Re: [OSL | CCIE_Voice] srst configuration for cbarge
I had some weirdness with the variant using auto-provision all (not auto-provision none as per the blog article) ! telephony-service srst mode auto-provision all ! In this case I expected CBarge and privacy-button to work out-of-the-box. (I have disabled single-button-barge on CUCM and configured the conference bridge to fallback to SRST) . In my testing this did not work: I had to bounce SRST mode, save the config (careful to reinput isdn bind-l3 ccm-manager), and reload the router. Now if the phones fall into SRST the ephone-template will take. Without the router reload the ephone-template seems to be ignored: i.e. privacy is on, privacy-button does not appear ephone-template 1 softkeys remote-in-use NewCall CBarge privacy off privacy-button Does it work for you folks immediately? ___ 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 configuration for cbarge
I am testing no pre-configuration and auto-provision all, i.e., no ephones and ephone-dns. In telephony-service use no privacy In ephone-template use privacy-button When it falls into CME as SRST, the ephones show privacy: 1 and there is no privacy-button - seems like the privacy and privacy-button settings are ignored. I need to save the config, revert to CUCM, reload the router, then try SRST again. Now everything works. I'm using 12.4(22)T. On Tue, Apr 10, 2012 at 11:38 PM, Vik Malhi vma...@ipexpert.com wrote: I have always disabled privacy on the ephone and in the case of the privacy button- either template or ephone. And this works right away. Are you saying that disabling privacy on the ephone template without it being disabled on the ephone takes effect? Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Apr 10, 2012, at 2:27 AM, Anthony Alba wrote: I had some weirdness with the variant using auto-provision all (not auto-provision none as per the blog article) ! telephony-service srst mode auto-provision all ! In this case I expected CBarge and privacy-button to work out-of-the-box. (I have disabled single-button-barge on CUCM and configured the conference bridge to fallback to SRST) . In my testing this did not work: I had to bounce SRST mode, save the config (careful to reinput isdn bind-l3 ccm-manager), and reload the router. Now if the phones fall into SRST the ephone-template will take. Without the router reload the ephone-template seems to be ignored: i.e. privacy is on, privacy-button does not appear ephone-template 1 softkeys remote-in-use NewCall CBarge privacy off privacy-button Does it work for you folks immediately? ___ 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] VPIM
Hello Juan, Try 8.0.2 for VPIM; it's included in the demo license. Of course, you should still use 7.0 for all other tasks. On Sat, Mar 10, 2012 at 12:13 AM, Juan Lopez lopez.hernandez.j...@gmail.com wrote: how do people train this part with their own HW if VPIM is not in the CUC demo license? Is there a workaround? cheers, Juan 2012/3/9 Cisco Nut rafayc...@gmail.com Hi Vik Its my CUC, I am able to add location in my CUE, when I send message to 2125002 from 3002, I hear its telling me sending message to 5002 location 212 but message never gets deliverd, instead I get a message in 3002 that message is not delivered to 5002. I guess its due to the fact CUC dont have VPIM license and it wont accept or send VPIM messages. Regards Rafay On Thu, Mar 8, 2012 at 11:05 PM, Vik Malhi vma...@ipexpert.com wrote: Is this your CUC or CUE? The demo license on CUC does not allow you to add VPIM locations. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Mar 8, 2012, at 4:51 PM, Cisco Nut wrote: Hi I am running a Demo license on my CUE server, when I add VPIM location it gives me an error that VPIM is a license feature, Please let me kow how you guys are working on VPIM in your home labs. Please see below exact error I get when I tried adding VPIM location. Regards Status [image: error] The requested operation would result in a license violation. [image: error] Unable to create VPIM Location ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com http://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com http://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] When does IPMA route point show as registered?
I would like to build-up a step-by-step IPMA Proxy mode checklist and verification. If you configure the IPMA route point (with DN a superset of Managers' DNs like 5XXX), configure the IPMA Service Parameters on both Pub/Sub and restart he IPMA service, ought the IPMA route point appear as registered? (at this stage I have no Managers or Assistants configured) In my brief testing this usually doesn't happen (I suspect my VMs), but I would like to confirm whether the IPMA route point should appear registered or unregistered without any user configuration (yet). ___ 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] CUCM-GK-CUCME-CUE Call failure.
Does your call from CUCM to B-ACD allow G.711? If not, then the suggested way is to use another incoming dial-peer with a prefix e.g. 44#3500. This matches G.729, strip the prefix 44#, then this should match dial-peer 3500 and invoke the transcoder. I hit exactly the same problem and this was the proposed soln. On 18 Feb 2012, at 22:38, J. Peralta jperalt...@gmail.com wrote: Ok... So I did a search on OSL archives and it seems like defining voice-class codec command which contains multiple codecs in it to a dial peer used for incoming calls is not recommended when there's a need to invoke the transcoder. So I created another dial-peer with incoming called-number . and assigned that the voice-class codec command to it. This also fix the issue with my SIP phones at the CUCME site using g711. Am i correct on this? I have changed my configuration to the following, however, I'm not able to call B-ACD from CUCM through the GK. I have enabled fast-start on my trunk to GK from CUCM. dial-peer voice 100 pots incoming called-number . direct-inward-dial ! dial-peer voice 3000 voip incoming called-number 3...$ dtmf-relay h245-alphanumeric ! dial-peer voice 3999 voip destination-pattern 3999$ session protocol sipv2 session target ipv4:10.60.100.240 dtmf-relay sip-notify sip-kpml codec g711ulaw no vad ! dial-peer voice 3500 voip service aa destination-pattern 3500 session target ipv4:10.10.110.3 incoming called-number 3500 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 3501 voip service aa-drop destination-pattern 3501 session target ipv4:10.10.110.3 incoming called-number 3501 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 5000 voip destination-pattern 5...$ session target ras incoming called-number . dtmf-relay h245-alphanumeric no vad ! dial-peer voice 10 voip voice-class codec 1 incoming called-number . dtmf-relay h245-alphanumeric On Sat, Feb 18, 2012 at 6:36 PM, J. Peralta jperalt...@gmail.com wrote: Hello Experts, I've been having this problem for a while and I'm not able to understand why the call is failing and I get a fast busy. I have a GK trunk from CUCM to CUCME where I have configured CUE. CUE works from my CUCME phone just fine. I'm able to make the call go through from CUCM with this configuration on my CUCME. (This configuration does not have the voice class codec 1 or 2 assigned to the incoming dial peer. When I assigned the voice class codec command to the dial-peer that's when the call fails) voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip no supplementary-service h225-notify cid-update h323 sip bind control source-interface Loopback0 bind media source-interface Loopback0 registrar server ! ! ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 ! voice class codec 2 codec preference 1 g729r8 codec preference 2 g711ulaw sccp local Loopback0 sccp ccm 10.10.110.3 identifier 1 version 7.0 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register xcoder keepalive retries 5 switchover method immediate switchback method immediate switchback interval 5 ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 5 associate application SCCP dial-peer voice 3000 voip incoming called-number 3...$ dtmf-relay h245-alphanumeric ! dial-peer voice 3999 voip destination-pattern 3999$ session protocol sipv2 session target ipv4:10.60.100.240 dtmf-relay sip-notify sip-kpml codec g711ulaw no vad telephony-service sdspfarm units 4 sdspfarm transcode sessions 40 sdspfarm tag 1 xcoder authentication credential admin cisco max-ephones 20 max-dn 20 no-reg ip source-address 10.10.110.3 port 2000 system message Cisco Unified CME - Spain url services http://10.60.100.240/voiceview/common/login.do url authentication http://10.60.100.1/CCMCIP/authenticate.asp voicemail 3999 max-conferences 8 gain -6 call-forward pattern .T moh music-on-hold.au web admin system name admin password cisco dn-webedit transfer-system full-consult transfer-pattern .T create cnf-files version-stamp 7960 Feb 12 2012 00:04:19 ___ 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 Session step Cisco's example script
Hi list, I am referring to the Cisco sample script that uses Session steps on pg 17-2 of Cisco Unified Contact Center Express Scripting and Development Series: Volume 1, Getting Started with Scripts 7.0(1) http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_7_0/user/guide/uccx70edgs.pdf The script starts as follows: Start Accept session = Get Contact Info (--Triggering Contact--, Session) Get Session Info (session) /* etc etc */ My question: since you are entering from the top of the script, under what routing conditions will the Session be a cached Session object and not just a new instantiated null populated object? The explantion states: The designer continues to build the sample script by adding a Get Session Info step, which evaluates the value of session, attempting to retrieve previous information collected from the caller, who may have been [1] disconnected during a previous call or [2] transferred back into the Cisco Unified CCX queue by an agent. A caller can be transferred back into the queue if the script fails, in which case the Cisco Cisco Unified CCX system falls back to the default script (see Using Default Scripts, page 17-32), or if an agent routes the call back to the route point. [1] I don't this applies here as there is a Get Session step further in the script that attempts to retrieve a cached Session using the Mapping ID [2] Does this mean that if the Agent does a call transfer back to the CTI Route Point (i.e, the caller was never disconnected) Get Contact Info ( Session) and Get Session Info will retrieve the current Session and not instantiate a new Session object? Side note: how does an agent route(s) the call back to the route point - does this mean consult transfer? Regards 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] Ebook Lab 4: Campus QoS Task 3.1 - 25% bandwidth cap per queue
Task 3.1 configure campus qos * egress media in the priority queue * egress signal in the second queue * do not permit any queue to exceed 25% of the total allowable bandwidth Solution: int f1/0/2 auto qos voip cisco-phone !--- adjust auto qos int f1/02 srr-queue bandw shape 4 4 4 4 priority-queue out service-policy input XXX Don't quite understand the solution here: From Vik's explanation I put an asterisks for the rate limiting requirement of 25% for each queue because priority queue command will allow Q1 to use 100% of the queue up to the service-policy applied there. However, the configuration of “4 4 4 4” means each queue is configured to not exceed 1/4 or 25% of the bandwidth. This para confuses me re queue and bandwidth:the PQ will get 100% of the queue, but I thought that the srr shape won't be able to limit Q1, since PQ trumps eveything else. Furthermore the service-policy is an input policy, what does this have to do with egress queueing? ___ 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] New Five Labs: UCCX Question
