't crash when you try to remote it... -Chase -- If winners never quit and quitters never win, then who coined the phrase, Quit while you’re still ahead.? -- Date: Sat, 19 May 2012 12:40:17 -0300 From: brun...@gmail.com To: ccie_voice@onlinestudylist.com CC: com...@gmail.com; cc...@corb.net Subject: Re: [OSL | CCIE_Voice] does MLP LFI break your wan? Hey guys, Has anyone discovered what goes wrong with the interfaces after applying MLP LFI? I am running the same problem on my lab3. When I configure MLP LFI on both sides, the virtual interface stays up/down: Virtual-Access110.10.111.2 YES TFTP updown I have already rebooted the router several times, but it does not come up. The auto qos is applied on both sides: HQ: interface Serial0/0/1:0.1 point-to-point bandwidth 384 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 201 ppp Virtual-Template200 class AutoQoS-FR-Se0/0/1:0-201 auto qos voip trust fr-atm ip rsvp bandwidth 112 BR1: interface Serial0/0/1:0.1 point-to-point bandwidth 384 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 101 ppp Virtual-Template200 class AutoQoS-FR-Se0/0/1:0-101 auto qos voip trust fr-atm ip rsvp bandwidth 112 Has anyone run into this issue? Thank you, Bruno On Sat, Mar 5, 2011 at 6:51 PM, adam compton com...@gmail.com wrote: Yeah I've had this problem before. First off, you have to configure it on both sides for it to come back up. It usually comes right back up once you run auto qos voip fr-atm on both sides. If it doesn't, wr mem and reboot both routers should fix it. In proctorlabs, I haven't had to reboot the routers in this situation. In GNS3, I have to reboot them every single time. On Fri, Mar 4, 2011 at 11:01 PM, CCIE Voice cc...@corb.net wrote: I am having the exact same problem. virtual-access1, virtual-access2, and virtual-template200 are all in a down/down or up/down state. Not sure how to rectify it. Anyone else experienced this and figure out what was wrong? On Tue, Nov 23, 2010 at 12:33 PM, Romain Mullier romain.mull...@gmail.com wrote: Hi guys, was working on lab3, RSVP configuration worked well but after applying MLP LFI between HQ and BR1, I cannot bring the virtual interfaces up. Has anyone seen this before? (Routers have been reloaded) Thanks for your help. HQ class-map match-any AutoQoS-VoIP-RTP-Trust match ip dscp ef class-map match-any AutoQoS-VoIP-Control-Trust match ip dscp cs3 match ip dscp af31 ! ! policy-map AutoQoS-Policy-Trust class AutoQoS-VoIP-RTP-Trust priority 56 compress header ip rtp class AutoQoS-VoIP-Control-Trust bandwidth 17 class class-default fair-queue ! interface Serial0/0/1:0 no ip address encapsulation frame-relay no fair-queue frame-relay traffic-shaping frame-relay lmi-type ansi ip rsvp bandwidth ! interface Serial0/0/1:0.1 point-to-point bandwidth 384 ip pim sparse-dense-mode ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 201 ppp Virtual-Template200 class AutoQoS-FR-Se0/0/1:0-101 ip rsvp bandwidth 112 ! ! interface Virtual-Template200 bandwidth 384 ip address 10.10.111.1 255.255.255.0 ip ospf mtu-ignore ppp multilink ppp multilink interleave ppp multilink fragment delay 10 service-policy output AutoQoS-Policy-Trust ip rsvp bandwidth 112 ! map-class frame-relay AutoQoS
[OSL | CCIE_Voice] [Vol2 Lab3] Call Routing doubt
Hello Experts, Maybe I am missing something really obvious here, but I have a doubt regarding the Dial Plan in Lab 3 Vol2. The questions are not clear regarding on how each site make outbound calls, except for Emergency. It is clear on how PSTN should receive the digits, but not on how users are supposed to dial. On questions 2.7 and 2.8 we should configure TEHO and International Calls to India. According to the solution guide, you should only setup route patterns using + dialing (\91.!). But how will users call these numbers? Only via + dialing from Missed/Received Calls? What if I want to test a call to India? Should I first make an inbound call and then use the Received Call list to place an outbound call? Thank you! Bruno ___ 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] CME Call Blocking - Dial Peer Exempt
Hello George, Thank you very much for your answer! :) Rgds, Bruno On Sat, Feb 11, 2012 at 7:13 AM, George Goglidze gogli...@gmail.com wrote: Hi Bruno, The command paramspace callsetup after-hours-exempt true is only when there is a call with that dial-peer as incoming leg, and then going out of another dial-peer. basically when you have call coming from somewhere else like CUCM, then choosing as incoming leg dialpeer one with that command, and going out by another dialpeer that has a pattern that is blocked. If you want to allow one patter, it should be NOT included in the blocked patter range. in another words, you will have to be more exact on your block pattern. for example, don't put just 0, put 0 hope this helps, Cheers, On Sat, Feb 11, 2012 at 12:46 AM, Bruno Nonogaki brun...@gmail.comwrote: Hello everybody, I was making some tests with After-Hours Call Blocking on CME, for example: telephony-services after-hours block pattern 1 0 (blocks pattern 0) after-hours day sun start 10:00 11:00 And then I had my dial-peers: dial-peer voice 1 voip destination-pattern 0190 session target ipv4:x.x.x.x dial-peer voice 2 voip destination-pattern 000T session target ipv4:x.x.x.x This is just an example... And the call block was working fine. When I dialed 0 on my phone, I got the fast busy tone... I could configure an exempt on ephone with a PIN (after-hours exempt), and even an override-code (after-hours override-code ). But I am trying to configure an exempt for my dial-peer as well. Let's suppose I don't want the dial-peer 1 to be blocked, never, because it is an emergency number. I tried to configure the line paramspace callsetup after-hours-exempt true on this dial-peer, according to CME documentation. But it is still getting blocked. Does anyone know how can I use this command to configure a dial-peer exempt for Call Blocking? Thank you, Bruno ___ 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] CME Call Blocking - Dial Peer Exempt
Hello everybody, I was making some tests with After-Hours Call Blocking on CME, for example: telephony-services after-hours block pattern 1 0 (blocks pattern 0) after-hours day sun start 10:00 11:00 And then I had my dial-peers: dial-peer voice 1 voip destination-pattern 0190 session target ipv4:x.x.x.x dial-peer voice 2 voip destination-pattern 000T session target ipv4:x.x.x.x This is just an example... And the call block was working fine. When I dialed 0 on my phone, I got the fast busy tone... I could configure an exempt on ephone with a PIN (after-hours exempt), and even an override-code (after-hours override-code ). But I am trying to configure an exempt for my dial-peer as well. Let's suppose I don't want the dial-peer 1 to be blocked, never, because it is an emergency number. I tried to configure the line paramspace callsetup after-hours-exempt true on this dial-peer, according to CME documentation. But it is still getting blocked. Does anyone know how can I use this command to configure a dial-peer exempt for Call Blocking? Thank you, Bruno ___ 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] Interesting Project
