[OSL | CCIE_Voice] Assistance with IPBlue Multilab Version
Hi I recently purchased IPBlue Multilab Version and am unable to get them registered with both the CME UCM. It receives the Failed to retrieve configuration from TFTP Server XX.XX.XX.XX. I have been running the demo version all this time was able to run them without these issues. Is there anything specific we need to do with the multilab version. Please advise at your earliest convenience as I am doing a Lab currently will appreciate your response. Thanks Mann ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Assistance with IPBlue Multilab Version
Got it resolved. It needs MAC without any dotted notation ( NO .. but just ). Sorry to bother. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] E1 doesn't accept more than one incoming call
Hi All, I have a weird problem. I configured the router voice controller E1 as one pri-group with H.323 the signaling protocol. The router accepts the incoming calls and forward them to CCM7. The problem is the router only accept one incoming call and it can process more than one outgoing call. Any other incoming call , the PSTN answers the call saying the number is busy at the moment. Any ideas?? I'm thinking of dividing the pri-group to two groups. Thanks in advance for any help. Here is my config: interface Serial0/2/0:15 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn incoming-voice voice isdn send-alerting isdn bchan-number-order ascending isdn sending-complete ! voice-port 0/2/0:15 cptone GB connection plar 0 bearer-cap Speech num-exp 0 888 to the contact center ! dial-peer voice 1 pots destination-pattern .T incoming called-number . direct-inward-dial --- the customer dosn't have this service at the moment port 0/2/0:15 ! dial-peer voice 12 voip destination-pattern ... voice-class h323 1 session target ipv4:192.168.77.105 dtmf-relay h245-alphanumeric codec g711alaw ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] E1 doesn't accept more than one incoming call
Hi, Have you done any debugs? The router sends the h225 setup to the CUCM or it just denies the call itself? deb h225 q931 Then you might need to see the traces of the Call Manager. What is the 888 on the call manager? is it a hunt group? particular extension? Tell us more about setup on CallManager side... Cheers, 2010/10/17 khaled Saholy khaled_sah...@hotmail.com Hi All, I have a weird problem. I configured the router voice controller E1 as one pri-group with H.323 the signaling protocol. The router accepts the incoming calls and forward them to CCM7. The problem is the router only accept one incoming call and it can process more than one outgoing call. Any other incoming call , the PSTN answers the call saying the number is busy at the moment. Any ideas?? I'm thinking of dividing the pri-group to two groups. Thanks in advance for any help. Here is my config: interface Serial0/2/0:15 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn incoming-voice voice isdn send-alerting isdn bchan-number-order ascending isdn sending-complete ! voice-port 0/2/0:15 cptone GB connection plar 0 bearer-cap Speech num-exp 0 888 to the contact center ! dial-peer voice 1 pots destination-pattern .T incoming called-number . direct-inward-dial --- the customer dosn't have this service at the moment port 0/2/0:15 ! dial-peer voice 12 voip destination-pattern ... voice-class h323 1 session target ipv4:192.168.77.105 dtmf-relay h245-alphanumeric codec g711alaw ___ 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] E1 doesn't accept more than one incoming call
Thanks Goerge for the reply. I haven't tried this debug of h225. I tried debug voip dialpeer all .I'll try it with tracing of the call manager . That number is for the Contact Center CTI port. Any incoming call will be answered by CUCCX by the Application triggered by that number (888). The setup on the cm7 side , router is configured as H323 gateway. Is there any other information you need to know about? Thanks and regards. Khaled Date: Sun, 17 Oct 2010 10:41:41 +0100 Subject: Re: [OSL | CCIE_Voice] E1 doesn't accept more than one incoming call From: gogli...@gmail.com To: khaled_sah...@hotmail.com CC: ccie_voice@onlinestudylist.com Hi, Have you done any debugs? The router sends the h225 setup to the CUCM or it just denies the call itself? deb h225 q931 Then you might need to see the traces of the Call Manager. What is the 888 on the call manager? is it a hunt group? particular extension? Tell us more about setup on CallManager side... Cheers, 2010/10/17 khaled Saholy khaled_sah...@hotmail.com Hi All, I have a weird problem. I configured the router voice controller E1 as one pri-group with H.323 the signaling protocol. The router accepts the incoming calls and forward them to CCM7. The problem is the router only accept one incoming call and it can process more than one outgoing call. Any other incoming call , the PSTN answers the call saying the number is busy at the moment. Any ideas?? I'm thinking of dividing the pri-group to two groups. Thanks in advance for any help. Here is my config: interface Serial0/2/0:15 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn incoming-voice voice isdn send-alerting isdn bchan-number-order ascending isdn sending-complete ! voice-port 0/2/0:15 cptone GB connection plar 0 bearer-cap Speech num-exp 0 888 to the contact center ! dial-peer voice 1 pots destination-pattern .T incoming called-number . direct-inward-dial --- the customer dosn't have this service at the moment port 0/2/0:15 ! dial-peer voice 12 voip destination-pattern ... voice-class h323 1 session target ipv4:192.168.77.105 dtmf-relay h245-alphanumeric codec g711alaw ___ 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] Vol2 Lab7 cue aar