The script populates the variable at runtime with Set Enterprise Call Info step; this happens in the Select Resource step before you connect the caller to the agent. Now, much to my surprise, I actually managed to get this to work and I saw the field get updated. The question I want to ask the list: does the telecaster application user need any specific User Group (e.g. Standard CTI Enabled) ? The 6.6 and 8.x CAD are quite skimpy on this: they state to create an application user telecaster/password telecaster and associate all Agent phones. They don't mention whether the telecaster user needs specific roles. Searches on this list turn up which state that the telecaster user is needed for Expanded Call Variables to work. On Mon, Jan 23, 2012 at 10:12 PM, John McGaughey (jomcgaug) jomcg...@cisco.com wrote: In lab 1 of the new 5 labs, question 7.3 it asks that the agent be able to see the ani and number of calls in queue. The DSG says to add a “salesinq” field to the default layout. The problem is that there is nothing telling IPPA what to populate this field with. ** ** So in my lab I see the following when I press CDATA. ** ** ANI: 4678124 callsinq: ** ** But the DSG is showing callsinq: 1. There must be a step missing from the DSG. How do we populate the callsinq field? ** ** John ___ 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] Voicemail during AAR Redux
Yes, the RDNIS of 5600 looks bad. When I dial a normal DN (1002 to 5002) I get CalledID 14087775002 and CallerID 4158881002 with no RDNIS. My Hunt Pilot is configured in AAR-GLOBAL with mask +14087775600. There is no 5600 DN around. The VM Pilot looks normal: Branch1# ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type 0xD is 0x2 0x1, Calling num 4158881002 ISDN Se0/0/0:23 Q931: Sending SETUP callref = 0x00F7 callID = 0x8089 switch = primary-ni interface = User ISDN Se0/0/0:23 Q931: TX - SETUP pd = 8 callref = 0x00F7 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98383 Exclusive, Channel 3 Display i = 'Thomas Jefferson' Calling Party Number Branch1# i = 0x2181, '4158881002' Plan:ISDN, Type:National Called Party Number i = 0xA1, '14087775600' Plan:ISDN, Type:National Redirecting Number i = 0x81, '5600' Plan:Unknown, Type:Unknown ISDN Se0/0/0:23 Q931: RX - CALL_PROC pd = 8 callref = 0x80F7 Channel ID i = 0xA98383 Exclusive, Channel 3 ISDN Se0/0/0:23 Q931: RX - ALERTING pd = 8 callref = 0x80F7 Progress Ind i = 0x8088 - In-band info or appropriate now available ISDN Se0/0/0:23 Q931: RX - CONNECT pd = 8 callref = 0x80F7 Display i = 'UC7' %ISDN-6-CONNECT: Interface Serial0/0/0:2 is now connected to 14087775600 N/A %ISDN-6-CONNECT: Interface Serial0/0/0:2 is now connected to 14087775600 N/A Branch1# ISDN Se0/0/0:23 Q931: TX - CONNECT_ACK pd = 8 callref = 0x00F7 Branch1# %ISDN-6-DISCONNECT: Interface Serial0/0/0:2 disconnected from 14087775600 , call lasted 2 seconds Branch1# ISDN Se0/0/0:23 Q931: TX - DISCONNECT pd = 8 callref = 0x00F7 Cause i = 0x8090 - Normal call clearing ISDN Se0/0/0:23 Q931: RX - RELEASE pd = 8 callref = 0x80F7 ISDN Se0/0/0:23 Q931: TX - RELEASE_COMP pd = 8 callref = 0x00F7 On Thu, Jan 19, 2012 at 4:26 PM, George Goglidze gogli...@gmail.com wrote: Hi Anthony, Case 1 if correct. That is the best way to configure it. In case 2, you should not need to configure that forwarding rule at all. As a matter of fact, I believe the call should not have RDNIS at all, it's a direct call so should only have DNIS/ANI. If I remember correctly at least. I'm pretty sure it's like that. What's the AAR configuration that you have, especially on VM Ports and BR1 Phones? As well make sure there is no DN 5600 floating around without being assigned to any device. Can you paste the q931 message from BR1 GW and coming into HQ gateway too? Cheers, On Thu, Jan 19, 2012 at 12:15 AM, Anthony Alba ascanio.al...@gmail.comwrote: Hello, this issue has surfaced in the past but no one email seems to summarize the exact requirements to get Voicemail to work during AAR. I'd like to give a go and get your feedback: Task: BR1, a H.323 GW, is in AAR, Voicemail must work 1. BR1 Ph2 dials Voicemail external PSTN DID directly: Note: HQ-RTR sees: CalledID 4087775600, CallerID 4158881002 Solution: Configure 415888100N as alternate extension for all BR1 lines 100N 2. BR1 Ph2 presses messages service button or dials 5600 (the VM pilot) Note: HQ-RTR sees: CalledID 4087775600, CallerID 4158881002, RDNIS 5600 Solution: This is the task that seems to cause the most confusion, you hit the Unity Connection Opening Greeting rather than the users Attempt Sign-In; this is due to the RDNIS 5600 which isn't a mailbox on the system. Unlike some reports which stated that the 10D CallerID as alternate extension worked for them. I found that the RDNIS matching wins, it is a non-mailbox extension, so I always get Unity Connection Opening Greeting. Can you guys confirm that this is the expected behaviour for RDNIS = 5600 (VM pilot) and CallerID = 4158881002 (1002 alternate extension). My solution is to add a Fowarded Routing Rule with Forwarding Station = 5600 and Send Call To = Attempt Sign-In I have only read one report that suggested this and I find I need this; yet nobody else seemed to need this. Hence I really like to hear your thoughts: is the Forwarded Routing Rule mandatory? 3. PSTN, Internal users call BR1 Ph2 Note: HQ-RTR sees CalledID 4087775600, CallerID 123456789, RDNIS 1002 Solution: this task works with no further configuration because the RDNIS is already correct. ___ 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] Voicemail during AAR Redux
Hello, this issue has surfaced in the past but no one email seems to summarize the exact requirements to get Voicemail to work during AAR. I'd like to give a go and get your feedback: Task: BR1, a H.323 GW, is in AAR, Voicemail must work 1. BR1 Ph2 dials Voicemail external PSTN DID directly: Note: HQ-RTR sees: CalledID 4087775600, CallerID 4158881002 Solution: Configure 415888100N as alternate extension for all BR1 lines 100N 2. BR1 Ph2 presses messages service button or dials 5600 (the VM pilot) Note: HQ-RTR sees: CalledID 4087775600, CallerID 4158881002, RDNIS 5600 Solution: This is the task that seems to cause the most confusion, you hit the Unity Connection Opening Greeting rather than the users Attempt Sign-In; this is due to the RDNIS 5600 which isn't a mailbox on the system. Unlike some reports which stated that the 10D CallerID as alternate extension worked for them. I found that the RDNIS matching wins, it is a non-mailbox extension, so I always get Unity Connection Opening Greeting. Can you guys confirm that this is the expected behaviour for RDNIS = 5600 (VM pilot) and CallerID = 4158881002 (1002 alternate extension). My solution is to add a Fowarded Routing Rule with Forwarding Station = 5600 and Send Call To = Attempt Sign-In I have only read one report that suggested this and I find I need this; yet nobody else seemed to need this. Hence I really like to hear your thoughts: is the Forwarded Routing Rule mandatory? 3. PSTN, Internal users call BR1 Ph2 Note: HQ-RTR sees CalledID 4087775600, CallerID 123456789, RDNIS 1002 Solution: this task works with no further configuration because the RDNIS is already correct. ___ 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] QoS question on Workbook2 Lab 10
...I just did a check: in Workbook 2 Lab 6, Tasks 7.1, 7.2 we are trusting the phones+HWIC-4ESW on both BR1 BR2 , the class-map used is class-map match-all wan-rtp match dscp ef etc. etc ...so as I thought, the DSG is not consistent here... On Wed, Jan 18, 2012 at 12:52 PM, Anthony Alba ascanio.al...@gmail.comwrote: Hello, This is what I thought the DSG was pointing too: the HWIC-4ESW is a cheapo low-end device and we're not sure what it does with the markings from the phone so let's re-classify and re-mark at BR1's WAN egress interface to be safe (i.e., don't depend on what phone + HWIC-4ESW passes to us) BTW, I have no knowledge that the HWIC-4ESW spoils markings so this is more a case of being paranoid. Now if you had your phones attached via another 3750 to BR1 then by all means use trust. (I'm not sure the DSG is entirely consistent about this: I'm sure there are other solutions where the phone+HWIC-4ESW is trusted.) On Tue, Jan 17, 2012 at 10:18 AM, John McGaughey (jomcgaug) jomcg...@cisco.com wrote: Hello, ** ** In Workbook 2, Lab 10, question 5.2 it asks you to setup MLP LFI between HQ and BR1. In the solution guide it has you use auto qos trust on the HQ side but does not use trust on the BR1 side. The DSG guide says the reason for not using the trust key word is because of the following: ** ** *Note that we have not done any prior QOS classification/marking on the ESW module therefore we will use class-based marking (no use of the trust keyword when running auto qos).* * * But the phones use the following markings by default. ** ** signaling (SCCP or SIP) - CoS 3 / cs3 media (RTP) - CoS 5 / DSCP 46 (EF) ** ** Why couldn’t we just use the trust keyword on BR1 as well since the phone is already marking the packets correctly? ** ** John ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] BACD - question
Hi Randall, You cannot control which file is used for the options:it is hardcoded in the TCL script and not exposed as a param for us to change; i.e., there is no way to point the configuration to use another options audio file. To change the options menu you have to replace _bacd_options_menu.au file. If you have multiple AAs, then Cisco recommends to record all the greetings and menu choices into the welcome file (one file per AA that you configure) and record 2 seconds of silence for _bacd_options_menu.au. Anthony On Tue, Jan 10, 2012 at 5:13 AM, Randall Crumm rrcr...@yahoo.com wrote: HI, I have configured BACD on my sc-rtr. It does work but, I do not know how to control the greetings. When I dial 02077353000 i get Thank you for calling. This is excpected. Then I am getting for sales press 1 for Customer Service press 2 to dial by extension press 3 for operator press 0 Here is my config: application service queue flash:/bacdprompts/app-b-acd-2.1.2.2.tcl param number-of-hunt-grps 2 param aa-hunt2 3002 param aa-hunt3 5010 param queue-len 15 param queue-manager-debugs 1 ! service aa flash:/bacdprompts/app-b-acd-aa-2.1.2.2.tcl paramspace english index 1 paramspace english language en paramspace english location flash:/bacdprompts/ param service-name queue param handoff-string aa param aa-pilot 3000 param welcome-prompt _bacd_welcome.au param number-of-hunt-grps 2 param dial-by-extension-option 1 param second-greeting-time 60 param call-retry-timer 15 param max-time-call-retry 700 param max-time-vm-retry 2 param voice-mail 3600 ! dial-peer voice 222 voip service aa destination-pattern 3000 session target ipv4:10.10.110.3 incoming called-number 3000 dtmf-relay h245-alphanumeric codec g711ulaw no vad Thanks, 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
[OSL | CCIE_Voice] Dummy CTI route points in Solutions Guide to CFA to Voice-mail
Hi, the solution guide uses dummy (unregistered) CTI route points in many tasks purely to forward calls (CFA) to Unity Connection, either mailboxes, live record, call handlers, greetings administrator etc Examples: Lab 1: Dummy CTI route point at DN 1113 for MeetMe task Why not just use a directory number (no device) to CFA to Voicemail? Is there any difference in using a directory number (no device) or a dummy CTI Route point? 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] CME Multicast Paging: does it work to non-connected subnets?