Hi Emanuel, I have just talked with Andre Mulato regarding this issue (we worked together some years ago! hehe). I have already seen this warnings before when I made an upgrade from 4.2 to 6.1.x, but the DMA could be restored without problems, even with these warnings. I still have the warnings and errors logs here for that particular DMA, and I do see the same message as you: Warning: Numplan (NULL) for User linhara1 is invalid. Ignoring the user Condition: Primary Extension for this user is invalid. It does not exist in the database Solution: Please ensure that the user is associated with a valid primary extension/numplan When you generate the tar file, you have a file inside called DMAResult.txt. Do you have any errors in there? In mine I have: Validation = Success with Warnings So I think the Warnings are not a big deal... maybe you have to fix one or other thing manually, but you will still be able to import the DMA to the new system. Didn't you get an error (not warning)? I can see the only error you got was that one on FAC table, but I think you have already solved that by removing special characters, right? I remember that when I made this upgrade, TAC suggested me to use the DMA version 6.1.3 or 6.1.5. He said 6.1.2 had a bug and the import would fail. So maybe it is a bug with the version you are using. Why don't you try to use 6.1.5 and import it to a CUCM 6.1.x? And then upgrade it to 8.5? Here are useful links provided by TAC when I made this upgrade: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/dma/6_1_3/dmaug613.html#wp37764 DMA user guide to DMA 6.1.3, but you can follow the exact same steps for 6.1.5: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/dma/6_1_3/dmaug613.html#wp37764 Additional info regarding DMA 6.1.5: http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cucm/dma/6_1_5/dma-rel_note-615.html Hope that helps... Rgs, Bruno On Wed, Feb 8, 2012 at 1:24 PM, Ki Wi kiwi.vo...@gmail.com wrote: Maybe there's better way but I would have staged it in lab environment. I have seen first error before but can't really remember how to resolved it. All I remember is some name with - or _ might not work. You might want to verify or just remove the device association under this user as well which is what the error message suggest. Basically after you try out, you need to run DMA again to see whether that certain error message goes away for that specific phone/user. I did manually fix before but that time was for 100+ users before. That was like 3 to 4 years ago since I encountered this type of issue. Surprising I have done 10 such upgrade, only one customer I encounter this. Sent from my iPhone Pls pardon my fat fingers. On 8 Feb, 2012, at 9:42 PM, Emanuel Damasceno aedamasc...@gmail.com wrote: Hello Expert. First I'd like to say this is not CCIE related. I decided to post it here, because the majority of us work with voice, and we might help each other. I received a project a few months ago where I have to upgrade from CUCM 4.1.3 to 8.5. There is no upgrade straight through 8.5. I have to go first to 7.X and then to 8.5. This customer has one big structure, where there is 1 CUCM Pub and 5 Subscribers. He also got 6 Gatekeepers and over 5k devices. Needless to say this customer has thousands of users, route patterns, partitions, CSSs etc. As I run DMA on the Publisher server, I receive a lot of warnings like the ones shown below: *02/01/2012 12:31:49 Warning: Device or DeviceProfile=001ef7297af9 for user=3028541 not found in the database. Condition: The Device found in the directory does not exist in the database Solution: Please verify that the user is associated to a valid device or a device profile. Please add the mentioned device or device profile through CCM Admin pages or manually delete this value from the users CCN profile *That's just one of the thousands I get with the same message. Now, another couple thousand of these: * 02/01/2012 12:36:18 Warning: Numplan (NULL) for User 3008935 is invalid. Ignoring the user Condition: Primary Extension for this user is invalid. It does not exist in the database Solution: Please ensure that the user is associated with a valid primary extension/numplan* And another couple thousand of these: * Row: 4705 SQL error: -746(37129) Error executing update FACInfo set Name='Aloísio Flávio' where pkid='2a07d93d-d281-412a-a066-ff9e28edd14a': [Informix][Informix ODBC Driver][Informix]Unspecified System Error = -746. sqlerrm(37129) Name Aloísio Flávio AuthorizationLevel0 Code 925046 pkid 2a07d93d-d281-412a-a066-ff9e28edd14a NOT MODIFIABLE Table: PhoneButton 2012-02-01 16:09:22* So, after a long time waiting for Cisco to give us an idea of how to handle this (they haven't answered us yet
Re: [OSL | CCIE_Voice] Interesting Project
Hi Emanuel, Yeah, small world... =) I will private message you later... Well, I didn't face that issue when I did my upgrade. Are you using the same IP Address, Subnet Mask, Hostname, when making the 7.1.5 installation? DNS are properly configured? I have seen these errors just when addind a Subscriber to the cluster, but never when installing a Publisher... Maybe the way will be to import everything manually... hehe! But I really don't remeber if we can export configurations with BAT on 4.x, just like we do on 7.1.x... And even if you can, you will have a lot of Excel work to make the import files... Good luck on your studies! :) Bruno On Wed, Feb 8, 2012 at 9:24 PM, Emanuel Damasceno aedamasc...@gmail.comwrote: Sorry, I forgot to attach the file... Here it goes... *Emanuel Damasceno* CCNP Voice On Wed, Feb 8, 2012 at 9:20 PM, Emanuel Damasceno aedamasc...@gmail.comwrote: Bruno!!! It's good to see more Brazilians on the study list. :) Small world Well, let me explain to you what happens. It's not about the DMA. I can get the tar file just fine. I tried with 7.0.2, and 7.1.5. Tomorrow I will try what you suggested me, but like I said, the problem is not the tar file. When we finish it, it gives me the file with warnings. Thousands of warnings. Ok, I start installing CUCM 7.1.5 just fine, it installs, it goes all the way until the end of the installation. But instead of finishing, it actually shows me what is in the picture attached. Every single time, no matter what I do. I will follow your advice and try that, but I am sick of making DMAs, and when it comes to that part, it gives me the same error, over and over again. * *I've sent the log files to Cisco Systems, and this is the response I got (btw, my first name is Antonio and I HATE that name lol):* Antonio, If you could send me the most diagnostics from the most recent attempt. Once I have these I will look at them to try and find what exactly is causing your current issue. If you have any questions or concerns, please feel free to email or call me. Thanks, +1 919-574-5984, 0900-1630 EST (GMT -5) Cisco Customer Support Engineer Unified Communications Infrastructure RTP, NC, USA* Ok, I removed the engineer's name. But that's not the kind of answer I was expecting of him. I've sent them over 6 log files of all the errors and they NEVER gave me a plausible answer. By the way, that is the 3rd engineer on the case. I am really not dealing with Cisco anymore. I have to go all over the DB and make an Excel file so I can export. Tomorrow will be my last attempt on this. I am giving my best, but I don't see any effort from Cisco to help me out. That's why I posted this here. Thanks for answering in such detail. I will go over them tomorrow. I still got my 5 hour study block to face and I just got home (9:20 pm). My exam is getting closer and I intend to nail it. But I do need to get this problem out of the way before I can study with peace of mind. Best regards. *Emanuel Damasceno* CCNP Voice On Wed, Feb 8, 2012 at 2:56 PM, Bruno Nonogaki brun...