Re: [OSL | CCIE_Voice] SRST Gatekeeper
I have solved the issue, There was a command - call fallback active - on SRST Router. When I have removed that command, the calls were made succesfuly to CUCME through GK. But what does that command means? On Tue, Nov 15, 2011 at 9:16 AM, Mohd Baqari baqari.voic...@gmail.comwrote: You need to configure dialpeer pointing to ras Regards, Mohammed Al Baqari Sent from my iPhone On Nov 15, 2011, at 12:04 AM, Matthew Saskin m...@saskin.net wrote: What does the rest of your configuration on the SRST router look like? Do you have the requisite dial peers in place to direct calls from SRST registered phones to the gatekeeper? Without that you're not going to get very far...the SRST router has no knowledge of the callmanager gatekeeper configuration. -matthew On Mon, Nov 14, 2011 at 1:18 PM, datucha123 datucha123 datucha...@gmail.com wrote: Hello everyone, I have a problem with SRST to Gatekeeper calls. When the IP Phone are registered with CUCM calls to CUCME through GK works just fine. But as soon as I will shutdown the CUCM Service on all servers, so that the IP Phones will register with SRST Router, the outgoing calls to CUCME through the GK fail. But the inbound call to SRST IP Phone through the GK from CUCME works fine. SRST Config: *interface Loopback0 ip address X.X.X.X 255.255.255.255 h323-gateway voip interface h323-gateway voip id CUCM ipaddr Y.Y.Y.Y 1719 h323-gateway voip h323-id srst h323-gateway voip tech-prefix 1# h323-gateway voip bind srcaddr Z.Z.Z.Z* * * *call-manager-fallback max-conferences 4 gain -6 transfer-system full-consult ip source-address X.X.X.X port 2000 max-ephones 10 max-dn 10 no-reg system message primary SRST moh music-on-hold.au multicast moh 239.5.5.6 port 16384 time-zone 32 time-format 24 date-format dd-mm-yy* GK Config: *gatekeeper zone local CUCME test.com A.A.A.A zone local CUCM test.com zone prefix CUCM 1... gw-priority 10 ccm_2 zone prefix CUCM 1... gw-priority 9 ccm_1 zone prefix CUCM 1... gw-priority 8 srst zone prefix CUCM 2... gw-priority 10 ccm_2 zone prefix CUCM 2... gw-priority 9 ccm_1* * zone prefix CUCM 2... gw-priority 8 srst zone prefix CUCME 3... zone prefix CUCME 5... gw-type-prefix 1#* default-technology gw-type-prefix 3#* gw ipaddr B.B.B.B 1720 gw-type-prefix 2#* arq reject-unknown-prefix no shutdown * Maybe there is a known issue with SRST and Gatekeeper finctionality. Or maybe SRST does not work with GK at all? ___ 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/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/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
[OSL | CCIE_Voice] Automated Reply Re: CCIE_Voice Digest, Vol 69, Issue 66
This is an automated reply to your message CCIE_Voice Digest, Vol 69, Issue 66 sent to stewart.mcfarl...@provista-uk.com. Dear ccie_voice-requ...@onlinestudylist.com Thank you for your email. Please note that due to unforseen circumstances I will be out of the office for the rest of this week (w/c 17/10/11). If you require any assistance please contact the provista office on 08456424642. For technical assistance please contact serviced...@provista-uk.com. Thanks Stewart ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Voice Class Codec Question
Hi Kshitij, Thanks for the great explanation.I have one doubt regarding case1. In case 1 (call coming from PSTN to branch phone) when we have not defined any codec on voip peer then by default g729 codec will be selected and it seems G729 codec finally used to connect to phone irrespective of fact that branch gateway and branch Ip phone are in same region. Show voice call summ on brach router shows that call connected using g729 and MOH to PSTN caller will not work in this case when CUCM is multicasting MOH (no multicating from branch router) as g729 codec is not supported. Same MOH/multicasting works when voip dial-peer codec is forced G711. PSTN GW --G729 --CUCMg729- IP PSTN-- GW-G729---IP Phone The reason for g729 codec used seems to be fact that this codec is used by GW voip dial-peer as default (only option) and now CUCM will negotiate codec with IP phone using sccp keeping in view that one leg is g729 so final selection will be g729. If this is the case transcoder should be invoked to send call to Ip phone. Has someone tested this in lab scenario. Regards, Brajesh. On Mon, Nov 14, 2011 at 6:06 PM, Kshitij Singhi martinian.ksin...