I am configuring multicast paging on CME. ephone-dn 8 number no-reg primary name Sales Page paging ip 239.3.10.1 port 2000 ephone XX paging-dn 8 Two directly connected phones Ph1 Ph2 receive multicast paging and the RTP stream shows to 239.3.10.1. However, two phones Ph5 Ph6, not directly connected to CME show unicast streams. I have connected these two CME phones of HQ-RTR and configure HQ-RTR as multicast router. The multicast path to Ph5 Ph6 is working for multicast MOH For the paging multicast route I do not see any attempt by the phones to join 239.3.10.1. There is also no mroute on HQ-RTR. 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] TEHO tasks: Would you configure local gateway backup?
Hello, In TEHO tasks, do you automatically configure a local backup, if not explicitly stated in the task? E.g Workbook 2 Lab 3 Previously we had configured 911/999, local, LD, international dialing at all sites. The TEHO task reads: Configure Tail End Hopoff wherever possible throughout your UCM cluster for calls to Spain and both sites within the US. The task doesn't explicitly mention having a local gateway as a fallback but would you do so as a matter of course. 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] IPMA: Mostly working except for Assistant transfer calls to Manager's VM
[Finally, got most of IPMA working after my earlier saga] The only thing not working is Assistant's TransVM softkey and Transfer to Voice Mail on Assistant Console. I believe all my CSS/partitions are correct; also the Manager phone can use the TrnsfVM. Every other function of IPMA seems to be working. On Assistant Console: Failed to transfer to Voice Mail On Assistant phone: nothing happens when I press the softkey (Redirect to the Manager's phone works). I must be overlooking something simple here. My CTI Route Point is registered to Sub as Sub IP address (primary IPMA Server). Is there any special consideration for the Assistant to transfer to Manager's VM? RPSM on Unity Connection doesn't show anything which means the call didn't even make it to the VM port. 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
Re: [OSL | CCIE_Voice] IPMA: Mostly working except for Assistant transfer calls to Manager's VM
Sorry for the noise: tracked the problem to proxy DN not having a vmbox pilot; IPMA uses the proxy line VM pilot, it does not extract the VM pilot from the Manager's line. On Fri, Dec 30, 2011 at 7:51 PM, Anthony Alba ascanio.al...@gmail.comwrote: [Finally, got most of IPMA working after my earlier saga] The only thing not working is Assistant's TransVM softkey and Transfer to Voice Mail on Assistant Console. I believe all my CSS/partitions are correct; also the Manager phone can use the TrnsfVM. Every other function of IPMA seems to be working. On Assistant Console: Failed to transfer to Voice Mail On Assistant phone: nothing happens when I press the softkey (Redirect to the Manager's phone works). I must be overlooking something simple here. My CTI Route Point is registered to Sub as Sub IP address (primary IPMA Server). Is there any special consideration for the Assistant to transfer to Manager's VM? RPSM on Unity Connection doesn't show anything which means the call didn't even make it to the VM port. 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
Re: [OSL | CCIE_Voice] IPMA: Mostly working except for Assistant transfer calls to Manager's VM
My error was the Proxy DN did not have a VM pilot set; I assumed that IPMA would use the VM pilot of the Manager, but the log4j trace showed VM pilot error. When I configured the Proxy DN with the same VM pilot as Manager then it worked. Finally, I have the full IPMA/Assistant Console lab'ed and working - whew - what a complicated business IPMA proxy line is :-(( On Fri, Dec 30, 2011 at 8:34 PM, datucha123 datucha123 datucha...@gmail.com wrote: Does the Assistants Proxy Line has the VM Pilot Partition in its CSS? On Fri, Dec 30, 2011 at 3:51 PM, Anthony Alba ascanio.al...@gmail.comwrote: [Finally, got most of IPMA working after my earlier saga] The only thing not working is Assistant's TransVM softkey and Transfer to Voice Mail on Assistant Console. I believe all my CSS/partitions are correct; also the Manager phone can use the TrnsfVM. Every other function of IPMA seems to be working. On Assistant Console: Failed to transfer to Voice Mail On Assistant phone: nothing happens when I press the softkey (Redirect to the Manager's phone works). I must be overlooking something simple here. My CTI Route Point is registered to Sub as Sub IP address (primary IPMA Server). Is there any special consideration for the Assistant to transfer to Manager's VM? RPSM on Unity Connection doesn't show anything which means the call didn't even make it to the VM port. 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 http://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] IPMA: Mostly working except for Assistant transfer calls to Manager's VM
Were you using Unity Connection as Default VM profile? I had a separate non-default Unity Connection VM profile so TransVm did not work out-of-the-box. On Fri, Dec 30, 2011 at 9:21 PM, datucha123 datucha123 datucha...@gmail.com wrote: Well, I also have configured the IPMA, and did not had the VM Profile assigned to Assistants Proxy Line (Even the Managers DN did not had a VM Profile assinged) and the TransfVM worked great. On Fri, Dec 30, 2011 at 4:42 PM, Anthony Alba ascanio.al...@gmail.comwrote: My error was the Proxy DN did not have a VM pilot set; I assumed that IPMA would use the VM pilot of the Manager, but the log4j trace showed VM pilot error. When I configured the Proxy DN with the same VM pilot as Manager then it worked. Finally, I have the full IPMA/Assistant Console lab'ed and working - whew - what a complicated business IPMA proxy line is :-(( On Fri, Dec 30, 2011 at 8:34 PM, datucha123 datucha123 datucha...@gmail.com wrote: Does the Assistants Proxy Line has the VM Pilot Partition in its CSS? On Fri, Dec 30, 2011 at 3:51 PM, Anthony Alba ascanio.al...@gmail.comwrote: [Finally, got most of IPMA working after my earlier saga] The only thing not working is Assistant's TransVM softkey and Transfer to Voice Mail on Assistant Console. I believe all my CSS/partitions are correct; also the Manager phone can use the TrnsfVM. Every other function of IPMA seems to be working. On Assistant Console: Failed to transfer to Voice Mail On Assistant phone: nothing happens when I press the softkey (Redirect to the Manager's phone works). I must be overlooking something simple here. My CTI Route Point is registered to Sub as Sub IP address (primary IPMA Server). Is there any special consideration for the Assistant to transfer to Manager's VM? RPSM on Unity Connection doesn't show anything which means the call didn't even make it to the VM port. 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 http://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] IOS version of Proctorlabs Routers
Gatekeeper license for 12.4-20T+ is very expensive. HQ-RTR can run 12.4-15T without us students noticing any difference except as you found out for the obvious cut-n-paste issues. Interestingly, I accidentally misconfigured my home PSTN with 15.1 - didn't notice any difference until I started getting gatekeeper license warnings and hit the voip-to-voip toll-fraud feature set of 15 (that's another long story...) BR1-RTR and BR2-RTR must run 12.4-20T+ for CME 7/SRST features. On Mon, Dec 26, 2011 at 2:47 AM, Ken Wyan kew...@gmail.com wrote: Dear All, In proctorlabs CCIE Voice Racks , they claim All routers run 12.4.22T IOS . https://proctorlabs.com/index.cfm/product/sku/CCIE_Voice_vRack_Online_Hardware_Rental_Session Actually they use older IOS images ( HQ , BR1 BR2 routers have different IOS versions ) . ( I can't copy sccp ccm 10.10.210.11 identifier 1 version 7 command between Routers. Routers expect version 5 , 6 7 depending on IOS) Why do they show false information on IOS before scheduling / purchasing rack rental sessions. ( I apologize if this is not the correct place for such complains ; but I'm sure relevant people will get my message from this list ) Thank You ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Campus QoS: mls qos trust and service-policy simultaneously clarification needed
Hi, What is the definitive requirement for having mls qos trust and service-policy on an interface to work. In the 12.44 command reference Classification using a port trust state (for example, *mls qos trust* [*cos * | *dscp* | *ip-precedence*] and a policy map (for example, *service-policy input* *policy-map-name*) are mutually exclusive. The last one configured overwrites the previous configuration. An earlier thread on the list http://onlinestudylist.com/archives/ccie_voice/2011-January/072385.html seems to indicate you can have both mls qos trust and service-policy input If you want to match on AF31 and re-mark to CS3: ! mls qos ! class-map AF31 match ip dscp AF31 ! policy-map REMARK class AF31 set ip dscp CS3 ! int fa0/1 service-policy input REMARK ! Will this configuration work? Note there is no mls qos trust statement. 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] IPMA: Service failed to go active
Hello, I'm having a problem with the IPMA tomcat service; during restart I'm getting the message IPMA Application not started Servlet Name:Cisco IP Manager Assistant Reason: Service failed to go active. Unable to create CTI provider App ID: Cisco Tomcat Service Any ideas? I'm stumped as this used to work from the same vanilla VMs; I haven't done anything different when restarting these VMs from snapshots. The symptoms are that the CTI Route Point used for IPMA doesn't register. 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
Re: [OSL | CCIE_Voice] IPMA: Service failed to go active
Everything is configured according to the checklist. I am only using Pub for CTI and IPMA just to be simple. Stop and Started IPMA and CTI on PUb I can get the phone TUI on both Manager and Assistant phone. Service Parameter for IPMA has Route Point (I created only one route point and there is only one choice when I configure the IPMA service) My problems: IPMA route point is not registered, IP address, state is Unknown Unknown Pub: %CCM_TOMCAT_APPS-JAVAAPPLICATIONS-0-IPMANotStarted: IPMA Application not started Servlet Name: Cisco IP Manager Assistant Reason: Service failed to go active. Unable to create CTI provider App ID: Cisco Tomcat Cluster ID: Node ID: cucmpub On Tue, Dec 27, 2011 at 3:43 AM, Mohd Baqari baqari.voic...@gmail.comwrote: First verify the configuration of IPMA. I have given a detailed explination for this in my blog (x-ccie.blogspot.com). Then lets know if you are still facing problem. Regards, Mohammed Al Baqari Sent from my iPhone On Dec 26, 2011, at 6:27 PM, Anthony Alba ascanio.al...@gmail.com wrote: Hello, I'm having a problem with the IPMA tomcat service; during restart I'm getting the message IPMA Application not started Servlet Name:Cisco IP Manager Assistant Reason: Service failed to go active. Unable to create CTI provider App ID: Cisco Tomcat Service Any ideas? I'm stumped as this used to work from the same vanilla VMs; I haven't done anything different when restarting these VMs from snapshots. The symptoms are that the CTI Route Point used for IPMA doesn't register. 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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Lab Attempt #1
