[OSL | CCIE_Voice] hi
HI sir I need ccie voice more information... ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 2 Wb lab 6 ---- HQ call BR-2 CUE failed
uncheck the farend under GK gateway. On 3/15/2011 9:32 PM, Erwan Erwan wrote: hi all, I am practicing WB lab 6 , that use Gatekeeper for call from HQ (UCM) to BR-2 (CME) - call from HQ-- 5001 to 3001 -- Br-2 and vice versa (thru Gatekeeper) , 4 digit ext works fine - call from HQ to CUE pilot point (3600) (thru Gatekeeper) in Br-2 also work fine - call to BR-2 (3001) phone from PSTN also hit the CUE and I can leave the message Issue = however when I call from HQ to 3001 in BR-2 and I did not pick up the 3001, it supposed to transfer to CUE (3600) so that I can leave message , but instead I heard the busy tone. * I suspect the BR-2 transcode is not triggering ,however *sh sccp* it said transcode register OK config = telephony-service privacy off sdspfarm units 1 sdspfarm transcode sessions 3 sdspfarm tag 1 sc-xcode voicemail 3600 ephone-dn 1 octo-line number 3001 no-reg both call-forward busy 3600 call-forward noan 3600 timeout 20 --- Is there any other possibility I need to check ? Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 2 Wb lab 6 ---- HQ call BR-2 CUE failed
Hi Erwan, Did you follow the instruction on the WB completely? Or you modified the question? The WB is requested a UCME integration with CUC with SIP, not CUE, and user mailbox should be located on CUC not CUE. You need a sip dialpeer to point to CUC with some additional configuration. So, can confirm this is a UCME + CUE or UCME + CUC ? Thanks Shingei. On Wed, Mar 16, 2011 at 2:30 PM, Shrini linuxbos...@gmail.com wrote: uncheck the farend under GK gateway. On 3/15/2011 9:32 PM, Erwan Erwan wrote: hi all, I am practicing WB lab 6 , that use Gatekeeper for call from HQ (UCM) to BR-2 (CME) - call from HQ-- 5001 to 3001 -- Br-2 and vice versa (thru Gatekeeper) , 4 digit ext works fine - call from HQ to CUE pilot point (3600) (thru Gatekeeper) in Br-2 also work fine - call to BR-2 (3001) phone from PSTN also hit the CUE and I can leave the message Issue = however when I call from HQ to 3001 in BR-2 and I did not pick up the 3001, it supposed to transfer to CUE (3600) so that I can leave message , but instead I heard the busy tone. * I suspect the BR-2 transcode is not triggering ,however *sh sccp* it said transcode register OK config = telephony-service privacy off sdspfarm units 1 sdspfarm transcode sessions 3 sdspfarm tag 1 sc-xcode voicemail 3600 ephone-dn 1 octo-line number 3001 no-reg both call-forward busy 3600 call-forward noan 3600 timeout 20 --- Is there any other possibility I need to check ? Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Question about sdspfarm units
Hello, You need one for each SCCP profile configured in router. So, if you configured Transcoder and Conference, use: ! telephony-service sdspfarm units 2 sdspfarm tag 1 xcoder sdspfarm tag 2 conference sdspfarm transcoder session X ! Note: The default value is 0 Best regards, Tamer Ismail From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Randall Crumm Sent: Tuesday, March 15, 2011 10:34 PM To: OSL Questions Subject: [OSL | CCIE_Voice] Question about sdspfarm units HI, Can the experts please explain sdspfarm units please? How should I configure this and do I need to increment the units as I add profiles? Or when would I increment the units? Thanks, Best Regards, Randall Crumm Voice and Video Architect Global Networking Telecom Description: ccvp_voice_sm Description: logo 1007 Gibraltar Drive Milpitas, CA 95035 Direct +1 408.576.7344 VoIP .100.7344 Creating value that increases customer competitiveness Legal Disclaimer: The information contained in this message may be privileged and confidential. It is intended to be read only by the individual or entity to whom it is addressed or by their designee. If the reader of this message is not the intended recipient, you are on notice that any distribution of this message, in any form, is strictly prohibited. If you have received this message in error, please immediately notify the sender and delete or destroy any copy of this message image001.gifimage002.gif___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Unity AA transfer call directly to Voice mail
Hi All As i am having the issue with Unity 5.0 Server,if some one calls to AA,after pressing the extension call is rolling to directly Voicemail,It is not ringing first . There is no call FWD all apply on DN Can any body suggests Regards Kuldeep ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Unity AA transfer call directly to Voice mail
