Re: [OSL | CCIE_Voice] HQ and Branch1 phones cannot call CUE.
Greetings Vikas, Can you call 4101 from either HQ or SB or do you get a rapid busy or if you call from HQ to SB to 4101 and no other calls up does the call reroute over the PSTN. If you get rapid busy or if your call immediately reroutes over the PSTN that isn't right. You could have a locations issue from what you explain I'm a little confused. Also it appear that you haven't put into place a rsvp bandwidth statement which is required to perform rsvp calls. On HQ and SC need the following statements: > interface Serial0/1/0:0.102 point-to-point >ip rsvp bandwidth 112 >(to allow 4 calls) Michael Sears Compucom Systems Western Region Senior Infrastructure Solution Consulting Office: +1.720.344.6833 Mobile: +1.303.328.5590 Fax: +1.978.863.0740 -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Friday, March 22, 2013 7:34 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 85, Issue 78 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: HQ and Branch1 phones cannot call CUE (Vikky Kumar) 2. Re: SRST to voicemail without Alternate Extension (Leslie Meade) -- Message: 1 Date: Fri, 22 Mar 2013 16:17:09 +0300 From: Vikky Kumar To: "Sears, Michael (msears)" Cc: "ccie_voice@onlinestudylist.com" Subject: Re: [OSL | CCIE_Voice] HQ and Branch1 phones cannot call CUE Message-ID: Content-Type: text/plain; charset="iso-8859-1" Hi Williams, Thanks for your email. Earlier message on the calling hq-phones was not enough bandwidth when i tried to place the call to 4220, after reload whole lab now its "Ring out" and fast busy. When i dial 4 digit extension of BR2 phone from HQ or BR1 phone call goes smooth no prob upto 4 calls after that , not enough bandwidth, Rerouting on display of phone and call goes via PSTN network I think this is expected ok This happens vice versa, sc to hq. I configured sdspfarm transcode session 4 under telephony-service. my configuration is CME-SRST My problem is HQ and BR1 Phones can't call CUE VM Pilot . as mentioned above I am using RSVP, its configures exactly how you have explained & working ok. Automated Alternate Routing in Service Parameters = TRUE. HQ and BR2 you need to configure MTP resources = Done configure Location RSVP setting to mandatory between HQ -->BR2 = Done I have not configured ip precedence under sccp, does that matters, I have added now but situation remain same. i dont have serial uplink its ethernet only, but works as per my testing 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 ! voice-card 0 dspfarm dsp services dspfarm ! interface Loopback0 ip address 172.16.30.254 255.255.255.0 ip pim dense-mode ip ospf network point-to-point ! interface FastEthernet0/0 ip address 192.168.1.4 255.255.255.0 duplex auto speed auto ip rsvp bandwidth 112 ! interface Service-Engine0/0 ip unnumbered Loopback0 service-module ip address 172.16.30.253 255.255.255.0 service-module ip default-gateway 172.16.30.254 ! interface Vlan31 ip address 172.22.30.254 255.255.255.0 ip pim dense-mode ! interface Vlan32 ip address 172.32.30.254 255.255.255.0 ! ! sccp local Loopback0 sccp ccm 172.20.10.12 identifier 1 priority 1 version 7.0 sccp ccm 172.20.10.11 identifier 2 priority 2 version 7.0 sccp ccm 172.22.30.254 identifier 3 priority 3 version 7.0 sccp ! sccp ccm group 1 bind interface Loopback0 associate ccm 1 priority 1 associate ccm 2 priority 2 associate ccm 3 priority 3 associate profile 3 register sc-conf associate profile 1 register sc-mtp associate profile 2 register sc-xcode ! dspfarm profile 2 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 4 associate application SCCP ! dspfarm profile 1 mtp codec g729r8 codec pass-through rsvp maximum sessions software 8 associate application SCCP ! Regards, Vikky On Thu, Mar 21, 2013 at 7:47 AM, Sears, Michael (msears) < michael.se...@compucom.com> wrote: > Greetings Vikas, > > First, do you get any message on the calling phone, like not enough > bandwidth when
Re: [OSL | CCIE_Voice] HQ and Branch1 phones cannot call CUE