Just curious: 1. You mean you RDP to the VM Lab PC and ssh from there to the servers? Can you confirm you don't have routable/SSH access from the exam candidate PC. 2. Does the candidate PC at least allow telnet access to the routers or ONLY reverse telnet to the console server? I have been always been advised that telnet access is much faster for debug output, i.e., term mon, no logging console but have not been clear whether the exam site allows telnet to the routers vty lines. Thank you. Anthony On Fri, Dec 16, 2011 at 1:20 AM, Chris Martin clm.c...@gmail.com wrote: Well guys, I took my first lab attempt on Tuesday and got my results, I failed. Looking at my score report it was extremely close, which is both encouraging and disheartening. Some things I would like to share from my experience: 1. The VM lab PC you have to work with and the SSH client is really poor. I found I had to click on the screen to get it to update, I lost a good few minutes figuring out why my screen wasn't updating. 2. Be very careful if it tells you to leave something open for the Proctor to grade. I had a section working and confident of my answer but I remember closing a window toward the end of the day, not realizing this until back at the hotel. 3. Double check every IP you enter, I lost some precious time by putting the wrong IP in one section. You don't want to add additional troubleshooting. Double check, it is worth it. 4. Try to verify every section, and don't take anything for granted. There was one section I did and was so sure of my work that I didn't go back and read the question and re-verify, this was a huge mistake. Overall I am trying to stay positive, failure is hard, but next time I will be more prepared for the environment and gained some knowledge on what to watch and lookout for. Going to take a break over the holidays and hit it again next month. Hope this helps someone. Chris ___ 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] Five-Lab Lab 2: To +442077966596 on Phone Display - utter weirdness
Tks Vik, I started practicing the lab using global dial plan first, and then switched to RP/RL due to this task. I expected the task to fail 'til I had done the conversion but to my surprise it still worked sorta halfway through. Weird, this one. P.S we are missing the Solutions to 4 5 in the Members area. Cheers Richard On Tue, Dec 13, 2011 at 1:26 PM, Vik Malhi vma...@ipexpert.com wrote: I'm familiar with this- I don't know if it is by design like the affects of using Called Transformations at both the Route Pattern and Route List. It's good to know about, I think it's just a lot easier to do it without gateway called party transformation patterns (kind of defeats the object of Called Party Transformation Patterns when you have to perform manipulations on the RP or RL in combination with gw called party transformations). Vik On Dec 12, 2011, at 6:04 AM, Anthony Alba wrote: Hi, This is Lab 2 in the Five-Lab Handbook. The the task: on SC (UK site +442077964XXX, MGCP gateway) the requirement is to plus dial from directory without EditDial +442077966596 but the phone display must show To +442077966596 Normally globalized dial plan will not work; if I have one \+.! route pattern and a gateway Called Party Transformation \+4420.! -- DDI (send 8D out to PSTN) then the caller will see To 77966596. To satisfy this type of task we should use Route Pattern and digit manipulation at Route List Details. E.g. Route Pattern: \+442077966596 The route list for this task has RG_SC has primary and RG_SB as backup SB is an H.323 gateway Route List Details: RG_SC use Mask RG_SB use Mask 90114420 Caller sees To: +442077966596 But during my testing I came across a strange result where I used globalized dialplan/gateway called party transformations but got the correct display !!?? I expected it to FAIL and show To 77966596 The weirdness: if you use both global dial-plan/Called Party Transformation and at the same time use Route Pattern / Route List Details; provided the manipulation at RL details and Called Party Transformation give *identical* results then the phone will show the number as at the Route Pattern stage. Is this a bug or feature?? Example: A. WRONG: Configure only globalized dial plan +442077966596 --- 7796596: See To: 77966596 B. CORRECT: Configure both globalized dialplan and an identical overlapping route-pattern/RL details see To: +442077966596 C. TESTING: We know that gateway Called Party Transformation trumps; so to test configure globalized dialplan (correct DNIS) and deliberately create a bad route-pattern/RL details Global Dialplan +442077966596 --- 77966596 Erroneous RL details: +442077966596 --- Since Called Party Transformations trumps, we get DNIS correct and the display shows To 77966596 Summary: gateway Called Party Transformation always trumps so we always get a 8 Digit DNIS; but if Route List Details digit manipulation gives the identical pattern to the Called Party Transformation then the caller's phone will see the DNIS at the Route Pattern stage. Have you folks ever heard of this behaviour?? 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 ___ 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] Five-Lab Lab3 Task 4.5: GK to remote GK succeeds (supposed to fail)
Hi, in this task, the call from SA GK to remote PSTN GK is supposed to fail (dialing 0119167) and we are supposed to use asn1 debugs to see the LRJ (no route to destination). BUT.. My call actually succeeded. My question: is the un-bug in the initial PSTN config that is too liberal? Should there be lrq reject-unknown-prefix in the initial configuration to achieve the aim of the task? gatekeeper zone local backbone ipexpert.com 10.10.100.2 zone remote US ipexpert.com 10.10.110.1 1719 zone prefix backbone 44* gw-type-prefix 1#* default-technology no shutdown !--- call actually succeeds; 01191* is routed to local zone ! 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
Re: [OSL | CCIE_Voice] Five-Lab Lab3 Task 4.5: GK to remote GK succeeds (supposed to fail)
That's it; my PSTN-WAN.txt did not disable gateway. ! International num-exp 001202 202 num-exp 001408 408 num-exp 009167 9167 num-exp 01144207 0207 num-exp 01144161 0161 num-exp 0119167 9167 num-exp 44207 0207 ! gateway ! clock timezone EST -5 clock summer-time EDT recurring ntp master 4 ntp update-calendar ! ! gatekeeper zone local backbone ipexpert.com 10.10.100.2 zone remote US ipexpert.com 10.10.110.1 1719 zone prefix backbone 44* gw-type-prefix 1#* default-technology no shutdown ! ! telephony-service no auto-reg-ephone load 7960-7940 P00308000500 max-ephones 42 max-dn 144 ip source-address 10.10.100.2 port 2000 caller-id block code *67 system message IPexpert PSTN Phone max-conferences 8 gain -6 transfer-system full-consult create cnf-files ! On Tue, Dec 13, 2011 at 11:58 PM, Vik Malhi vma...@ipexpert.com wrote: The PSTN should have the command no gateway configured. Does your base config not include this command? Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Dec 13, 2011, at 7:09 AM, Anthony Alba wrote: Hi, in this task, the call from SA GK to remote PSTN GK is supposed to fail (dialing 0119167) and we are supposed to use asn1 debugs to see the LRJ (no route to destination). BUT.. My call actually succeeded. My question: is the un-bug in the initial PSTN config that is too liberal? Should there be lrq reject-unknown-prefix in the initial configuration to achieve the aim of the task? gatekeeper zone local backbone ipexpert.com 10.10.100.2 zone remote US ipexpert.com 10.10.110.1 1719 zone prefix backbone 44* gw-type-prefix 1#* default-technology no shutdown !--- call actually succeeds; 01191* is routed to local zone ! 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] Five-Lab Lab 2: To +442077966596 on Phone Display - utter weirdness
Hi, This is Lab 2 in the Five-Lab Handbook. The the task: on SC (UK site +442077964XXX, MGCP gateway) the requirement is to plus dial from directory without EditDial +442077966596 but the phone display must show To +442077966596 Normally globalized dial plan will not work; if I have one \+.! route pattern and a gateway Called Party Transformation \+4420.! -- DDI (send 8D out to PSTN) then the caller will see To 77966596. To satisfy this type of task we should use Route Pattern and digit manipulation at Route List Details. E.g. Route Pattern: \+442077966596 The route list for this task has RG_SC has primary and RG_SB as backup SB is an H.323 gateway Route List Details: RG_SC use Mask RG_SB use Mask 90114420 Caller sees To: +442077966596 But during my testing I came across a strange result where I used globalized dialplan/gateway called party transformations but got the correct display !!?? I expected it to FAIL and show To 77966596 The weirdness: if you use both global dial-plan/Called Party Transformation and at the same time use Route Pattern / Route List Details; provided the manipulation at RL details and Called Party Transformation give *identical* results then the phone will show the number as at the Route Pattern stage. Is this a bug or feature?? Example: A. WRONG: Configure only globalized dial plan +442077966596 --- 7796596: See To: 77966596 B. CORRECT: Configure both globalized dialplan and an identical overlapping route-pattern/RL details see To: +442077966596 C. TESTING: We know that gateway Called Party Transformation trumps; so to test configure globalized dialplan (correct DNIS) and deliberately create a bad route-pattern/RL details Global Dialplan +442077966596 --- 77966596 Erroneous RL details: +442077966596 --- Since Called Party Transformations trumps, we get DNIS correct and the display shows To 77966596 Summary: gateway Called Party Transformation always trumps so we always get a 8 Digit DNIS; but if Route List Details digit manipulation gives the identical pattern to the Called Party Transformation then the caller's phone will see the DNIS at the Route Pattern stage. Have you folks ever heard of this behaviour?? 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
Re: [OSL | CCIE_Voice] CCIE Voice Lab Guidance Req !
Suggest you use rack rentals to familiarize yourself with all the hardware and do a few labs. Then get some local hardware phones connecting to the pod remotely. You can then decide whether to do it yourself as it is expensive to put everything together. The rack rental guide shows all the hw sw needed. Good luck with your studies. On 7 Dec 2011, at 01:05, Muhammad Zubair muhammad.zub...@gmail.com wrote: Hi All i am planning to start studying for CCIE Voice. i need help to prepare my own lab at home. but no idea how to do that. i had prepare something on VM and GNS3 but PSTN is the problem. if i want to prepare in real means with real equipments than how many routers required and how to make that. i can arrange one 2821 with FXO, 2600 and one PoE switch. is this enough ? can anyone guide me how to start and how to prepare a good lab which can fulfill this need. -- Muhammad Zubair Mob : +973-36613334 / +973-66338466 Email: muhammad.zub...@gmail.com Skype: muhammad.zubairr ___ 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] MOH to HW Conference Bridge - Unicast-only MRGL required?
If we allow a participant to put a conference on hold with MOH, i.e., Suppress MOH to Conference Bridge == FALSE does the Conference Bridge need a unicast-only MRGL to reflect MOH back to all the other participants? I find that if the device pool MRGL has a multicast-enabled MRG then although a multicast MOH resource is activated there is no MOH heard by the participants. I'm guessing it's not possible for the conference bridge to take the multicast stream and remix to the participants. 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] SNUR requires 'Urgent Priority' on globalized translation pattern \+.! ????