@gmail.com wrote: Hi Sonu, So for a call on the Branch site, I can think of a few scenarios: 1. Call coming in for an IP Phone at the branch site from the PSTN. 2. Call being made from a phone at Branch Site 1 to another phone at Branch Site 1. 3. Call being made from a phone at Branch Site X to a phone at Branch Site 1. 4. Call coming in for an IP Phone at the branch site from the PSTN rolling over to Unity. 5. Call coming in via a GK to a phone at Branch Site 1 For 2. and 3. we don't need to worry about the dial peers. This is SCCP signalling within CUCM itself and the codec selected is going to be governed by the Region settings. I am assuming that we have the following regions created (all sites are CUCM sites - if there is a GK involved, there might be a CME site in which case one of the Regions will not be there): Reg-SiteA Reg-SiteB Reg-SiteC Reg-GK Reg-MOH Within the same region, the relationship is G.711. Inter-region relationships are G.729. The only exception to this rule is the MOH region which is G.711 throughout. For 5. the dial-peer with a session target of ras shouldn't have any codec defined on it. That would invoke G.729r8 on such calls. For 1 and 2 we have the dial peer set up as you have described. In such a case, the Destination phone will be in Reg-SiteB and the ingress GW will also be in the same Region. So it doesn't really matter how we specify the voice class codec since this is not a call between sites. For 4, Unity should be in Reg-SiteA and the IP Phone/Ingress GW in Reg-SiteB. Thus, even though G711ulaw will be advertised in the TCS to CUCM, only G.729 will be negotiated due to the Inter-region relationship defined. What we should be looking at are calls from Site A TEHO to a Site B PSTN phone (Or something similar). According to me, this is a call between sites and once again, we needn't worry about the preference of codecs in the voice class command since: Site A IP Phone (Reg-SiteA) will be calling the egress GW at Site B (Reg-SiteB). If the incoming dial peer at Site B has both codecs defined, the GW will send an H.245 TCS to CUCM advertising both codecs. However, CUCM will enforce the region relationship(s) mentioned previously and will thus negotiate only G.729. Note that the preference of the codecs in the voice class codec becomes a matter of concern only when something like this happens: 1. An H.323 endpoint advertises G.711 as first preference and then G.729 2. GW advertises G729 as first preference and then G.711. In this scenario, the MSD will be the tie breaker. In most cases, for a CUCM scenario, CUCM becomes the master and wins, so to speak although I don't know of any way to define a codec preference on CUCM as such. On Mon, Nov 14, 2011 at 12:18 PM, ccie_voice-requ...@onlinestudylist.com wrote: Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: [VM Ware Question] (Adam Thompson) 2. Voice Class Codec Question. (Pradeep Kumar Sharma) 3. Re: can not save script in script repository (=?gbk?B?YnJ1bm8=?=) 4. Re: Number of IP Phones in the lab (Google) 5. uccx Unified CM Telephony Subsystem gray out (=?gbk?B?YnJ1bm8=?=)
Re: [OSL | CCIE_Voice] Voice Class Codec Question
Hi Brajesh, You are right - if we don't have any codec defined then the call will use G.729 irrespective of the fact that the ingress GW and the destination IP Phone are in the same Region. The tree diagram (approximate - not all messages enumerated) for this is given below: PSTN - - - - - - - - - - - - -GW - - - - -- - - - - - - - -CUCM | setup---| proceeding-- | |H..225Setup |---H.225 Proc |---H.225 Alerting alerting| |--- H.225 Connect connect/ | |TCS with ONLY G.729 advertised |--- TCS from CUCM (finally, G.729 negotiated once OLC/MSD and ACKs have gone through) Hence, in this scenario MOH might not work. I remember facing something similar while practicing (i.e. the call connected but MOH did not work) and it was because of the fact that I did not specify the codec under the VoIP dial-peer. For unicast, this was resolved by adding G.729 capabilities for MOH. However, for multicast from the routers' flash, I needed to modify the port used for MOH in call-manager-fallback (assuming we are incrementing on the basis of port and not IP address). It would be best to add a voice class codec with G.711 and G.729 on the outgoing VoIP dial peer of the branch site. Hope this helps. On Wed, Nov 16, 2011 at 6:10 PM, brajesh kumaR brjku...