Change the call transfer rule on the persons mailbox to ring this extension. By default it goes to the greeting. -Original Message- From: kuldeep tiwari kuldeep.tiwari1...@gmail.com Sender: ccie_voice-boun...@onlinestudylist.com Date: Wed, 16 Mar 2011 17:12:53 To: ccie_voice@onlinestudylist.com; ccie_voice-requ...@onlinestudylist.com Subject: [OSL | CCIE_Voice] Unity AA transfer call directly to Voice mail ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Unity AA transfer call directly to Voice mail
Remember too, that whenever a call goes through Unity first, it always will search itself for the call routing rules before going to CCM. In this case, it is going to the entity that has the extension you are forwarding to (mailbox or call handler) and looks at the call transfer rules. From: kuldeep tiwari Sent: Wednesday, March 16, 2011 7:42 AM To: ccie_voice@onlinestudylist.com ; ccie_voice-requ...@onlinestudylist.com Subject: [OSL | CCIE_Voice] Unity AA transfer call directly to Voice mail Hi All As i am having the issue with Unity 5.0 Server,if some one calls to AA,after pressing the extension call is rolling to directly Voicemail,It is not ringing first . There is no call FWD all apply on DN Can any body suggests Regards Kuldeep ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] hi
Cisco.com duy ccie #27737 voice tmobile g2 On Mar 16, 2011 1:13 AM, Ajay Singh sony199...@gmail.com wrote: HI sir I need ccie voice more information... ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 2 Wb lab 6 ---- HQ call BR-2 CUE failed
I refer to the Guide, it is CUE and dialpeer point to CUE 3600 --- On Wed, 3/16/11, ShinGei Yong shingei.y...@gmail.com wrote: From: ShinGei Yong shingei.y...@gmail.com Subject: Re: [OSL | CCIE_Voice] Vol 2 Wb lab 6 HQ call BR-2 CUE failed To: Erwan Erwan e_er...@yahoo.com, ccie_voice@onlinestudylist.com Received: Wednesday, March 16, 2011, 4:20 PM Hi Erwan, Did you follow the instruction on the WB completely? Or you modified the question? The WB is requested a UCME integration with CUC with SIP, not CUE, and user mailbox should be located on CUC not CUE. You need a sip dialpeer to point to CUC with some additional configuration. So, can confirm this is a UCME + CUE or UCME + CUC ? Thanks Shingei. On Wed, Mar 16, 2011 at 2:30 PM, Shrini linuxbos...@gmail.com wrote: uncheck the farend under GK gateway. On 3/15/2011 9:32 PM, Erwan Erwan wrote: hi all, I am practicing WB lab 6 , that use Gatekeeper for call from HQ (UCM) to BR-2 (CME) - call from HQ-- 5001 to 3001 -- Br-2 and vice versa (thru Gatekeeper) , 4 digit ext works fine - call from HQ to CUE pilot point (3600) (thru Gatekeeper) in Br-2 also work fine - call to BR-2 (3001) phone from PSTN also hit the CUE and I can leave the message Issue = however when I call from HQ to 3001 in BR-2 and I did not pick up the 3001, it supposed to transfer to CUE (3600) so that I can leave message , but instead I heard the busy tone. * I suspect the BR-2 transcode is not triggering ,however sh sccp it said transcode register OK config = telephony-service privacy off sdspfarm units 1 sdspfarm transcode sessions 3 sdspfarm tag 1 sc-xcode voicemail 3600 ephone-dn 1 octo-line number 3001 no-reg both call-forward busy 3600 call-forward noan 3600 timeout 20 --- Is there any other possibility I need to check ? Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 2 Wb lab 6 ---- HQ call BR-2 CUE failed
Hi Erwan, I hope we are referring to the same WB. The question is asked: Question 6.2: Integrate UCME with Unity Connection using SIP and create mailbox for br2 phone 1 and phone 2.(4pts) Yes, the dp pointed to 3600,but did you look at the address? is that 210.13 or 202.2 Can tell me which page of the doc you are looking at? Shingei On Wed, Mar 16, 2011 at 11:08 PM, Erwan Erwan e_er...@yahoo.com wrote: I refer to the Guide, it is CUE and dialpeer point to CUE 3600 --- On *Wed, 3/16/11, ShinGei Yong shingei.y...@gmail.com* wrote: From: ShinGei Yong shingei.y...@gmail.com Subject: Re: [OSL | CCIE_Voice] Vol 2 Wb lab 6 HQ call BR-2 CUE failed To: Erwan Erwan e_er...@yahoo.com, ccie_voice@onlinestudylist.com Received: Wednesday, March 16, 2011, 4:20 PM Hi Erwan, Did you follow the instruction on the WB completely? Or you modified the question? The WB is requested a UCME integration with CUC with SIP, not CUE, and user mailbox should be located on CUC not CUE. You need a sip dialpeer to point to CUC with some additional configuration. So, can confirm this is a UCME + CUE or UCME + CUC ? Thanks Shingei. On Wed, Mar 16, 2011 at 2:30 PM, Shrini linuxbos...@gmail.comhttp://ca.mc1205.mail.yahoo.com/mc/compose?to=linuxbos...