Hi Experts, I got my problem solved buy reloading the br2-config , publisher and subscriber. can you please advise me how to tackle the requirement of reload in CCIE Lab exam this is very important for me because my Presence also donot work unless i restart my whole lab please advice... Regards, Vikky On Fri, Mar 22, 2013 at 4:17 PM, Vikky Kumar wrote: > Hi Williams, > Thanks for your email. > > Earlier message on the calling hq-phones was not enough bandwidth when i > tried to place the call to 4220, after reload whole lab now its "Ring out" > and fast busy. When i dial 4 digit extension of BR2 phone from HQ or BR1 > phone call goes smooth no prob upto 4 calls after that , not enough > bandwidth, Rerouting on display of phone and call goes via PSTN network I > think this is expected ok This happens vice versa, sc to hq. > > I configured sdspfarm transcode session 4 under telephony-service. my > configuration is CME-SRST > > My problem is HQ and BR1 Phones can't call CUE VM Pilot . > > as mentioned above I am using RSVP, its configures exactly how you have > explained & working ok. > > > Automated Alternate Routing in Service Parameters = TRUE. > HQ and BR2 you need to configure MTP resources = Done > configure Location RSVP setting to mandatory between HQ -->BR2 = Done > > I have not configured ip precedence under sccp, does that matters, I have > added now but situation remain same. > > i dont have serial uplink its ethernet only, but works as per my testing > > 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 > ! > voice-card 0 > dspfarm > dsp services dspfarm > ! > interface Loopback0 > ip address 172.16.30.254 255.255.255.0 > ip pim dense-mode > ip ospf network point-to-point > ! > interface FastEthernet0/0 > ip address 192.168.1.4 255.255.255.0 > duplex auto > speed auto > ip rsvp bandwidth 112 > ! > interface Service-Engine0/0 > ip unnumbered Loopback0 > service-module ip address 172.16.30.253 255.255.255.0 > service-module ip default-gateway 172.16.30.254 > ! > interface Vlan31 > ip address 172.22.30.254 255.255.255.0 > ip pim dense-mode > ! > interface Vlan32 > ip address 172.32.30.254 255.255.255.0 > > ! > ! > sccp local Loopback0 > sccp ccm 172.20.10.12 identifier 1 priority 1 version 7.0 > sccp ccm 172.20.10.11 identifier 2 priority 2 version 7.0 > sccp ccm 172.22.30.254 identifier 3 priority 3 version 7.0 > sccp > ! > sccp ccm group 1 > bind interface Loopback0 > associate ccm 1 priority 1 > associate ccm 2 priority 2 > associate ccm 3 priority 3 > associate profile 3 register sc-conf > > associate profile 1 register sc-mtp > associate profile 2 register sc-xcode > ! > dspfarm profile 2 transcode > codec g711ulaw > codec g711alaw > codec g729ar8 > codec g729abr8 > codec g729r8 > maximum sessions 4 > associate application SCCP > ! > dspfarm profile 1 mtp > codec g729r8 > codec pass-through > rsvp > maximum sessions software 8 > associate application SCCP > ! > > > Regards, > > Vikky > > > > On Thu, Mar 21, 2013 at 7:47 AM, Sears, Michael (msears) < > michael.se...@compucom.com> wrote: > >> Greetings Vikas, >> >> First, do you get any message on the calling phone, like not enough >> bandwidth when you try and place the call. What happens when you try and >> dial 4 digit extension of BR2 phone from HQ or BR1 phone, Rapid busy, dead >> air, not enough bandwidth on display of phone, Rerouting? >> >> First this could just be a simple case of not configuring sdspfarm >> transcode session 10 under telephony-service. It all depends on your >> configuration and if SRST is involved. >> >> I just read in your header HQ and BR1 Phones can't call CUE VM Pilot so >> the following may not help and your problem is local on BR2 router. How >> are you trying to trigger CAC? >> >> Are you in fact trying to configure RSVP based CAC or plain simple >> locations based CAC? It is unclear what it is your trying to accomplish. >> If your trying to perform RSVP Based CAC how many calls do you want to >> permit. Let's say for example you want to permit 4 calls then reroute >> across the PSTN using AAR. >> >> In this case you would need to turn on Automated