Re: [OSL | CCIE_Voice] Cisco Unity 7 vmware install error
I have a IBM x346 server runs on windows2003 server. I put vmware server 2 on it and runs virtual machine with 2G memory and 80G hard drive. I have successfully installed PUB, SUB , Precense and IPCC7 on vmware already. Cheers, Jerry Date: Thu, 23 Jul 2009 12:22:54 +0530 Subject: Re: [OSL | CCIE_Voice] Cisco Unity 7 vmware install error From: lakpr...@gmail.com To: jerry...@hotmail.com CC: ccie_voice@onlinestudylist.com Hey Jerry, U r installing on AMD processor based machine ?? On Thu, Jul 23, 2009 at 12:20 PM, Bai Min jerry...@hotmail.com wrote: Hi Guys, Did you experience problem when install Unity connection 7 on vmware? I got internal critical error and halt installation. Please find this error screenshot in attachment. Thanks in advance for your advise. Cheers, Jerry NEW mobile Hotmail. Optimized for YOUR phone. Click here. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Ravindra Lakpriya +94 773 532 094 _ Windows Live™ SkyDrive™: Store, access, and share your photos. See how. http://windowslive.com/Online/SkyDrive?ocid=TXT_TAGLM_WL_CS_SD_photos_072009___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Hilarity with CUCME 4.3/7.0
CCME 7.0(1) What did the phone say? -Original Message- From: Jonathan Charles [mailto:jonv...@gmail.com] Sent: Thursday, July 23, 2009 12:37 AM To: Michael Ciarfello Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Hilarity with CUCME 4.3/7.0 Ub CUCME 4.1 and earlier, there was sw conf... in 4.3, it says, on page 709 of the CUCME7 admin guide: SCCP: Configuring Multi-Party Ad Hoc and Meet-Me Conferencing in Cisco Unified CME 4.1 and Later Versions: Prerequisites * Cisco Unified CME 4.1 or a later version * You must have a PVDM2-8, PVDM2-16, PVDM2-32, or PVDM2-64 high-density packet voice digital signal processor module hosted on the motherboard or on a module such as the NM-HDV2 or NM-HD-2VE. * For Cisco Unified IP Phone 7985, firmware version 4-1-2-0 or a later version I configured it, and tested the old-style, max-conferences command under telephony services and was told no by the phone. My telephony-service version is 7.0(1) what is yours? Jonathan On Wed, Jul 22, 2009 at 9:46 PM, Michael Ciarfellomciarfe...@iplogic.com wrote: Looks good here. SIP to SIP 3-way conference, SCCP to SCCP and SCCP to/from SIP. Also admin guide page 680 Conferencing Overview. No software meet-me (never was sw meetme.) From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jonathan Charles [jonv...@gmail.com] Sent: Wednesday, July 22, 2009 7:15 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Hilarity with CUCME 4.3/7.0 Conferencing, check it out, no ad-hoc software conferencing any more... Jonathan ___ 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] Cisco Unity 7 vmware install error
I had a somewhat similar error message when I tried to load both CUCM 7 and UC7 on VMWARE ESXi 4 (linux version), the install failed with a System 'halt' error. Both loaded without any issues on VMWARE GSX server (Windows version) so as a workaround I export the GSX VM image, convert to OVA format then import into ESXi. Jeff Knuckle, Network Engineer From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Bai Min Sent: Thursday, July 23, 2009 2:50 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Cisco Unity 7 vmware install error Hi Guys, Did you experience problem when install Unity connection 7 on vmware? I got internal critical error and halt installation. Please find this error screenshot in attachment. Thanks in advance for your advise. Cheers, Jerry file:///C:\DOCUME~1\ADMINI~1\LOCALS~1\Temp\moz-screenshot-2.png file:///C:\DOCUME~1\ADMINI~1\LOCALS~1\Temp\moz-screenshot-3.png NEW mobile Hotmail. Optimized for YOUR phone. Click here. http://windowslive.com/Mobile?ocid=TXT_TAGLM_WL_CS_MB_new_hotmail_07200 9 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP endpoints on the other side of an H.323 trunk...
Fast Start did not change the behavior in any way... On Wed, Jul 22, 2009 at 10:32 PM, vineet sanghivineet_san...@yahoo.com wrote: this works if you choose faststart inbound outbound on ccm h323 trunk with mtp. The issue is SIP phones are configured for early offer so h323 trunk should match. - Original Message From: Jonathan Charles jonv...@gmail.com To: ccie_voice@onlinestudylist.com; cisco-v...@puck.nether.net Sent: Thursday, July 23, 2009 11:55:22 AM Subject: Re: [OSL | CCIE_Voice] SIP endpoints on the other side of an H.323 trunk... OK, so if SIP endpoints aren't supported, does that mean the CUE is not supported across the GK? On Wed, Jul 22, 2009 at 6:21 PM, Jonathan Charlesjonv...@gmail.com wrote: So, does anyone know if this is true? I am actually seeing that SIP cannot call phones across an H.323 trunk It looks like everything is fine, the call sets up, but then no audio cuts through and the SIP phone keeps ringing... And then I see this when I hang up the cucm phone Jul 22 23:19:48.514: //-1//SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 500 Internal Server Error Reason: Q.850;cause=16 Date: Wed, 22 Jul 2009 23:19:40 GMT From: Phone1 sip:3...@192.168.29.254;tag=00169def16e500dfdb252a73-c3694981 Allow-Events: telephone-event Content-Length: 0 To: sip:5...@192.168.29.254;tag=BCCA62C-BD3 Call-ID: 00169def-16e50019-a4734327-02d0c...@192.168.29.102 Via: SIP/2.0/UDP 192.168.29.102:5060;branch=z9hG4bKc0c82a6f CSeq: 101 INVITE Server: Cisco-SIPGateway/IOS-12.x Jul 22 23:19:48.522: //-1//SIP/Msg/ccsipDisplayMsg: Received: BR2#ACK sip:5...@192.168.29.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.29.102:5060;branch=z9hG4bKc0c82a6f From: Phone1 sip:3...@192.168.29.254;tag=00169def16e500dfdb252a73-c3694981 To: sip:5...@192.168.29.254;tag=BCCA62C-BD3 Call-ID: 00169def-16e50019-a4734327-02d0c...@192.168.29.102 Date: Wed, 22 Jul 2009 23:19:44 GMT CSeq: 101 ACK Content-Length: 0 On Wed, Jun 10, 2009 at 1:30 AM, Jonathan Charlesjonv...@gmail.com wrote: Found a weird statement in a Cisco doc: SIP endpoints are not supported on H.323 trunks. SIP endpoints are supported on SIP trunks only This was in the SRST System Admin guide under Octo-Lines... I thought the CUBE fixed this? Jonathan ___ 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] Hilarity with CUCME 4.3/7.0
Cannot Complete Conference. On Thu, Jul 23, 2009 at 8:20 AM, Michael Ciarfellomciarfe...@iplogic.com wrote: CCME 7.0(1) What did the phone say? -Original Message- From: Jonathan Charles [mailto:jonv...@gmail.com] Sent: Thursday, July 23, 2009 12:37 AM To: Michael Ciarfello Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Hilarity with CUCME 4.3/7.0 Ub CUCME 4.1 and earlier, there was sw conf... in 4.3, it says, on page 709 of the CUCME7 admin guide: SCCP: Configuring Multi-Party Ad Hoc and Meet-Me Conferencing in Cisco Unified CME 4.1 and Later Versions: Prerequisites * Cisco Unified CME 4.1 or a later version * You must have a PVDM2-8, PVDM2-16, PVDM2-32, or PVDM2-64 high-density packet voice digital signal processor module hosted on the motherboard or on a module such as the NM-HDV2 or NM-HD-2VE. * For Cisco Unified IP Phone 7985, firmware version 4-1-2-0 or a later version I configured it, and tested the old-style, max-conferences command under telephony services and was told no by the phone. My telephony-service version is 7.0(1) what is yours? Jonathan On Wed, Jul 22, 2009 at 9:46 PM, Michael Ciarfellomciarfe...@iplogic.com wrote: Looks good here. SIP to SIP 3-way conference, SCCP to SCCP and SCCP to/from SIP. Also admin guide page 680 Conferencing Overview. No software meet-me (never was sw meetme.) From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jonathan Charles [jonv...@gmail.com] Sent: Wednesday, July 22, 2009 7:15 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Hilarity with CUCME 4.3/7.0 Conferencing, check it out, no ad-hoc software conferencing any more... Jonathan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] MWI for SIP phones on CCME and CUE 7
Not lighting up... ! ! voice register global mode cme source-address 192.168.29.254 port 5060 max-dn 8 max-pool 16 load 7961 term61.default authenticate register authenticate realm cisco.com timezone 42 time-format 24 date-format D/M/Y voicemail 3100 tftp-path flash: create profile sync 0029906034594977 ntp-server 192.168.29.254 mode directedbroadcast ! voice register session-server 1 ! voice register dn 1 number 3005 call-forward b2bua busy 3100 call-forward b2bua mailbox 3005 call-forward b2bua noan 3100 timeout 12 allow watch name Phone1 ! voice register dn 2 number 3006 call-forward b2bua busy 3100 call-forward b2bua mailbox 3006 call-forward b2bua noan 3100 timeout 12 mwi ! voice register dn 8 mwi ! voice register template 1 voicemail 3100 timeout 12 ! voice register dialplan 1 type 7940-7960-others pattern 1 3... ! voice register pool 1 id mac 0016.9DEF.16E5 type 7961 number 1 dn 1 template 1 presence call-list dtmf-relay rtp-nte sip-notify username 3005 password cisco description 3214-3005 codec g711ulaw blf-speed-dial 2 3001 label BLFto3001 sip-ua retry invite 2 timers trying 200 mwi-server ipv4:192.168.29.230 expires 3600 port 5060 transport udp ! ! On the CUE... ccn subsystem sip gateway address 192.168.29.254 mwi sip outcall sub-notify end subsystem Any ideas? Jonathan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] UCCX and meet-me
Hello Experts, I was wondering if you know how to use UCCX to request a password before a user is allowed to establish a meet-me conference call do you know if this is possible ? thanks Cris ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCCX and meet-me
Yes. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cristobal Priego Sent: Thursday, July 23, 2009 12:53 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] UCCX and meet-me Hello Experts, I was wondering if you know how to use UCCX to request a password before a user is allowed to establish a meet-me conference call do you know if this is possible ? thanks Cris ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCCX and meet-me
Now THERE's an expert response! lol On Thu, Jul 23, 2009 at 12:15 PM, Michael Ciarfello mciarfe...@iplogic.comwrote: Yes. *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Cristobal Priego *Sent:* Thursday, July 23, 2009 12:53 PM *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] UCCX and meet-me Hello Experts, I was wondering if you know how to use UCCX to request a password before a user is allowed to establish a meet-me conference call do you know if this is possible ? thanks Cris ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Jeffrey W. Hall Jeffrey W. Hall, LLC. CCSI #31661, CCVP, CCSP, CCIP, CCNP, CCDP, CQS, CCNA (V,S,W), MCT, MCITP (Cell) 901-490-4140 (Email) layer8...@gmail.com (Blog) http://layer8man.ccieblog.com Sent from Olive Branch, MS, United States ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] AC console in 7.0 lab
I have a copy I let me check how big is the file 2009/7/23 Art Sandborgh asandbo...@hotmail.com All, I attempted to set up AC on the new 7.0 V3 rack today and came across a roadblock. For 7.0 and beyond Cisco requires that you buy a different console if you are a New install and they no longer have the application on the downloads page. Apparantly if you upgrade from a previous release it does appear on the page (it automatically grandfathers you in), but if you install new you don't get it. From what I have been reading Cisco did have it posted for a time out on the 7.0 CCM s/w download page, but unfortunately they have pulled it from there now as well. I saw information indicating that they will only support the new (charged for) versions in 8.0. Does anyone know where I can get a copy of the 7.0 version? Thanks, Art -- Windows Live™ SkyDrive™: Store, access, and share your photos. See how.http://windowslive.com/Online/SkyDrive?ocid=TXT_TAGLM_WL_CS_SD_photos_072009 ___ 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] UCCX and meet-me