You need to set the EPNM on the CTI ports to point to the number of the CTI RP for CUE. This is since the call can not go directly to the CTI ports, it has to first be sent to the CTI RP, then on to the CTI port. Sincerely Roger Källberg CCIE #26199 (Voice) Consultant Cygate AB Från: David A [david.a...@gmail.com] Skickat: den 16 oktober 2010 19:18 Till: ccie_voice@onlinestudylist.com Ämne: [OSL | CCIE_Voice] Vol2 Lab7 cue aar Hi all, I always get a busy when I configure AAR for cue and dial from HQ or SiteB. I have aar group on all the phones and lines. cue external mask is same as the sietc phones. cti ports and cti rp have aar css and aar group. I do not see the call go out of any of the gateways. Anyone face similar issues. Thanks, DA___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] maximum sessions 0
hi, this is my first mail to the group i have a problem with conference bridge my H/w is cisco2811 12.4.22.T with a PVDM2-16 nothing configured yet and i got this Router(config-dspfarm-profile)#maximum sessions ? 0-0 Number of sessions assigned to this profile my config is: Router(config-dspfarm-profile)#do sh run hostname Router ! boot-start-marker boot system flash:c2800nm-adventerprisek9_ivs_li-mz.124-22.T.bin memory-size iomem 5 no network-clock-participate wic 3 ! ip cef ! ! no ip domain lookup ! voice-card 0 dsp services dspfarm ! ! ! ! ! controller T1 0/3/0 ! controller T1 0/3/1 ! ! ! ! ! interface FastEthernet0/0 ip address 192.168.1.10 255.255.255.0 duplex auto speed auto ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! interface Serial0/0/0 no ip address shutdown clock rate 200 ! ! ! ! voice-port 0/2/0 ! voice-port 0/2/1 ! ccm-manager fax protocol cisco ! mgcp fax t38 ecm mgcp behavior g729-variants static-pt ! sccp local FastEthernet0/0 sccp ccm 192.168.1.20 identifier 1 priority 1 version 7.0 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register GW_CFB ! dspfarm profile 1 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 shutdown ! ! ! ! ! gatekeeper shutdown ! ! line con 0 line aux 0 line vty 0 4 login ! scheduler allocate 2 1000 end can anyone tell me what is happening with my router ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 56, Issue 139
: /archives/ccie_voice/attachments/20101017/ee287636/attachment.html -- ___ CCIE_Voice mailing list CCIE_Voice@onlinestudylist.com http://onlinestudylist.com/mailman/listinfo/ccie_voice End of CCIE_Voice Digest, Vol 56, Issue 139 *** This email and any files transmitted with it are confidential and are intended for the sole use of the individual(s) to whom they are properly addressed. Any use, dissemination or forwarding of this email and any files transmitted with it by anyone other than the intended recipient(s) is strictly prohibited. If you have received this email in error, please notify the sender by replying to this email. Black Box Corporation and its affiliates reserve the right to scan and monitor all e-mail traffic. winmail.dat___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] maximum sessions 0
Hello, Knowing that installing any voice card on router, consuming the DSPs. Try to plug out the Te card and try again (just to be sure). Tamer, -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mohammad Dewan Sent: Sunday, October 17, 2010 1:50 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] maximum sessions 0 hi, this is my first mail to the group i have a problem with conference bridge my H/w is cisco2811 12.4.22.T with a PVDM2-16 nothing configured yet and i got this Router(config-dspfarm-profile)#maximum sessions ? 0-0 Number of sessions assigned to this profile my config is: Router(config-dspfarm-profile)#do sh run hostname Router ! boot-start-marker boot system flash:c2800nm-adventerprisek9_ivs_li-mz.124-22.T.bin memory-size iomem 5 no network-clock-participate wic 3 ! ip cef ! ! no ip domain lookup ! voice-card 0 dsp services dspfarm ! ! ! ! ! controller T1 0/3/0 ! controller T1 0/3/1 ! ! ! ! ! interface FastEthernet0/0 ip address 192.168.1.10 255.255.255.0 duplex auto speed auto ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! interface Serial0/0/0 no ip address shutdown clock rate 200 ! ! ! ! voice-port 0/2/0 ! voice-port 0/2/1 ! ccm-manager fax protocol cisco ! mgcp fax t38 ecm mgcp behavior g729-variants static-pt ! sccp local FastEthernet0/0 sccp ccm 192.168.1.20 identifier 1 priority 1 version 7.0 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register GW_CFB ! dspfarm profile 1 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 shutdown ! ! ! ! ! gatekeeper shutdown ! ! line con 0 line aux 0 line vty 0 4 login ! scheduler allocate 2 1000 end can anyone tell me what is happening with my router ___ 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] Website saying LAB not Scheduled
Could someone from Proctor Labs please get back to me, I gave up so much to LAB today including tickets for a New York Giants Football Game. If someone out there has the Tech Support Hotline Number that would be great, it is notg posted on the Web Site, live Chat is only available apparently only when you are in a LAB. My receipt info for Today is Below. Help I have a Proctor LABS Voice Lab scheduled today 10/17 at 8:00AM and it is telling me no LAB scheduled. Here is my receipt 56434521101017800E89D9 Voice - Oct 17, 2010 at 08:00EST Order#: pli110983572 What is the Technical support number for Proctor Labs, unless you are on a LAB the website does not show this. -Original Message- From: ccie_voice-boun...@onlinestudylist.com on behalf of ccie_voice-requ...@onlinestudylist.com Sent: Sun 10/17/2010 8:36 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 56, Issue 140 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: E1 doesn't accept more than one incoming call (George Goglidze) 2. Re: E1 doesn't accept more than one incoming call (khaled Saholy) 3. Re: Vol2 Lab7 cue aar (Roger K?llberg) 4. maximum sessions 0 (Mohammad Dewan) 5. Re: CCIE_Voice Digest, Vol 56, Issue 139 (Stern, Larry) -- Message: 1 Date: Sun, 17 Oct 2010 10:41:41 +0100 From: George Goglidze gogli...