@gmail.com wrote: Hi Emanuel, I have just talked with Andre Mulato regarding this issue (we worked together some years ago! hehe). I have already seen this warnings before when I made an upgrade from 4.2 to 6.1.x, but the DMA could be restored without problems, even with these warnings. I still have the warnings and errors logs here for that particular DMA, and I do see the same message as you: Warning: Numplan (NULL) for User linhara1 is invalid. Ignoring the user Condition: Primary Extension for this user is invalid. It does not exist in the database Solution: Please ensure that the user is associated with a valid primary extension/numplan When you generate the tar file, you have a file inside called DMAResult.txt. Do you have any errors in there? In mine I have: Validation = Success with Warnings So I think the Warnings are not a big deal... maybe you have to fix one or other thing manually, but you will still be able to import the DMA to the new system. Didn't you get an error (not warning)? I can see the only error you got was that one on FAC table, but I think you have already solved that by removing special characters, right? I remember that when I made this upgrade, TAC suggested me to use the DMA version 6.1.3 or 6.1.5. He said 6.1.2 had a bug and the import would fail. So maybe it is a bug with the version you are using. Why don't you try to use 6.1.5 and import it to a CUCM 6.1.x? And then upgrade it to 8.5? Here are useful links provided by TAC when I made this upgrade: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/dma/6_1_3/dmaug613.html#wp37764 DMA user guide to DMA 6.1.3, but you can follow the exact same steps for 6.1.5: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/dma/6_1_3/dmaug613.html#wp37764 Additional info regarding DMA 6.1.5: http://www.cisco.com/en/US/partner
[OSL | CCIE_Voice] how to dial-by-name in cue
i have configure dial-by-name on cue. but i don't how to dial. i configure user br2ph1 for 3001 and br2ph2 for 3002. AA number is 3100,when i dial 3100 ,press 2 ,then i don't know how to dial-by-name? -- Best Regards, Bruno___ 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 dspfarm or dsparm under voice-card
voice-card 0 no dspfarm dsp services dspfarm even i have learn ccie voice for long time .i am always confuse with no dspfarm and dsparm under voice-card. Is there anyone can tell the different ? -- Best Regards, Bruno___ 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] how to retrieve the directed callpark
i have config below ,when i use 3002 call 3001 ,3001 tranfer to directed call park slot 3020. how do i retrieve the call . dial 3020 from 3001 ? or use *3020 . boto can't retrieve the call. SITEC(config)#telephony-service SITEC(config-telephony)#call-park system application SITEC(config-telephony)#max-dn 30 no-reg primary SITEC(config-telephony)#! SITEC(config-telephony)#ephone-dn 11 SITEC(config-ephone-dn)# number 3011 no-reg primary SITEC(config-ephone-dn)# park-slot reserved-for 3001 timeout 60 limit 2 recall SITEC(config-ephone-dn)# description park slot for SCPH1 SITEC(config-ephone-dn)#! SITEC(config-ephone-dn)#ephone-dn 12 SITEC(config-ephone-dn)# number 3012 no-reg primary SITEC(config-ephone-dn)# park-slot timeout 60 limit 2 recall SITEC(config-ephone-dn)# description park slot for All Users SITEC(config-ephone-dn)#! SITEC(config-ephone-dn)#ephone-dn 20 SITEC(config-ephone-dn)# number 3020 SITEC(config-ephone-dn)# park-slot directed SITEC(config-ephone-dn)# description park-slot directed -- Best Regards, Bruno___ 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] can not save script in script repository
sorry for delay reply. it works . thx . -- Best Regards, Bruno -- Original -- From: Ken Wyankew...@gmail.com; Date: Sun, Nov 13, 2011 11:12 AM To: CCIEVoiceKPccievoic...@gmail.com; Cc: brunobruno.juni...@gmail.com; CCIE-V邮件列表ccie_voice@onlinestudylist.com; Subject: Re: [OSL | CCIE_Voice] can not save script in script repository It's always better to import script from Script Management page. But then it's shown as SCRIPT only default scripts are shown as SSCRIPT. After importing to UCCX we needn't worry about folder locations. On Sat, Nov 12, 2011 at 11:27 PM, CCIEVoiceKP ccievoic...@gmail.com wrote: Not sure if it's a bug or by design, but you need to import a script first on the Script Management page ... Then the default folder will show up ... Kevin Prater 907.301.6519 Sent from my iPhone and I have big thumbs ... So please excuse the typos. On Nov 11, 2011, at 8:12 PM, bruno bruno.juni...@gmail.com wrote: after i have intergrate uccx with cucm. I success login uccx editor with uccxadmin.after edit script ,i want to save the script in script repository. i can't find the default folder. long time ago ,i can find it. what happen . -- Best Regards, Bruno ___ 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] uccx Unified CM Telephony Subsystem gray out
i have creat Unified CM Telephony Group #0 \ application \trigger etc on uccx ,and check sync ,everything is fine. But when i check the them on cucm ,i found none of them register. so i go to uccx control center, i found the Unified CM Telephony Subsystem gray out,whatever i restart or reboot the server. it just can't up . what 's prob do u think ? Unified CM Telephony Users jtapi_1 -- Ok Unified CM Telephony Port Groups Port Group 0 CTI ports on node 1 -- Ok CCM Data Device Pool--Ok Media Resource Group List-- Ok PresenceGroup-- Ok Partition-- Ok User Hold Audio Source-- Ok Alerting Name Ascii-- Ok Require DTMF Reception-- Ok NetWork Hold Audio Source-- Ok CallPickUpGroup-- Ok VoiceMailProfile-- Ok AAR Group-- Ok Location-- Ok Calling Search Space-- Ok Unified CM Telephony Triggers ICD ICD -- Ok CCM Data Forward Busy Calling Search Space-- Ok Device Pool-- Ok Alerting Name Ascii-- Ok PresenceGroup-- Ok Partition-- Ok CallPickUpGroup-- Ok VoiceMailProfile-- Ok Location-- Ok Calling Search Space-- Ok -- Best Regards, Bruno___ 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] can cuc demo license run vpim
thx for the info. -- Best Regards, Bruno -- Original -- From: Anthony Albaascanio.al...@gmail.com; Date: Fri, Nov 11, 2011 10:57 AM To: brunobruno.juni...@gmail.com; Cc: CCIE-V邮件列表ccie_voice@onlinestudylist.com; Subject: Re: [OSL | CCIE_Voice] can cuc demo license run vpim Hi Bruno, I can confirm that Unity Express 7 to Unity Connection 8.02c works with VPIM using the demo license of CUC8. I was able to do the Vol2 Lab2 Q8.3 VPIM task per the solution guide. I did not notice any difference compared with the Proctor Labs rack (CUC7). I saw this in the license file: INCREMENT LicVPIMIsLicensed cisco 8.0 permanent 1 HOSTID=ANY \ NOTICE=LicFileIDCUCdemo.lic/LicFileIDLicLineID11/LicLineID \ PAKdummyPak/PAK SIGN=FADA8C243098 Good luck with your VPIM studies. Anthony___ 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] can not save script in script repository
after i have intergrate uccx with cucm. I success login uccx editor with uccxadmin.after edit script ,i want to save the script in script repository. i can't find the default folder. long time ago ,i can find it. what happen . -- Best Regards, Bruno___ 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] can cuc demo license run vpim
When I attempt to add a VPIM location is Unity Connection I receive the following license error. Anyone attempt VPIM in these labs yet? Status The requested operation would result in a license violation. Unable to create VPIM Location -- Best Regards, Bruno___ 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] can cuc demo license run vpim