I answered the question he was asking at the time. Lol CRS script: You can play a prompt if you wish, you will use the get digit string to collect the password from the user then use the IF step to compare it to the accepted password, if they are equal then call redirect to your meetme number. If not then do what you want (hang up, re-prompt user, etc.) From: Jeffrey Hall [mailto:layer8...@gmail.com] Sent: Thursday, July 23, 2009 2:24 PM To: Michael Ciarfello Cc: Cristobal Priego; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] UCCX and meet-me Now THERE's an expert response! lol On Thu, Jul 23, 2009 at 12:15 PM, Michael Ciarfello mciarfe...@iplogic.commailto:mciarfe...@iplogic.com wrote: Yes. From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cristobal Priego Sent: Thursday, July 23, 2009 12:53 PM To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] UCCX and meet-me Hello Experts, I was wondering if you know how to use UCCX to request a password before a user is allowed to establish a meet-me conference call do you know if this is possible ? thanks Cris ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com -- Jeffrey W. Hall Jeffrey W. Hall, LLC. CCSI #31661, CCVP, CCSP, CCIP, CCNP, CCDP, CQS, CCNA (V,S,W), MCT, MCITP (Cell) 901-490-4140 (Email) layer8...@gmail.commailto:layer8...@gmail.com (Blog) http://layer8man.ccieblog.com Sent from Olive Branch, MS, United States ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] AC console in 7.0 lab
Art you can download the file from here http://rapidshare.com/files/259206841/CiscoAttendantConsoleClient.rar.html MD5: 41A0E94DC6889361719CFE2E4EF2E917 Cris 2009/7/23 Cristobal Priego cristobalpri...@gmail.com I have a copy I let me check how big is the file 2009/7/23 Art Sandborgh asandbo...@hotmail.com All, I attempted to set up AC on the new 7.0 V3 rack today and came across a roadblock. For 7.0 and beyond Cisco requires that you buy a different console if you are a New install and they no longer have the application on the downloads page. Apparantly if you upgrade from a previous release it does appear on the page (it automatically grandfathers you in), but if you install new you don't get it. From what I have been reading Cisco did have it posted for a time out on the 7.0 CCM s/w download page, but unfortunately they have pulled it from there now as well. I saw information indicating that they will only support the new (charged for) versions in 8.0. Does anyone know where I can get a copy of the 7.0 version? Thanks, Art -- Windows Live™ SkyDrive™: Store, access, and share your photos. See how.http://windowslive.com/Online/SkyDrive?ocid=TXT_TAGLM_WL_CS_SD_photos_072009 ___ 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] UCCX and meet-me
thank you, however the meet me needs to be initiated first, right and then the callers needs to authenticate or is there a way to authenticate the initiator only and then have the remains users just to join the bridge without authentication 2009/7/23 Michael Ciarfello mciarfe...@iplogic.com I answered the question he was asking at the time. Lol CRS script: You can play a prompt if you wish, you will use the get digit string to collect the password from the user then use the IF step to compare it to the accepted password, if they are equal then call redirect to your meetme number. If not then do what you want (hang up, re-prompt user, etc.) *From:* Jeffrey Hall [mailto:layer8...@gmail.com] *Sent:* Thursday, July 23, 2009 2:24 PM *To:* Michael Ciarfello *Cc:* Cristobal Priego; ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] UCCX and meet-me Now THERE's an expert response! lol On Thu, Jul 23, 2009 at 12:15 PM, Michael Ciarfello mciarfe...@iplogic.com wrote: Yes. *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Cristobal Priego *Sent:* Thursday, July 23, 2009 12:53 PM *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] UCCX and meet-me Hello Experts, I was wondering if you know how to use UCCX to request a password before a user is allowed to establish a meet-me conference call do you know if this is possible ? thanks Cris ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Jeffrey W. Hall Jeffrey W. Hall, LLC. CCSI #31661, CCVP, CCSP, CCIP, CCNP, CCDP, CQS, CCNA (V,S,W), MCT, MCITP (Cell) 901-490-4140 (Email) layer8...@gmail.com (Blog) http://layer8man.ccieblog.com Sent from Olive Branch, MS, United States ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] PR labs scheduling
Gent's, How the scheduling for the PR Voice labs? I'm in the process of trying to decide whether I should invest in some lab gear for my home office or spend the $ for the PR labs. My biggest concern is that the time slots available. If you work a normal job and get home at say 6:00 PM, how do you schedule that ? Or, do you do a full 7.5 hr session on Saturday or Sunday.which, to me would be the most popular and the hardest to get into.. Thoughts? Thomas J Koch Owner/Consultant Digitones, LLC Cell: 630-808-4910 E-mail: digito...@comcast.net BEGIN:VCARD VERSION:2.1 N:Koch;Thomas FN:Thomas J Koch (digito...@comcast.net) ORG:Digitones, LLC TITLE:Owner/Consultant TEL;CELL;VOICE:(630) 808-4910 TEL;WORK;FAX:(630) 243-8971 ADR;WORK:;;461 Kromray Road;Lemont;IL;60439;United States of America LABEL;WORK;ENCODING=QUOTED-PRINTABLE:461 Kromray Road=0D=0ALemont, IL 60439=0D=0AUnited States of America EMAIL;PREF;INTERNET:digito...@comcast.net REV:20090702T193658Z END:VCARD ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCCX and meet-me
The conference controller must manually and at his/her phone setup the meet-me. There's no authentication (pick up phone, more, meet-me, meet-me directory number). You may also have to design PTs and CSSs, possible translation patterns so that the conference controller can initiate the meet-me and the callers join. After the meet-me is setup, you want to only authenticate the callers dialing in. They will probably dial another number, go through the IPCC script, authenticate then the script transfers to the meet-me number. Your initiator and your IPCC ports will need to have access to the meet-me number. The callers / gateway(s) will need to have access to the IPCC RP. I might have confused you by combining too many possibilities into one message. Let me know your exact requirements and I'd be glad to clean it up for you. From: Cristobal Priego [mailto:cristobalpri...@gmail.com] Sent: Thursday, July 23, 2009 2:44 PM To: Michael Ciarfello Cc: Jeffrey Hall; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] UCCX and meet-me thank you, however the meet me needs to be initiated first, right and then the callers needs to authenticate or is there a way to authenticate the initiator only and then have the remains users just to join the bridge without authentication 2009/7/23 Michael Ciarfello mciarfe...@iplogic.commailto:mciarfe...@iplogic.com I answered the question he was asking at the time. Lol CRS script: You can play a prompt if you wish, you will use the get digit string to collect the password from the user then use the IF step to compare it to the accepted password, if they are equal then call redirect to your meetme number. If not then do what you want (hang up, re-prompt user, etc.) From: Jeffrey Hall [mailto:layer8...@gmail.commailto:layer8...@gmail.com] Sent: Thursday, July 23, 2009 2:24 PM To: Michael Ciarfello Cc: Cristobal Priego; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] UCCX and meet-me Now THERE's an expert response! lol On Thu, Jul 23, 2009 at 12:15 PM, Michael Ciarfello mciarfe...@iplogic.commailto:mciarfe...@iplogic.com wrote: Yes. From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cristobal Priego Sent: Thursday, July 23, 2009 12:53 PM To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] UCCX and meet-me Hello Experts, I was wondering if you know how to use UCCX to request a password before a user is allowed to establish a meet-me conference call do you know if this is possible ? thanks Cris ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com -- Jeffrey W. Hall Jeffrey W. Hall, LLC. CCSI #31661, CCVP, CCSP, CCIP, CCNP, CCDP, CQS, CCNA (V,S,W), MCT, MCITP (Cell) 901-490-4140 (Email) layer8...@gmail.commailto:layer8...@gmail.com (Blog) http://layer8man.ccieblog.com Sent from Olive Branch, MS, United States ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCCX and meet-me
I think I got it Is there a way to do a consult transfer with the Script. that's the exact scenario that we want. once the meet-me is setup internally we want only to authenticate the callers dialing in from the pstn using the script. so i was thinking of a consult transfer just in case the meet-me isn't setup yet they won't get a fast busy and try to do something else. or should i just send the call to a call handler and do exactly what Christopher told me to do thanks Michael for your time 2009/7/23 Michael Ciarfello mciarfe...@iplogic.com The conference controller must manually and at his/her phone setup the meet-me. There’s no authentication (pick up phone, more, meet-me, meet-me directory number). You may also have to design PTs and CSSs, possible translation patterns so that the conference controller can initiate the meet-me and the callers join. After the meet-me is setup, you want to only authenticate the callers dialing in. They will probably dial another number, go through the IPCC script, authenticate then the script transfers to the meet-me number. Your initiator and your IPCC ports will need to have access to the meet-me number. The callers / gateway(s) will need to have access to the IPCC RP. I might have confused you by combining too many possibilities into one message. Let me know your exact requirements and I’d be glad to clean it up for you. *From:* Cristobal Priego [mailto:cristobalpri...@gmail.com] *Sent:* Thursday, July 23, 2009 2:44 PM *To:* Michael Ciarfello *Cc:* Jeffrey Hall; ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] UCCX and meet-me thank you, however the meet me needs to be initiated first, right and then the callers needs to authenticate or is there a way to authenticate the initiator only and then have the remains users just to join the bridge without authentication 2009/7/23 Michael Ciarfello mciarfe...@iplogic.com I answered the question he was asking at the time. Lol CRS script: You can play a prompt if you wish, you will use the get digit string to collect the password from the user then use the IF step to compare it to the accepted password, if they are equal then call redirect to your meetme number. If not then do what you want (hang up, re-prompt user, etc.) *From:* Jeffrey Hall [mailto:layer8...@gmail.com] *Sent:* Thursday, July 23, 2009 2:24 PM *To:* Michael Ciarfello *Cc:* Cristobal Priego; ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] UCCX and meet-me Now THERE's an expert response! lol On Thu, Jul 23, 2009 at 12:15 PM, Michael Ciarfello mciarfe...@iplogic.com wrote: Yes. *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Cristobal Priego *Sent:* Thursday, July 23, 2009 12:53 PM *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] UCCX and meet-me Hello Experts, I was wondering if you know how to use UCCX to request a password before a user is allowed to establish a meet-me conference call do you know if this is possible ? thanks Cris ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Jeffrey W. Hall Jeffrey W. Hall, LLC. CCSI #31661, CCVP, CCSP, CCIP, CCNP, CCDP, CQS, CCNA (V,S,W), MCT, MCITP (Cell) 901-490-4140 (Email) layer8...@gmail.com (Blog) http://layer8man.ccieblog.com Sent from Olive Branch, MS, United States ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCCX and meet-me