@gmail.com To: khaled Saholy khaled_sah...@hotmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] E1 doesn't accept more than one incoming call Message-ID: aanlktimecuwq6p+2yj0wap+v4ii+iqyso75sbe9ne...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi, Have you done any debugs? The router sends the h225 setup to the CUCM or it just denies the call itself? deb h225 q931 Then you might need to see the traces of the Call Manager. What is the 888 on the call manager? is it a hunt group? particular extension? Tell us more about setup on CallManager side... Cheers, 2010/10/17 khaled Saholy khaled_sah...@hotmail.com Hi All, I have a weird problem. I configured the router voice controller E1 as one pri-group with H.323 the signaling protocol. The router accepts the incoming calls and forward them to CCM7. The problem is the router only accept one incoming call and it can process more than one outgoing call. Any other incoming call , the PSTN answers the call saying the number is busy at the moment. Any ideas?? I'm thinking of dividing the pri-group to two groups. Thanks in advance for any help. Here is my config: interface Serial0/2/0:15 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn incoming-voice voice isdn send-alerting isdn bchan-number-order ascending isdn sending-complete ! voice-port 0/2/0:15 cptone GB connection plar 0 bearer-cap Speech num-exp 0 888 to the contact center ! dial-peer voice 1 pots destination-pattern .T incoming called-number . direct-inward-dial --- the customer dosn't have this service at the moment port 0/2/0:15 ! dial-peer voice 12 voip destination-pattern ... voice-class h323 1 session target ipv4:192.168.77.105 dtmf-relay h245-alphanumeric codec g711alaw ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20101017/a8d7b2ee/attachment-0001.html -- Message: 2 Date: Sun, 17 Oct 2010 13:27:27 +0300 From: khaled Saholy khaled_sah...@hotmail.com To: gogli...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] E1 doesn't accept more than one incoming call Message-ID: blu146-w127c7291704da47e094ad09f...@phx.gbl Content-Type: text/plain; charset=windows-1256 Thanks Goerge for the reply. I haven't tried this debug of h225. I tried debug voip dialpeer all .I'll try it with tracing of the call manager . That number is for the Contact Center CTI port. Any incoming call will be answered by CUCCX by the Application triggered by that number (888). The setup on the cm7 side , router is configured as H323 gateway. Is there any other information you need to know about? Thanks and regards. Khaled Date: Sun, 17 Oct 2010 10:41:41 +0100 Subject: Re: [OSL | CCIE_Voice] E1
[OSL | CCIE_Voice] Lab 8, Q2.3
Kind of stuck here and really cannot find the issue. Calls out to BR2 work fine, 323 to the cube, sip to br2. (Task 2.2). However, calls to UCM are getting reorder. I don't see the incoming calls on CDRs. Running ccapi and 245 debugs seems to point to a q.850, cause 57. The dialp inout shows correct ingress and egress on the peers. The ICT on UCM has inbound faststart enabled. Any ideas where to look? I've been all over the PG and I'm sure I'm compliant with their solution, but getting frustrated : ) Paul Dardinski (RS/Sec #16842) ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol2 Lab7 cue aar
Hi Roger, I have the EPNM on the CTI ports set to the CTI-RP/VM pilot ie 3600 and EPMN is 02077353600. I have no idea why the call doesnt pass thru to the gateway. Thanks, DA 2010/10/17 Roger Källberg roger.kallb...@cygate.se You need to set the EPNM on the CTI ports to point to the number of the CTI RP for CUE. This is since the call can not go directly to the CTI ports, it has to first be sent to the CTI RP, then on to the CTI port. Sincerely *Roger Källberg* CCIE #26199 (Voice) Consultant Cygate AB -- *Från:* David A [david.a...@gmail.com] *Skickat:* den 16 oktober 2010 19:18 *Till:* ccie_voice@onlinestudylist.com *Ämne:* [OSL | CCIE_Voice] Vol2 Lab7 cue aar Hi all, I always get a busy when I configure AAR for cue and dial from HQ or SiteB. I have aar group on all the phones and lines. cue external mask is same as the sietc phones. cti ports and cti rp have aar css and aar group. I do not see the call go out of any of the gateways. Anyone face similar issues. Thanks, DA ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] maximum sessions 0
I am assuming you see the PVDM in sh diag. A conference bridge uses 240 MIPS ie a single PVDM-16. In the config you need to use voice-card 0 no dspfarm dsp serviced dspfarm Also make sure none of the dsp's are being used ie in a pri or else you would not see any resources if you have a single pvdm-16. HTH Prashant On Sun, Oct 17, 2010 at 8:44 AM, Tamer Ismail tih...@gmail.com wrote: Hello, Knowing that installing any voice card on router, consuming the DSPs. Try to plug out the Te card and try again (just to be sure). Tamer, -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mohammad Dewan Sent: Sunday, October 17, 2010 1:50 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] maximum sessions 0 hi, this is my first mail to the group i have a problem with conference bridge my H/w is cisco2811 12.4.22.T with a PVDM2-16 nothing configured yet and i got this Router(config-dspfarm-profile)#maximum sessions ? 0-0 Number of sessions assigned to this profile my config is: Router(config-dspfarm-profile)#do sh run hostname Router ! boot-start-marker boot system flash:c2800nm-adventerprisek9_ivs_li-mz.124-22.T.bin memory-size iomem 5 no network-clock-participate wic 3 ! ip cef ! ! no ip domain lookup ! voice-card 0 dsp services dspfarm ! ! ! ! ! controller T1 0/3/0 ! controller T1 0/3/1 ! ! ! ! ! interface FastEthernet0/0 ip address 192.168.1.10 255.255.255.0 duplex auto speed auto ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! interface Serial0/0/0 no ip address shutdown clock rate 200 ! ! ! ! voice-port 0/2/0 ! voice-port 0/2/1 ! ccm-manager fax protocol cisco ! mgcp fax t38 ecm mgcp behavior g729-variants static-pt ! sccp local FastEthernet0/0 sccp ccm 192.168.1.20 identifier 1 priority 1 version 7.0 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register GW_CFB ! dspfarm profile 1 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 shutdown ! ! ! ! ! gatekeeper shutdown ! ! line con 0 line aux 0 line vty 0 4 login ! scheduler allocate 2 1000 end can anyone tell me what is happening with my router ___ 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] CCIE_Voice Digest, Vol 56, Issue 141