@gmail.com wrote: uncheck the farend under GK gateway. On 3/15/2011 9:32 PM, Erwan Erwan wrote: hi all, I am practicing WB lab 6 , that use Gatekeeper for call from HQ (UCM) to BR-2 (CME) - call from HQ-- 5001 to 3001 -- Br-2 and vice versa (thru Gatekeeper) , 4 digit ext works fine - call from HQ to CUE pilot point (3600) (thru Gatekeeper) in Br-2 also work fine - call to BR-2 (3001) phone from PSTN also hit the CUE and I can leave the message Issue = however when I call from HQ to 3001 in BR-2 and I did not pick up the 3001, it supposed to transfer to CUE (3600) so that I can leave message , but instead I heard the busy tone. * I suspect the BR-2 transcode is not triggering ,however *sh sccp* it said transcode register OK config = telephony-service privacy off sdspfarm units 1 sdspfarm transcode sessions 3 sdspfarm tag 1 sc-xcode voicemail 3600 ephone-dn 1 octo-line number 3001 no-reg both call-forward busy 3600 call-forward noan 3600 timeout 20 --- Is there any other possibility I need to check ? Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Unity AA transfer call directly to Voice mail
still not working On Wed, Mar 16, 2011 at 5:52 PM, CCIE for Me cciefo...@hotmail.com wrote: Remember too, that whenever a call goes through Unity first, it always will search itself for the call routing rules before going to CCM. In this case, it is going to the entity that has the extension you are forwarding to (mailbox or call handler) and looks at the call transfer rules. *From:* kuldeep tiwari kuldeep.tiwari1...@gmail.com *Sent:* Wednesday, March 16, 2011 7:42 AM *To:* ccie_voice@onlinestudylist.com ; ccie_voice-requ...@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] Unity AA transfer call directly to Voice mail Hi All As i am having the issue with Unity 5.0 Server,if some one calls to AA,after pressing the extension call is rolling to directly Voicemail,It is not ringing first . There is no call FWD all apply on DN Can any body suggests Regards Kuldeep -- ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Regards Kuldeep Tiwari +91-9324390008 +91-9326775437 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Need suggestion on MVA
Hi All, Finally My MVA started working ! Need your suggestion for following question PSTN phone line 4 can call in and make outbound Intl call. also allow SiteB for PSTN phone line 4 to call MVA , I guess SNR for number which is assigned to line 4 should be working and to allow siteB , do i need to create bogus remote destination number ie SNR for all site B user ? i am not sure about 2nd part ? what is your approach ? thx, Rahul ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] OT : Missing normal softkeys from 9971.
I'm hoping i missed something simple. I just got a 9971 registered on a CUCM 7.x server. it works great but I noticed there are no available softkeys for hold, park , etc during a call. It's my first SIP phone I'm using, so is there something I'm missing regarding supplemental services? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] OT : Missing normal softkeys from 9971.
9971 are not evaluated in CCIE Lab as far as I know therefore, this is not the right place for this question. Try doing some google or post this question at Cisco Support forums. http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/9971_9951_8961/8_5/english/user_guide/book_911/callingfeatures_rt.html#wp1027556 Regards, Roger Carpio. On Wed, Mar 16, 2011 at 1:41 PM, ccielab...@gmail.com wrote: I'm hoping i missed something simple. I just got a 9971 registered on a CUCM 7.x server. it works great but I noticed there are no available softkeys for hold, park , etc during a call. It's my first SIP phone I'm using, so is there something I'm missing regarding supplemental services? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] OT : Missing normal softkeys from 9971.
Right, thats why I included OT : (Off topic) in the subject line. That way, for people who don't want to be bothered can simply ignore it. On Wed, Mar 16, 2011 at 4:14 PM, Roger Carpio roger.car...@gmail.comwrote: 9971 are not evaluated in CCIE Lab as far as I know therefore, this is not the right place for this question. Try doing some google or post this question at Cisco Support forums. http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/9971_9951_8961/8_5/english/user_guide/book_911/callingfeatures_rt.html#wp1027556 Regards, Roger Carpio. On Wed, Mar 16, 2011 at 1:41 PM, ccielab...@gmail.com wrote: I'm hoping i missed something simple. I just got a 9971 registered on a CUCM 7.x server. it works great but I noticed there are no available softkeys for hold, park , etc during a call. It's my first SIP phone I'm using, so is there something I'm missing regarding supplemental services? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] OT : Missing normal softkeys from 9971.