Alternate Routing in >> Service Parameters. Then on HQ and BR2 you need to configure MTP >> resources. In addition you need to configure Location RSVP setting to >> mandatory between HQ -->BR2. >> >> You also need to configure on HQ and BR2: >> >> interface Serial0/1/0:0.102 point-to-point >> ip rsvp bandwidth 112 (to allow 4 calls) >> ! >> sccp local Loopback0 >> sccp ccm [ip address] identifier 1 priority 1 version 6.0 >> sccp ccm [ip address] identifier 2 priority 2 version 6.0 >> sccp ccm [ip address] identifier 3 priority 3 version 6.0 >> sccp ip precedence 3 >> sccp >> ! >> sccp ccm group 1 >> description sccp ccm group 1 >> bind interface Loopback0 >> associate ccm 1 priority 1
Re: [OSL | CCIE_Voice] HQ and Branch1 phones cannot call CUE
Hi Williams, Thanks for your email. Earlier message on the calling hq-phones was not enough bandwidth when i tried to place the call to 4220, after reload whole lab now its "Ring out" and fast busy. When i dial 4 digit extension of BR2 phone from HQ or BR1 phone call goes smooth no prob upto 4 calls after that , not enough bandwidth, Rerouting on display of phone and call goes via PSTN network I think this is expected ok This happens vice versa, sc to hq. I configured sdspfarm transcode session 4 under telephony-service. my configuration is CME-SRST My problem is HQ and BR1 Phones can't call CUE VM Pilot . as mentioned above I am using RSVP, its configures exactly how you have explained & working ok. Automated Alternate Routing in Service Parameters = TRUE. HQ and BR2 you need to configure MTP resources = Done configure Location RSVP setting to mandatory between HQ -->BR2 = Done I have not configured ip precedence under sccp, does that matters, I have added now but situation remain same. i dont have serial uplink its ethernet only, but works as per my testing 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 ! voice-card 0 dspfarm dsp services dspfarm ! interface Loopback0 ip address 172.16.30.254 255.255.255.0 ip pim dense-mode ip ospf network point-to-point ! interface FastEthernet0/0 ip address 192.168.1.4 255.255.255.0 duplex auto speed auto ip rsvp bandwidth 112 ! interface Service-Engine0/0 ip unnumbered Loopback0 service-module ip address 172.16.30.253 255.255.255.0 service-module ip default-gateway 172.16.30.254 ! interface Vlan31 ip address 172.22.30.254 255.255.255.0 ip pim dense-mode ! interface Vlan32 ip address 172.32.30.254 255.255.255.0 ! ! sccp local Loopback0 sccp ccm 172.20.10.12 identifier 1 priority 1 version 7.0 sccp ccm 172.20.10.11 identifier 2 priority 2 version 7.0 sccp ccm 172.22.30.254 identifier 3 priority 3 version 7.0 sccp ! sccp ccm group 1 bind interface Loopback0 associate ccm 1 priority 1 associate ccm 2 priority 2 associate ccm 3 priority 3 associate profile 3 register sc-conf associate profile 1 register sc-mtp associate profile 2 register sc-xcode ! dspfarm profile 2 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 4 associate application SCCP ! dspfarm profile 1 mtp codec g729r8 codec pass-through rsvp maximum sessions software 8 associate application SCCP ! Regards, Vikky On Thu, Mar 21, 2013 at 7:47 AM, Sears, Michael (msears) < michael.se...@compucom.com> wrote: > Greetings Vikas, > > First, do you get any message on the calling phone, like not enough > bandwidth when you try and place the call. What happens when you try and > dial 4 digit extension of BR2 phone from HQ or BR1 phone, Rapid busy, dead > air, not enough bandwidth on display of phone, Rerouting? > > First this could just be a simple case of not configuring sdspfarm > transcode session 10 under telephony-service. It all depends on your > configuration and if SRST is involved. > > I just read in your header HQ and BR1 Phones can't call CUE VM Pilot so > the following may not help and your problem is local on BR2 router. How > are you trying to trigger CAC? > > Are you in fact trying to configure RSVP based CAC or plain simple > locations based CAC? It is unclear what it is your trying to accomplish. > If your trying to perform RSVP Based CAC how many