how can i get it for labbing . i do it in my home lab -- Best Regards, Bruno -- Original -- From: Farkas Péterwormh...@sch.bme.hu; Date: Wed, Nov 9, 2011 07:50 PM To: brunobruno.juni...@gmail.com; Cc: CCIE-V邮件列表ccie_voice@onlinestudylist.com; Subject: Re: [OSL | CCIE_Voice] can cuc demo license run vpim No, demo license not cover VPIM so it requries VPIM license to be added. However proctorlabs should have. Peter - Original Message - From: bruno bruno.juni...@gmail.com Date: Wednesday, November 9, 2011 11:18 am Subject: [OSL | CCIE_Voice] can cuc demo license run vpim To: CCIE-V邮件列表 ccie_voice@onlinestudylist.com When I attempt to add a VPIM location is Unity Connection I receive the following license error. Anyone attempt VPIM in these labs yet? Status The requested operation would result in a license violation. Unable to create VPIM Location -- Best Regards, Bruno ___ 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] h323 gw question
thx kshitij, very detail . very helpfull. -- Best Regards, Bruno -- Original -- From: Kshitij Singhimartinian.ksin...@gmail.com; Date: Sun, Nov 6, 2011 09:23 PM To: ccie_voiceccie_voice@onlinestudylist.com; bruno.juniperbruno.juni...@gmail.com; brjkumarbrjku...@gmail.com; Subject: Re: h323 gw question The reason why an incoming dial-peer is created is because PID 0 (or dial peer 0) has certain characteristics which cannot be modified and don't work too well when matched. For a VoIP dial peer 0: Supports any codec No DTMF relay IP precedence 0 VAD-enabled No RSVP support Fax-rate voice This effectively means that DTMF might not work and QoS might be messed by. Also, VAD is BAD hence it's best not to take a risk. Despite the statement supports any codec, there have been a number of scenarios where a call fails simply because it's matching dial peer 0 and media negotiation doesn't occur correctly. For a POTS dial peer, PID 0 is a strict no-no since it has the following characteristics: No applications No DID No DID effectively means that we will get dial-tone when an incoming call via a PRI hits PID 0. That is definitely not desirable. To add to the problems with matching PID 0, we won't be able to perform any translations on the gateway on the inbound call leg if PID 0 is matched simply because PID 0 is non-modifiable. Would strongly recommend adding an inbound dial peer, although in the real world there are situations where PID 0 is matched but things are working fine. It's not a mandate to match an incoming dial-peer, but more of a very sound and logical recommendation which can help in avoiding issues. Here is the document that explains the same: http://docwiki.cisco.com/wiki/Cisco_IOS_Voice_Troubleshooting_and_Monitoring_--_Call_Flow_Overview#Dial_Peer_0 Hope this helps. On Sun, Nov 6, 2011 at 1:29 PM, ccie_voice-requ...@onlinestudylist..com wrote: 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. h323 gw question (=?gbk?B?YnJ1bm8=?=) 2. Re: h323 gw question (brajesh kumaR) 3. Re: CUCM 8.6 and CUPS 8.6 NTP issue (Shrini) 4. Re: CUCM and UCCX - JTAPI issue (Shrini) 5. Re: h323 gw question (Shrini) 6. Re: transfer file to router with no tftp or ftp server (Shrini) -- Message: 1 Date: Sun, 6 Nov 2011 13:00:57 +0800 From: =?gbk?B?YnJ1bm8=?= bruno.juni...@gmail.com To: =?gbk?B?Q0NJRS1W08q8/sHQse0=?= ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] h323 gw question Message-ID: tencent_0b862c4933d61e430c6da...@qq.com Content-Type: text/plain; charset=gbk hello guys, regarding the h323 gw , why we need incoming called-number . under dial-peer ? i saw vik create another dial-peer for match any inbound . dial-peer voice 3000 voip destination-pattern ^3...$ voice-class codec 1 voice-class h323 1 session target ipv4:192.168.10.211 incoming called-number . dtmf-relay h245-alphanumeric no vad ! dial-peer voice 3001 voip preference 1 destination-pattern ^3...$ voice-class codec 1 voice-class h323 1 session target ipv4:192.168.10.210 dtmf-relay h245-alphanumeric no vad -- Best Regards, Bruno -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/2006/3f5d1f91/attachment-0001..html -- Message: 2 Date: Sun, 6 Nov 2011 12:08:20 +0530 From: brajesh kumaR brjku...@gmail.com To: bruno bruno.juni...@gmail.com Cc: CCIE-V ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] h323 gw question Message-ID: CAKVC5rc09B_1Cc1QoQj=longnijkbh2_vyuwjrxsijgy3vb...@mail.gmail.com Content-Type: text/plain; charset=ISO-8859-1 This is to match inbound call coming from CUCM. Dial-peer 3000 voip will be used to matched inbound voip calls as well as outbound voip calls. Inbound in terms to call coming from cucm to h323 GW and outbound in terms of call coming from PSTN/GW to CUCM for inbound phone calls. 2011/11/6 bruno bruno.juni...@gmail.com: hello guys, regarding the h323 gw , why we need incoming called-number . under dial-peer ?? i saw vik create another dial-peer for match any inbound . dial-peer voice 3000 voip ?destination-pattern ^3...$ ?voice-class codec 1 ?voice-class h323 1 ?session target ipv4:192.168.10.211 ?incoming called-number . ?dtmf-relay h245-alphanumeric ?no vad ! dial-peer voice 3001 voip
[OSL | CCIE_Voice] h323 gw question
hello guys, regarding the h323 gw , why we need incoming called-number . under dial-peer ? i saw vik create another dial-peer for match any inbound . dial-peer voice 3000 voip destination-pattern ^3...$ voice-class codec 1 voice-class h323 1 session target ipv4:192.168.10.211 incoming called-number . dtmf-relay h245-alphanumeric no vad ! dial-peer voice 3001 voip preference 1 destination-pattern ^3...$ voice-class codec 1 voice-class h323 1 session target ipv4:192.168.10.210 dtmf-relay h245-alphanumeric no vad -- Best Regards, Bruno___ 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] unifiedfx phoneview xml error problem
hello expert , i use unifiedfx phoneview software on cme and follow their guide. In PhoneView if you goto Group-Add and use the following credentials to tie in the CME config provided: Admin User: pvadmin Admin Password: cisco Phone User: pvphone Phone Password: cisco cme config: ixi transport http response size 64 no shutdown request outstanding 1 ixi application cme no shutdown telephony-service xml user pvadmin password cisco 15 url authentication http://[CME IP Address]/CCMCIP/authenticate.asp pvphone cisco it's ok on cucm . but with cme , i saw xml error response from phone from log. i try to reset the phone. still not work. is there anyone workround this ? best regards, Bruno___ 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] how to make UCSInstall_UCOS_8.5.1.10000-26bootable
thx. it works for me . -- Original -- From: khaled Saholykhaled_sah...@hotmail.com; Date: Fri, Aug 5, 2011 11:00 PM To: bruno.juniperbruno.juni...@gmail.com; gwenzitgwen...@gmail.com; ccie_voiceccie_voice@onlinestudylist.com; Subject: RE: [OSL | CCIE_Voice] how to make UCSInstall_UCOS_8.5.1.1-26bootable Hi. Try this link: http://htluo.blogspot.com/2010/04/how-to-make-non-bootable-iso-image.html Regards. Khaled Al-Saholy From: bruno.juni...@gmail.com To: gwen...@gmail.com; ccie_voice@onlinestudylist.com Date: Fri, 5 Aug 2011 18:10:14 +0800 Subject: Re: [OSL | CCIE_Voice] how to make UCSInstall_UCOS_8.5.1.1-26 bootable Hello， thx for ur reply . could u give me a link. best regards Bruno -- Original -- From: gwen...@gmail.comgwen...@gmail.com; Date: Fri, Aug 5, 2011 06:06 PM To: brunobruno.juni...@gmail.com; CCIE-V邮件列表ccie_voice@onlinestudylist.com; Subject: Re: [OSL | CCIE_Voice] how to make UCSInstall_UCOS_8.5.1.1-26 bootable Have u tried to use the Cisco VMWare image file instead of generic. It can be downloaded from Wikipedia. Also I have done this and it worked no prob. You may not have names the ISO file correctly. Sent from my HTC on the Now Network from Sprint! - Reply message - From: bruno bruno.juni...