nevermind I got it sorry 2009/7/23 Cristobal Priego cristobalpri...@gmail.com I think I got it Is there a way to do a consult transfer with the Script. that's the exact scenario that we want. once the meet-me is setup internally we want only to authenticate the callers dialing in from the pstn using the script. so i was thinking of a consult transfer just in case the meet-me isn't setup yet they won't get a fast busy and try to do something else. or should i just send the call to a call handler and do exactly what Christopher told me to do thanks Michael for your time 2009/7/23 Michael Ciarfello mciarfe...@iplogic.com The conference controller must manually and at his/her phone setup the meet-me. There’s no authentication (pick up phone, more, meet-me, meet-me directory number). You may also have to design PTs and CSSs, possible translation patterns so that the conference controller can initiate the meet-me and the callers join. After the meet-me is setup, you want to only authenticate the callers dialing in. They will probably dial another number, go through the IPCC script, authenticate then the script transfers to the meet-me number. Your initiator and your IPCC ports will need to have access to the meet-me number. The callers / gateway(s) will need to have access to the IPCC RP. I might have confused you by combining too many possibilities into one message. Let me know your exact requirements and I’d be glad to clean it up for you. *From:* Cristobal Priego [mailto:cristobalpri...@gmail.com] *Sent:* Thursday, July 23, 2009 2:44 PM *To:* Michael Ciarfello *Cc:* Jeffrey Hall; ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] UCCX and meet-me thank you, however the meet me needs to be initiated first, right and then the callers needs to authenticate or is there a way to authenticate the initiator only and then have the remains users just to join the bridge without authentication 2009/7/23 Michael Ciarfello mciarfe...@iplogic.com I answered the question he was asking at the time. Lol CRS script: You can play a prompt if you wish, you will use the get digit string to collect the password from the user then use the IF step to compare it to the accepted password, if they are equal then call redirect to your meetme number. If not then do what you want (hang up, re-prompt user, etc.) *From:* Jeffrey Hall [mailto:layer8...@gmail.com] *Sent:* Thursday, July 23, 2009 2:24 PM *To:* Michael Ciarfello *Cc:* Cristobal Priego; ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] UCCX and meet-me Now THERE's an expert response! lol On Thu, Jul 23, 2009 at 12:15 PM, Michael Ciarfello mciarfe...@iplogic.com wrote: Yes. *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Cristobal Priego *Sent:* Thursday, July 23, 2009 12:53 PM *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] UCCX and meet-me Hello Experts, I was wondering if you know how to use UCCX to request a password before a user is allowed to establish a meet-me conference call do you know if this is possible ? thanks Cris ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Jeffrey W. Hall Jeffrey W. Hall, LLC. CCSI #31661, CCVP, CCSP, CCIP, CCNP, CCDP, CQS, CCNA (V,S,W), MCT, MCITP (Cell) 901-490-4140 (Email) layer8...@gmail.com (Blog) http://layer8man.ccieblog.com Sent from Olive Branch, MS, United States ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCCX and meet-me
Yea, you can use Unity's supervised transfer or IPCC's consult (supervised) transfer. In IPCC, if the transfer fails, you can play a prompt to the caller with something intelligent (your conference bridge isn't setup right not, call back in a little while or call xxx-). I'm pretty sure you should be able to play the same in Unity also. From: Cristobal Priego [mailto:cristobalpri...@gmail.com] Sent: Thursday, July 23, 2009 3:23 PM To: Michael Ciarfello Cc: Jeffrey Hall; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] UCCX and meet-me nevermind I got it sorry 2009/7/23 Cristobal Priego cristobalpri...@gmail.commailto:cristobalpri...@gmail.com I think I got it Is there a way to do a consult transfer with the Script. that's the exact scenario that we want. once the meet-me is setup internally we want only to authenticate the callers dialing in from the pstn using the script. so i was thinking of a consult transfer just in case the meet-me isn't setup yet they won't get a fast busy and try to do something else. or should i just send the call to a call handler and do exactly what Christopher told me to do thanks Michael for your time 2009/7/23 Michael Ciarfello mciarfe...@iplogic.commailto:mciarfe...@iplogic.com The conference controller must manually and at his/her phone setup the meet-me. There's no authentication (pick up phone, more, meet-me, meet-me directory number). You may also have to design PTs and CSSs, possible translation patterns so that the conference controller can initiate the meet-me and the callers join. After the meet-me is setup, you want to only authenticate the callers dialing in. They will probably dial another number, go through the IPCC script, authenticate then the script transfers to the meet-me number. Your initiator and your IPCC ports will need to have access to the meet-me number. The callers / gateway(s) will need to have access to the IPCC RP. I might have confused you by combining too many possibilities into one message. Let me know your exact requirements and I'd be glad to clean it up for you. From: Cristobal Priego [mailto:cristobalpri...@gmail.commailto:cristobalpri...@gmail.com] Sent: Thursday, July 23, 2009 2:44 PM To: Michael Ciarfello Cc: Jeffrey Hall; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] UCCX and meet-me thank you, however the meet me needs to be initiated first, right and then the callers needs to authenticate or is there a way to authenticate the initiator only and then have the remains users just to join the bridge without authentication 2009/7/23 Michael Ciarfello mciarfe...@iplogic.commailto:mciarfe...@iplogic.com I answered the question he was asking at the time. Lol CRS script: You can play a prompt if you wish, you will use the get digit string to collect the password from the user then use the IF step to compare it to the accepted password, if they are equal then call redirect to your meetme number. If not then do what you want (hang up, re-prompt user, etc.) From: Jeffrey Hall [mailto:layer8...@gmail.commailto:layer8...@gmail.com] Sent: Thursday, July 23, 2009 2:24 PM To: Michael Ciarfello Cc: Cristobal Priego; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] UCCX and meet-me Now THERE's an expert response! lol On Thu, Jul 23, 2009 at 12:15 PM, Michael Ciarfello mciarfe...@iplogic.commailto:mciarfe...@iplogic.com wrote: Yes. From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cristobal Priego Sent: Thursday, July 23, 2009 12:53 PM To: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] UCCX and meet-me Hello Experts, I was wondering if you know how to use UCCX to request a password before a user is allowed to establish a meet-me conference call do you know if this is possible ? thanks Cris ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com -- Jeffrey W. Hall Jeffrey W. Hall, LLC. CCSI #31661, CCVP, CCSP, CCIP, CCNP, CCDP, CQS, CCNA (V,S,W), MCT, MCITP (Cell) 901-490-4140 (Email) layer8...@gmail.commailto:layer8...@gmail.com (Blog) http://layer8man.ccieblog.com Sent from Olive Branch, MS, United States ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP endpoints on the other side of an H.323 trunk...
I have found Inbound FastStart should fix the problem. Also check the codec within the voice register pool. Voice-class codec is not supported in the voice register pool so try using g711 end to end (voice register pool/outbound dial-peer/SIP Trunk DP). -- Vik Malhi CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Jonathan Charles jonv...@gmail.com Date: Thu, 23 Jul 2009 11:00:12 -0500 To: Padmanabhan, Padhu pa...@ti.com Cc: OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SIP endpoints on the other side of an H.323 trunk... On Thu, Jul 23, 2009 at 9:01 AM, Padmanabhan, Padhupa...@ti.com wrote: Can u pls post your configs?. Cheers,Padhu -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jonathan Charles Sent: Wednesday, July 22, 2009 6:21 PM To: ccie_voice@onlinestudylist.com; cisco-v...@puck.nether.net Subject: Re: [OSL | CCIE_Voice] SIP endpoints on the other side of an H.323 trunk... So, does anyone know if this is true? I am actually seeing that SIP cannot call phones across an H.323 trunk It looks like everything is fine, the call sets up, but then no audio cuts through and the SIP phone keeps ringing... And then I see this when I hang up the cucm phone Jul 22 23:19:48.514: //-1//SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 500 Internal Server Error Reason: Q.850;cause=16 Date: Wed, 22 Jul 2009 23:19:40 GMT From: Phone1 sip:3...@192.168.29.254;tag=00169def16e500dfdb252a73-c3694981 Allow-Events: telephone-event Content-Length: 0 To: sip:5...@192.168.29.254;tag=BCCA62C-BD3 Call-ID: 00169def-16e50019-a4734327-02d0c...@192.168.29.102 Via: SIP/2.0/UDP 192.168.29.102:5060;branch=z9hG4bKc0c82a6f CSeq: 101 INVITE Server: Cisco-SIPGateway/IOS-12.x Jul 22 23:19:48.522: //-1//SIP/Msg/ccsipDisplayMsg: Received: BR2#ACK sip:5...@192.168.29.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.29.102:5060;branch=z9hG4bKc0c82a6f From: Phone1 sip:3...@192.168.29.254;tag=00169def16e500dfdb252a73-c3694981 To: sip:5...@192.168.29.254;tag=BCCA62C-BD3 Call-ID: 00169def-16e50019-a4734327-02d0c...@192.168.29.102 Date: Wed, 22 Jul 2009 23:19:44 GMT CSeq: 101 ACK Content-Length: 0 On Wed, Jun 10, 2009 at 1:30 AM, Jonathan Charlesjonv...@gmail.com wrote: Found a weird statement in a Cisco doc: SIP endpoints are not supported on H.323 trunks. SIP endpoints are supported on SIP trunks only This was in the SRST System Admin guide under Octo-Lines... I thought the CUBE fixed this? Jonathan ___ 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] DTMF and MWI issue when using 7961 SIP with CUE7.0
Thanks Aamir! The DTMF part is working now but the MWI still doesn't work. From: Aamir Panjwani aamir.panjw...@ivision.com.au To: P N png_sanj...@yahoo.com; ccie_voice@onlinestudylist.com Sent: Wednesday, July 22, 2009 8:24:32 PM Subject: RE: [OSL | CCIE_Voice] DTMF and MWI issue when using 7961 SIP with CUE7.0 MWI: mwi sip sub-notify under ccn subsystem sip DTMF: dtmf-relay rtp-nte sip-notify under voice register pool From:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of P N Sent: Thursday, 23 July 2009 1:21 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] DTMF and MWI issue when using 7961 SIP with CUE7.0 Hi all, I tried to use a 7961 SIP phone to use with the CUE 7.0, I got the following dtmf and mwi issues: 1) The dtmf on the phone is not working when it is interacting with the CUE, what kind of config should be used for dtmf setting? 2) The MWI is not on on this phone when a message is left 3) I tried calling 39993003 to turn on MWI (3999 is my MWI ON DN) but nothing happen, if I call this MWI number on other SCCP phone, the MWI light can be turned on. Voicemail and MWI all working for other SCCP phones on the same CME 7.0. Any idea? Thanks Patrick Ng __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] vRack dsp modules question
We have 2 x PVDM2-16. There is no sharing when you configure a conference so if you had a single PVDM2-16 you would need to remove the pri-group and xcoders. As to what is in the lab- don¹t know but I imagine something 16. -- Vik Malhi CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Nara Shikamaru shikam...@kagadis.com Date: Wed, 22 Jul 2009 23:13:48 -0700 To: OSL Group ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] vRack dsp modules question I have a home lab that I use for after-hours practice with my IPexpert material and, in working through module 7 tonight, realized that my PVDM2-16 module in BR1 and BR2 routers were not be enough to set up a CFB session. My HQ router happened to have a PVDM2-32 module that allowed me to configure a conference bridge with a max session of 1 (which was enough for the exercise). Will PVDM2-32 modules be enough for the mock labs and modules? What modules do the vRacks ISRs have? -- -Shikamaru ___ 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] SIP endpoints on the other side of an H.323 trunk...