Hi Dewan, From your configuration i can see that your Dsp farm profile is shutdown, do a no shutdown on your conference dspfarm profile. Pithog oil Message: 1 Date: Sun, 17 Oct 2010 14:44:21 +0200 From: Tamer Ismail tih...@gmail.com To: 'Mohammad Dewan' mohamad.de...@gmail.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CUE Msg Notification via CLI configurations - Lab 8 4.6
Hi Experts, I am working on Vol 2 Lab 8 with has CME integrated with CUE. I can configure almost the requirements on this lab via CUE CLI, but cannot configure CLI commands to turn on the Msg Notification for user to ring out the PSTN phone when he got a new message. I only got a portion of it, but not complete the task without access the GUI to configure the 'cell phone number' and the 'schedule'. I did an output of the CLI configurations before and after the completion via GUI, but could not find any addition CLI commands that configured for the 'cell phone number' and 'the schedule' for msg notifications. Have anyone try to configure this in CLI before and have it work? If so, please share the configurations or shed the light. I know that access GUI is very quick way to complete this task via VM Msg Notification tab, but WHAT IF WE CANNOT ACCESS GUI FOR SOME REASON OR NO GUI ALLOW IN THE REAL LAB. Here is my portion of the CLI command to turn on the msg notification, but not the whole thing to turn on the cell phone and the schedule. voicemail notification owner scphn4 enable voicemail notification enable voicemail notification preference all voicemail notification allow-login username scphn4 phonenumberE164 032141891 Thanks, TN. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] E1 doesn't accept more than one incoming call
What session limit did you put on the Application trigger? It has to be increased maybe? CTI Group that 888 App trigger is using, should have enough CTI Ports too. I'd go through traces in the following order: 1) Voice Gateway - deb h225 q931 if it's Call Manager that sends release then step 2... 2) You might want to leave CUCM traces for the last, as probably it has something to do with resource allocation on UCCX. Therefore check UCCX Traces, try to look for that application trigger, search by name/number. It will give you a clue about what's happening. 3) On CCM make sure locations are not limiting bandwidth to only one call between h323 gateway and UCCX App Route Point. Hope this helps, 2010/10/17 khaled Saholy khaled_sah...@hotmail.com Thanks Goerge for the reply. I haven't tried this debug of h225. I tried debug voip dialpeer all .I'll try it with tracing of the call manager . That number is for the Contact Center CTI port. Any incoming call will be answered by CUCCX by the Application triggered by that number (888). The setup on the cm7 side , router is configured as H323 gateway. Is there any other information you need to know about? Thanks and regards. Khaled -- Date: Sun, 17 Oct 2010 10:41:41 +0100 Subject: Re: [OSL | CCIE_Voice] E1 doesn't accept more than one incoming call From: gogli...@gmail.com To: khaled_sah...@hotmail.com CC: ccie_voice@onlinestudylist.com Hi, Have you done any debugs? The router sends the h225 setup to the CUCM or it just denies the call itself? deb h225 q931 Then you might need to see the traces of the Call Manager. What is the 888 on the call manager? is it a hunt group? particular extension? Tell us more about setup on CallManager side... Cheers, 2010/10/17 khaled Saholy khaled_sah...@hotmail.com Hi All, I have a weird problem. I configured the router voice controller E1 as one pri-group with H.323 the signaling protocol. The router accepts the incoming calls and forward them to CCM7. The problem is the router only accept one incoming call and it can process more than one outgoing call. Any other incoming call , the PSTN answers the call saying the number is busy at the moment. Any ideas?? I'm thinking of dividing the pri-group to two groups. Thanks in advance for any help. Here is my config: interface Serial0/2/0:15 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn incoming-voice voice isdn send-alerting isdn bchan-number-order ascending isdn sending-complete ! voice-port 0/2/0:15 cptone GB connection plar 0 bearer-cap Speech num-exp 0 888 to the contact center ! dial-peer voice 1 pots destination-pattern .T incoming called-number . direct-inward-dial --- the customer dosn't have this service at the moment port 0/2/0:15 ! dial-peer voice 12 voip destination-pattern ... voice-class h323 1 session target ipv4:192.168.77.105 dtmf-relay h245-alphanumeric codec g711alaw ___ 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] maximum sessions 0