@gmail.com Date: Fri, Aug 5, 2011 5:26 am Subject: [OSL | CCIE_Voice] how to make UCSInstall_UCOS_8.5.1.1-26 bootable To: CCIE-V邮件列表 ccie_voice@onlinestudylist.com Hello all, I downloaded three files of UNREST version from cisco download , merged them and built an iso image however when trying to boot a vm machine on an esx UCS servers it complains and instead looking at DHCP for network location to find an image. This is when i am trying to boot it from local CD/DVD drive or from a loca storage source. Also what is confusing about these files is that in release notes for 8.5(1) it says this iso can be used for fresh new install as well as upgrade but then on : http://www.cisco.com/en/US/products/ps7273/prod_release_notes_list.html page 4 Before You Begin In Cisco Unified Communications Manager 8.5(1), the image available for download from Cisco.com is a bootable image that can be burned to DVD and used for both upgrades and fresh installs. Cisco Unified Communications Manager 8.5(1) upgrade DVDs ordered from Cisco are also bootable for use with upgrades or fresh installs. Then on page 8 it says : Note Because the UCSInstall_UCOS_8.5.1.1-26 build specifies a nonbootable ISO, the build proves useful only for upgrades. You cannot use this build for new installations. it can boot on vmware but not esxi . I am still working to figure it if its really an issue with iso image i created or VMware. i saw some post in internet ,ppl said we can make it bootable ,how to do it ? ___ 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] how to make UCSInstall_UCOS_8.5.1.10000-26
Hello all, I downloaded three files of UNREST version from cisco download , merged them and built an iso image however when trying to boot a vm machine on an esx UCS servers it complains and instead looking at DHCP for network location to find an image. This is when i am trying to boot it from local CD/DVD drive or from a loca storage source. Also what is confusing about these files is that in release notes for 8.5(1) it says this iso can be used for fresh new install as well as upgrade but then on : http://www.cisco.com/en/US/products/ps7273/prod_release_notes_list.html page 4 Before You Begin In Cisco Unified Communications Manager 8.5(1), the image available for download from Cisco.com is a bootable image that can be burned to DVD and used for both upgrades and fresh installs. Cisco Unified Communications Manager 8.5(1) upgrade DVDs ordered from Cisco are also bootable for use with upgrades or fresh installs. Then on page 8 it says : Note Because the UCSInstall_UCOS_8.5.1.1-26 build specifies a nonbootable ISO, the build proves useful only for upgrades. You cannot use this build for new installations. it can boot on vmware but not esxi . I am still working to figure it if its really an issue with iso image i created or VMware. i saw some post in internet ,ppl said we can make it bootable ,how to do it ? any input will be highly appreciated.___ 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] how to make UCSInstall_UCOS_8.5.1.10000-26 bootable
Hello all, I downloaded three files of UNREST version from cisco download , merged them and built an iso image however when trying to boot a vm machine on an esx UCS servers it complains and instead looking at DHCP for network location to find an image. This is when i am trying to boot it from local CD/DVD drive or from a loca storage source. Also what is confusing about these files is that in release notes for 8.5(1) it says this iso can be used for fresh new install as well as upgrade but then on : http://www.cisco.com/en/US/products/ps7273/prod_release_notes_list.html page 4 Before You Begin In Cisco Unified Communications Manager 8.5(1), the image available for download from Cisco.com is a bootable image that can be burned to DVD and used for both upgrades and fresh installs. Cisco Unified Communications Manager 8.5(1) upgrade DVDs ordered from Cisco are also bootable for use with upgrades or fresh installs. Then on page 8 it says : Note Because the UCSInstall_UCOS_8.5.1.1-26 build specifies a nonbootable ISO, the build proves useful only for upgrades. You cannot use this build for new installations. it can boot on vmware but not esxi . I am still working to figure it if its really an issue with iso image i created or VMware. i saw some post in internet ,ppl said we can make it bootable ,how to do it ?___ 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] how to make UCSInstall_UCOS_8.5.1.10000-26
use cco ,can download 8.5 and 8.6 non-bootable version only.it 's use for upgrade. -- Original -- From: Bryan Byrneccie.25...@gmail.com; Date: Fri, Aug 5, 2011 06:18 PM To: brunobruno.juni...@gmail.com; Cc: ccie_voiceccie_voice@onlinestudylist.com; Subject: Re: [OSL | CCIE_Voice] how to make UCSInstall_UCOS_8.5.1.1-26 There is a bootable and a non-bootable version of the ISO. My guess is you got the non-bootable version and I do not believe there is a way to make the DVD bootable. You just need to download the correct version. -Bryan On Aug 5, 2011, at 3:23 AM, bruno wrote: Hello all, I downloaded three files of UNREST version from cisco download , merged them and built an iso image however when trying to boot a vm machine on an esx UCS servers it complains and instead looking at DHCP for network location to find an image. This is when i am trying to boot it from local CD/DVD drive or from a loca storage source. Also what is confusing about these files is that in release notes for 8.5(1) it says this iso can be used for fresh new install as well as upgrade but then on : http://www.cisco.com/en/US/products/ps7273/prod_release_notes_list.html page 4 Before You Begin In Cisco Unified Communications Manager 8.5(1), the image available for download from Cisco.com is a bootable image that can be burned to DVD and used for both upgrades and fresh installs. Cisco Unified Communications Manager 8.5(1) upgrade DVDs ordered from Cisco are also bootable for use with upgrades or fresh installs. Then on page 8 it says : Note Because the UCSInstall_UCOS_8.5.1.1-26 build specifies a nonbootable ISO, the build proves useful only for upgrades. You cannot use this build for new installations. it can boot on vmware but not esxi . I am still working to figure it if its really an issue with iso image i created or VMware. i saw some post in internet ,ppl said we can make it bootable ,how to do it ? any input will be highly appreciated. ___ 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] unitycn problem
hello nirvair, thx. i follow u now it work. Bruno -- Original -- From: Nirvair Sahotanirvair.sah...@sbcglobal.net; Date: Mon, May 16, 2011 04:21 AM To: CCIE-V邮件列表ccie_voice@onlinestudylist.com; brunobruno.juni...@gmail.com; Subject: Re: [OSL | CCIE_Voice] unitycn problem Bruno, Have you tried to unckeck the option Request Entry of User ID after Failed Password Entry from Known Extension in unity connection in: Advanced -- Converations settings? Nirvair --- On Sat, 5/14/11, bruno bruno.juni...@gmail.com wrote: From: bruno bruno.juni...@gmail.com Subject: [OSL | CCIE_Voice] unitycn problem To: CCIE-V邮件列表 ccie_voice@onlinestudylist.com Date: Saturday, May 14, 2011, 11:53 AM Hello all, one of my client want to achive it. When a VM user presses VM button and enters a wrong PIN,Unity should again prompt for PIN instead of userid. i can 't found it in unitycn. could someone help me ? best Regards, bruno -Inline Attachment Follows- ___ 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] Unity Connection Voice View