calls do you want to > permit. Let's say for example you want to permit 4 calls then reroute > across the PSTN using AAR. > > In this case you would need to turn on Automated Alternate Routing in > Service Parameters. Then on HQ and BR2 you need to configure MTP > resources. In addition you need to configure Location RSVP setting to > mandatory between HQ -->BR2. > > You also need to configure on HQ and BR2: > > interface Serial0/1/0:0.102 point-to-point > ip rsvp bandwidth 112 (to allow 4 calls) > ! > sccp local Loopback0 > sccp ccm [ip address] identifier 1 priority 1 version 6.0 > sccp ccm [ip address] identifier 2 priority 2 version 6.0 > sccp ccm [ip address] identifier 3 priority 3 version 6.0 > sccp ip precedence 3 > sccp > ! > sccp ccm group 1 > description sccp ccm group 1 > bind interface Loopback0 > associate ccm 1 priority 1 > associate ccm 2 priority 2 > associate ccm 3 priority 3 > associate profile 1 register sc-mtp > associate profile 2 register sc-xcode > associate profile 3 register sc-conf > registration timeout 3 > registration retri 3 > keepalive timeout 3 > keepalive retri 3 > switchback met imm > switchback interval 15 > switchover met imm > ! > dspfarm profile 1 mtp > description dspfarm profile 1 mtp > codec pass-through (I always use codec pass-through some do not it seem > to work both ways) > codec g729r8 > rsvp > maximum sessions software 10 > associate application SCCP > ! > dspfar
Re: [OSL | CCIE_Voice] HQ and Branch1 phones cannot call CUE
Greetings Vikas, First, do you get any message on the calling phone, like not enough bandwidth when you try and place the call. What happens when you try and dial 4 digit extension of BR2 phone from HQ or BR1 phone, Rapid busy, dead air, not enough bandwidth on display of phone, Rerouting? First this could just be a simple case of not configuring sdspfarm transcode session 10 under telephony-service. It all depends on your configuration and if SRST is involved. I just read in your header HQ and BR1 Phones can't call CUE VM Pilot so the following may not help and your problem is local on BR2 router. How are you trying to trigger CAC? Are you in fact trying to configure RSVP based CAC or plain simple locations based CAC? It is unclear what it is your trying to accomplish. If your trying to perform RSVP Based CAC how many calls do you want to permit. Let's say for example you want to permit 4 calls then reroute across the PSTN using AAR. In this case you would need to turn on Automated Alternate Routing in Service Parameters. Then on HQ and BR2 you need to configure MTP resources. In addition you need to configure Location RSVP setting to mandatory between HQ -->BR2. You also need to configure on HQ and BR2: interface Serial0/1/0:0.102 point-to-point ip rsvp bandwidth 112 (to allow 4 calls) ! sccp local Loopback0 sccp ccm [ip address] identifier 1 priority 1 version 6.0 sccp ccm [ip address] identifier 2 priority 2 version 6.0 sccp ccm [ip address] identifier 3 priority 3 version 6.0 sccp ip precedence 3 sccp ! sccp ccm group 1 description sccp ccm group 1 bind interface Loopback0 associate ccm 1 priority 1 associate ccm 2 priority 2 associate ccm 3 priority 3 associate profile 1 register sc-mtp associate profile 2 register sc-xcode associate profile 3 register sc-conf registration timeout 3 registration retri 3 keepalive timeout 3 keepalive retri 3 switchback met imm switchback interval 15 switchover met imm ! dspfarm profile 1 mtp description dspfarm profile 1 mtp codec pass-through (I always use codec pass-through some do not it seem to work both ways) codec g729r8 rsvp maximum sessions software 10 associate application SCCP ! dspfarm profile 2 transcode description dspfarm profile 2 transcode ! dspfarm profile 3 conference description dspfarm profile 2 conference ! To reroute the calls once the upper limit of four is reached you will need to do the following: Create aar group Create pt-aar Create css-aar containing pt-aar Create two route lists one for HQ and one for BR2 Create two route patterns using the appropriate partitions and apply correct route list Calls will by default use the External Phone Number Mask to reroute the call by matching the route pattern, which is assigned to appropriate route list directing the call out the appropriate gateway. You might want to paste in your configuration so that everyone can have a look :) Michael Sears CCIE 38404 ___ 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] HQ and Branch1 phones cannot call CUE