Hello, Mobile Connect (SNUR) issue: ** using E.164 for remote destination e.g. +12123941234 ** using globalized dial plan with one route pattern \+.! ** using one translation pattern \+.! (for plus dialing from directory) whose CSS sees the global route pattern. I do not want the devices to see the route pattern directly; every dialed number goes via a translation pattern. If the \+.! translation pattern does not have 'Urgent Priority' then mobile connect cannot route out using this pattern. Is this expected behaviour? Actually, I do not want the translation pattern \+.! to have 'Urgent Priority' because this causes issues with SIP phones and + dialing from directory. Is there some reference to this behaviour of SNUR? Tks. 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
Re: [OSL | CCIE_Voice] Introducing myself
Bonjour Nicholas I am in the same position, RS trying to move to Voice, I have just passed the written. I have also built a lab and I use rack rentals to see the configurations and deliberate 'bugs'. My lessons learnt so far for home rack: ** use Intel for your VMware server; the versions of the servers don't install nice on AMD ** use Unity Connection 8 for home VPIM, as VPIM is in the demo license. I am running both CUC 7 and 8. I had problems installing 8 on VMware; in the end I gave it unlimited memory during install and reduced to 3gb after install. ** try to get at least a pair of 7965 so you see the actual lab phone and needed to test iLBC tasks ** if you access your rack with local hardware phones to your pod by l2l VPN you can get multicast moh to work over a gre tunnel to your home phones. Since you've done RS you'll know what I mean: switch to PIM sparse mode. I don't use another router at home as the tunnel peer but just a Linux box and Xorp. I use HQ router as tunnel endpoint and PIM RP. It works very well. ** if your phones are separated by VPN from servers always set up a local TFTP servers for firmware and use loadServer in CUCM to tell phones to grab local firmware. For CME phones I switch firmware using CUCM and loadServer first. TFTP over the WAN is very very slow. Anthony On Friday, December 2, 2011, Nicolas MICHEL mcl.nico...@gmail.com wrote: Hey Michael. Thanks again for all the help provided with the CCIE RS when I was studying for it :) How far from the CCIE Voice are you now ? I m just starting, building a phones lab and then I'll be using IPX and some other vendors as well I guess I remember you were building a rack with friends, how far are you from there ? :) Seeing that this Mailing list is far more active than the RS one ! Cheers !! Nic 2011/12/2 Michael Miller kf4...@gmail.com Hello Nicholas, Its nice to see some familiar faces from the RS OSL boards. =) Congrats on passing the RS, and good luck on the Voice! Thanks, Michael On Fri, Dec 2, 2011 at 11:30 AM, Emanuel Damasceno aedamasc...@gmail.com wrote: Welcome Michel! You will see this is just a nice journey as RS was for you. I am the opposite, I started into Voice, and when I get my CCIE, I will start on my CCNP, and CCIE RS... =) Welcome to the UC world. I really love it and I am hoping you will love it too! Best regards, brother. Emanuel Damasceno On Fri, Dec 2, 2011 at 8:11 AM, Nicolas MICHEL mcl.nico...@gmail.com wrote: Hey There guys. I'm a french network engineer mainly focused into RS but as of now I m starting to deploy UC solutions and so far so good I like it. This is why I decided to pursue my 2nd CCIE into Voice and can't wait to be there yet :) I actually finished the CCNA book and the CBT nuggets for that series and now digging into CCNP stuff. I got CVOICE, CIPT1,CIPT2, Presence, CUCM+unity, Unity books. and then I ll start to read the SRND which looks awesome. I'm also building a lab to use some remote racks. If you guys have any advices, I d be glad to hear them :P Thanks for your help and cant wait to have the knowledge to ask question and answer on the OSL :) 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 -- Nicolas MICHEL Ingenieur Réseaux CCIE #29410 Tel: +33 6 08 72 75 97 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Vol 2 Lab 2: cannot get inbound H.323 trunk calls to CME SIP phone to work G.729 to G.711
Tks Ash! That explains the solution guide where there are no voice classes on thIs trunk, although in trunks to PSTN breakout Vik invariably puts the voice-class in the dial peer. On 1 Dec 2011, at 15:41, Ashraf Ayyash ash.ayy...@gmail.com wrote: Hello Anthony , You cannot Transcode call that Hit Dial peer with Voice class codec , it make sense as the router though that he can support Both codecs I hope this clarify the issue you saw Ash On Wed, Nov 30, 2011 at 8:36 PM, Anthony Alba ascanio.al...@gmail.com wrote: Very strange: I can now get both inbound and outbound calls to CME SIP working with transcoder invoked at BR2-RTR. I cannot use voice-class codec 1 under the dial-peer. This surprises me: why would voice class codec hurt the task? voice class codec 1 codec pref 1 g729r8 codec pref 2 g711ulaw If I put this under any of the dial-peers it breaks CME SIP. ___ 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] Vol 2 Lab 2: cannot get inbound H.323 trunk calls to CME SIP phone to work G.729 to G.711
Hi, I'm working on Vol 2 Lab 2: H.323 GK controlled trunk between CUCM to CME. This is different from Vol 2 Lab 1 task where we used a CUBE between CUCM and CME. My problem: I cannot get inbound G.729 calls from CUCM to CME SIP phones to work: it clearly is some sort of codec issue; when I reconfigure to allow G.711 everywhere it works. On inbound, BR2-RTR just doesn't invoke any transcoder. One consequence is CFNA to CUE fails too (G.711 issues). To CME SCCP phones, it is the phone that switches to G.729 and does not need a transcoder (inbound or outbound). Outbound calls to CUCM work: BR2-RTR invokes a transcoder G.711 - G.729 to CUCM. All PSTN calls work; H.323 trunk calls to CME SCCP phones work (except of course CF to VM). !--- SIP Phone voice register pool 2 id mac 0064.40B4.5F35 type 7962 number 1 dn 2 dtmf-relay rtp-nte username 3005 password cisco description 932143005 codec g711ulaw !--- inbound dial peer for G.729 voice class codec 1 codec preference 1 g729r8 codec preference 2 g711ulaw ! dial-peer voice 1001 voip voice-class codec 1 incoming called-number 3...$ dtmf-relay h245-alphanumeric no vad ! !--- H.323 to SIP on CME voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip fax protocol cisco h323 sip bind control source-interface Vlan400 bind media source-interface Vlan400 registrar server !--- transcoder configured Branch2#show sdspfarm units mtp-1 Device:BR2-XCODE TCP socket:[1] REGISTERED in SCCP ver 17/10 actual_stream:16 max_stream 16 IP:1 64002 MTP Dixieland keepalive 879 Supported codec: G711Ulaw G711Alaw G729a G729ab Universal Xcoder max-mtps:1, max-streams:16, alloc-streams:16, act-streams:0 ___ 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] Question about Called Number displayed on phone.
The IOS gateway cam also use H.225 back to CUCM to affect the Called Number display. Amy covers this in the audio: if you want to control the display on the phone, then do manipulation at the RP, even though this DM may be trumped by RL details. You also have to disable the H.225 notification on the IOS gateway back to CUCM or this will overwrite the called number. On Tue, Nov 29, 2011 at 1:06 PM, ccielabrat ccielab...@gmail.com wrote: Can someone help me understand what determines what gets displayed on the phone display when calling outbound. I have a setup where I have a h323 Gw and MGCP Gw in a single RL. I create a route pattern of 9.2345678 and assign it to the RL. If it goes to the H323 GW , I don't drop the 9 prefix in the RL and it displays 92345678 on the phone. If it goes to the MGCP GW, the 9 prefix is dropped in the RL and it displays 2345678 on the phone. So I figured the display value must be based on what gets sent to the GW, but this doesn't seem to be true either. If I adjust my dial-peers on H323 to match on 2345678 (no 9 prefix) , and drop the 9 in the RL , I still see the 9 prefix as dialed on the phone. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Vol 2 Lab 2: cannot get inbound H.323 trunk calls to CME SIP phone to work G.729 to G.711
I have changed the transcoder to dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 8 associate application SCCP now inbound calls to CUE and CME SIP work and I see the transcoder invoked. Tks! Now I seem to have broken something else...outbound CME SIP is now b0rked...I get reorder when CUCM phone picks up. Can you see anything wrong with dial-peer voice 3600 voip destination-pattern 3[16]00 b2bua session protocol sipv2 session target ipv4:10.10.250.254 dtmf-relay sip-notify codec g711ulaw no vad dial-peer voice 3601 voip incoming called-number 399[89] codec g711ulaw !--- outbound to GK dial-peer voice 1000 voip destination-pattern [15]...$ voice-class codec 1 voice-class h323 1 session target ras tech-prefix 1# dtmf-relay h245-alphanumeric no vad !--- inbound to CME dial-peer voice 2 voip incoming called-number 3...$ dtmf-relay h245-alphanumeric CME SIP outbound to CUCM is now broken On Thu, Dec 1, 2011 at 8:11 AM, Mohammed Al Baqari baqari.voic...@gmail.com wrote: You are using g729r8 codec on inbound dial-peer but this isn’t included in your DSP profile. Instead you are putting g729a G729ab. ** ** Regards, Mohammed Al Baqari ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Anthony Alba *Sent:* Thursday, December 01, 2011 3:38 AM *To:* CCIE Study *Subject:* [OSL | CCIE_Voice] Vol 2 Lab 2: cannot get inbound H.323 trunk calls to CME SIP phone to work G.729 to G.711 ** ** Hi, I'm working on Vol 2 Lab 2: H.323 GK controlled trunk between CUCM to CME. This is different from Vol 2 Lab 1 task where we used a CUBE between CUCM and CME. My problem: I cannot get inbound G.729 calls from CUCM to CME SIP phones to work: it clearly is some sort of codec issue; when I reconfigure to allow G.711 everywhere it works. On inbound, BR2-RTR just doesn't invoke any transcoder. One consequence is CFNA to CUE fails too (G.711 issues). To CME SCCP phones, it is the phone that switches to G.729 and does not need a transcoder (inbound or outbound). Outbound calls to CUCM work: BR2-RTR invokes a transcoder G.711 - G.729 to CUCM. All PSTN calls work; H.323 trunk calls to CME SCCP phones work (except of course CF to VM). !--- SIP Phone voice register pool 2 id mac 0064.40B4.5F35 type 7962 number 1 dn 2 dtmf-relay rtp-nte username 3005 password cisco description 932143005 codec g711ulaw !--- inbound dial peer for G.729 voice class codec 1 codec preference 1 g729r8 codec preference 2 g711ulaw ! dial-peer voice 1001 voip voice-class codec 1 incoming called-number 3...$ dtmf-relay h245-alphanumeric no vad ! !--- H.323 to SIP on CME voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip fax protocol cisco h323 sip bind control source-interface Vlan400 bind media source-interface Vlan400 registrar server !--- transcoder configured Branch2#show sdspfarm units mtp-1 Device:BR2-XCODE TCP socket:[1] REGISTERED in SCCP ver 17/10 actual_stream:16 max_stream 16 IP:1 64002 MTP Dixieland keepalive 879 Supported codec: G711Ulaw G711Alaw G729a G729ab Universal Xcoder max-mtps:1, max-streams:16, alloc-streams:16, act-streams:0 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Vol 2 Lab 2: cannot get inbound H.323 trunk calls to CME SIP phone to work G.729 to G.711
Very strange: I can now get both inbound and outbound calls to CME SIP working with transcoder invoked at BR2-RTR. I cannot use voice-class codec 1 under the dial-peer. This surprises me: why would voice class codec hurt the task? voice class codec 1 codec pref 1 g729r8 codec pref 2 g711ulaw If I put this under any of the dial-peers it breaks CME SIP. ___ 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] CUPS License
Hi, the demo license lasts for 90 days; it might be easiest to snapshot the VM before integration - you will reset the 90 day clock each time you revert to the pre-integration snapshot. On Tue, Nov 15, 2011 at 11:56 AM, Cisco Nut rafayc...@gmail.com wrote: Hello- Any one knows how I can obtain CUPS license for IE Voice Lab, Right now I am running it under Eval License. Regards ___ 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] can cuc demo license run vpim
Hi Bruno, Alas Unity Connection 7 has a separate VPIM license SKU (aka UNITYCN7-VPIM) which is not covered by the demo license. The rack rentals will have the VPIM license installed. 8.x versions will have VPIM covered in the demo license so if you clone VM/upgrade to 8.x you might be able to practice VPIM. (There is no SKU UNITYCN8-VPIM) Of course, I wouldn't recommend doing all the labs on 8.x as that is not the lab version, but I doubt the VPIM feature will differ significantly. 2011/11/9 bruno bruno.juni...@gmail.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] 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
Re: [OSL | CCIE_Voice] UCCX RmCm resources
Hello Duncan, In System -- Enterprise Parameters -- CRS Application Parameters do you see IPCC Express Installed = true Auto Attendant Installed = true ? This is necessary to get IPCC Extension to be assignable. Sometimes CUCM and UCCX get out of sync on whether they have been integrated. This (very useful) blog post shows how to resync UCCX and CUCM; it has worked for me when my CUCM and UCCX got out of whack. http://ccie-musketeers.blogspot.com/2011/01/uccx-not-properly-integrated-with-cucm.html Anthony On Thu, Nov 10, 2011 at 3:10 AM, Duncan Hamilton-Walker dun...@rosethorn.plus.com wrote: Dear All, ** ** Can someone please help me out... ** ** Under RmCm Configuration, Resources.. ** ** I can’t see the two users i have created in CM But i feel that its linked to the fact i can’t add IPCC extension to the end user page... Because its not there as an option... I have associated the DN with the users... but still nothing I know am missing something... but cant see it ** ** Please advise ** ** Thanks Duncan ___ 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] VMware snapshots for Pub Sub?