It is already 711... BR2#sh voice register dial-peers dial-peer voice 40001 voip destination-pattern 3005 session target ipv4:192.168.29.101:5060 session protocol sipv2 dtmf-relay rtp-nte sip-notify digit collect kpml codec g711ulaw bytes 160 call-fwd-mbox3005 call-fwd-busy3100 call-fwd-noan-timeou 12 call-fwd-noan3100 after-hours-exempt FALSE J On Thu, Jul 23, 2009 at 2:32 PM, Vik Malhivma...@ipexpert.com wrote: I have found Inbound FastStart should fix the problem. Also check the codec within the voice register pool. Voice-class codec is not supported in the voice register pool so try using g711 end to end (voice register pool/outbound dial-peer/SIP Trunk DP). -- Vik Malhi CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Jonathan Charles jonv...@gmail.com Date: Thu, 23 Jul 2009 11:00:12 -0500 To: Padmanabhan, Padhu pa...@ti.com Cc: OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SIP endpoints on the other side of an H.323 trunk... On Thu, Jul 23, 2009 at 9:01 AM, Padmanabhan, Padhupa...@ti.com wrote: Can u pls post your configs?. Cheers,Padhu -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jonathan Charles Sent: Wednesday, July 22, 2009 6:21 PM To: ccie_voice@onlinestudylist.com; cisco-v...@puck.nether.net Subject: Re: [OSL | CCIE_Voice] SIP endpoints on the other side of an H.323 trunk... So, does anyone know if this is true? I am actually seeing that SIP cannot call phones across an H.323 trunk It looks like everything is fine, the call sets up, but then no audio cuts through and the SIP phone keeps ringing... And then I see this when I hang up the cucm phone Jul 22 23:19:48.514: //-1//SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 500 Internal Server Error Reason: Q.850;cause=16 Date: Wed, 22 Jul 2009 23:19:40 GMT From: Phone1 sip:3...@192.168.29.254;tag=00169def16e500dfdb252a73-c3694981 Allow-Events: telephone-event Content-Length: 0 To: sip:5...@192.168.29.254;tag=BCCA62C-BD3 Call-ID: 00169def-16e50019-a4734327-02d0c...@192.168.29.102 Via: SIP/2.0/UDP 192.168.29.102:5060;branch=z9hG4bKc0c82a6f CSeq: 101 INVITE Server: Cisco-SIPGateway/IOS-12.x Jul 22 23:19:48.522: //-1//SIP/Msg/ccsipDisplayMsg: Received: BR2#ACK sip:5...@192.168.29.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.29.102:5060;branch=z9hG4bKc0c82a6f From: Phone1 sip:3...@192.168.29.254;tag=00169def16e500dfdb252a73-c3694981 To: sip:5...@192.168.29.254;tag=BCCA62C-BD3 Call-ID: 00169def-16e50019-a4734327-02d0c...@192.168.29.102 Date: Wed, 22 Jul 2009 23:19:44 GMT CSeq: 101 ACK Content-Length: 0 On Wed, Jun 10, 2009 at 1:30 AM, Jonathan Charlesjonv...@gmail.com wrote: Found a weird statement in a Cisco doc: SIP endpoints are not supported on H.323 trunks. SIP endpoints are supported on SIP trunks only This was in the SRST System Admin guide under Octo-Lines... I thought the CUBE fixed this? Jonathan ___ 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] Documentation at voice lab exam
No response yet. 3rd time's a charm. Anyone? Thanks, Kevin Damisch Vital Support Systems -Original Message- From: Kevin Damisch Sent: Thursday, July 16, 2009 12:23 AM To: OSL Group Subject: RE: Documentation at voice lab exam Was this mentioned at Networkers? Thanks, Kevin -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kevin Damisch Sent: Friday, July 10, 2009 8:33 PM To: OSL Group Subject: [OSL | CCIE_Voice] Documentation at voice lab exam Has there been any confirmation of what documentation is available during the lab? I've heard mixed comments about it only being the config guides and SRNDs. But, then others have mentioned that the technotes and configuration examples documents are available as well. I'm taking it in San Jose. Thanks, Kevin This communication (including any attachments) is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If you are not the intended recipient, any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify Vital Support Systems at 515 334 5700 and delete or destroy all copies and the original document. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Documentation at voice lab exam
Cisco product documentation is not available on the web in the lab anymore? Respectfully Charles George Date: Thu, 23 Jul 2009 16:54:14 -0400 From: earl.ho...@sarcom.com To: kevin.dami...@vitalsite.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Documentation at voice lab exam At Networkers in the CCIE Voice techtorial, there were 4 docs which were mentioned as being available on the candidate PC's desktop. I believe the 4 are: Unified Communications 7.0 SRND Enterprise QOS SRND Unified Communications Manager Express SRND -- not 100% sure on this one Unified Contact Center Express 7.0 SRND -- not 100% sure on this one Earl Hough CCIE #16508 (RS/Security) -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kevin Damisch Sent: Thursday, July 23, 2009 3:52 PM To: OSL Group Subject: Re: [OSL | CCIE_Voice] Documentation at voice lab exam No response yet. 3rd time's a charm. Anyone? Thanks, Kevin Damisch Vital Support Systems -Original Message- From: Kevin Damisch Sent: Thursday, July 16, 2009 12:23 AM To: OSL Group Subject: RE: Documentation at voice lab exam Was this mentioned at Networkers? Thanks, Kevin -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kevin Damisch Sent: Friday, July 10, 2009 8:33 PM To: OSL Group Subject: [OSL | CCIE_Voice] Documentation at voice lab exam Has there been any confirmation of what documentation is available during the lab? I've heard mixed comments about it only being the config guides and SRNDs. But, then others have mentioned that the technotes and configuration examples documents are available as well. I'm taking it in San Jose. Thanks, Kevin This communication (including any attachments) is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If you are not the intended recipient, any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify Vital Support Systems at 515 334 5700 and delete or destroy all copies and the original document. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ The information contained in this transmission is confidential. It is intended solely for the use of the individual(s) or organization(s) to whom it is addressed. Any disclosure, copying or further distribution is not permitted unless such privilege is explicitly granted in writing by SARCOM, Inc. Furthermore, SARCOM, Inc. is not responsible for the proper and complete transmission of the substance of this communication, nor for any delay in its receipt. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ NEW mobile Hotmail. Optimized for YOUR phone. Click here. http://windowslive.com/Mobile?ocid=TXT_TAGLM_WL_CS_MB_new_hotmail_072009___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Documentation at voice lab exam
The web equivalents of the old Doc CD are available, but global search from the main website I believe has been disabled. The docs I mentioned are the ones which are directly accessible from the desktop. The other documentation has to be navigated through the web site. From: c george [mailto:cisco...@hotmail.com] Sent: Thursday, July 23, 2009 4:57 PM To: Hough, Earl; kevin.dami...@vitalsite.com; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Documentation at voice lab exam Cisco product documentation is not available on the web in the lab anymore? Respectfully Charles George Date: Thu, 23 Jul 2009 16:54:14 -0400 From: earl.ho...@sarcom.com To: kevin.dami...@vitalsite.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Documentation at voice lab exam At Networkers in the CCIE Voice techtorial, there were 4 docs which were mentioned as being available on the candidate PC's desktop. I believe the 4 are: Unified Communications 7.0 SRND Enterprise QOS SRND Unified Communications Manager Express SRND -- not 100% sure on this one Unified Contact Center Express 7.0 SRND -- not 100% sure on this one Earl Hough CCIE #16508 (RS/Security) -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kevin Damisch Sent: Thursday, July 23, 2009 3:52 PM To: OSL Group Subject: Re: [OSL | CCIE_Voice] Documentation at voice lab exam No response yet. 3rd time's a charm. Anyone? Thanks, Kevin Damisch Vital Support Systems -Original Message- From: Kevin Damisch Sent: Thursday, July 16, 2009 12:23 AM To: OSL Group Subject: RE: Documentation at voice lab exam Was this mentioned at Networkers? Thanks, Kevin -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kevin Damisch Sent: Friday, July 10, 2009 8:33 PM To: OSL Group Subject: [OSL | CCIE_Voice] Documentation at voice lab exam Has there been any confirmation of what documentation is available during the lab? I've heard mixed comments about it only being the config guides and SRNDs. But, then others have mentioned that the technotes and configuration examples documents are available as well. I'm taking it in San Jose. Thanks, Kevin This communication (including any attachments) is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If you are not the intended recipient, any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify Vital Support Systems at 515 334 5700 and delete or destroy all copies and the original document. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ The information contained in this transmission is confidential. It is intended solely for the use of the individual(s) or organization(s) to whom it is addressed. Any disclosure, copying or further distribution is not permitted unless such privilege is explicitly granted in writing by SARCOM, Inc. Furthermore, SARCOM, Inc. is not responsible for the proper and complete transmission of the substance of this communication, nor for any delay in its receipt. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com NEW mobile Hotmail. Optimized for YOUR phone. Click here. http://windowslive.com/Mobile?ocid=TXT_TAGLM_WL_CS_MB_new_hotmail_07200 9 _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ The information contained in this transmission is confidential. It is intended solely for the use of the individual(s) or organization(s) to whom it is addressed. Any disclosure, copying or further distribution is not permitted unless such privilege is explicitly granted in writing by SARCOM, Inc. Furthermore, SARCOM, Inc. is not responsible for the proper and complete transmission of the substance of this communication, nor for any delay in its receipt. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Documentation at voice lab exam
gotcha Respectfully Charles George Subject: RE: [OSL | CCIE_Voice] Documentation at voice lab exam Date: Thu, 23 Jul 2009 16:59:12 -0400 From: earl.ho...@sarcom.com To: cisco...@hotmail.com; kevin.dami...@vitalsite.com; ccie_voice@onlinestudylist.com The web equivalents of the old Doc CD are available, but global search from the main website I believe has been disabled. The docs I mentioned are the ones which are directly accessible from the desktop. The other documentation has to be navigated through the web site. From: c george [mailto:cisco...@hotmail.com] Sent: Thursday, July 23, 2009 4:57 PM To: Hough, Earl; kevin.dami...@vitalsite.com; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Documentation at voice lab exam Cisco product documentation is not available on the web in the lab anymore? Respectfully Charles George Date: Thu, 23 Jul 2009 16:54:14 -0400 From: earl.ho...@sarcom.com To: kevin.dami...@vitalsite.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Documentation at voice lab exam At Networkers in the CCIE Voice techtorial, there were 4 docs which were mentioned as being available on the candidate PC's desktop. I believe the 4 are: Unified Communications 7.0 SRND Enterprise QOS SRND Unified Communications Manager Express SRND -- not 100% sure on this one Unified Contact Center Express 7.0 SRND -- not 100% sure on this one Earl Hough CCIE #16508 (RS/Security) -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kevin Damisch Sent: Thursday, July 23, 2009 3:52 PM To: OSL Group Subject: Re: [OSL | CCIE_Voice] Documentation at voice lab exam No response yet. 3rd time's a charm. Anyone? Thanks, Kevin Damisch Vital Support Systems -Original Message- From: Kevin Damisch Sent: Thursday, July 16, 2009 12:23 AM To: OSL Group Subject: RE: Documentation at voice lab exam Was this mentioned at Networkers? Thanks, Kevin -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kevin Damisch Sent: Friday, July 10, 2009 8:33 PM To: OSL Group Subject: [OSL | CCIE_Voice] Documentation at voice lab exam Has there been any confirmation of what documentation is available during the lab? I've heard mixed comments about it only being the config guides and SRNDs. But, then others have mentioned that the technotes and configuration examples documents are available as well. I'm taking it in San Jose. Thanks, Kevin This communication (including any attachments) is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If you are not the intended recipient, any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify Vital Support Systems at 515 334 5700 and delete or destroy all copies and the original document. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ The information contained in this transmission is confidential. It is intended solely for the use of the individual(s) or organization(s) to whom it is addressed. Any disclosure, copying or further distribution is not permitted unless such privilege is explicitly granted in writing by SARCOM, Inc. Furthermore, SARCOM, Inc. is not responsible for the proper and complete transmission of the substance of this communication, nor for any delay in its receipt. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com NEW mobile Hotmail. Optimized for YOUR phone. Click here. _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ The information contained in this transmission is confidential. It is intended solely for the use of the individual(s) or organization(s) to whom it is addressed. Any disclosure, copying or further distribution is not permitted unless such privilege is explicitly granted in writing by SARCOM, Inc. Furthermore, SARCOM, Inc. is not responsible for the proper and complete transmission of the substance of this communication, nor for any delay in its receipt. _ Bing™ brings you maps, menus, and reviews organized in one place. Try it now.