I advise against using codec pass through on the MTP. I'd recommend something like the following: dspfarm profile 2 mtp no codec g711u codec g729r8 max sess softw assoc app sccp no shut dspfarm profile 3 transcod codec g729r8 max sess assoc app sccp no shut ! ccm group 1 ..stuff.. assoc prof 2 register sc-rsvp assoc prof 3 register sc-xocder ..stuff.. ! -Bill -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 20, 2013, at 7:20 PM, Vikky Kumar wrote: > Willam, > BR2 is CUCM site, and there is integration b/w CUE - CUCM > > I have configured RSVP, rsvp bandwidth = 136 kbps on both sides > > When i call HQ phone to BR2-CUE it gives fast busy tone and give "Ring out" > display on HQ Phones > > > pt. b. transcoder in a MRG and MRGL that is assigned to CUE CTI devices (use > Device Pool) ... Done Already > pt. c. i want codec g729 between sites, hence under MTP i selected only > codec g729r8 + codec pass thru > > > ?? still prob.. > > Regards, > > Vikas > > > > > On Thu, Mar 21, 2013 at 12:53 AM, William Bell wrote: > Vikky, > > Please clarify. You say you have configured Branch 2 as CME and CUE. Then you > say CUE is registered to CUCM. That doesn't jive for me. Perhaps I am > mis-reading or mis-interpreting that part of the question. > > You also say you are doing CAC between HQ and Branch 2. Is this locations > based CAC or RSVP? > > Finally, when you say you can't call CUE. What does that mean? Do you get a > fast busy? Annunciator? Does it ring and fail? > > > Others have touched on the key points and the natural inclination is to look > at CODEC since Branch 2 phones <--> CUE work fine. > > If Branch 2 is a CUCM site then you have to: > > a. Create transcoder at Branch 2. Looks like you have done this > > b. Make sure you have that transcoder in a MRG and MRGL that is assigned to > CUE CTI devices (use Device Pool) > > c. If you are using RSVP. Make sure you provision the same codec under the > software MTP resource as you expect to have on the WAN and that matches one > of the codecs supported by the transcoder. > > > The "allow connections" under voice service voip shouldn't come into play in > a CUE-CUCM integration because the CUE-CUCM integration isn't SIP nor is it > H323. It is TAPI. > > > > > > > > -- > William Bell > blog: http://ucguerrilla.com > twitter: @ucguerrilla > > > > On Mar 20, 2013, at 2:16 PM, Vikky Kumar wrote: > >> Hi Experts, >> >> I configured branch 2 CME/CUE working normal for Voice mails. >> >> CUE is registered with CUCM but I call not call CUE(6220) from HQ and >> Branch1 but branch2 phones 4001 and 4002 working normal/ringing from every >> where. >> >> FYI. I have also configured CAC between on Br2 site - HQ site >> >> Please hel. >> >> Regards >> >> Vikky >> >> ___ >> 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] HQ and Branch1 phones cannot call CUE
Willam, BR2 is CUCM site, and there is integration b/w CUE - CUCM I have configured RSVP, rsvp bandwidth = 136 kbps on both sides When i call HQ phone to BR2-CUE it gives fast busy tone and give "Ring out" display on HQ Phones pt. b. transcoder in a MRG and MRGL that is assigned to CUE CTI devices (use Device Pool) ... Done Already pt. c. i want codec g729 between sites, hence under MTP i selected only codec g729r8 + codec pass thru ?? still prob.. Regards, Vikas On Thu, Mar 21, 2013 at 12:53 AM, William Bell wrote: > Vikky, > > Please clarify. You say you have configured Branch 2 as CME and CUE. Then > you say CUE is registered to CUCM. That doesn't jive for me. Perhaps I am > mis-reading or mis-interpreting that part of the question. > > You also say you are doing CAC between HQ and Branch 