If you look at the phone you will observe that it has HARD KEYS for the functions you are looking for :) On Wed, Mar 16, 2011 at 2:41 PM, ccielab...@gmail.com wrote: I'm hoping i missed something simple. I just got a 9971 registered on a CUCM 7.x server. it works great but I noticed there are no available softkeys for hold, park , etc during a call. It's my first SIP phone I'm using, so is there something I'm missing regarding supplemental services? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] OT : Missing normal softkeys from 9971.
It's difficult to ignore it if it is addressed to a specialized mailing list. It also takes extra time to sort out the crap that does not belong from the important emails. I'm sure you will get some attention at the cisco support forums and 0 rejection. El mié, 16-03-2011 a las 17:35 -0400, ccielabrat escribió: Right, thats why I included OT : (Off topic) in the subject line. That way, for people who don't want to be bothered can simply ignore it. On Wed, Mar 16, 2011 at 4:14 PM, Roger Carpio roger.car...@gmail.com wrote: 9971 are not evaluated in CCIE Lab as far as I know therefore, this is not the right place for this question. Try doing some google or post this question at Cisco Support forums. http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/9971_9951_8961/8_5/english/user_guide/book_911/callingfeatures_rt.html#wp1027556 Regards, Roger Carpio. On Wed, Mar 16, 2011 at 1:41 PM, ccielab...@gmail.com wrote: I'm hoping i missed something simple. I just got a 9971 registered on a CUCM 7.x server. it works great but I noticed there are no available softkeys for hold, park , etc during a call. It's my first SIP phone I'm using, so is there something I'm missing regarding supplemental services? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Need suggestion on MVA
I don't understand the 2nd part either. Are you sure you are paraphrasing it correctly? Cheers. El mié, 16-03-2011 a las 23:58 +0530, Rahul Kapor escribió: Hi All, Finally My MVA started working ! Need your suggestion for following question PSTN phone line 4 can call in and make outbound Intl call. also allow SiteB for PSTN phone line 4 to call MVA , I guess SNR for number which is assigned to line 4 should be working and to allow siteB , do i need to create bogus remote destination number ie SNR for all site B user ? i am not sure about 2nd part ? what is your approach ? thx, Rahul ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] OT : Missing normal softkeys from 9971.
Randall Roger, Are you kidding me? I admit it was a dumb question and I tried to be polite by marking it OT: (Off Topic) I was stuck in a lab with no direct internet access or a manual for a phone I never used before. I leveraged the mailer group on my phone looking for a hand. A group, which for the most part, is helpful whether or not it has anything specifically to do with the voice lab. I got several quick replies with useful information. The thread could have died 5 minutes after I posted it. In the time it took to reply, both of you could have deleted it and moved on with your day. I don't see either of you replying back to morons who blatantly ask for NDA information. *Your arrogance is laughable*, to think either of you have a place to suggest what does or doesn't belong on this mailing list. If you have a problem with it , take it up with the mailer admin. Or email me a picture of your mailer police badge , and then I'll consider your email. I honestly hope if you are ever in the same position, you both get a helpful reply instead of the nonsense you both provided. On Wed, Mar 16, 2011 at 9:21 PM, Randall Saborío Cubero ill2...@gmail.comwrote: It's difficult to ignore it if it is addressed to a specialized mailing list. It also takes extra time to sort out the crap that does not belong from the important emails. I'm sure you will get some attention at the cisco support forums and 0 rejection. El mié, 16-03-2011 a las 17:35 -0400, ccielabrat escribió: Right, thats why I included OT : (Off topic) in the subject line. That way, for people who don't want to be bothered can simply ignore it. On Wed, Mar 16, 2011 at 4:14 PM, Roger Carpio roger.car...@gmail.com wrote: 9971 are not evaluated in CCIE Lab as far as I know therefore, this is not the right place for this question. Try doing some google or post this question at Cisco Support forums. http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/9971_9951_8961/8_5/english/user_guide/book_911/callingfeatures_rt.html#wp1027556 Regards, Roger Carpio. On Wed, Mar 16, 2011 at 1:41 PM, ccielab...@gmail.com wrote: I'm hoping i missed something simple. I just got a 9971 registered on a CUCM 7.x server. it works great but I noticed there are no available softkeys for hold, park , etc during a call. It's my first SIP phone I'm using, so is there something I'm missing regarding supplemental services? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