Re: [OSL | CCIE_Voice] Calling Party Tranformation Patterns
This is going to sound dumb, but I think I got the calling party transformations figured out, except for retaining the 4-digit ANI between sites... how do I do that? J On Wed, Jul 22, 2009 at 11:19 PM, Michael Ciarfellomciarfe...@iplogic.com wrote: Here is my testing summary for Route Pattern, Route List and Transformation Pattern transformations. Compared against 7.0(1) and 7.1(2a). We are testing 7.1(2a) to see if behavior changed (if there was a possible bug in 7.0(1). The CSS for the xform pattern (HQ_XFORM_CSS) only contained one PT (HQ_XFORM_PT) for the xform pattern. I have a 5-digit dial plan and don't use IPExpert's. Sorry for any intended confusion. I probably should switch back to using theirs. I just wanted a more complex dialplan. My dialplan has conflicting and overlapping numbers (45XXX for HQ, 64XXX for BR1 and 44XXX for BR2) It also follows some real cisco numbers in NYC, San Jose and Tokyo. 7.0(1) SIP: HQ phone 1 (45001) dials 9.19001234567 Prefix digits on the route pattern (111) and prefix digits on the RL (222) Calling Transform Pattern 45XXX with prefix digits 333 PSTN Phone shows 22245001. Doesn't hit the transform pattern and uses the RL transformation (which overwrites the route pattern transformation.) The calling party number in the trace file shows 11145001. So I changed the transform pattern to 11145XXX Now the PSTN phone shows 33311145001. I would intrepret this as the route pattern's prefix got prepended, then the transform pattern matched and prepended the 333 to the resulting transformation of the route pattern. 07/18/2009 01:30:06.202 CCM|SPROC DATransformMatch - matchNumber [11145001] transformCSSPkid [45ece7a3-264c-0e86-4c9a-b7c9f108dfc8] transformationCss [HQ_911_XFORM_PT] patternUsage [15] paternNodeID [42cf1e0b-a2c0-6d54-5067-195be977104a] OutpulsedNum.nd [33311145001] tn [0] pi [1] npi [0]|CLID::StandAloneClusterNID::142.6.64.12LVL::ArbitraryMASK::0800 -- 7.0(1) MGCP/H323: BR1 Phone 2 (64002) dials 9.19001234567 Prefix digits on the route pattern (444) and prefix digits on the RL (555) Calling Transform Pattern 64XXX with prefix digits 666 PSTN phone shows 4002. So this hits the transform pattern. Trace file shows it. 07/18/2009 01:34:39.719 CCM|SPROC DATransformMatch - matchNumber [64002] transformCSSPkid [aca1739f-ec61-1f46-87e1-6bd991d15678] transformationCss [BR1_911_XFORM_PT] patternUsage [15] paternNodeID [795003d7-ea44-9cb8-e450-e41ebca818d3] OutpulsedNum.nd [4002] tn [0] pi [1] npi [0]|CLID::StandAloneClusterNID::142.6.64.12LVL::ArbitraryMASK::0800 The calling party number in the trace files shows 44464002. (I didn't paste it) So I changed the transform pattern to 44464XXX Now the PSTN phone shows 55564002. It didn't go through the transform pattern. Becasue there's no patternNodeID??? Anyways, we took the correct RL transformation. 07/18/2009 01:25:32.261 CCM|SPROC DATransformMatch - matchNumber [64002] transformCSSPkid [aca1739f-ec61-1f46-87e1-6bd991d15678] transformationCss [BR1_911_XFORM_PT] patternUsage [2] paternNodeID [] OutpulsedNum.nd [55564002] tn [0] pi [1] npi [0]|CLID::StandAloneClusterNID::142.6.64.12LVL::ArbitraryMASK::0800 Two different behaviors for two different trunk types. Let's see what 7.1(2a) does. We repeat the experiment. == 7.1(2a) SIP: HQ phone 1 (45001) dials 9.19001234567 Prefix digits on the route pattern (111) and prefix digits on the RL (222) Calling Transform Pattern 45XXX with prefix digits 333 PSTN Phone shows 33345001, so uses the transform. The calling party number in the trace files shows 11145001. So I changed the transform pattern to 11145XXX Now the PSTN phone shows 22245001. Skips (didn't match) the xform patt and uses the RL transform 7.1(2a) MGCP/H323: BR1 Phone 2 (64002) dials 9.19001234567 Prefix digits on the route pattern (444) and prefix digits on the RL (555) Calling Transform Pattern 64XXX with prefix digits 666 PSTN phone shows 4002 The calling party number in the trace files shows 55564002. So I changed the transform pattern to 55564XXX Now the PSTN phone shows 55564002. Skips (didn't match) the xform patt and uses the RL transform So 7.1(2a) seems to make sense and has consistent behavior for different trunk types. - NOW, the translation pattern trick is also different. I didn't document it here for 7.0(1). HQ Phone 1 dialing through the translation pattern trick and the PSTN phone shows 33345001. Trace files showed CallingPartyNumber=11145001, so we change the xform pattern to 11145XXX. The PSTN phone now shows 22245001 If add a prefix (777) on the translation pattern, (xform pattern is back to 45XXX) get 22277745001. Trace files show: 07/23/2009 00:05:38.768 CCM||PretransformCallingPartyNumber=77745001 |CallingPartyNumber=11177745001 Change the xform pattern to 77745XXX get 33377745001 Change the xform pattern to 11177745XXX get
Re: [OSL | CCIE_Voice] SIP endpoints on the other side of an H.323 trunk...
Enable MTP on the SIP trunk. -- Vik Malhi CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Jonathan Charles jonv...@gmail.com Date: Thu, 23 Jul 2009 14:51:00 -0500 To: Vik Malhi vma...@ipexpert.com Cc: Padmanabhan, Padhu pa...@ti.com, OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SIP endpoints on the other side of an H.323 trunk... It is already 711... BR2#sh voice register dial-peers dial-peer voice 40001 voip destination-pattern 3005 session target ipv4:192.168.29.101:5060 session protocol sipv2 dtmf-relay rtp-nte sip-notify digit collect kpml codec g711ulaw bytes 160 call-fwd-mbox3005 call-fwd-busy3100 call-fwd-noan-timeou 12 call-fwd-noan3100 after-hours-exempt FALSE J On Thu, Jul 23, 2009 at 2:32 PM, Vik Malhivma...@ipexpert.com wrote: I have found Inbound FastStart should fix the problem. Also check the codec within the voice register pool. Voice-class codec is not supported in the voice register pool so try using g711 end to end (voice register pool/outbound dial-peer/SIP Trunk DP). -- Vik Malhi CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Jonathan Charles jonv...@gmail.com Date: Thu, 23 Jul 2009 11:00:12 -0500 To: Padmanabhan, Padhu pa...@ti.com Cc: OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SIP endpoints on the other side of an H.323 trunk... On Thu, Jul 23, 2009 at 9:01 AM, Padmanabhan, Padhupa...@ti.com wrote: Can u pls post your configs?. Cheers,Padhu -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jonathan Charles Sent: Wednesday, July 22, 2009 6:21 PM To: ccie_voice@onlinestudylist.com; cisco-v...@puck.nether.net Subject: Re: [OSL | CCIE_Voice] SIP endpoints on the other side of an H.323 trunk... So, does anyone know if this is true? I am actually seeing that SIP cannot call phones across an H.323 trunk It looks like everything is fine, the call sets up, but then no audio cuts through and the SIP phone keeps ringing... And then I see this when I hang up the cucm phone Jul 22 23:19:48.514: //-1//SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 500 Internal Server Error Reason: Q.850;cause=16 Date: Wed, 22 Jul 2009 23:19:40 GMT From: Phone1 sip:3...@192.168.29.254;tag=00169def16e500dfdb252a73-c3694981 Allow-Events: telephone-event Content-Length: 0 To: sip:5...@192.168.29.254;tag=BCCA62C-BD3 Call-ID: 00169def-16e50019-a4734327-02d0c...@192.168.29.102 Via: SIP/2.0/UDP 192.168.29.102:5060;branch=z9hG4bKc0c82a6f CSeq: 101 INVITE Server: Cisco-SIPGateway/IOS-12.x Jul 22 23:19:48.522: //-1//SIP/Msg/ccsipDisplayMsg: Received: BR2#ACK sip:5...@192.168.29.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.29.102:5060;branch=z9hG4bKc0c82a6f From: Phone1 sip:3...@192.168.29.254;tag=00169def16e500dfdb252a73-c3694981 To: sip:5...@192.168.29.254;tag=BCCA62C-BD3 Call-ID: 00169def-16e50019-a4734327-02d0c...@192.168.29.102 Date: Wed, 22 Jul 2009 23:19:44 GMT CSeq: 101 ACK Content-Length: 0 On Wed, Jun 10, 2009 at 1:30 AM, Jonathan Charlesjonv...@gmail.com wrote: Found a weird statement in a Cisco doc: SIP endpoints are not supported on H.323 trunks. SIP endpoints are supported on SIP trunks only This was in the SRST System Admin guide under Octo-Lines... I thought the CUBE fixed this? Jonathan ___ 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] Calling Party Tranformation Patterns
Phones should NOT have use Calling Party Transformation Pattern CSS checked. -- Vik Malhi CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Jonathan Charles jonv...@gmail.com Date: Thu, 23 Jul 2009 16:04:14 -0500 To: Michael Ciarfello mciarfe...@iplogic.com Cc: OSL Group ccie_voice@onlinestudylist.com, Otto Sanchez otto.sanc...@daxa.com.ve Subject: Re: [OSL | CCIE_Voice] Calling Party Tranformation Patterns This is going to sound dumb, but I think I got the calling party transformations figured out, except for retaining the 4-digit ANI between sites... how do I do that? J On Wed, Jul 22, 2009 at 11:19 PM, Michael Ciarfellomciarfe...@iplogic.com wrote: Here is my testing summary for Route Pattern, Route List and Transformation Pattern transformations. Compared against 7.0(1) and 7.1(2a). We are testing 7.1(2a) to see if behavior changed (if there was a possible bug in 7.0(1). The CSS for the xform pattern (HQ_XFORM_CSS) only contained one PT (HQ_XFORM_PT) for the xform pattern. I have a 5-digit dial plan and don't use IPExpert's. Sorry for any intended confusion. I probably should switch back to using theirs. I just wanted a more complex dialplan. My dialplan has conflicting and overlapping numbers (45XXX for HQ, 64XXX for BR1 and 44XXX for BR2) It also follows some real cisco numbers in NYC, San Jose and Tokyo. 7.0(1) SIP: HQ phone 1 (45001) dials 9.19001234567 Prefix digits on the route pattern (111) and prefix digits on the RL (222) Calling Transform Pattern 45XXX with prefix digits 333 PSTN Phone shows 22245001. Doesn't hit the transform pattern and uses the RL transformation (which overwrites the route pattern transformation.) The calling party number in the trace file shows 11145001. So I changed the transform pattern to 11145XXX Now the PSTN phone shows 33311145001. I would intrepret this as the route pattern's prefix got prepended, then the transform pattern matched and prepended the 333 to the resulting transformation of the route pattern. 07/18/2009 01:30:06.202 CCM|SPROC DATransformMatch - matchNumber [11145001] transformCSSPkid [45ece7a3-264c-0e86-4c9a-b7c9f108dfc8] transformationCss [HQ_911_XFORM_PT] patternUsage [15] paternNodeID [42cf1e0b-a2c0-6d54-5067-195be977104a] OutpulsedNum.nd [33311145001] tn [0] pi [1] npi [0]|CLID::StandAloneClusterNID::142.6.64.12LVL::ArbitraryMASK::0800 -- 7.0(1) MGCP/H323: BR1 Phone 2 (64002) dials 9.19001234567 Prefix digits on the route pattern (444) and prefix digits on the RL (555) Calling Transform Pattern 64XXX with prefix digits 666 PSTN phone shows 4002. So this hits the transform pattern. Trace file shows it. 07/18/2009 01:34:39.719 CCM|SPROC DATransformMatch - matchNumber [64002] transformCSSPkid [aca1739f-ec61-1f46-87e1-6bd991d15678] transformationCss [BR1_911_XFORM_PT] patternUsage [15] paternNodeID [795003d7-ea44-9cb8-e450-e41ebca818d3] OutpulsedNum.nd [4002] tn [0] pi [1] npi [0]|CLID::StandAloneClusterNID::142.6.64.12LVL::ArbitraryMASK::0800 The calling party number in the trace files shows 44464002. (I didn't paste it) So I changed the transform pattern to 44464XXX Now the PSTN phone shows 55564002. It didn't go through the transform pattern. Becasue there's no patternNodeID??? Anyways, we took the correct RL transformation. 07/18/2009 01:25:32.261 CCM|SPROC DATransformMatch - matchNumber [64002] transformCSSPkid [aca1739f-ec61-1f46-87e1-6bd991d15678] transformationCss [BR1_911_XFORM_PT] patternUsage [2] paternNodeID [] OutpulsedNum.nd [55564002] tn [0] pi [1] npi [0]|CLID::StandAloneClusterNID::142.6.64.12LVL::ArbitraryMASK::0800 Two different behaviors for two different trunk types. Let's see what 7.1(2a) does. We repeat the experiment. == 7.1(2a) SIP: HQ phone 1 (45001) dials 9.19001234567 Prefix digits on the route pattern (111) and prefix digits on the RL (222) Calling Transform Pattern 45XXX with prefix digits 333 PSTN Phone shows 33345001, so uses the transform. The calling party number in the trace files shows 11145001. So I changed the transform pattern to 11145XXX Now the PSTN phone shows 22245001. Skips (didn't match) the xform patt and uses the RL transform 7.1(2a) MGCP/H323: BR1 Phone 2 (64002) dials 9.19001234567 Prefix digits on the route pattern (444) and prefix digits on the RL (555) Calling Transform Pattern 64XXX with prefix digits 666 PSTN phone shows 4002 The calling party number in the trace files shows 55564002. So I changed the
Re: [OSL | CCIE_Voice] SIP endpoints on the other side of an H.323 trunk...