2. Is this locations > based CAC or RSVP? > > Finally, when you say you can't call CUE. What does that mean? Do you get > a fast busy? Annunciator? Does it ring and fail? > > > Others have touched on the key points and the natural inclination is to > look at CODEC since Branch 2 phones <--> CUE work fine. > > If Branch 2 is a CUCM site then you have to: > > a. Create transcoder at Branch 2. Looks like you have done this > > b. Make sure you have that transcoder in a MRG and MRGL that is assigned > to CUE CTI devices (use Device Pool) > > c. If you are using RSVP. Make sure you provision the same codec under the > software MTP resource as you expect to have on the WAN and that matches one > of the codecs supported by the transcoder. > > > The "allow connections" under voice service voip shouldn't come into play > in a CUE-CUCM integration because the CUE-CUCM integration isn't SIP nor is > it H323. It is TAPI. > > > > > > > > -- > William Bell > blog: http://ucguerrilla.com > twitter: @ucguerrilla > > > > On Mar 20, 2013, at 2:16 PM, Vikky Kumar wrote: > > Hi Experts, > > I configured branch 2 CME/CUE working normal for Voice mails. > > CUE is registered with CUCM but I call not call CUE(6220) from HQ and > Branch1 but branch2 phones 4001 and 4002 working normal/ringing from every > where. > > FYI. I have also configured CAC between on Br2 site - HQ site > > Please hel. > > Regards > > Vikky > > ___ > 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] HQ and Branch1 phones cannot call CUE
Vikky, Please clarify. You say you have configured Branch 2 as CME and CUE. Then you say CUE is registered to CUCM. That doesn't jive for me. Perhaps I am mis-reading or mis-interpreting that part of the question. You also say you are doing CAC between HQ and Branch 2. Is this locations based CAC or RSVP? Finally, when you say you can't call CUE. What does that mean? Do you get a fast busy? Annunciator? Does it ring and fail? Others have touched on the key points and the natural inclination is to look at CODEC since Branch 2 phones <--> CUE work fine. If Branch 2 is a CUCM site then you have to: a. Create transcoder at Branch 2. Looks like you have done this b. Make sure you have that transcoder in a MRG and MRGL that is assigned to CUE CTI devices (use Device Pool) c. If you are using RSVP. Make sure you provision the same codec under the software MTP resource as you expect to have on the WAN and that matches one of the codecs supported by the transcoder. The "allow connections" under voice service voip shouldn't come into play in a CUE-CUCM integration because the CUE-CUCM integration isn't SIP nor is it H323. It is TAPI. -- William Bell blog: http://ucguerrilla.com twitter: @ucguerrilla On Mar 20, 2013, at 2:16 PM, Vikky Kumar wrote: > Hi Experts, > > I configured branch 2 CME/CUE working normal for Voice mails. > > CUE is registered with CUCM but I call not call CUE(6220) from HQ and > Branch1 but branch2 phones 4001 and 4002 working normal/ringing from every > where. > > FYI. I have also configured CAC between on Br2 site - HQ site > > Please hel. > > Regards > > Vikky > > ___ > 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] HQ and Branch1 phones cannot call CUE
Ram, I added g729r8 but with no luck. Regards, On Wed, Mar 20, 2013 at 10:51 PM, Ramcharan Arya wrote: > Hi, > > You need to add codec g729r8 in your transcoder profile. > > > Codec Configuration > Codec : g711ulaw, Maximum Packetization Period : 30 > Codec : g711alaw, Maximum Packetization Period : 30 > Codec : g729ar8, Maximum Packetization Period : 60 > Codec : g729abr8, Maximum Packetization Period : 60 > > > Thanks, > Ramcharan Arya > > > > On Wed, Mar 20, 2013 at 2:46 PM, Vikky Kumar wrote: > >> Hi Experts, >> >> I have configured : >> >> 1. transcoder on CUCM for HQ & BR2 (Cisco IOS Enhanced Media Termination >> Point ) >> and also Router of BR2 (dspfarm profile 2 transcode , max sess 4 .. ) >> >> 2. above hq-xcode,, sc-xcode is registered > included in respective > >> mrg > mrgl > DP >> >> 3. voice service voip , allow connection h t