Xcoder an conf require 1 dsp to configure. You need more dsp's in order to configure your t1 up to 16 ports and the second dsp will be used for your conf and Xcoder Sent from my iPhone On Oct 17, 2010, at 8:44 AM, Tamer Ismail tih...@gmail.com wrote: Hello, Knowing that installing any voice card on router, consuming the DSPs. Try to plug out the Te card and try again (just to be sure). Tamer, -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mohammad Dewan Sent: Sunday, October 17, 2010 1:50 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] maximum sessions 0 hi, this is my first mail to the group i have a problem with conference bridge my H/w is cisco2811 12.4.22.T with a PVDM2-16 nothing configured yet and i got this Router(config-dspfarm-profile)#maximum sessions ? 0-0 Number of sessions assigned to this profile my config is: Router(config-dspfarm-profile)#do sh run hostname Router ! boot-start-marker boot system flash:c2800nm-adventerprisek9_ivs_li-mz.124-22.T.bin memory-size iomem 5 no network-clock-participate wic 3 ! ip cef ! ! no ip domain lookup ! voice-card 0 dsp services dspfarm ! ! ! ! ! controller T1 0/3/0 ! controller T1 0/3/1 ! ! ! ! ! interface FastEthernet0/0 ip address 192.168.1.10 255.255.255.0 duplex auto speed auto ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! interface Serial0/0/0 no ip address shutdown clock rate 200 ! ! ! ! voice-port 0/2/0 ! voice-port 0/2/1 ! ccm-manager fax protocol cisco ! mgcp fax t38 ecm mgcp behavior g729-variants static-pt ! sccp local FastEthernet0/0 sccp ccm 192.168.1.20 identifier 1 priority 1 version 7.0 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register GW_CFB ! dspfarm profile 1 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 shutdown ! ! ! ! ! gatekeeper shutdown ! ! line con 0 line aux 0 line vty 0 4 login ! scheduler allocate 2 1000 end can anyone tell me what is happening with my router ___ 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] Vol2 Lab7 cue aar
Hi David, Sounds like your call never gets to the VGW. Have you verified that your RP used for AAR will match to send the call to the gateway? You might also want to verify that you have reset the RL used, pretty common thing to forget. What kind of VGW do you use, H.323 or MGCP. With the first you might want to run debug voip dialpeer, to see what ingress dp that is used. That is if the call even gets sent to the VGW. If not the problem is within the UCM. Might be pretty obvious, but have you activated AAR in the SP? About AAR group, you should only set that on the line, setting it on the device might cause some unpredicted behaviour. Sincerely Roger Källberg CCIE #26199 (Voice) Consultant Cygate AB Från: David A [david.a...@gmail.com] Skickat: den 17 oktober 2010 16:10 Till: Roger Källberg Kopia: ccie_voice@onlinestudylist.com Ämne: Re: [OSL | CCIE_Voice] Vol2 Lab7 cue aar Hi Roger, I have the EPNM on the CTI ports set to the CTI-RP/VM pilot ie 3600 and EPMN is 02077353600. I have no idea why the call doesnt pass thru to the gateway. Thanks, DA 2010/10/17 Roger Källberg roger.kallb...@cygate.semailto:roger.kallb...@cygate.se You need to set the EPNM on the CTI ports to point to the number of the CTI RP for CUE. This is since the call can not go directly to the CTI ports, it has to first be sent to the CTI RP, then on to the CTI port. Sincerely Roger Källberg CCIE #26199 (Voice) Consultant Cygate AB Från: David A [david.a...@gmail.commailto:david.a...@gmail.com] Skickat: den 16 oktober 2010 19:18 Till: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Ämne: [OSL | CCIE_Voice] Vol2 Lab7 cue aar Hi all, I always get a busy when I configure AAR for cue and dial from HQ or SiteB. I have aar group on all the phones and lines. cue external mask is same as the sietc phones. cti ports and cti rp have aar css and aar group. I do not see the call go out of any of the gateways. Anyone face similar issues. Thanks, DA ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Mgcp outbound call fails.
Hello guys, I experienced a problem where i cannot make any outbound call from an cucm through an mgcp gateway. My mgcp gateway was registered to d cucm, right pt/css were applied to d calling phone and correct partition to the route pattern. When 911 is called, a disconnect tone is heard, when i call any other number it says ''number you have dialed cannot be ...'' note that inbound calls are working fine. Anybody seen dis issue before? Tjs. Sent from my Nokia phone ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] (no subject)
Hi, In PL WB 1 Lab 6 and onwards in the config tasks -prerequisites I see a NOTE: If you are using your own hw cisco 7961 phones instead of 7962 phones, please perform the following: first delete all 7962 phones and then run the BAT tool for phone install. We have laready preprovisioned a file that you simply need to import (and change MAC add) containing the 7961 phones types. My question is where do I find this file to import? I am not getting any response from ipexpert support team on this. Currently I just delete the 7962s and add my 7961s. Pls advise Thanks, Bhushan. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Cme background image
Make sure the filename is exactly as requested in the tftp request from phone. HTH Prashant On Sun, Oct 17, 2010 at 2:01 PM, fatai_adeku...@yahoo.com fatai_adeku...@yahoo.com wrote: I worked on putting a background image on a cucme router. I uploaded the background image successfully n configure ''tftp server flash ..'' on the cme. I created cnf files in telephony service, reloaded d router and checked if d image is available for the phone but to know avail. Anybody with an idea of what i am doing wrong? Tks. Sent from my Nokia phone ___ 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] Digit Manipulation on H323
hello all, I'm working on the workbook 1 lab 5 and i noticed what when i do digit manipulation either on the RP, RL or by using transformation patterns, the changes aren't sent to the GW, if my protocol is H.323 usually I need to create some dial rules on the Voice Gateway when I'm using MGCP i have no problem i was wondering if there is a setting on the ccm that will allow ccm to send the digit manipulation to the GW or does it has to be manually done at the GW level ? could you please explain a bit for me thank you ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Mgcp outbound call fails.