@gmail.com wrote: Hi Kshitij, Thanks for the great explanation.I have one doubt regarding case1. In case 1 (call coming from PSTN to branch phone) when we have not defined any codec on voip peer then by default g729 codec will be selected and it seems G729 codec finally used to connect to phone irrespective of fact that branch gateway and branch Ip phone are in same region. Show voice call summ on brach router shows that call connected using g729 and MOH to PSTN caller will not work in this case when CUCM is multicasting MOH (no multicating from branch router) as g729 codec is not supported. Same MOH/multicasting works when voip dial-peer codec is forced G711. PSTN GW --G729 --CUCMg729- IP PSTN-- GW-G729---IP Phone The reason for g729 codec used seems to be fact that this codec is used by GW voip dial-peer as default (only option) and now CUCM will negotiate codec with IP phone using sccp keeping in view that one leg is g729 so final selection will be g729. If this is the case transcoder should be invoked to send call to Ip phone. Has someone tested this in lab scenario. Regards, Brajesh. On Mon, Nov 14, 2011 at 6:06 PM, Kshitij Singhi martinian.ksin...@gmail.com wrote: Hi Sonu, So for a call on the Branch site, I can think of a few scenarios: 1. Call coming in for an IP Phone at the branch site from the PSTN. 2. Call being made from a phone at Branch Site 1 to another phone at Branch Site 1. 3. Call being made from a phone at Branch Site X to a phone at Branch Site 1. 4. Call coming in for an IP Phone at the branch site from the PSTN rolling over to Unity. 5. Call coming in via a GK to a phone at Branch Site 1 For 2. and 3. we don't need to worry about the dial peers. This is SCCP signalling within CUCM itself and the codec selected is going to be governed by the Region settings. I am assuming that we have the following regions created (all sites are CUCM sites - if there is a GK involved, there might be a CME site in which case one of the Regions will not be there): Reg-SiteA Reg-SiteB Reg-SiteC Reg-GK Reg-MOH Within the same region, the relationship is G.711. Inter-region relationships are G.729. The only exception to this rule is the MOH region which is G.711 throughout. For 5. the dial-peer with a session target of ras shouldn't have any codec defined on it. That would invoke G.729r8 on such calls. For 1 and 2 we have the dial peer set up as you have described. In such a case, the Destination phone will be in Reg-SiteB and the ingress GW will also be in the same Region. So it doesn't really matter how we specify the voice class codec since this is not a call between sites. For 4, Unity should be in Reg-SiteA and the IP Phone/Ingress GW in Reg-SiteB. Thus, even though G711ulaw will be advertised in the TCS to CUCM, only G.729 will be negotiated due to the Inter-region relationship defined. What we should be looking at are calls from Site A TEHO to a Site B PSTN phone (Or something similar). According to me, this is a
Re: [OSL | CCIE_Voice] CCM Integration with Microsoft OCS for Presence of the Phones without CUPS
Hi, Can any one help me on this. thanks and regards Kaja Hanumantharao On Tue, Nov 15, 2011 at 5:12 PM, Kaja Hanumantharao vasuhan...@gmail.comwrote: Hi, We have Cisco Call Manager 7.1.3 and want to integrate with Microsoft OCS for presence. If Cisco Phone is on call, We need to have the status on OCS as busy. I heard that we can do by CUIMOC. Please let me know the procedure to that or any other procedure to get the same without CUPS. We already have SIP connectivity to Microsoft OCS for Voice dialing from Cisco to OCS. http://www.cisco.com/en/US/docs/voice_ip_comm/cucimoc/7_1/english/integrat/guide/config_servers.html#wp1053151 Thanks and Regards Kaja Hanumantharao -- Kaja Uma Satya Hanumantharao Hexaware Technologies Limited Phone: +919677167332 ___ 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] ICT vs H225 GK controlled Trunk