h,allow connection h t s, >> allow connection s t h >> >> When call from HQ i can notice randomly (6221, 6222, 6223) , very very >> small ring/tone, Display "Ring out" with busy tone >> >> Please advise... >> >> [ >> Router BR2 >sh dspfarm all > >> >> Dspfarm Profile Configuration >> >> Profile ID = 2, Service = TRANSCODING, Resource ID = 1 >> Profile Description : >> Profile Service Mode : Non Secure >> Profile Admin State : UP >> Profile Operation State : ACTIVE >> Application : SCCP Status : ASSOCIATED >> Resource Provider : FLEX_DSPRM Status : UP >> Number of Resource Configured : 4 >> Number of Resource Available : 4 >> Codec Configuration >> Codec : g711ulaw, Maximum Packetization Period : 30 >> Codec : g711alaw, Maximum Packetization Period : 30 >> Codec : g729ar8, Maximum Packetization Period : 60 >> Codec : g729abr8, Maximum Packetization Period : 60 >> >> ] >> >> >> Regards, >> >> Vikky >> >> >> >> On Wed, Mar 20, 2013 at 9:54 PM, Ramcharan Arya > > wrote: >> >>> Hi, >>> >>> You have to add on your CME side. >>> >>> voice service voip >>> allow connection h t h >>> allow connection h t s >>> allow connection s t h >>> >>> Thanks, >>> Ramcharan Arya >>> >>> >>> >>> >>> >>> On Wed, Mar 20, 2013 at 1:16 PM, Vikky Kumar wrote: >>> Hi Experts, I configured branch 2 CME/CUE working normal for Voice mails. CUE is registered with CUCM but I call not call CUE(6220) from HQ and Branch1 but branch2 phones 4001 and 4002 working normal/ringing from every where. FYI. I have also configured CAC between on Br2 site - HQ site Please hel. Regards Vikky ___ 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] HQ and Branch1 phones cannot call CUE
Hi, You need to add codec g729r8 in your transcoder profile. Codec Configuration Codec : g711ulaw, Maximum Packetization Period : 30 Codec : g711alaw, Maximum Packetization Period : 30 Codec : g729ar8, Maximum Packetization Period : 60 Codec : g729abr8, Maximum Packetization Period : 60 Thanks, Ramcharan Arya On Wed, Mar 20, 2013 at 2:46 PM, Vikky Kumar wrote: > Hi Experts, > > I have configured : > > 1. transcoder on CUCM for HQ & BR2 (Cisco IOS Enhanced Media Termination > Point ) > and also Router of BR2 (dspfarm profile 2 transcode , max sess 4 .. ) > > 2. above hq-xcode,, sc-xcode is registered > included in respective > mrg > > mrgl > DP > > 3. voice service voip , allow connection h t h,allow connection h t s, > allow connection s t h > > When call from HQ i can notice randomly (6221, 6222, 6223) , very very > small ring/tone, Display "Ring out" with busy tone > > Please advise... > > [ > Router BR2 >sh dspfarm all > > > Dspfarm Profile Configuration > > Profile ID = 2, Service = TRANSCODING, Resource ID = 1 > Profile Description : > Profile Service Mode : Non Secure > Profile Admin State : UP > Profile Operation State : ACTIVE > Application : SCCP Status : ASSOCIATED > Resource Provider : FLEX_DSPRM Status : UP > Number of Resource Configured : 4 > Number of Resource Available : 4 > Codec Configuration > Codec : g711ulaw, Maximum Packetization Period : 30 > Codec : g711alaw, Maximum Packetization Period : 30 > Codec : g729ar8, Maximum Packetization Period : 60 > Codec : g729abr8, Maximum Packetization Period : 60 > > ] > > > Regards, > > Vikky > > > > On Wed, Mar 20, 2013 at 9:54 PM, Ramcharan Arya > wrote: > >> Hi, >> >> You have to add on your CME side. >> >> voice service voip >> allow connection h t h >> allow connection h t s >> allow connection s t h >> >> Thanks, >> Ramcharan Arya >> >> >> >> >> >> On Wed, Mar 20, 2013 at 1:16 PM, Vikky Kumar wrote: >> >>> Hi Experts, >>> >>> I configured branch 2 CME/CUE working normal for Voice mails. >>> >>> CUE is registered with CUCM but I call not call CUE(6220) from HQ and >>> Branch1 but branch2 phones 4001 and 4002 working normal/ringing from every >>> where. >>> >>> FYI. I have also configured CAC between on Br2 site - HQ site >>> >>> Please hel. >>> >>> Regards >>> >>> Vikky >>> >>> >>> ___ >>> 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] HQ and Branch1 phones cannot call CUE