do you see your call hitting the gateway ? do you have a pri ? check your digit manipulation 2010/10/17 fatai_adeku...@yahoo.com fatai_adeku...@yahoo.com Hello guys, I experienced a problem where i cannot make any outbound call from an cucm through an mgcp gateway. My mgcp gateway was registered to d cucm, right pt/css were applied to d calling phone and correct partition to the route pattern. When 911 is called, a disconnect tone is heard, when i call any other number it says ''number you have dialed cannot be ...'' note that inbound calls are working fine. Anybody seen dis issue before? Tjs. Sent from my Nokia phone ___ 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] Digit Manipulation on H323
Hi C.P, Digit manipulation will be done on CUCM and will be sent to H323 as well, and the preference would be on the manipulations done within RL rather than on RP. So, i.e. RP is 91608.[2-9]XX and - if you put pre-dot and prefix 608 under RP, - and then you also do pre-dot and prefix 9 for specific RG (for your h323 gw) under RL, then your h323 gw will receive 9[2-9]XX hence, dial-peer pots on your h323 gw needed to terminate this call should have the same/similar destination-pattern configured, i.e: dial-peer voice 9 pots destination-pattern 9[2-9]..$ port 0/1/0:23 Now, the real trick comes if you want to actually influence your calling phone LCD digit presentations of DNIS (so, not ANI on the receiving end, but the actual dialed number on the calling end being presented on your phone from which you are dialing those digits - this is where the difference between mgcp and h323 gw can be seen). mgcp will present whatever manipulations you've done using RP (will not present back to calling phone LCD what you have done withing RG/RL manipulations though it will use those manipulations to send to the GW). however, in case of h323 gw, manipulations on DNIS done withing RG/RL will be also presented back to calling phone LCD. Now, since that is H323, you can still have one more chance to do your digits manipulations and influence back presenting of dialed digits to calling phone - voice transformation rules/profiles attached to pots dial-peer (or forward-digits under dial-peer but that one will not influence LCD DNIS presentation on the calling phone) i.e. if for above example we want to actually show 9 in front of local number, we can just put 'forward-digits 7' under above pots and that's it. dial-peer voice 9 pots destination-pattern 9[2-9]..$ port 0/1/0:23 forward-digits 7 But, if we would like to show ONLY local number, without leading 9 back to the caller on his ip phone LCD, then we would have to strip that 9 inside voice translation-rule, i.e: voice translation-rule 9 rule 1 /^9\([2-9]..$\)/ /\1/ p any sub t any sub voice translation-profile 9 translate called 9 and then add that to above dp: dial-peer voice 9 pots translation-profile out 9 destination-pattern 9[2-9]..$ port 0/1/0:23 so this will result in showing only 7 digits back to LCD of the calling phone. (if dialed number was 91234567, it will show back only 1234567). here, you can also include forward-digits as well, but translation-profile will still have precedence dial-peer voice 9 pots translation-profile out 9 destination-pattern 9[2-9]..$ port 0/1/0:23 forward 7 in both cases you are sending 7 digits to PSTN, just the difference is what you will present back to the caller who actually dialed this number. and that is the difference with mgcp, as you don't have that extra step to manipulate DNIS - all needs to be done on the CUCM withing RP, RG of RL or CalledPartTransformationPattern attached to outgoing mgcp gw. hope this will give you some clues how it works... cheers, G. On Sun, Oct 17, 2010 at 8:17 PM, Cristobal Priego cristobalpri...@gmail.com wrote: hello all, I'm working on the workbook 1 lab 5 and i noticed what when i do digit manipulation either on the RP, RL or by using transformation patterns, the changes aren't sent to the GW, if my protocol is H.323 usually I need to create some dial rules on the Voice Gateway when I'm using MGCP i have no problem i was wondering if there is a setting on the ccm that will allow ccm to send the digit manipulation to the GW or does it has to be manually done at the GW level ? could you please explain a bit for me thank you ___ 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] Mgcp outbound call fails.
do debug isdn q931, place the call and let us know what do you get On Sun, Oct 17, 2010 at 8:35 PM, Cristobal Priego cristobalpri...@gmail.com wrote: do you see your call hitting the gateway ? do you have a pri ? check your digit manipulation 2010/10/17 fatai_adeku...@yahoo.com fatai_adeku...@yahoo.com Hello guys, I experienced a problem where i cannot make any outbound call from an cucm through an mgcp gateway. My mgcp gateway was registered to d cucm, right pt/css were applied to d calling phone and correct partition to the route pattern. When 911 is called, a disconnect tone is heard, when i call any other number it says ''number you have dialed cannot be ...'' note that inbound calls are working fine. Anybody seen dis issue before? Tjs. Sent from my Nokia phone ___ 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] Cme background image
what do you get on the phone when you try to click on background images selection? did you enable http server on your router? also, folder path is very important as for the phone types you are using. i recommend to use this document for that as it lists formats/folders for different phone types (and that is still valid for CME as well) http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_configuration_example09186a0080b3690c.shtml On Sun, Oct 17, 2010 at 8:04 PM, Prashant Patel prashantpatel...@gmail.comwrote: Make sure the filename is exactly as requested in the tftp request from phone. HTH Prashant On Sun, Oct 17, 2010 at 2:01 PM, fatai_adeku...@yahoo.com fatai_adeku...@yahoo.com wrote: I worked on putting a background image on a cucme router. I uploaded the background image successfully n configure ''tftp server flash ..'' on the cme. I created cnf files in telephony service, reloaded d router and checked if d image is available for the phone but to know avail. Anybody with an idea of what i am doing wrong? Tks. Sent from my Nokia phone ___ 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] Digit Manipulation on H323