I do that - I have set up a basic Pub Sub with all the System stuff (CM
servers, date/time, device pools etc) preconfigured (to save time to jump
into any lab) but no phones, route plan, users ('cept for uccx admin user)
or anything else. When replication is stable I shutdown down both VMs and
snapshot.
On Power On I have no issues with replication.
On Thu, Nov 10, 2011 at 6:22 AM, James R boost36d...@gmail.com wrote:
I'm working on my home lab with PUB SUB running on VMware Workstation 7.
The past few study sessions I seem to always run into issues where the
database starts getting replication errors. I'll see a phone registered to
my SUB yet it wont show up in the database. Unified Reporting System
verifies that there are errors in database each time I see this. Sometimes
it takes up to an hour for the cluster to resync after running 'utils
dbreplication reset'. The replication issues get so bad I just scrap it and
do a reinstall which is time consuming. Ive been taking one snapshot at the
end of a lab for example workbook 1 lab 1 on the pub and on the sub as well
while both are running. This is so I can come back the next sesion and
begin with next lab 2 without having to start from the very beginning. Is
this the wrong way to do it? Should they be shutdown first before taking a
snapshot?
James
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[OSL | CCIE_Voice] (New to Unity Connection) Call Handler Take Message - where did it go?
Hi, Looking at Unity Connection System Call Handlers: when a system call handler takes a message where does the message go to? How do you retrieve the message if the System Call Handler is used as, e.g., an AA? Thanks 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
Re: [OSL | CCIE_Voice] transfer file to router with no tftp or ftp server
For pure text files there is the well documented tcl trick
tclsh
puts [open flash:myfile.txt a+] {
!paste text file
}
tclquit
On 5 Nov 2011, at 01:57, John Smith cci...@yahoo.com wrote:
Forgive my ignorance, but if you needed to transfer a file from a PC to a
router and had no tftp server or ftp server on the PC, how could accomplish
that? Thank you.
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Re: [OSL | CCIE_Voice] cRTP in nested CB traffic shaping policy
I have encountered this too Are you able ( does iOS allow it) to put the compress rtp config in the parent shaper class? On 4 Nov 2011, at 21:42, Nicolaers Luk luk.nicola...@quentris-gdfsuez.be wrote: Hi, I'm trying to setup CB traffic shaping with CRTP. This is the config of the policy maps: policy-map VOIP class RTP priority 28 compress header ip rtp class SIG bandwidth 16 policy-map shape class class-default shape average 365600 3656 service-policy VOIP The policy-map shape is activated on the frame-relay map-class: map-class frame-relay toBR2 frame-relay fragment 480 service-policy output shape This is configured on the HQ router and the Branch office router When I make a call I can see that RTP traffic is matched but the rtp compression counters always remain at 0 BRANCH#show policy-map interface serial 0/1/0.200 Serial0/1/0.200: DLCI 200 - Service-policy output: shape Class-map: class-default (match-any) 403 packets, 27349 bytes 5 minute offered rate 4000 bps, drop rate 0 bps Match: any Queueing queue limit 64 packets (queue depth/total drops/no-buffer drops) 0/0/0 (pkts output/bytes output) 403/27349 shape (average) cir 365600, bc 3656, be 3656 target shape rate 365600 lower bound cir 0, adapt to fecn 0 Service-policy : VOIP queue stats for all priority classes: Queueing queue limit 64 packets (queue depth/total drops/no-buffer drops) 0/0/0 (pkts output/bytes output) 252/15834 Class-map: RTP (match-any) 252 packets, 15834 bytes 5 minute offered rate 4000 bps, drop rate 0 bps Match: access-group name RTP 252 packets, 15834 bytes 5 minute rate 4000 bps Priority: 28 kbps, burst bytes 1500, b/w exceed drops: 0 compress: header ip rtp UDP/RTP (compression on, Cisco, RTP) Sent:0 total, 0 compressed, 0 bytes saved, 0 bytes sent rate 0 bps Anyone that has an idea why the compression counters remain at 0? Thanks Luk ___ 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] Triggering +
Current firmware 9.1(1)SR1 and later does plus dial for 7965G (and all other type B phones). Press * for 1 second. This is not the lab version, though it should be compatible with CUCM 7, so you can test it out. http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/firmware/9_1_1/english/release/notes/7900_911.html#wp45395 None of the lab phones (7965G) or Proctorlab phones do plus dial from the keypad using stock firmware, i.e. 8.3 or 8.4. They can plus dial from the Directories list. On Thu, Oct 27, 2011 at 10:25 PM, Emanuel Damasceno aedamasc...@gmail.comwrote: Hello Experts, I am on Lab 5A and I am now wondering how I trigger the + on the phones. I was told that 7940s don't support + dial, but I don't want to know that, I want to know what I need to press so it becomes a +. On my cell phone, I press 0 and hold it. Is it the same with Cisco Phones? Thanks *Antonio Emanuel Damasceno* CCNA, CCNA Voice, CCNP Voice, CCIE Voice (written) CompTIA Network+ ___ 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] Vol2 Lab7: DISA Dialing can't get 3.1 and 3.2 to play together
Problem: in Vol2 Lab7 DISA Dialing tasks 3.1 and 3.2 I can get either of them to work separately but not together. Any thoughts? Task 3.1: when the Remote Destination calls in, show the CLID as the Remote Destination# and not the Mobility User 4D extension. Solution: Configure a special CSS-SNR-3002 for the RDP CSS: this CSS contains the partition PT-SNR-3002 which has a sole translation pattern: pattern: partition: PT-SNR-3002 css: CSS-internal use Calling Party External Mask: check digit manipulation: --nil-- When the RD calls in to an internal DN, the pattern is matched and the CLID is changed from 3002 to +447976852817 (the RD#). Task 3.2: Configure DISA dialing at the IVR menu. Solution: once the RDP CSS contains the translation pattern from 3.1 it breaks any sort of DISA dialing. If I change the RDP CSS to the CSS of the Branch2 phones (no 3.1 translation pattern), all onward dialing works - internal DNs all the way to international calls. The moment I have any sort of translation pattern in the RDP CSS I cannot invoke onward dialing: I get through the IVR, press 1, dial NUMBER# and nothing - the connection is severed. I.e., the RDP CSS has the translation pattern, to allow onward dialing I add the same partitions in the Branch2 phones Device CSS. Theoretically I should now be able to onward dial to internal DNs and all external patterns. But this fails completely. If I remove the translation pattern CSS of 3.1 and replace it with CSS-internal that sees all internal DNs, everything works. So somehow the translation pattern which swaps out the ANI is interfering with onward dialing. BTW in System Parameters I have changed the Remote Destination CSS setting to RDP+Line CSS as recommended in the DSG. ___ 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] Vol2 Lab6: MVA by MGCP hairpin outbound calls don't work
Hi Vik Thank you for your reply. I am getting closer...I can now place calls via MVA to HQ locations G.711 but WAN calls do not connect media strems. HQ-RTR#show call active voice compact callID A/O FAX Tsec Codec typePeer Address IP Rip:udp Total call-legs: 6 1874 ORG T48g711ulawVOIPP 10.10.200.3:17084 1873 ORG T48g711ulawTELEP 1875 ANS T48g711ulawVOIPP5002 10.10.200.3:19252 1878 ORG T27g711ulawVOIPP5011 10.10.200.3:17834 1879 ORG T27g711ulawVOIPP 192.168.1.65:27708 1880 ORG T27g711ulawVOIPP 10.10.200.3:17712 When I dial to 1002@BR1 signaling connects but there is no audio. I have a transcoder at both HQ and BR1; I put the H.323 gateway in DP HQ per the DSG. No transcoder is invoked. (Tried forcing MTP and/or putting gateway in a G711-everywhere device pool; did not help. That made things worse, it could not connect signaling to 1002.) HQ-RTR#show call active voice compact callID A/O FAX Tsec Codec typePeer Address IP Rip:udp Total call-legs: 6 1901 ORG T34g711ulawVOIPP 10.10.200.3:18734 1900 ORG T34g711ulawTELEP 1902 ANS T34g711ulawVOIPP5002 10.10.200.3:18116 1905 ORG T25g729r8 VOIPP5011 10.10.200.3:17004 1906 ORG T25g729r8 VOIPP 192.168.1.81:18786 1907 ORG T25g729r8 VOIPP 10.10.200.3:16618 dial-peer voice 5 voip translation-profile incoming ToRDP service cmm destination-pattern 5011 voice-class codec 1 voice-class h323 1 session target ipv4:10.10.210.10 incoming called-number 5010 dtmf-relay h245-alphanumeric On Wed, Oct 26, 2011 at 12:37 AM, Vik Malhi vma...@ipexpert.com wrote: Its easier to make the DID number you call for MVA and the MVA DN a DIFFERENT number. Also you have a codec problem. Keep the DID# 5010. Change the MVA DN 5011. This is under the Media Resource menu in the ccmadmin page. Change the dial-peer to look like this: dial-peer voice 5 voip service cmm destination-pattern 5011 * equal to MVA DN* session target ipv4:10.10.210.10 incoming called-number 5010 * equal to DID* dtmf-relay h245-alphanumeric voice-class codec 1 * you need to support 729 and 711 since you are making a call over the WAN* no vad Make sure that you have the h323-g voip bind src in the interface you are using when you add the H323 gw. Reset the H323 gw too. Keep the MVA DN in the None partition. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Oct 24, 2011, at 10:18 PM, Anthony Alba wrote: Hello, When I try to use MVA with MGCP hairpin I cannot make calls. When I try to make a call I get the IVR menu again Dial 3945010 get IVR menu Enter PIN # Enter 1 1002# ( to make a call) ...instead of being connected I get back to the IVR menu. I seem to be trapped in some sort of loop. Any ideas? (When I change HQ-RTR to a H.323 gateway everything works including making calls. I think this means that my RDP CSS is looking good.) I have configured Mobile Voice Access as per the Solution Guide in Vol2 Lab6. HQ-RTR is running H.323 solely to provide VXML support. CUCM is configured to hairpin the call to HQ-RTR. application service cmm http://10.10.210.10:8080/ccmivr/pages/IVRMainpage.vxml dial-peer voice 5 voip service cmm destination-pattern 5010 session target ipv4:10.10.210.10 incoming called-number 5010 dtmf-relay h245-alphanumeric codec g711ulaw no vad ___ 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] Vol2 Lab6: MVA by MGCP hairpin outbound calls don't work