MTP is required is checked on the H.323 trunk... this is not a SIP trunk... J On Thu, Jul 23, 2009 at 4:17 PM, Vik Malhivma...@ipexpert.com wrote: Enable MTP on the SIP trunk. -- Vik Malhi CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Jonathan Charles jonv...@gmail.com Date: Thu, 23 Jul 2009 14:51:00 -0500 To: Vik Malhi vma...@ipexpert.com Cc: Padmanabhan, Padhu pa...@ti.com, OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SIP endpoints on the other side of an H.323 trunk... It is already 711... BR2#sh voice register dial-peers dial-peer voice 40001 voip destination-pattern 3005 session target ipv4:192.168.29.101:5060 session protocol sipv2 dtmf-relay rtp-nte sip-notify digit collect kpml codec g711ulaw bytes 160 call-fwd-mbox 3005 call-fwd-busy 3100 call-fwd-noan-timeou 12 call-fwd-noan 3100 after-hours-exempt FALSE J On Thu, Jul 23, 2009 at 2:32 PM, Vik Malhivma...@ipexpert.com wrote: I have found Inbound FastStart should fix the problem. Also check the codec within the voice register pool. Voice-class codec is not supported in the voice register pool so try using g711 end to end (voice register pool/outbound dial-peer/SIP Trunk DP). -- Vik Malhi CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Jonathan Charles jonv...@gmail.com Date: Thu, 23 Jul 2009 11:00:12 -0500 To: Padmanabhan, Padhu pa...@ti.com Cc: OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SIP endpoints on the other side of an H.323 trunk... On Thu, Jul 23, 2009 at 9:01 AM, Padmanabhan, Padhupa...@ti.com wrote: Can u pls post your configs?. Cheers,Padhu -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jonathan Charles Sent: Wednesday, July 22, 2009 6:21 PM To: ccie_voice@onlinestudylist.com; cisco-v...@puck.nether.net Subject: Re: [OSL | CCIE_Voice] SIP endpoints on the other side of an H.323 trunk... So, does anyone know if this is true? I am actually seeing that SIP cannot call phones across an H.323 trunk It looks like everything is fine, the call sets up, but then no audio cuts through and the SIP phone keeps ringing... And then I see this when I hang up the cucm phone Jul 22 23:19:48.514: //-1//SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 500 Internal Server Error Reason: Q.850;cause=16 Date: Wed, 22 Jul 2009 23:19:40 GMT From: Phone1 sip:3...@192.168.29.254;tag=00169def16e500dfdb252a73-c3694981 Allow-Events: telephone-event Content-Length: 0 To: sip:5...@192.168.29.254;tag=BCCA62C-BD3 Call-ID: 00169def-16e50019-a4734327-02d0c...@192.168.29.102 Via: SIP/2.0/UDP 192.168.29.102:5060;branch=z9hG4bKc0c82a6f CSeq: 101 INVITE Server: Cisco-SIPGateway/IOS-12.x Jul 22 23:19:48.522: //-1//SIP/Msg/ccsipDisplayMsg: Received: BR2#ACK sip:5...@192.168.29.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.29.102:5060;branch=z9hG4bKc0c82a6f From: Phone1 sip:3...@192.168.29.254;tag=00169def16e500dfdb252a73-c3694981 To: sip:5...@192.168.29.254;tag=BCCA62C-BD3 Call-ID: 00169def-16e50019-a4734327-02d0c...@192.168.29.102 Date: Wed, 22 Jul 2009 23:19:44 GMT CSeq: 101 ACK Content-Length: 0 On Wed, Jun 10, 2009 at 1:30 AM, Jonathan Charlesjonv...@gmail.com wrote: Found a weird statement in a Cisco doc: SIP endpoints are not supported on H.323 trunks. SIP endpoints are supported on SIP trunks only This was in the SRST System Admin guide under Octo-Lines... I thought the CUBE fixed this? Jonathan ___ 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] SIP endpoints on the other side of an H.323 trunk...
Ok- make sure that you are using an MTP which is NOT the RSVP Call Agent. Give the SIP Trunk an MRGL which contains the HQ Xcoder but not the RSVP-enabled MTP. -- Vik Malhi CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Jonathan Charles jonv...@gmail.com Date: Thu, 23 Jul 2009 16:18:34 -0500 To: Vik Malhi vma...@ipexpert.com Cc: Padmanabhan, Padhu pa...@ti.com, OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SIP endpoints on the other side of an H.323 trunk... MTP is required is checked on the H.323 trunk... this is not a SIP trunk... J On Thu, Jul 23, 2009 at 4:17 PM, Vik Malhivma...@ipexpert.com wrote: Enable MTP on the SIP trunk. -- Vik Malhi CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Jonathan Charles jonv...@gmail.com Date: Thu, 23 Jul 2009 14:51:00 -0500 To: Vik Malhi vma...@ipexpert.com Cc: Padmanabhan, Padhu pa...@ti.com, OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SIP endpoints on the other side of an H.323 trunk... It is already 711... BR2#sh voice register dial-peers dial-peer voice 40001 voip destination-pattern 3005 session target ipv4:192.168.29.101:5060 session protocol sipv2 dtmf-relay rtp-nte sip-notify digit collect kpml codec g711ulaw bytes 160 call-fwd-mbox 3005 call-fwd-busy 3100 call-fwd-noan-timeou 12 call-fwd-noan 3100 after-hours-exempt FALSE J On Thu, Jul 23, 2009 at 2:32 PM, Vik Malhivma...@ipexpert.com wrote: I have found Inbound FastStart should fix the problem. Also check the codec within the voice register pool. Voice-class codec is not supported in the voice register pool so try using g711 end to end (voice register pool/outbound dial-peer/SIP Trunk DP). -- Vik Malhi CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Jonathan Charles jonv...@gmail.com Date: Thu, 23 Jul 2009 11:00:12 -0500 To: Padmanabhan, Padhu pa...@ti.com Cc: OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SIP endpoints on the other side of an H.323 trunk... On Thu, Jul 23, 2009 at 9:01 AM, Padmanabhan, Padhupa...@ti.com wrote: Can u pls post your configs?. Cheers,Padhu -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jonathan Charles Sent: Wednesday, July 22, 2009 6:21 PM To: ccie_voice@onlinestudylist.com; cisco-v...@puck.nether.net Subject: Re: [OSL | CCIE_Voice] SIP endpoints on the other side of an H.323 trunk... So, does anyone know if this is true? I am actually seeing that SIP cannot call phones across an H.323 trunk It looks like everything is fine, the call sets up, but then no audio cuts through and the SIP phone keeps ringing... And then I see this when I hang up the cucm phone Jul 22 23:19:48.514: //-1//SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 500 Internal Server Error Reason: Q.850;cause=16 Date: Wed, 22 Jul 2009 23:19:40 GMT From: Phone1 sip:3...@192.168.29.254;tag=00169def16e500dfdb252a73-c3694981 Allow-Events: telephone-event Content-Length: 0 To: sip:5...@192.168.29.254;tag=BCCA62C-BD3 Call-ID: 00169def-16e50019-a4734327-02d0c...@192.168.29.102 Via: SIP/2.0/UDP 192.168.29.102:5060;branch=z9hG4bKc0c82a6f CSeq: 101 INVITE Server: Cisco-SIPGateway/IOS-12.x Jul 22 23:19:48.522: //-1//SIP/Msg/ccsipDisplayMsg: Received: BR2#ACK sip:5...@192.168.29.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.29.102:5060;branch=z9hG4bKc0c82a6f From: Phone1 sip:3...@192.168.29.254;tag=00169def16e500dfdb252a73-c3694981 To:
Re: [OSL | CCIE_Voice] SIP endpoints on the other side of an H.323 trunk...
Ok- make sure that you are using an MTP which is NOT the RSVP Call Agent. Give the H323 Trunk an MRGL which contains the HQ Xcoder but not the RSVP-enabled MTP. Yeah- not SIP Trunk- my mistake. If you are not using RSVP Locations CAC then obviously this response will not help you out. -- Vik Malhi CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Jonathan Charles jonv...@gmail.com Date: Thu, 23 Jul 2009 16:18:34 -0500 To: Vik Malhi vma...@ipexpert.com Cc: Padmanabhan, Padhu pa...@ti.com, OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SIP endpoints on the other side of an H.323 trunk... MTP is required is checked on the H.323 trunk... this is not a SIP trunk... J On Thu, Jul 23, 2009 at 4:17 PM, Vik Malhivma...@ipexpert.com wrote: Enable MTP on the SIP trunk. -- Vik Malhi CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Jonathan Charles jonv...@gmail.com Date: Thu, 23 Jul 2009 14:51:00 -0500 To: Vik Malhi vma...@ipexpert.com Cc: Padmanabhan, Padhu pa...@ti.com, OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SIP endpoints on the other side of an H.323 trunk... It is already 711... BR2#sh voice register dial-peers dial-peer voice 40001 voip destination-pattern 3005 session target ipv4:192.168.29.101:5060 session protocol sipv2 dtmf-relay rtp-nte sip-notify digit collect kpml codec g711ulaw bytes 160 call-fwd-mbox 3005 call-fwd-busy 3100 call-fwd-noan-timeou 12 call-fwd-noan 3100 after-hours-exempt FALSE J On Thu, Jul 23, 2009 at 2:32 PM, Vik Malhivma...@ipexpert.com wrote: I have found Inbound FastStart should fix the problem. Also check the codec within the voice register pool. Voice-class codec is not supported in the voice register pool so try using g711 end to end (voice register pool/outbound dial-peer/SIP Trunk DP). -- Vik Malhi CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Jonathan Charles jonv...@gmail.com Date: Thu, 23 Jul 2009 11:00:12 -0500 To: Padmanabhan, Padhu pa...@ti.com Cc: OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SIP endpoints on the other side of an H.323 trunk... On Thu, Jul 23, 2009 at 9:01 AM, Padmanabhan, Padhupa...@ti.com wrote: Can u pls post your configs?. Cheers,Padhu -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jonathan Charles Sent: Wednesday, July 22, 2009 6:21 PM To: ccie_voice@onlinestudylist.com; cisco-v...@puck.nether.net Subject: Re: [OSL | CCIE_Voice] SIP endpoints on the other side of an H.323 trunk... So, does anyone know if this is true? I am actually seeing that SIP cannot call phones across an H.323 trunk It looks like everything is fine, the call sets up, but then no audio cuts through and the SIP phone keeps ringing... And then I see this when I hang up the cucm phone Jul 22 23:19:48.514: //-1//SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 500 Internal Server Error Reason: Q.850;cause=16 Date: Wed, 22 Jul 2009 23:19:40 GMT From: Phone1 sip:3...@192.168.29.254;tag=00169def16e500dfdb252a73-c3694981 Allow-Events: telephone-event Content-Length: 0 To: sip:5...@192.168.29.254;tag=BCCA62C-BD3 Call-ID: 00169def-16e50019-a4734327-02d0c...@192.168.29.102 Via: SIP/2.0/UDP 192.168.29.102:5060;branch=z9hG4bKc0c82a6f CSeq: 101 INVITE Server: Cisco-SIPGateway/IOS-12.x Jul 22 23:19:48.522: //-1//SIP/Msg/ccsipDisplayMsg: Received: BR2#ACK sip:5...@192.168.29.254 SIP/2.0 Via: SIP/2.0/UDP
Re: [OSL | CCIE_Voice] SIP endpoints on the other side of an H.323 trunk...