thank you very much it really make sense and what would happen to the call type ? if i use a called party transformation pattern to set the called type to: national, subscriber, intl, on the MGCP gateway it's no problem the call is sent out on the PRI with the proper digit manipulation and the proper call type on the other hand on H323. stripping digits from ccm are passed to the h323 gw, but that's it, no call type at all this is when the translation rules comes in place, right ? 2010/10/17 Goran Selthofer seltho...@gmail.com Hi C.P, Digit manipulation will be done on CUCM and will be sent to H323 as well, and the preference would be on the manipulations done within RL rather than on RP. So, i.e. RP is 91608.[2-9]XX and - if you put pre-dot and prefix 608 under RP, - and then you also do pre-dot and prefix 9 for specific RG (for your h323 gw) under RL, then your h323 gw will receive 9[2-9]XX hence, dial-peer pots on your h323 gw needed to terminate this call should have the same/similar destination-pattern configured, i.e: dial-peer voice 9 pots destination-pattern 9[2-9]..$ port 0/1/0:23 Now, the real trick comes if you want to actually influence your calling phone LCD digit presentations of DNIS (so, not ANI on the receiving end, but the actual dialed number on the calling end being presented on your phone from which you are dialing those digits - this is where the difference between mgcp and h323 gw can be seen). mgcp will present whatever manipulations you've done using RP (will not present back to calling phone LCD what you have done withing RG/RL manipulations though it will use those manipulations to send to the GW). however, in case of h323 gw, manipulations on DNIS done withing RG/RL will be also presented back to calling phone LCD. Now, since that is H323, you can still have one more chance to do your digits manipulations and influence back presenting of dialed digits to calling phone - voice transformation rules/profiles attached to pots dial-peer (or forward-digits under dial-peer but that one will not influence LCD DNIS presentation on the calling phone) i.e. if for above example we want to actually show 9 in front of local number, we can just put 'forward-digits 7' under above pots and that's it. dial-peer voice 9 pots destination-pattern 9[2-9]..$ port 0/1/0:23 forward-digits 7 But, if we would like to show ONLY local number, without leading 9 back to the caller on his ip phone LCD, then we would have to strip that 9 inside voice translation-rule, i.e: voice translation-rule 9 rule 1 /^9\([2-9]..$\)/ /\1/ p any sub t any sub voice translation-profile 9 translate called 9 and then add that to above dp: dial-peer voice 9 pots translation-profile out 9 destination-pattern 9[2-9]..$ port 0/1/0:23 so this will result in showing only 7 digits back to LCD of the calling phone. (if dialed number was 91234567, it will show back only 1234567). here, you can also include forward-digits as well, but translation-profile will still have precedence dial-peer voice 9 pots translation-profile out 9 destination-pattern 9[2-9]..$ port 0/1/0:23 forward 7 in both cases you are sending 7 digits to PSTN, just the difference is what you will present back to the caller who actually dialed this number. and that is the difference with mgcp, as you don't have that extra step to manipulate DNIS - all needs to be done on the CUCM withing RP, RG of RL or CalledPartTransformationPattern attached to outgoing mgcp gw. hope this will give you some clues how it works... cheers, G. On Sun, Oct 17, 2010 at 8:17 PM, Cristobal Priego cristobalpri...@gmail.com wrote: hello all, I'm working on the workbook 1 lab 5 and i noticed what when i do digit manipulation either on the RP, RL or by using transformation patterns, the changes aren't sent to the GW, if my protocol is H.323 usually I need to create some dial rules on the Voice Gateway when I'm using MGCP i have no problem i was wondering if there is a setting on the ccm that will allow ccm to send the digit manipulation to the GW or does it has to be manually done at the GW level ? could you please explain a bit for me thank you ___ 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] Digit Manipulation on H323
To set Called and Calling type on H323 GW it is preferred to use voice-translation rules on the gateway so that if you go into srst you dont have to do it again. However if no srst you can set it in CUCM. HTH, Prashant On Sun, Oct 17, 2010 at 3:02 PM, Cristobal Priego cristobalpri...@gmail.com wrote: thank you very much it really make sense and what would happen to the call type ? if i use a called party transformation pattern to set the called type to: national, subscriber, intl, on the MGCP gateway it's no problem the call is sent out on the PRI with the proper digit manipulation and the proper call type on the other hand on H323. stripping digits from ccm are passed to the h323 gw, but that's it, no call type at all this is when the translation rules comes in place, right ? 2010/10/17 Goran Selthofer seltho...@gmail.com Hi C.P, Digit manipulation will be done on CUCM and will be sent to H323 as well, and the preference would be on the manipulations done within RL rather than on RP. So, i.e. RP is 91608.[2-9]XX and - if you put pre-dot and prefix 608 under RP, - and then you also do pre-dot and prefix 9 for specific RG (for your h323 gw) under RL, then your h323 gw will receive 9[2-9]XX hence, dial-peer pots on your h323 gw needed to terminate this call should have the same/similar destination-pattern configured, i.e: dial-peer voice 9 pots destination-pattern 9[2-9]..$ port 0/1/0:23 Now, the real trick comes if you want to actually influence your calling phone LCD digit presentations of DNIS (so, not ANI on the receiving end, but the actual dialed number on the calling end being presented on your phone from which you are dialing those digits - this is where the difference between mgcp and h323 gw can be seen). mgcp will present whatever manipulations you've done using RP (will not present back to calling phone LCD what you have done withing RG/RL manipulations though it will use those manipulations to send to the GW). however, in case of h323 gw, manipulations on DNIS done withing RG/RL will be also presented back to calling phone LCD. Now, since that is H323, you can still have one more chance to do your digits manipulations and influence back presenting of dialed digits to calling phone - voice transformation rules/profiles attached to pots dial-peer (or forward-digits under dial-peer but that one will not influence LCD DNIS presentation on the calling phone) i.e. if for above example we want to actually show 9 in front of local number, we can just put 'forward-digits 7' under above pots and that's it. dial-peer voice 9 pots destination-pattern 9[2-9]..$ port 0/1/0:23 forward-digits 7 But, if we would like to show ONLY local number, without leading 9 back to the caller on his ip phone LCD, then we would have to strip that 9 inside voice translation-rule, i.e: voice translation-rule 9 rule 1 /^9\([2-9]..$\)/ /\1/ p any sub t any sub voice translation-profile 9 translate called 9 and then add that to above dp: dial-peer voice 9 pots translation-profile out 9 destination-pattern 9[2-9]..