hello all, I have Voice View working great but in Cisco CME with Unity Express I think this feature can be accessed by a service. With Unity Connection the only way I see to access this is to dial into unity, put in password, choose option 5 and then select the way you want the messages to be viewed. Is there a way to make this a service??? Best Regards, bruno___ 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] unitycn problem
Hello all, one of my client want to achive it. When a VM user presses VM button and enters a wrong PIN,Unity should again prompt for PIN instead of userid. i can 't found it in unitycn. could someone help me ? best Regards, bruno___ 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] 回复： use 26xm to setup ccie voice lab
thank u . -- 原始邮件 -- 发件人: Shrinilinuxbos...@gmail.com; 发送时间: 2011年4月14日(星期四) 中午12:50 收件人: brunobruno.juni...@gmail.com; 抄送: ccie_voiceccie_voice@onlinestudylist.com; 主题: Re: [OSL | CCIE_Voice] use 26xm to setup ccie voice lab They are modular router but will work. 26xxXM NM-HD-2VE ( it has builtin PVDMs) VWIC-2MFT-T1 or 1MFT based on your site requirement. VWIC-2MFT-E1 Or also you can use VWIC-1MFT-E1/T1 For framerelay: NM-4T (on PSTN router) NM-1T on Site routers Serial cables - 3 VWIC-4ESW - 2 3750 Switch On 4/12/2011 10:24 PM, bruno wrote: do some guys have any experience to use 26XX xm to setup ccie voice lab ? could u kindly give the hareware and software list. include e1 t1 card /pvdm stuff. as detail as possible.thanks in advance. BR, bruno ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] use 26xm to setup ccie voice lab
do some guys have any experience to use 26XX xm to setup ccie voice lab ? could u kindly give the hareware and software list. include e1 t1 card /pvdm stuff. as detail as possible.thanks in advance. BR, bruno___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] 回复： 回复： how to configure cucme to support phoneview
dear all, this blog is for uc phoneview .not for unified fx phoneview. http://blog.ipexpert.com/2010/11/17/setting-up-phone-view/ Best Regards bruno -- 原始邮件 -- 发件人: Randall Saborí Cuberoill2...@gmail.com; 发送时间: 2011年3月30日(星期三) 上午10:05 收件人: brunobruno.juni...@gmail.com; 抄送: Bo Gaobga...@gmail.com; ccie_voiceccie_voice@onlinestudylist.com; 主题: Re: [OSL | CCIE_Voice] 回复： how to configure cucme to support phoneview Try this: http://lmgtfy.com/?q=ipexpert+blog+phoneview Welcome to the Internet. El mié, 30-03-2011 a las 09:13 +0800, bruno escribió: hello bo, I have check the ipexpert blog ,can't find it. did u have the link? Best Regards, Bruno -- 原始邮件 -- 发件人: Bo Gaobga...@gmail.com; 发送时间: 2011年3月29日(星期二) 晚上9:15 收件人: brunobruno.juni...@gmail.com; 抄送: ccie_voiceccie_voice@onlinestudylist.com; 主题: Re: [OSL | CCIE_Voice] how to configure cucme to support phoneview Bruno, Please check IP Experts' website, I remember Vik had a blog about how to configure phoneview on it. Bo 2011/3/28 bruno bruno.juni...@gmail.com hello guys, ?? great news Unified FX release their lab version. how to configure cucme to support phoneview??? i can not find any tutorial on their website. could someone help？ ?? Best Regards, bruno ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] 回复： how to configure cucme to support phoneview
hello bo, I have check the ipexpert blog ,can't find it. did u have the link? Best Regards, Bruno -- 原始邮件 -- 发件人: Bo Gaobga...@gmail.com; 发送时间: 2011年3月29日(星期二) 晚上9:15 收件人: brunobruno.juni...@gmail.com; 抄送: ccie_voiceccie_voice@onlinestudylist.com; 主题: Re: [OSL | CCIE_Voice] how to configure cucme to support phoneview Bruno, Please check IP Experts' website, I remember Vik had a blog about how to configure phoneview on it. Bo 2011/3/28 bruno bruno.juni...@gmail.com hello guys, ?? great news Unified FX release their lab version. how to configure cucme to support phoneview??? i can not find any tutorial on their website. could someone help？ ?? Best Regards, bruno ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] how to configure cucme to support phoneview
hello guys, great news Unified FX release their lab version. how to configure cucme to support phoneview? i can not find any tutorial on their website. could someone help？ Best Regards, bruno___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] About Channel Selection Order
Hello expert, I see this The Channel Selection Order which defines the order the system “hunts” for available channels. The logic here is reversed from a numerical perspective, as the channels are considered to be top at 1 and bottom at 24, therefore Top Down selection will select channel 1 first in some doc. i want to know what 's best pratice in real life or in ccie lab. top-down or bottom-up? Best Regards, Bruno___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] why trunk mode on ESW?
thanks for discuss. anyone can give me some confirm? -- Original -- From: Rogers Ochiengrogersochi...@gmail.com; Date: Wed, Jan 19, 2011 09:26 PM To: Shrinilinuxbos...@gmail.com; Cc: Roger Kllberroger.kallb...@cygate.se; brunobruno.juni...@gmail.com; ccie_voiceccie_voice@onlinestudylist.com; Subject: Re: [OSL | CCIE_Voice] why trunk mode on ESW? All recommendations i see prefers the new way, even though out of our scope it's more secure as compared to trunk mode. On 19 January 2011 14:12, Shrini linuxbos...@gmail.com wrote: Hi Roger, I agree with legacy word, but I prefer trunk for our purpose, reason is below link. access mode and trunk mode both explained well here. http://www.cisco.com/en/US/docs/ios/lanswitch/configuration/guide/lsw_hwic_ethsw_ic_ps6441_TSD_Products_Configuration_Guide_Chapter.html#wp1049866 -S From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roger Kllberg Sent: Wednesday, January 19, 2011 2:13 AM To: bruno; ccie_voice Subject: Re: [OSL | CCIE_Voice] why trunk mode on ESW? Hi Bruno, First of all I do not speak for IPX, but my understanding is that the reason for why the vol1 has the old way of configuring the ports on ESW module is because of at the time of when it was written this was the way these ports were configured. I can from my own experience say that you definitely can configure these ports with the access port mode. This is what I did after I realized that this was now supported. But it might be good to practise on both methods, you never know if you will get that as a requirement in the lab. Sincerely Roger Kllberg CCIE #26199 (Voice) Consultant Cygate AB Eric Perssons vg 21, SE-217 62 MALM Frn: bruno [bruno.juni...@gmail.com] Skickat: den 19 januari 2011 07:18 Till: ccie_voice mne: [OSL | CCIE_Voice] why trunk mode on ESW? Dear all in vol1 network infrastrure, why we need to configure trunk mode on esw,why not access mode .i have test the access mode is ok. SITEB(config)#int range f0/1/0 -3 SITEB(config-if-range)# switchport trunk native vlan 602 SITEB(config-if-range)# switchport mode trunk SITEB(config-if-range)# switchport voice vlan 502 SITEB(config-if-range)#description ***CONNECT TO IP PHONE*** SITEC#show interfaces f0/1/0 switchport Name: Fa0/1/0 Switchport: Enabled Administrative Mode: trunk Operational Mode: trunk Administrative Trunking Encapsulation: dot1q Operational Trunking Encapsulation: dot1q Negotiation of Trunking: Disabled Access Mode VLAN: 0 ((Inactive)) Trunking Native Mode VLAN: 602 (DATA-VLAN) Trunking VLANs Enabled: ALL Trunking VLANs Active: 1,502,602 Protected: false Priority for untagged frames: 0 Override vlan tag priority: FALSE Voice VLAN: 502 Appliance trust: none Best Regards, bruno ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] why trunk mode on ESW?