Hi Experts, I have configured : 1. transcoder on CUCM for HQ & BR2 (Cisco IOS Enhanced Media Termination Point ) and also Router of BR2 (dspfarm profile 2 transcode , max sess 4 .. ) 2. above hq-xcode,, sc-xcode is registered > included in respective > mrg > mrgl > DP 3. voice service voip , allow connection h t h,allow connection h t s, allow connection s t h When call from HQ i can notice randomly (6221, 6222, 6223) , very very small ring/tone, Display "Ring out" with busy tone Please advise... [ Router BR2 >sh dspfarm all > Dspfarm Profile Configuration Profile ID = 2, Service = TRANSCODING, Resource ID = 1 Profile Description : Profile Service Mode : Non Secure Profile Admin State : UP Profile Operation State : ACTIVE Application : SCCP Status : ASSOCIATED Resource Provider : FLEX_DSPRM Status : UP Number of Resource Configured : 4 Number of Resource Available : 4 Codec Configuration Codec : g711ulaw, Maximum Packetization Period : 30 Codec : g711alaw, Maximum Packetization Period : 30 Codec : g729ar8, Maximum Packetization Period : 60 Codec : g729abr8, Maximum Packetization Period : 60 ] Regards, Vikky On Wed, Mar 20, 2013 at 9:54 PM, Ramcharan Arya wrote: > Hi, > > You have to add on your CME side. > > voice service voip > allow connection h t h > allow connection h t s > allow connection s t h > > Thanks, > Ramcharan Arya > > > > > > On Wed, Mar 20, 2013 at 1:16 PM, Vikky Kumar wrote: > >> Hi Experts, >> >> I configured branch 2 CME/CUE working normal for Voice mails. >> >> CUE is registered with CUCM but I call not call CUE(6220) from HQ and >> Branch1 but branch2 phones 4001 and 4002 working normal/ringing from every >> where. >> >> FYI. I have also configured CAC between on Br2 site - HQ site >> >> Please hel. >> >> Regards >> >> Vikky >> >> >> ___ >> 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] HQ and Branch1 phones cannot call CUE
Hi, You have to add on your CME side. voice service voip allow connection h t h allow connection h t s allow connection s t h Thanks, Ramcharan Arya On Wed, Mar 20, 2013 at 1:16 PM, Vikky Kumar wrote: > Hi Experts, > > I configured branch 2 CME/CUE working normal for Voice mails. > > CUE is registered with CUCM but I call not call CUE(6220) from HQ and > Branch1 but branch2 phones 4001 and 4002 working normal/ringing from every > where. > > FYI. I have also configured CAC between on Br2 site - HQ site > > Please hel. > > Regards > > Vikky > > > ___ > 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] HQ and Branch1 phones cannot call CUE
Check the codecs, transcoders, MRG, MRGL, etc On Wed, Mar 20, 2013 at 11:16 AM, Vikky Kumar wrote: > Hi Experts, > > I configured branch 2 CME/CUE working normal for Voice mails. > > CUE is registered with CUCM but I call not call CUE(6220) from HQ and > Branch1 but branch2 phones 4001 and 4002 working normal/ringing from every > where. > > FYI. I have also configured CAC between on Br2 site - HQ site > > Please hel. > > Regards > > Vikky > > > ___ > 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 > -- duy CCIE #27737 Voice ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] HQ and Branch1 phones cannot call CUE
CUE only does G711, do you have a transcoder? Sent from my iPhone On Mar 20, 2013, at 12:39 PM, "Vikky Kumar" wrote: > Hi Experts, > > I configured branch 2 CME/CUE working normal for Voice mails. > > CUE is registered with CUCM but I call not call CUE(6220) from HQ and > Branch1 but branch2 phones 4001 and 4002 working normal/ringing from every > where. > > FYI. I have also configured CAC between on Br2 site - HQ site > > Please hel. > > Regards > > Vikky > > ___ > 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] HQ and Branch1 phones cannot call CUE
Hi Experts, I configured branch 2 CME/CUE working normal for Voice mails. CUE is registered with CUCM but I call not call CUE(6220) from HQ and Branch1 but branch2 phones 4001 and 4002 working normal/ringing from every where. FYI. I have also configured CAC between on Br2 site - HQ site Please hel. Regards Vikky ___ 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