Now, when HQ answers, the call just drops Jonathan On Thu, Jul 23, 2009 at 4:32 PM, Vik Malhivma...@ipexpert.com wrote: Ok- make sure that you are using an MTP which is NOT the RSVP Call Agent. Give the H323 Trunk an MRGL which contains the HQ Xcoder but not the RSVP-enabled MTP. Yeah- not SIP Trunk- my mistake. If you are not using RSVP Locations CAC then obviously this response will not help you out. -- Vik Malhi CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Jonathan Charles jonv...@gmail.com Date: Thu, 23 Jul 2009 16:18:34 -0500 To: Vik Malhi vma...@ipexpert.com Cc: Padmanabhan, Padhu pa...@ti.com, OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SIP endpoints on the other side of an H.323 trunk... MTP is required is checked on the H.323 trunk... this is not a SIP trunk... J On Thu, Jul 23, 2009 at 4:17 PM, Vik Malhivma...@ipexpert.com wrote: Enable MTP on the SIP trunk. -- Vik Malhi CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Jonathan Charles jonv...@gmail.com Date: Thu, 23 Jul 2009 14:51:00 -0500 To: Vik Malhi vma...@ipexpert.com Cc: Padmanabhan, Padhu pa...@ti.com, OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SIP endpoints on the other side of an H.323 trunk... It is already 711... BR2#sh voice register dial-peers dial-peer voice 40001 voip destination-pattern 3005 session target ipv4:192.168.29.101:5060 session protocol sipv2 dtmf-relay rtp-nte sip-notify digit collect kpml codec g711ulaw bytes 160 call-fwd-mbox 3005 call-fwd-busy 3100 call-fwd-noan-timeou 12 call-fwd-noan 3100 after-hours-exempt FALSE J On Thu, Jul 23, 2009 at 2:32 PM, Vik Malhivma...@ipexpert.com wrote: I have found Inbound FastStart should fix the problem. Also check the codec within the voice register pool. Voice-class codec is not supported in the voice register pool so try using g711 end to end (voice register pool/outbound dial-peer/SIP Trunk DP). -- Vik Malhi CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Jonathan Charles jonv...@gmail.com Date: Thu, 23 Jul 2009 11:00:12 -0500 To: Padmanabhan, Padhu pa...@ti.com Cc: OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SIP endpoints on the other side of an H.323 trunk... On Thu, Jul 23, 2009 at 9:01 AM, Padmanabhan, Padhupa...@ti.com wrote: Can u pls post your configs?. Cheers,Padhu -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jonathan Charles Sent: Wednesday, July 22, 2009 6:21 PM To: ccie_voice@onlinestudylist.com; cisco-v...@puck.nether.net Subject: Re: [OSL | CCIE_Voice] SIP endpoints on the other side of an H.323 trunk... So, does anyone know if this is true? I am actually seeing that SIP cannot call phones across an H.323 trunk It looks like everything is fine, the call sets up, but then no audio cuts through and the SIP phone keeps ringing... And then I see this when I hang up the cucm phone Jul 22 23:19:48.514: //-1//SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 500 Internal Server Error Reason: Q.850;cause=16 Date: Wed, 22 Jul 2009 23:19:40 GMT From: Phone1 sip:3...@192.168.29.254;tag=00169def16e500dfdb252a73-c3694981 Allow-Events: telephone-event Content-Length: 0 To: sip:5...@192.168.29.254;tag=BCCA62C-BD3 Call-ID: 00169def-16e50019-a4734327-02d0c...@192.168.29.102 Via: SIP/2.0/UDP 192.168.29.102:5060;branch=z9hG4bKc0c82a6f CSeq: 101 INVITE Server: Cisco-SIPGateway/IOS-12.x Jul 22 23:19:48.522:
Re: [OSL | CCIE_Voice] Cisco Unity 7 vmware install error
Hi Juliana, You can place an order via cisco marketplace for NFR. It includes everything you want for ccie voice lab. Jerry From: juliana...@hotmail.com To: jerry...@hotmail.com Subject: RE: [OSL | CCIE_Voice] Cisco Unity 7 vmware install error Date: Thu, 23 Jul 2009 12:39:34 + Hi, I'm looking for uccx 7 software, would you please kindly inform where you got the software? Thanks in advance Juliana From: jerry...@hotmail.com To: lakpr...@gmail.com; ccie_voice@onlinestudylist.com Date: Thu, 23 Jul 2009 07:02:03 + Subject: Re: [OSL | CCIE_Voice] Cisco Unity 7 vmware install error I have a IBM x346 server runs on windows2003 server. I put vmware server 2 on it and runs virtual machine with 2G memory and 80G hard drive. I have successfully installed PUB, SUB , Precense and IPCC7 on vmware already. Cheers, Jerry Date: Thu, 23 Jul 2009 12:22:54 +0530 Subject: Re: [OSL | CCIE_Voice] Cisco Unity 7 vmware install error From: lakpr...@gmail.com To: jerry...@hotmail.com CC: ccie_voice@onlinestudylist.com Hey Jerry, U r installing on AMD processor based machine ?? On Thu, Jul 23, 2009 at 12:20 PM, Bai Min jerry...@hotmail.com wrote: Hi Guys, Did you experience problem when install Unity connection 7 on vmware? I got internal critical error and halt installation. Please find this error screenshot in attachment. Thanks in advance for your advise. Cheers, Jerry NEW mobile Hotmail. Optimized for YOUR phone. Click here. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Ravindra Lakpriya +94 773 532 094 Windows Live™ SkyDrive™: Store, access, and share your photos. See how. Share your memories online with anyone you want anyone you want. _ Windows Live™ SkyDrive™: Store, access, and share your photos. See how. http://windowslive.com/Online/SkyDrive?ocid=TXT_TAGLM_WL_CS_SD_photos_072009___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Documentation at voice lab exam
Not to beat a dead horse here, but so I am clear. I should assume the doc cd URL equivalent will be available for me. i.e. where I click product-voice and uniied communication-IP Telephony-call control---etc, etc - this will be there in some form? From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of c george Sent: Thursday, July 23, 2009 5:00 PM To: earl.ho...@sarcom.com; kevin.dami...@vitalsite.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Documentation at voice lab exam gotcha Respectfully Charles George Subject: RE: [OSL | CCIE_Voice] Documentation at voice lab exam Date: Thu, 23 Jul 2009 16:59:12 -0400 From: earl.ho...@sarcom.com To: cisco...@hotmail.com; kevin.dami...@vitalsite.com; ccie_voice@onlinestudylist.com The web equivalents of the old Doc CD are available, but global search from the main website I believe has been disabled. The docs I mentioned are the ones which are directly accessible from the desktop. The other documentation has to be navigated through the web site. From: c george [mailto:cisco...@hotmail.com] Sent: Thursday, July 23, 2009 4:57 PM To: Hough, Earl; kevin.dami...@vitalsite.com; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Documentation at voice lab exam Cisco product documentation is not available on the web in the lab anymore? Respectfully Charles George Date: Thu, 23 Jul 2009 16:54:14 -0400 From: earl.ho...@sarcom.com To: kevin.dami...@vitalsite.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Documentation at voice lab exam At Networkers in the CCIE Voice techtorial, there were 4 docs which were mentioned as being available on the candidate PC's desktop. I believe the 4 are: Unified Communications 7.0 SRND Enterprise QOS SRND Unified Communications Manager Express SRND -- not 100% sure on this one Unified Contact Center Express 7.0 SRND -- not 100% sure on this one Earl Hough CCIE #16508 (RS/Security) -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kevin Damisch Sent: Thursday, July 23, 2009 3:52 PM To: OSL Group Subject: Re: [OSL | CCIE_Voice] Documentation at voice lab exam No response yet. 3rd time's a charm. Anyone? Thanks, Kevin Damisch Vital Support Systems -Original Message- From: Kevin Damisch Sent: Thursday, July 16, 2009 12:23 AM To: OSL Group Subject: RE: Documentation at voice lab exam Was this mentioned at Networkers? Thanks, Kevin -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kevin Damisch Sent: Friday, July 10, 2009 8:33 PM To: OSL Group Subject: [OSL | CCIE_Voice] Documentation at voice lab exam Has there been any confirmation of what documentation is available during the lab? I've heard mixed comments about it only being the config guides and SRNDs. But, then others have mentioned that the technotes and configuration examples documents are available as well. I'm taking it in San Jose. Thanks, Kevin This communication (including any attachments) is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If you are not the intended recipient, any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify Vital Support Systems at 515 334 5700 and delete or destroy all copies and the original document. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ The information contained in this transmission is confidential. It is intended solely for the use of the individual(s) or organization(s) to whom it is addressed. Any disclosure, copying or further distribution is not permitted unless such privilege is explicitly granted in writing by SARCOM, Inc. Furthermore, SARCOM, Inc. is not responsible for the proper and complete transmission of the substance of this communication, nor for any delay in its receipt. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com NEW mobile Hotmail. Optimized for YOUR phone. Click here. http://windowslive.com/Mobile?ocid=TXT_TAGLM_WL_CS_MB_new_hotmail_07200 9 _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ The information contained in this transmission is confidential. It is intended solely for the use of the individual(s) or organization(s) to whom it
Re: [OSL | CCIE_Voice] Documentation at voice lab exam
That is my understanding. I haven't seen the Voice lab yet, but that was the way it was in previous years in other tracks. Also, this was what Ben Ng said during the CCIE Voice techtorial at Networkers. I don't know if or when anything may come of it, but in the session I attended, the question was brought up regarding would it be possible for Cisco to publish the documentation site as it will be viewed from the candidates (i.e., with all external links and forbidden configuration documents blocked). This way people wouldn't get used to something that won't be in the actual lab. This suggestion was taken down by Ben, but as I said...I don't know if anything will come of this any time soon. From: Steve Sarrick [mailto:ssarr...@drsllc.net] Sent: Thursday, July 23, 2009 11:03 PM To: c george; Hough, Earl; kevin.dami...@vitalsite.com; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Documentation at voice lab exam Not to beat a dead horse here, but so I am clear. I should assume the doc cd URL equivalent will be available for me. i.e. where I click product-voice and uniied communication-IP Telephony-call control---etc, etc - this will be there in some form? From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of c george Sent: Thursday, July 23, 2009 5:00 PM To: earl.ho...@sarcom.com; kevin.dami...@vitalsite.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Documentation at voice lab exam gotcha Respectfully Charles George Subject: RE: [OSL | CCIE_Voice] Documentation at voice lab exam Date: Thu, 23 Jul 2009 16:59:12 -0400 From: earl.ho...@sarcom.com To: cisco...@hotmail.com; kevin.dami...@vitalsite.com; ccie_voice@onlinestudylist.com The web equivalents of the old Doc CD are available, but global search from the main website I believe has been disabled. The docs I mentioned are the ones which are directly accessible from the desktop. The other documentation has to be navigated through the web site. From: c george [mailto:cisco...@hotmail.com] Sent: Thursday, July 23, 2009 4:57 PM To: Hough, Earl; kevin.dami...@vitalsite.com; ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Documentation at voice lab exam Cisco product documentation is not available on the web in the lab anymore? Respectfully Charles George Date: Thu, 23 Jul 2009 16:54:14 -0400 From: earl.ho...