A bit of follow-up: If I put BR1 in a G.711 region calls connect perfectly. I have added a transcoder to HQ-RTR/telephony-service but that didn't help when BR1 is in a G.729 region. So I'm having issues to get the hairpin gateway to invoke a transcoder. On Wed, Oct 26, 2011 at 6:22 AM, Anthony Alba ascanio.al...@gmail.comwrote: Hi Vik Thank you for your reply. I am getting closer...I can now place calls via MVA to HQ locations G.711 but WAN calls do not connect media strems. HQ-RTR#show call active voice compact callID A/O FAX Tsec Codec typePeer Address IP Rip:udp Total call-legs: 6 1874 ORG T48g711ulawVOIPP 10.10.200.3:17084 1873 ORG T48g711ulawTELEP 1875 ANS T48g711ulawVOIPP5002 10.10.200.3:19252 1878 ORG T27g711ulawVOIPP5011 10.10.200.3:17834 1879 ORG T27g711ulawVOIPP 192.168.1.65:27708 1880 ORG T27g711ulawVOIPP 10.10.200.3:17712 When I dial to 1002@BR1 signaling connects but there is no audio. I have a transcoder at both HQ and BR1; I put the H.323 gateway in DP HQ per the DSG. No transcoder is invoked. (Tried forcing MTP and/or putting gateway in a G711-everywhere device pool; did not help. That made things worse, it could not connect signaling to 1002.) HQ-RTR#show call active voice compact callID A/O FAX Tsec Codec typePeer Address IP Rip:udp Total call-legs: 6 1901 ORG T34g711ulawVOIPP 10.10.200.3:18734 1900 ORG T34g711ulawTELEP 1902 ANS T34g711ulawVOIPP5002 10.10.200.3:18116 1905 ORG T25g729r8 VOIPP5011 10.10.200.3:17004 1906 ORG T25g729r8 VOIPP 192.168.1.81:18786 1907 ORG T25g729r8 VOIPP 10.10.200.3:16618 dial-peer voice 5 voip translation-profile incoming ToRDP service cmm destination-pattern 5011 voice-class codec 1 voice-class h323 1 session target ipv4:10.10.210.10 incoming called-number 5010 dtmf-relay h245-alphanumeric On Wed, Oct 26, 2011 at 12:37 AM, Vik Malhi vma...@ipexpert.com wrote: Its easier to make the DID number you call for MVA and the MVA DN a DIFFERENT number. Also you have a codec problem. Keep the DID# 5010. Change the MVA DN 5011. This is under the Media Resource menu in the ccmadmin page. Change the dial-peer to look like this: dial-peer voice 5 voip service cmm destination-pattern 5011 * equal to MVA DN* session target ipv4:10.10.210.10 incoming called-number 5010 * equal to DID* dtmf-relay h245-alphanumeric voice-class codec 1 * you need to support 729 and 711 since you are making a call over the WAN* no vad Make sure that you have the h323-g voip bind src in the interface you are using when you add the H323 gw. Reset the H323 gw too. Keep the MVA DN in the None partition. Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com On Oct 24, 2011, at 10:18 PM, Anthony Alba wrote: Hello, When I try to use MVA with MGCP hairpin I cannot make calls. When I try to make a call I get the IVR menu again Dial 3945010 get IVR menu Enter PIN # Enter 1 1002# ( to make a call) ...instead of being connected I get back to the IVR menu. I seem to be trapped in some sort of loop. Any ideas? (When I change HQ-RTR to a H.323 gateway everything works including making calls. I think this means that my RDP CSS is looking good.) I have configured Mobile Voice Access as per the Solution Guide in Vol2 Lab6. HQ-RTR is running H.323 solely to provide VXML support. CUCM is configured to hairpin the call to HQ-RTR. application service cmm http://10.10.210.10:8080/ccmivr/pages/IVRMainpage.vxml dial-peer voice 5 voip service cmm destination-pattern 5010 session target ipv4:10.10.210.10 incoming called-number 5010 dtmf-relay h245-alphanumeric codec g711ulaw no vad ___ 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] Vol2 Lab6: MVA by MGCP hairpin outbound calls don't work
Hello, When I try to use MVA with MGCP hairpin I cannot make calls. When I try to make a call I get the IVR menu again Dial 3945010 get IVR menu Enter PIN # Enter 1 1002# ( to make a call) ...instead of being connected I get back to the IVR menu. I seem to be trapped in some sort of loop. Any ideas? (When I change HQ-RTR to a H.323 gateway everything works including making calls. I think this means that my RDP CSS is looking good.) I have configured Mobile Voice Access as per the Solution Guide in Vol2 Lab6. HQ-RTR is running H.323 solely to provide VXML support. CUCM is configured to hairpin the call to HQ-RTR. application service cmm http://10.10.210.10:8080/ccmivr/pages/IVRMainpage.vxml dial-peer voice 5 voip service cmm destination-pattern 5010 session target ipv4:10.10.210.10 incoming called-number 5010 dtmf-relay h245-alphanumeric codec g711ulaw no vad ___ 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] HOWTO?: Unified CM User Options /ccmuser page from Unity Connection not working
How do you allow a user to get to the ccmuser Cisco Unified CM User Options page from Unity Connection (*not* CUCM)? https://10.10.210.13:8443/ccmuser/showHome.do The username/password is accepted but I just get bounced back to the login page. All Feature Services/Network Services are running. Voicemail and the PCA page https://10.10.210.13:8443/ciscopca/home.do are both working From CUCM, the CM User Options page works; https://10.10.210.10:8443/ccmuser/showHome.do I just can't seem to enter from Unity Connection. 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
Re: [OSL | CCIE_Voice] HOWTO?: Unified CM User Options /ccmuser
That's what I thought too...but once in a while in previous labbing by accident I managed to get a login. I want to make it reproducible...but can't recall the magic words! On 23 Oct 2011, at 00:53, Inder Singh ising...@gmail.com wrote: Hi Anthony, I don't think it is possible to get to the UCM user page from the CUC server...it must be accessed from the UCM server. Regards. Inder. Subject: [OSL | CCIE_Voice] HOWTO?: Unified CM User Options /ccmuser page from Unity Connection not working Message-ID: CADkWibdTMPUzVfKAay9G-re=yt0oug7uac7oka5pmhehko8...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 How do you allow a user to get to the ccmuser Cisco Unified CM User Options page from Unity Connection (*not* CUCM)? https://10.10.210.13:8443/ccmuser/showHome.do The username/password is accepted but I just get bounced back to the login page. All Feature Services/Network Services are running. Voicemail and the PCA page https://10.10.210.13:8443/ciscopca/home.do are both working From CUCM, the CM User Options page works; https://10.10.210.10:8443/ccmuser/showHome.do I just can't seem to enter from Unity Connection. 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 ___ 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] Call Routing
If you use an anchored regex ^ it may not consider the 9 as explicitly matched digit. What happens if you use 9[2-9]..$ ? On 21 Oct 2011, at 18:59, Ccie Voice v.c...@yahoo.com wrote: Hi all, I have very strange problem, and I need someone to help me to understand why? I am trying to study call routing, local calls I have the following setup SCCP Phone RP Local RL H.323 GW PSTN in the GW I added the following dial-peer: dial-peer voice 15 pots translation-profile outgoing loc destination-pattern ^9[2-9]..$ port 0/1/0:23 while the dial-peer is pots so the 9 should be stripped and remaining will be send to pstn but the call is not working unless I added forward-digits 7 anybody can help me to understand why?? ___ 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] thoughts on plan / type / name requirements forani / dnis
Slightly OT: is the US shifting to allowing 10D or 1+10D for all calls or is it still highly telco specific? Is 7D still used? I thought that many cities require 10D even for local calls. Obviously-not-living-in-the-US Anthony On 16 Oct 2011, at 16:38, Brian btmulg...@gmail.com wrote: thanks a lot Kshitij, i am not too familiar with US telco requirement so this is good to know. Sent from my iPad On 15 Oct 2011, at 20:39, Kshitij Singhi martinian.ksin...@gmail.com wrote: Hi Brian, That's quite a common requirement for Telcos. (not so much for 11 digit calls but definitely for the international calls) There are quite a few Service Providers who expect one of the following (pertaining only to international calls) If 011 is prefixed, then they will pass the call only if it is flagged as unknown. If 011 is not being sent across, then they will pass the call only if it is flagged as international. From a Gateway perspective, what we usually do is make use of the following command to set the plan/type on the Serial Interface: interface serial x/y/z:23 isdn map address ^011 plan isdn type unknown (or plan unknown type unknown depending on what the Telco expects) (this obviously pertains to calls where the 011 is being sent across - be careful with the command since it changes the calling/called number plan/type). For calls where the 011 is not sent across, it's safer to use translation profiles on the outgoing dial peers. For MGCP, it's all done on the CUCM and translations are eventually leveraged for sending the correct plan/type/number of digits in SRST. On Sun, Oct 16, 2011 at 12:59 AM, Brian Mulgrew btmulg...@gmail.com wrote: Thanks for the response Kshitij just curious - but what is the thinking behind sending type as unknown when not presenting leading digits in the US? Brian On Sat, Oct 15, 2011 at 8:27 AM, Kshitij Singhi martinian.ksin...