Dear all in vol1 network infrastrure, why we need to configure trunk mode on esw,why not access mode .i have test the access mode is ok. SITEB(config)#int range f0/1/0 -3 SITEB(config-if-range)# switchport trunk native vlan 602 SITEB(config-if-range)# switchport mode trunk SITEB(config-if-range)# switchport voice vlan 502 SITEB(config-if-range)#description ***CONNECT TO IP PHONE*** SITEC#show interfaces f0/1/0 switchport Name: Fa0/1/0 Switchport: Enabled Administrative Mode: trunk Operational Mode: trunk Administrative Trunking Encapsulation: dot1q Operational Trunking Encapsulation: dot1q Negotiation of Trunking: Disabled Access Mode VLAN: 0 ((Inactive)) Trunking Native Mode VLAN: 602 (DATA-VLAN) Trunking VLANs Enabled: ALL Trunking VLANs Active: 1,502,602 Protected: false Priority for untagged frames: 0 Override vlan tag priority: FALSE Voice VLAN: 502 Appliance trust: none Best Regards, bruno___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] GK and cube problem
yes ,i have config dialpeer ! dial-peer voice 1 voip incoming called-number 01132T dtmf-relay h245-alphanumeric no vad ! dial-peer voice 2 voip destination-pattern .T session target ras dtmf-relay h245-alphanumeric no vad ! -- Original -- From: Cristobal Priegocristobalpri...@gmail.com; Date: Thu, Nov 4, 2010 11:44 PM To: bruno.juniperbruno.juni...@gmail.com; Cc: ccie_voiceccie_voice@onlinestudylist.com; Subject: Re: [OSL | CCIE_Voice] GK and cube problem do you have your Voip Dial peers configured? 2010/11/4 bruno.juniper bruno.juni...@gmail.com hello mate, ? I have some problem regarding GK and CUBE. HQ is gk also a cube.HQ user call 01132* go through gk. my config is below. the call didn't succeed. it show gk_rassrv_sep_arq: LRQ suspension point failed (return code = 0x4009). can anyone help me? ? interface?FastEthernet0/0.102 ?description?***VOICE?VLAN*** ?encapsulation?dot1Q?102 ?ip?address?142.102.64.254?255.255.255.0 ?ip?helper-address?142.100.64.11 ?h323-gateway?voip?interface ?h323-gateway?voip?id?VGK?ipaddr?142.1.64.254?1719 ?h323-gateway?voip?h323-id?CUBE ?h323-gateway?voip?bind?srcaddr?142.102.64.254 ! ! gatekeeper ?zone?local?GK?cisco.com?142.1.64.254 ?zone?local?VGK?cisco.com ?zone?remote?BBGK?cisco.com?157.1.26.253?1719?outvia?VGK ?zone?prefix?BBGK?01132* ?no?shutdown ! ? HQ-RTR#debug gatek ma 10 Nov? 3 13:39:17.072: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Nov? 3 13:39:17.076: ////GK/gk_rassrv_arq: arqp=0x4A2DE644,crv=0xB, answerCall=0 Nov? 3 13:39:17.076: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC Nov? 3 13:39:17.076: //809F22BF0B00/809F22BF0B00/GK/gk_dns_query: No Name servers Nov? 3 13:39:17.076: //809F22BF0B00/809F22BF0B00/GK/rassrv_get_addrinfo: (0113212345678) Tech-prefix match failed. Nov? 3 13:39:17.076: //809F22BF0B00/809F22BF0B00/GK/rassrv_get_addrinfo: (0113212345678) Matched zone prefix 01132 and remainder 12345678 Nov? 3 13:39:17.076: ////GK/gk_rassrv_get_ingress_network: returning default ingress network = 1 Nov? 3 13:39:17.076: //809F22BF0B00/809F22BF0B00/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x4A04AC50 Nov? 3 13:39:17.076: //809F22BF0B00/809F22BF0B00/GK/rassrv_arq_select_viazone: matched zone is GK, and z_invianamelen=0 Nov? 3 13:39:17.076: //809F22BF0B00/809F22BF0B00/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x495E8FC4 Nov? 3 13:39:17.076: //809F22BF0B00/809F22BF0B00/GK/rassrv_arq_select_viazone: matched zone is BBGK, and z_outvianamelen=3 Nov? 3 13:39:17.076: //809F22BF0B00/809F22BF0B00/GK/rassrv_arq_select_viazone? and z_outvianamep=VGK Nov? 3 13:39:17.076: //809F22BF0B00/809F22BF0B00/GK/rassrv_arq_select_viazone: Received ARQ for a zone (BBGK) that has an outviazone (VGK) specified.? Pick an IP-IP gateway in that viazone. Nov? 3 13:39:17.076: ////GK/gk_gw_select_ipipgw_random: zonep: 0x4A297F40, tpp: 0x0, current_endpt: 1 Nov? 3 13:39:17.076: ////GK/gk_gw_select_ipipgw_random: Selecting any IPIPGW. qelemp.head=0x49E0F3FC, use_count=1, current_endpt=1 Nov? 3 13:39:17.076: ////GK/gk_gw_select_ipipgw_random: Gateway selection will start at the top of the linked list. use_count=1, current_endpt=0 Nov? 3 13:39:17.076: ////GK/gk_gw_select_ipipgw_random: qelemp=0x49E0F3FC, loop_count=0 Nov? 3 13:39:17.076: ////GK/gk_gw_select_ipipgw_random: Examining tgwp 0x49E1F0D8, g_supp_prots: 0x50 qelemp: 0x49E0F3FC, loop_count: 1 Nov? 3 13:39:17.076: ////GK/gk_gw_select_ipipgw_random: Found an IPIPGW. tgwp: 0x49E1F0D8, endptsigIP: 142.102.64.254, endptrasIP: 142.102.64.254, zone: VGK Nov? 3 13:39:17.076: ////GK/gk_gw_select_ipipgw_random: Selected an IPIPGW. Nov? 3 13:39:17.076: //809F22BF0B00/809F22BF0B00/GK/rassrv_get_addrinfo: (0113212345678) successfully resolved IPIPGW and returning with return code 0 Nov? 3 13:39:17.092: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Nov? 3 13:39:17.092: ////GK/gk_rassrv_arq: arqp=0x4A2DE644,crv=0x28, answerCall=1 Nov? 3 13:39:17.092: //809F22BF0B00/809F22BF0B00/GK/gk_rassrv_dep_arq: ARQ Didn't use GK_AAA_PROC Nov? 3 13:39:17.108: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Nov? 3 13:39:17.112: ////GK/gk_rassrv_arq: arqp=0x4A281EEC,crv=0x29, answerCall=0 Nov? 3 13:39:17.112: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC Nov? 3 13:39:17.112: //809F22BF0B00/809F22BF0B00/GK/gk_dns_query: No Name servers Nov? 3 13:39:17.112: //809F22BF0B00/809F22BF0B00/GK/rassrv_get_addrinfo:
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 57, Issue 19