@sarcom.com To: kevin.dami...@vitalsite.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Documentation at voice lab exam At Networkers in the CCIE Voice techtorial, there were 4 docs which were mentioned as being available on the candidate PC's desktop. I believe the 4 are: Unified Communications 7.0 SRND Enterprise QOS SRND Unified Communications Manager Express SRND -- not 100% sure on this one Unified Contact Center Express 7.0 SRND -- not 100% sure on this one Earl Hough CCIE #16508 (RS/Security) -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kevin Damisch Sent: Thursday, July 23, 2009 3:52 PM To: OSL Group Subject: Re: [OSL | CCIE_Voice] Documentation at voice lab exam No response yet. 3rd time's a charm. Anyone? Thanks, Kevin Damisch Vital Support Systems -Original Message- From: Kevin Damisch Sent: Thursday, July 16, 2009 12:23 AM To: OSL Group Subject: RE: Documentation at voice lab exam Was this mentioned at Networkers? Thanks, Kevin -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kevin Damisch Sent: Friday, July 10, 2009 8:33 PM To: OSL Group Subject: [OSL | CCIE_Voice] Documentation at voice lab exam Has there been any confirmation of what documentation is available during the lab? I've heard mixed comments about it only being the config guides and SRNDs. But, then others have mentioned that the technotes and configuration examples documents are available as well. I'm taking it in San Jose. Thanks, Kevin This communication (including any attachments) is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If you are not the intended recipient, any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify Vital Support Systems at 515 334 5700 and delete or destroy all copies and the original document. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ The information contained in this transmission is
Re: [OSL | CCIE_Voice] Documentation at voice lab exam
As an example, by going to: http://www.cisco.com/cisco/web/psa/default.html?mode=prod Then choose: Products Voice and Unified Communications IP Telephony Call Control Cisco Unified Communications Manager Express You are then at the CUCME doc page: http://www.cisco.com/en/US/partner/products/sw/voicesw/ps4625/tsd_products_support_series_home.html My question is - Will we have access to the docs on that page? Such as: Command/Technical References Design Guides/Technotes Install/Upgrade Guides Configuration Examples/Technotes/Guides Feature/Programming Guides etc... Thanks, Kevin Damisch From: Vik Malhi [mailto:vma...@ipexpert.com] Sent: Thursday, July 23, 2009 10:33 PM To: Steve Sarrick Cc: c george; earl.ho...@sarcom.com; Kevin Damisch; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Documentation at voice lab exam Yes (I believe). The pdf's on the desktop are: - ucm/qos srnd - cme admin guide v7.0 - cad installation guide for uccx There was talk at networkers of cisco making a link available to candidates giving them a web equivalent of the documentation that is available within the lab exam. This seems to be very sensible and I hope they follow that up with some action. Vik Malhi - CCIE#13890 Senior Technical Instructor - IPexpert Inc Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.commailto:vma...@ipexpert.com Join IPexpert's Free CCIE Peer Groups Study Communities at www.IPexpert.com/communitieshttp://www.IPexpert.com/communities On Jul 24, 2009, at 4:03 AM, Steve Sarrick ssarr...@drsllc.netmailto:ssarr...@drsllc.net wrote: Not to beat a dead horse here, but so I am clear. I should assume the doc cd URL equivalent will be available for me. i.e. where I click product—voice and uniied communication—IP Telephony—call control---etc, etc – this will be there in some form? From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of c george Sent: Thursday, July 23, 2009 5:00 PM To: earl.ho...@sarcom.commailto:earl.ho...@sarcom.com; kevin.dami...@vitalsite.commailto:kevin.dami...@vitalsite.com; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Documentation at voice lab exam gotcha Respectfully Charles George Subject: RE: [OSL | CCIE_Voice] Documentation at voice lab exam Date: Thu, 23 Jul 2009 16:59:12 -0400 From: earl.ho...@sarcom.commailto:earl.ho...@sarcom.com To: cisco...@hotmail.commailto:cisco...@hotmail.com; kevin.dami...@vitalsite.commailto:kevin.dami...@vitalsite.com; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com The web equivalents of the old Doc CD are available, but global search from the main website I believe has been disabled. The docs I mentioned are the ones which are directly accessible from the desktop. The other documentation has to be navigated through the web site. From: c george [mailto:cisco...@hotmail.com] Sent: Thursday, July 23, 2009 4:57 PM To: Hough, Earl; kevin.dami...@vitalsite.commailto:kevin.dami...@vitalsite.com; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] Documentation at voice lab exam Cisco product documentation is not available on the web in the lab anymore? Respectfully Charles George Date: Thu, 23 Jul 2009 16:54:14 -0400 From: earl.ho...@sarcom.commailto:earl.ho...@sarcom.com To: kevin.dami...@vitalsite.commailto:kevin.dami...@vitalsite.com; ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Documentation at voice lab exam At Networkers in the CCIE Voice techtorial, there were 4 docs which were mentioned as being available on the candidate PC's desktop. I believe the 4 are: Unified Communications 7.0 SRND Enterprise QOS SRND Unified Communications Manager Express SRND -- not 100% sure on this one Unified Contact Center Express 7.0 SRND -- not 100% sure on this one Earl Hough CCIE #16508 (RS/Security) -Original Message- From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kevin Damisch Sent: Thursday, July 23, 2009 3:52 PM To: OSL Group Subject: Re: [OSL | CCIE_Voice] Documentation at voice lab exam No response yet. 3rd time's a charm. Anyone? Thanks, Kevin Damisch Vital Support Systems -Original Message- From: Kevin Damisch Sent: Thursday, July 16, 2009 12:23 AM To: OSL Group Subject: RE: Documentation at voice lab exam Was this mentioned at Networkers? Thanks, Kevin -Original Message- From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kevin Damisch Sent: Friday, July 10, 2009 8:33 PM To: OSL Group Subject:
Re: [OSL | CCIE_Voice] Hilarity with CUCME 4.3/7.0
ahhh. Got ya. Welcome to Cisco's wonderful documentation. SW Conf for UP TO (and including) 3 people is on the page I mentioned (no meet-me) HW Conf for 3-8 people plus meet-me is on your Page 709. Page 685 shows how to configure both types. Don't know why it's not working for you though. I confirmed 7.0(1) on 12.4(22)T2. I have no DSPs configured, no sdspfarm commands, etc. Just max-conferences 8 gain -6. SIP seems to conference locally says in conference, locally mixed From: Jonathan Charles [jonv...@gmail.com] Sent: Thursday, July 23, 2009 12:39 PM To: Michael Ciarfello Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Hilarity with CUCME 4.3/7.0 Cannot Complete Conference. On Thu, Jul 23, 2009 at 8:20 AM, Michael Ciarfellomciarfe...@iplogic.com wrote: CCME 7.0(1) What did the phone say? -Original Message- From: Jonathan Charles [mailto:jonv...@gmail.com] Sent: Thursday, July 23, 2009 12:37 AM To: Michael Ciarfello Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Hilarity with CUCME 4.3/7.0 Ub CUCME 4.1 and earlier, there was sw conf... in 4.3, it says, on page 709 of the CUCME7 admin guide: SCCP: Configuring Multi-Party Ad Hoc and Meet-Me Conferencing in Cisco Unified CME 4.1 and Later Versions: Prerequisites * Cisco Unified CME 4.1 or a later version * You must have a PVDM2-8, PVDM2-16, PVDM2-32, or PVDM2-64 high-density packet voice digital signal processor module hosted on the motherboard or on a module such as the NM-HDV2 or NM-HD-2VE. * For Cisco Unified IP Phone 7985, firmware version 4-1-2-0 or a later version I configured it, and tested the old-style, max-conferences command under telephony services and was told no by the phone. My telephony-service version is 7.0(1) what is yours? Jonathan On Wed, Jul 22, 2009 at 9:46 PM, Michael Ciarfellomciarfe...@iplogic.com wrote: Looks good here. SIP to SIP 3-way conference, SCCP to SCCP and SCCP to/from SIP. Also admin guide page 680 Conferencing Overview. No software meet-me (never was sw meetme.) From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jonathan Charles [jonv...@gmail.com] Sent: Wednesday, July 22, 2009 7:15 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Hilarity with CUCME 4.3/7.0 Conferencing, check it out, no ad-hoc software conferencing any more... Jonathan ___ 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] Hilarity with CUCME 4.3/7.0
Yes, I got that too... couldn't get the sw conf to work J On Thu, Jul 23, 2009 at 11:28 PM, Michael Ciarfellomciarfe...@iplogic.com wrote: ahhh. Got ya. Welcome to Cisco's wonderful documentation. SW Conf for UP TO (and including) 3 people is on the page I mentioned (no meet-me) HW Conf for 3-8 people plus meet-me is on your Page 709. Page 685 shows how to configure both types. Don't know why it's not working for you though. I confirmed 7.0(1) on 12.4(22)T2. I have no DSPs configured, no sdspfarm commands, etc. Just max-conferences 8 gain -6. SIP seems to conference locally says in conference, locally mixed From: Jonathan Charles [jonv...@gmail.com] Sent: Thursday, July 23, 2009 12:39 PM To: Michael Ciarfello Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Hilarity with CUCME 4.3/7.0 Cannot Complete Conference. On Thu, Jul 23, 2009 at 8:20 AM, Michael Ciarfellomciarfe...@iplogic.com wrote: CCME 7.0(1) What did the phone say? -Original Message- From: Jonathan Charles [mailto:jonv...@gmail.com] Sent: Thursday, July 23, 2009 12:37 AM To: Michael Ciarfello Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Hilarity with CUCME 4.3/7.0 Ub CUCME 4.1 and earlier, there was sw conf... in 4.3, it says, on page 709 of the CUCME7 admin guide: SCCP: Configuring Multi-Party Ad Hoc and Meet-Me Conferencing in Cisco Unified CME 4.1 and Later Versions: Prerequisites * Cisco Unified CME 4.1 or a later version * You must have a PVDM2-8, PVDM2-16, PVDM2-32, or PVDM2-64 high-density packet voice digital signal processor module hosted on the motherboard or on a module such as the NM-HDV2 or NM-HD-2VE. * For Cisco Unified IP Phone 7985, firmware version 4-1-2-0 or a later version I configured it, and tested the old-style, max-conferences command under telephony services and was told no by the phone. My telephony-service version is 7.0(1) what is yours? Jonathan On Wed, Jul 22, 2009 at 9:46 PM, Michael Ciarfellomciarfe...@iplogic.com wrote: Looks good here. SIP to SIP 3-way conference, SCCP to SCCP and SCCP to/from SIP. Also admin guide page 680 Conferencing Overview. No software meet-me (never was sw meetme.) From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jonathan Charles [jonv...@gmail.com] Sent: Wednesday, July 22, 2009 7:15 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Hilarity with CUCME 4.3/7.0 Conferencing, check it out, no ad-hoc software conferencing any more... Jonathan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Call Manager Multiple Directories
Hi I would like to find out if anyone has the xml code for an external ldap direcory search for Call Manager. I have mulitple LDAP directoies in my lab and would like to get a phone service to point to them instead of its own internal one. Paul ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