Hello, Can anybody please explain the difference between the Intercluster Trunk (Gatekepeer-Controlled) and H225 Trunk (Gatekepeer-Controlled). And which one must be used on the LAB exam for connecting CUCM to Gatekepeer? ___ 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] Fwd: B2BUA mailbox
-- Forwarded message -- From: datucha123 datucha123 datucha...@gmail.com Date: Mon, Jul 11, 2011 at 12:45 AM Subject: B2BUA mailbox To: ccie_voice@onlinestudylist.com Hello, everyone Can anybody please tell me how does the b2bua mailbox command works for SIP IP Phones registered to CUCME? I have searched for that command, but unfortunately I cannot get the idea. What's the logic for that command? ___ 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] ICT vs H225 GK controlled Trunk
From CCM4.x SRND •H.225 trunk (gatekeeper controlled) Always use this configuration to communicate with any combination Cisco Unified CallManager (Release 3.2 or later) or other H.323 endpoints that are all controlled by a common set of gatekeeper(s). Cisco Unified CallManager will use the H.225-based Intercluster Trunk Protocol between clusters and use standard H.225 to other H.323 endpoints. The autodetection feature will invoke the appropriate behavior. Do not use this trunk for connection to any version of Cisco CallManager prior to Release 3.2 because earlier versions will not send the UUIE and the connection will use the standard H.225 protocol. •Intercluster trunk (gatekeeper controlled) This trunk must be used only for communication to Cisco Unified CallManagers. The trunk always utilizes the Intercluster Trunk Protocol, which will not be understood by a standard H.323 endpoint. Configure this type of trunk only when interoperability is required with versions of Cisco CallManager prior to Release 3.2. On Wed, Nov 16, 2011 at 12:10 PM, datucha123 datucha123 datucha...@gmail.com wrote: Hello, Can anybody please explain the difference between the Intercluster Trunk (Gatekepeer-Controlled) and H225 Trunk (Gatekepeer-Controlled). And which one must be used on the LAB exam for connecting CUCM to Gatekepeer? ___ 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] ICT vs H225 GK controlled Trunk
Also... Intercluster Trunk (Gatekeeper Controlled) In a distributed call-processing network with gatekeepers, use an intercluster trunk with gatekeeper control to configure connections between clusters of Cisco CallManager systems. Gatekeepers provide call admission control and address resolution for intercluster calls. A single intercluster trunk can communicate with all remote clusters. On Wed, Nov 16, 2011 at 12:55 PM, Abel ... midga...@gmail.com wrote: From CCM4.x SRND •H.225 trunk (gatekeeper controlled) Always use this configuration to communicate with any combination Cisco Unified CallManager (Release 3.2 or later) or other H.323 endpoints that are all controlled by a common set of gatekeeper(s). Cisco Unified CallManager will use the H.225-based Intercluster Trunk Protocol between clusters and use standard H.225 to other H.323 endpoints. The autodetection feature will invoke the appropriate behavior. Do not use this trunk for connection to any version of Cisco CallManager prior to Release 3.2 because earlier versions will not send the UUIE and the connection will use the standard H.225 protocol. •Intercluster trunk (gatekeeper controlled) This trunk must be used only for communication to Cisco Unified CallManagers. The trunk always utilizes the Intercluster Trunk Protocol, which will not be understood by a standard H.323 endpoint. Configure this type of trunk only when interoperability is required with versions of Cisco CallManager prior to Release 3.2. On Wed, Nov 16, 2011 at 12:10 PM, datucha123 datucha123 datucha...@gmail.com wrote: Hello, Can anybody please explain the difference between the Intercluster Trunk (Gatekepeer-Controlled) and H225 Trunk (Gatekepeer-Controlled). And which one must be used on the LAB exam for connecting CUCM to Gatekepeer? ___ 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] Query about Online Rack Rental