HQ-RTR#debug voice ipipgw Nov 5 04:35:39.739: //1/00E5B6220100/H323/setup_ind: Receive bearer cap infoXRate 16, rateMult 0 Nov 5 04:35:39.747: //1/00E5B6220100/H323/cch323_set_h245_state_mc_mode_incoming: h245 state m/c mode=0x10F, h323_ctl=0x2F Nov 5 04:35:39.767: //-1//H323/cch245_event_handler: callID=1 Nov 5 04:35:39.767: //-1//H323/cch245_event_handler: Event CC_EV_H245_SET_MODE: data ptr=0x4A07CD18 Nov 5 04:35:39.767: //1/00E5B6220100/H323/cch323_set_mode: callID=1, flow Mode=1 spi_mode=0x1 Nov 5 04:35:39.767: //1/00E5B6220100/H323/cch323_do_call_proceeding: set_mode NOT called yet...saved deferred CALL_PROC Nov 5 04:35:39.767: //-1//H323/cch323_get_dp_pref_mask: cch323_get_dp_pref_mask:IPIPGW(2):setting mask for 729ar8also as 729 is configured Nov 5 04:35:39.767: //2/00E5B6220100/H323/cch323_set_h245_state_mc_mode_outgoing: call_spi_mode = 1 Nov 5 04:35:39.767: //2/00E5B6220100/H323/cch323_set_h245_state_mc_mode_outgoing: h245 state m/c mode=0x1AF0, h323_ctl=0x0 Nov 5 04:35:39.767: //2/00E5B6220100/H323/cch323_get_peer_info: Entry Nov 5 04:35:39.767: //2/00E5B6220100/H323/cch323_get_peer_info: Have peer Nov 5 04:35:39.767: //2/00E5B6220100/H323/cch323_set_pref_codec_list: First preferred codec(bytes)=16(20) Nov 5 04:35:39.771: //2/00E5B6220100/H323/cch323_get_peer_info: Flow Mode set to FLOW_THROUGH Nov 5 04:35:39.771: //2/00E5B6220100/H323/cch323_set_h323_control_options_outgoing: h245 sm mode = 6896 Nov 5 04:35:39.771: //2/00E5B6220100/H323/cch323_set_h323_control_options_outgoing: h323_ctl=0x2F Nov 5 04:35:39.771: //1/00E5B6220100/H323/cch323_process_set_mode: Setting inbound leg mode flags to 0x1AF0, flow-mode to FLOW_THROUGH Nov 5 04:35:39.771: //1/00E5B6220100/H323/cch323_process_set_mode: Sending deferred CALL_PROC Nov 5 04:35:39.771: //1/00E5B6220100/H323/cch323_do_call_proceeding: set_mode called so we can proceed with CALLPROC -- Original -- From: ccie_voice-requestccie_voice-requ...@onlinestudylist.com; Date: Fri, Nov 5, 2010 01:01 AM To: ccie_voiceccie_voice@onlinestudylist.com; Subject: CCIE_Voice Digest, Vol 57, Issue 19 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. Re: GK and cube problem (givemeccievoice2...@gmail.com) -- Message: 1 Date: Thu, 4 Nov 2010 10:01:15 -0700 From: givemeccievoice2...@gmail.com To: 'ccie_voice' ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] GK and cube problem Message-ID: 000601cb7c41$e5381f10$afa85d...@com Content-Type: text/plain; charset=us-ascii Could you also send the output from the command debug voice ipipgw when you attempt a call. This would also help find the problem from a CUBE standpoint. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cristobal Priego Sent: Thursday, November 04, 2010 8:45 AM To: bruno.juniper Cc: ccie_voice Subject: Re: [OSL | CCIE_Voice] GK and cube problem do you have your Voip Dial peers configured? 2010/11/4 bruno.juniper bruno.juni...@gmail.com hello mate, I have some problem regarding GK and CUBE. HQ is gk also a cube.HQ user call 01132* go through gk. my config is below. the call didn't succeed. it show gk_rassrv_sep_arq: LRQ suspension point failed (return code = 0x4009). can anyone help me? interface FastEthernet0/0.102 description ***VOICE VLAN*** encapsulation dot1Q 102 ip address 142.102.64.254 255.255.255.0 ip helper-address 142.100.64.11 h323-gateway voip interface h323-gateway voip id VGK ipaddr 142.1.64.254 1719 h323-gateway voip h323-id CUBE h323-gateway voip bind srcaddr 142.102.64.254 ! ! gatekeeper zone local GK cisco.com 142.1.64.254 zone local VGK cisco.com zone remote BBGK cisco.com 157.1.26.253 1719 outvia VGK zone prefix BBGK 01132* no shutdown ! HQ-RTR#debug gatek ma 10 Nov 3 13:39:17.072: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Nov 3 13:39:17.076: ////GK/gk_rassrv_arq: arqp=0x4A2DE644,crv=0xB, answerCall=0 Nov 3 13:39:17.076: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC Nov 3 13:39:17.076: //809F22BF0B00/809F22BF0B00/GK/gk_dns_query: No Name servers Nov 3 13:39:17.076: //809F22BF0B00/809F22BF0B00/GK/rassrv_get_addrinfo: (0113212345678) Tech-prefix match failed. Nov 3 13:39:17.076:
Re: [OSL | CCIE_Voice] Call Forward Unregistered
hello vik, you said The only way to manipulate the Redirecting # in UCM is using the VM Profile. if u add a VM profile. and apple it to line 3001 . i think it will affect when the phone go back the normal. when the phone 3001 register the ucm. the phone 3001 hit the voicemail button. it won't hear the the system ask him input pin code . he will hear hello ,welcome to cisco unity bruno___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUE 7.0.3 ivr license installation failed
hello mate , how did u finally fix the problem? brunoHi , I got a strange problem when trying to install ivr license file , anybody face this before ? Downloading ftp cue-vm-license_4port_ivr_7.0.3.pkg Bytes downloaded : 3319 Validating package signature ... done compatibility mode Validating installed manifests .complete. The system will be brought to offline state for a brief period and will be brought back to online state automatically No work order produced. The system is back in online state 55296+0 records in 108+0 records out ERROR - Hot Installation failed. Cyrus -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20090801/b0c3fbab/attachment-0003.html___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] E1 card can ring no voice
hello mate , i met some problem. the e1 card work wired . IPphone can ring but no sound (no ringback , no voice ,no busytone) IPphone--2811 ---3640 IPphone config of 2811 isdn switch-type primary-net5 ! network-clock-participate wic 0 network-clock-select 1 E1 0/0/0 ! controller E1 0/0/0 framing NO-CRC4 pri-group timeslots 1-3,16 ! interface Serial0/0/0:15 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn incoming-voice voice isdn outgoing display-ie isdn outgoing ie redirecting-number no cdp enable voice-port 0/0/0:15___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] cbarge key is not active here
hello mate， when i do the cbarge feature on cme ,i press the cbarge softkey ,it show the key is not active here.how did i active this key? could somebody give me a config. bruno___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