@gmail.com wrote: Here is what I had followed: ALWAYS send a plan of ISDN. If it has been stated that the Telco expects a specific plan/type, then ensure that the plan/type is sent in the ani/dnis. Also, for an outgoing call ensure that you take care of the digits seen on the calling phone ONLY if specified in the paper. If it has not been specifically stated, then for called number: 1. Send a plan of ISDN and type of subscriber for 7 digit calls (US) 2. Send a plan of ISDN a type of national for 11 digit calls (US) (assuming the Telco does NOT expect the leading 1) 3. Send a plan of ISDN and type of international for International calls (US) (assuming the Telco does not expect the leading 011) 4. Send a plan of ISDN and type of unknown for 11 digit calls (US) (assuming the Telco expects the leading 1) 5. Send a plan of ISDN and type of unknown for international calls (US) (assuming the Telco expects the leading 011) 6. Send a plan of ISDN and type of unknown for 911 calls. For non-US sites: 1. Send a plan of ISDN and type of subscriber for local calls 2. Send a plan of ISDN and type of international for international calls 3. For emergency numbers, send a plan of ISDN and type of unknown. For the calling number: 1. Send a plan of ISDN and type of subscriber for 7 digit calls. (US) 2. Send a plan of ISDN and type of national for 11 digit calls. (US) 3. Send a plan of ISDN and type of international for international calls. (US) 4. Send a plan of ISDN and type of national for emergency calls For non US sites: 1. Send a plan of ISDN and type of subscriber for emergency calls. 2. Send a plan of ISDN and type of subscriber for local calls 3. Send a plan of ISDN and type of international for international calls. On Sat, Oct 15, 2011 at 7:21 AM, 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. Re: thoughts on plan / type / name requirements forani / dnis (Jason Langenfeld) 2. Re: MOH multicast (Ccie Voice) 3. IP Blue (running multiple instances) (Jeferson Guardia) 4. Re: MOH multicast (Mohammed Al baqari) 5. ipcc express video training (donny f) 6. Re: IP Blue (running multiple instances) (Randall Crumm) -- Message: 1 Date: Fri, 14 Oct 2011 19:33:21 + From: Jason Langenfeld jlangenf...@prosysis.com To: CCIE for Me cciefo...@hotmail.com, OSL
Re: [OSL | CCIE_Voice] SIP phones supp services on RAS GK trunk: revisited (WB2, Lab2)
Hi Vik Within the CME cluster everything functions: I am using the firmware SIP41.8-4-1S. 3001 SCCP / 7961G 3005 SIP / 7691G 3002 SCCP / CIPC 1. CME can transfer between all 3XXX DNs/SIP,SCCP for an active call to CUCM 2. CME SCCP can be holder, be held, and transferred within CUCM 3. CME SIP cannot hold/be held across the trunk; call failure occurs when trying to Resume - no media 4. If CUCM tries to transfer the CME SIP phone, the transfer does not complete and CUCM phone still shows the two DNs on the display. When both CUCM phones have gone on hook CME SIP still shows connected. When 5001 puts 3005 on hold, I get a 501 message: //-1//SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:3005@10.10.202.50:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.10.202.1:5060;branch=z9hG4bK1E326DA Remote-Party-ID: HQ Phone1 sip:5001@10.10.202.1 ;party=calling;screen=yes;privacy=off From: sip:5001@10.10.202.1;tag=B55E4E0-10D4 To: BR2 Phone2 sip:3005@10.10.202.1 ;tag=0017e08967fd002865d6a9e0-a4732b54 Date: Thu, 13 Oct 2011 09:11:48 GMT Call-ID: 0017e089-67fd0014-d37c5e50-189e54e4@10.10.202.50 Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE: 1800 Cisco-Guid: 825456687-4105310688-2245099869-3422373919 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 102 INVITE Max-Forwards: 70 Timestamp: 1318497108 Contact: sip:5001@10.10.202.1:5060 Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 235 v=0 o=CiscoSystemsSIP-GW-UserAgent 2535 2636 IN IP4 10.10.202.1 s=SIP Call c=IN IP4 0.0.0.0 t=0 0 m=audio 19506 RTP/AVP 0 101 c=IN IP4 0.0.0.0 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=inactive //-1//SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.202.1:5060;branch=z9hG4bK1E326DA From: sip:5001@10.10.202.1;tag=B55E4E0-10D4 To: BR2 Phone2 sip:3005@10.10.202.1 ;tag=0017e08967fd002865d6a9e0-a4732b54 Call-ID: 0017e089-67fd0014-d37c5e50-189e54e4@10.10.202.50 Date: Thu, 13 Oct 2011 09:11:46 GMT CSeq: 102 INVITE Server: Cisco-CP7961G/8.4.0 Contact: sip:3005@10.10.202.50:5060;transport=udp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE Remote-Party-ID: BR2 Phone2 sip:3005@10.10.202.1 ;party=calling;id-type=subscriber;privacy=off;screen=yes Allow-Events: dialog Content-Length: 202 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 8179 1 IN IP4 10.10.202.50 s=SIP Call t=0 0 m=audio 22568 RTP/AVP 0 101 c=IN IP4 10.10.202.50 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=inactive //-1//SIP/Msg/ccsipDisplayMsg: Sent: Branch2#ACK sip:3005@10.10.202.50:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.10.202.1:5060;branch=z9hG4bK1E4E65 From: sip:5001@10.10.202.1;tag=B55E4E0-10D4 To: BR2 Phone2 sip:3005@10.10.202.1 ;tag=0017e08967fd002865d6a9e0-a4732b54 Date: Thu, 13 Oct 2011 09:11:48 GMT Call-ID: 0017e089-67fd0014-d37c5e50-189e54e4@10.10.202.50 Max-Forwards: 70 CSeq: 102 ACK Allow-Events: telephone-event Content-Length: 0 //-1//SIP/Msg/ccsipDisplayMsg: Sent: Branch2#SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 10.10.202.50:5060;branch=z9hG4bK62a26070 From: BR2 Phone2 sip:3005@10.10.202.1 ;tag=0017e08967fd002865d6a9e0-a4732b54 To: sip:5001@10.10.202.1;tag=B55E4E0-10D4 Date: Thu, 13 Oct 2011 09:11:48 GMT Call-ID: 0017e089-67fd0014-d37c5e50-189e54e4@10.10.202.50 CSeq: 101 INVITE Allow-Events: telephone-event Reason: Q.850;cause=65 Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 ___ 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 phones supp services on RAS GK trunk: revisited (WB2, Lab2)
For (4) can you confirm the H323 Trunk in UCM has the following settings: inbound FastStart is enabled, Wait for far end H245 TCS is disabled and MTP Required is enabled. Also use g711 end to end (g711-DP/Region) and it should work. Performed the following test: 1. GK Device Pool/Region allows G.711 everywhere. 2. CME to CUCM: forced G.711 codec on the dial-peer to [15]... dial-peer voice 101 voip destination-pattern [15]... session target ras tech-prefix 1# dtmf-relay h245-alphanumeric codec g711ulaw no vad It works: no transcoders on either Br2 or HQ, MTP active on HQ. Even the Unicast MOH stream worked to Br2/SIP. For CUCM to CME, I'm not able to force G.711 yet ; the phone still seem to pick G.729. I've shifted the trunk to the HQ Device pool. On Thu, Oct 13, 2011 at 5:35 PM, Vik Malhi vma...@ipexpert.com wrote: For (3) I still believe this is a firmware issue in the SIP phone. For (4) can you confirm the H323 Trunk in UCM has the following settings: inbound FastStart is enabled, Wait for far end H245 TCS is disabled and MTP Required is enabled. Also use g711 end to end (g711-DP/Region) and it should work. On Thu, Oct 13, 2011 at 10:20 AM, Anthony Alba ascanio.al...@gmail.comwrote: Hi Vik Within the CME cluster everything functions: I am using the firmware SIP41.8-4-1S. 3001 SCCP / 7961G 3005 SIP / 7691G 3002 SCCP / CIPC 1. CME can transfer between all 3XXX DNs/SIP,SCCP for an active call to CUCM 2. CME SCCP can be holder, be held, and transferred within CUCM 3. CME SIP cannot hold/be held across the trunk; call failure occurs when trying to Resume - no media 4. If CUCM tries to transfer the CME SIP phone, the transfer does not complete and CUCM phone still shows the two DNs on the display. When both CUCM phones have gone on hook CME SIP still shows connected. When 5001 puts 3005 on hold, I get a 501 message: //-1//SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:3005@10.10.202.50:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.10.202.1:5060;branch=z9hG4bK1E326DA Remote-Party-ID: HQ Phone1 sip:5001@10.10.202.1 ;party=calling;screen=yes;privacy=off From: sip:5001@10.10.202.1;tag=B55E4E0-10D4 To: BR2 Phone2 sip:3005@10.10.202.1 ;tag=0017e08967fd002865d6a9e0-a4732b54 Date: Thu, 13 Oct 2011 09:11:48 GMT Call-ID: 0017e089-67fd0014-d37c5e50-189e54e4@10.10.202.50 Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE: 1800 Cisco-Guid: 825456687-4105310688-2245099869-3422373919 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 102 INVITE Max-Forwards: 70 Timestamp: 1318497108 Contact: sip:5001@10.10.202.1:5060 Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 235 v=0 o=CiscoSystemsSIP-GW-UserAgent 2535 2636 IN IP4 10.10.202.1 s=SIP Call c=IN IP4 0.0.0.0 t=0 0 m=audio 19506 RTP/AVP 0 101 c=IN IP4 0.0.0.0 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=inactive //-1//SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.202.1:5060;branch=z9hG4bK1E326DA From: sip:5001@10.10.202.1;tag=B55E4E0-10D4 To: BR2 Phone2 sip:3005@10.10.202.1 ;tag=0017e08967fd002865d6a9e0-a4732b54 Call-ID: 0017e089-67fd0014-d37c5e50-189e54e4@10.10.202.50 Date: Thu, 13 Oct 2011 09:11:46 GMT CSeq: 102 INVITE Server: Cisco-CP7961G/8.4.0 Contact: sip:3005@10.10.202.50:5060;transport=udp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE Remote-Party-ID: BR2 Phone2 sip:3005@10.10.202.1 ;party=calling;id-type=subscriber;privacy=off;screen=yes Allow-Events: dialog Content-Length: 202 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 8179 1 IN IP4 10.10.202.50 s=SIP Call t=0 0 m=audio 22568 RTP/AVP 0 101 c=IN IP4 10.10.202.50 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=inactive //-1//SIP/Msg/ccsipDisplayMsg: Sent: Branch2#ACK sip:3005@10.10.202.50:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.10.202.1:5060;branch=z9hG4bK1E4E65 From: sip:5001@10.10.202.1;tag=B55E4E0-10D4 To: BR2 Phone2 sip:3005@10.10.202.1 ;tag=0017e08967fd002865d6a9e0-a4732b54 Date: Thu, 13 Oct 2011 09:11:48 GMT Call-ID: 0017e089-67fd0014-d37c5e50-189e54e4@10.10.202.50 Max-Forwards: 70 CSeq: 102 ACK Allow-Events: telephone-event Content-Length: 0 //-1//SIP/Msg/ccsipDisplayMsg: Sent: Branch2#SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 10.10.202.50:5060;branch=z9hG4bK62a26070 From: BR2 Phone2 sip:3005@10.10.202.1 ;tag=0017e08967fd002865d6a9e0-a4732b54 To: sip:5001@10.10.202.1;tag=B55E4E0-10D4 Date: Thu, 13 Oct 2011 09:11:48 GMT Call-ID: 0017e089-67fd0014-d37c5e50-189e54e4@10.10.202.50 CSeq: 101 INVITE Allow-Events: telephone-event Reason: Q.850;cause=65 Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0