Hi All Can someone help me understand how the online Rack Rental procedure Any videos or documentation would help Trail / Demo lab access would be much appreciated. I am trying to practice some labs by taking PODS on rental In which I want to make some Local, onnet and offnet calls using ANALOG PHONES via POTS connections on the router Want to know how a call can be made with an analog phone online??? Please help Cheers Dev ___ 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] ICT vs H225 GK controlled Trunk
Great answer, thank you very much. So as I understand, for CUCM 7.x, I have to use H225 Trunk always, even if connecting to other CUCM cluster through GK, or connecting to CUCME through GK, am I right? On Wed, Nov 16, 2011 at 8:58 PM, Abel ... midga...@gmail.com wrote: Also... Intercluster Trunk (Gatekeeper Controlled) In a distributed call-processing network with gatekeepers, use an intercluster trunk with gatekeeper control to configure connections between clusters of Cisco CallManager systems. Gatekeepers provide call admission control and address resolution for intercluster calls. A single intercluster trunk can communicate with all remote clusters. On Wed, Nov 16, 2011 at 12:55 PM, Abel ... midga...@gmail.com wrote: From CCM4.x SRND •H.225 trunk (gatekeeper controlled) Always use this configuration to communicate with any combination Cisco Unified CallManager (Release 3.2 or later) or other H.323 endpoints that are all controlled by a common set of gatekeeper(s). Cisco Unified CallManager will use the H.225-based Intercluster Trunk Protocol between clusters and use standard H.225 to other H.323 endpoints. The autodetection feature will invoke the appropriate behavior. Do not use this trunk for connection to any version of Cisco CallManager prior to Release 3.2 because earlier versions will not send the UUIE and the connection will use the standard H.225 protocol. •Intercluster trunk (gatekeeper controlled) This trunk must be used only for communication to Cisco Unified CallManagers. The trunk always utilizes the Intercluster Trunk Protocol, which will not be understood by a standard H.323 endpoint. Configure this type of trunk only when interoperability is required with versions of Cisco CallManager prior to Release 3.2. On Wed, Nov 16, 2011 at 12:10 PM, datucha123 datucha123 datucha...@gmail.com wrote: Hello, Can anybody please explain the difference between the Intercluster Trunk (Gatekepeer-Controlled) and H225 Trunk (Gatekepeer-Controlled). And which one must be used on the LAB exam for connecting CUCM to Gatekepeer? ___ 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] Query about Online Rack Rental
As I know, there are not Analog Phones any more on the LAB. So why are you wondering for Analog Phones? On Wed, Nov 16, 2011 at 10:11 PM, Devakanth Gangavarapu devakanth2...@gmail.com wrote: Hi All Can someone help me understand how the online Rack Rental procedure Any videos or documentation would help Trail / Demo lab access would be much appreciated. I am trying to practice some labs by taking PODS on rental In which I want to make some Local, onnet and offnet calls using ANALOG PHONES via POTS connections on the router Want to know how a call can be made with an analog phone online??? Please help Cheers Dev ___ 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] Fwd: B2BUA mailbox
If other forwarding fails , it transfers to mailbox On Wed, Nov 16, 2011 at 9:55 PM, datucha123 datucha123 datucha...@gmail.com wrote: -- Forwarded message -- From: datucha123 datucha123 datucha...@gmail.com Date: Mon, Jul 11, 2011 at 12:45 AM Subject: B2BUA mailbox To: ccie_voice@onlinestudylist.com Hello, everyone Can anybody please tell me how does the b2bua mailbox command works for SIP IP Phones registered to CUCME? I have searched for that command, but unfortunately I cannot get the idea. What's the logic for that command? ___ 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