$ port 0/1/0:23 so this will result in showing only 7 digits back to LCD of the calling phone. (if dialed number was 91234567, it will show back only 1234567). here, you can also include forward-digits as well, but translation-profile will still have precedence dial-peer voice 9 pots translation-profile out 9 destination-pattern 9[2-9]..$ port 0/1/0:23 forward 7 in both cases you are sending 7 digits to PSTN, just the difference is what you will present back to the caller who actually dialed this number. and that is the difference with mgcp, as you don't have that extra step to manipulate DNIS - all needs to be done on the CUCM withing RP, RG of RL or CalledPartTransformationPattern attached to outgoing mgcp gw. hope this will give you some clues how it works... cheers, G. On Sun, Oct 17, 2010 at 8:17 PM, Cristobal Priego cristobalpri...@gmail.com wrote: hello all, I'm working on the workbook 1 lab 5 and i noticed what when i do digit manipulation either on the RP, RL or by using transformation patterns, the changes aren't sent to the GW, if my protocol is H.323 usually I need to create some dial rules on the Voice Gateway when I'm using MGCP i have no problem i was wondering if there is a setting on the ccm that will allow ccm to send the digit manipulation to the GW or does it has to be manually done at the GW level ? could you please explain a bit for me thank you ___ 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
Re: [OSL | CCIE_Voice] Digit Manipulation on H323
thank you all for your comments I've been debugging the gateway i have digit manipulation on the rp, rl. I also have a called pattern transformation css applied to the h323 gateway i've reset the GW multiple times however when i make the call this is what i see Exclusive, Channel 1 Display i = 'HQ Phone 2' Calling Party Number HQ-RTR(config-gk)# i = 0x0081, '12123945002' Plan:Unknown, Type:Unknown Called Party Number i = 0x80, '16178632683' Plan:Unknown, Type:Unknown My calling party transformation CSS is being hit by the GW 10/17/2010 16:01:47.979 CCM|SPROC DATransformMatch - matchNumber [5002]transformCSSPkid [c85a253f-f88d-7fbb-50b0-ff8aaacde5c3] transformationCss [gw-hq-pt] patternUsage [15] paternNodeID [b071ad82-eaf4-c3c5-09df-6acba1ad0832] OutpulsedNum.nd [+12123945002] tn [2] pi [1] npi [0]|CLID::StandAloneClusterNID::10.10.210.11LVL::ArbitraryMASK::0800 however i didn't see the called party transformation pattern being hit and i can't get the call type to work properly nor the + dialing without Dialing Rules with dialing rules works great however i was wondering if it could be done from CUCM 2010/10/17 Prashant Patel prashantpatel...@gmail.com To set Called and Calling type on H323 GW it is preferred to use voice-translation rules on the gateway so that if you go into srst you dont have to do it again. However if no srst you can set it in CUCM. HTH, Prashant On Sun, Oct 17, 2010 at 3:02 PM, Cristobal Priego cristobalpri...@gmail.com wrote: thank you very much it really make sense and what would happen to the call type ? if i use a called party transformation pattern to set the called type to: national, subscriber, intl, on the MGCP gateway it's no problem the call is sent out on the PRI with the proper digit manipulation and the proper call type on the other hand on H323. stripping digits from ccm are passed to the h323 gw, but that's it, no call type at all this is when the translation rules comes in place, right ? 2010/10/17 Goran Selthofer seltho...@gmail.com Hi C.P, Digit manipulation will be done on CUCM and will be sent to H323 as well, and the preference would be on the manipulations done within RL rather than on RP. So, i.e. RP is 91608.[2-9]XX and - if you put pre-dot and prefix 608 under RP, - and then you also do pre-dot and prefix 9 for specific RG (for your h323 gw) under RL, then your h323 gw will receive 9[2-9]XX hence, dial-peer pots on your h323 gw needed to terminate this call should have the same/similar destination-pattern configured, i.e: dial-peer voice 9 pots destination-pattern 9[2-9]..$ port 0/1/0:23 Now, the real trick comes if you want to actually influence your calling phone LCD digit presentations of DNIS (so, not ANI on the receiving end, but the actual dialed number on the calling end being presented on your phone from which you are dialing those digits - this is where the difference between mgcp and h323 gw can be seen). mgcp will present whatever manipulations you've done using RP (will not present back to calling phone LCD what you have done withing RG/RL manipulations though it will use those manipulations to send to the GW). however, in case of h323 gw, manipulations on DNIS done withing RG/RL will be also presented back to calling phone LCD. Now, since that is H323, you can still have one more chance to do your digits manipulations and influence back presenting of dialed digits to calling phone - voice transformation rules/profiles attached to pots dial-peer (or forward-digits under dial-peer but that one will not influence LCD DNIS presentation on the calling phone) i.e. if for above example we want to actually show 9 in front of local number, we can just put 'forward-digits 7' under above pots and that's it. dial-peer voice 9 pots destination-pattern 9[2-9]..$ port 0/1/0:23 forward-digits 7 But, if we would like to show ONLY local number, without leading 9 back to the caller on his ip phone LCD, then we would have to strip that 9 inside voice translation-rule, i.e: voice translation-rule 9 rule 1 /^9\([2-9]..$\)/ /\1/ p any sub t any sub voice translation-profile 9 translate called 9 and then add that to above dp: dial-peer voice 9 pots translation-profile out 9 destination-pattern 9[2-9]..$ port 0/1/0:23 so this will result in showing only 7 digits back to LCD of the calling phone. (if dialed number was 91234567, it will show back only 1234567). here, you can also include forward-digits as well, but translation-profile will still have precedence dial-peer voice 9 pots translation-profile out 9 destination-pattern 9[2-9]..$ port 0/1/0:23 forward 7 in both cases you are sending 7 digits to PSTN, just the difference is what you will present back to the caller who actually dialed this number. and that is the difference with
[OSL | CCIE_Voice] maximum sessions 0
Hello, I am going to assume that you have a FXO or FXS voice card, given that, you will need a second PVDM to be able to use both conference bridges and the voice FXO/FXS in your Cisco 2811. You can also migrate this voice card to a network module with DSP resources to free up more DSP resources for your router. I know PVDM’s are not inexpensive but if you need hardware conference ability and FXO/FXS cards you will need multiple PVDM’s http://www.cisco.com/cgi-bin/Support/DSP/cisco_prodsel.pl Sincerely, Bill ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Call Forward Unregistered
Scenario: In SRST mode: HQ MGCP gateway and Site-B H322 gateway Site-C H322 gateway cme HQ and Site C phones are being able to call SiteB Phone 1 using 4 digits dialing in SRST.(Wan failure) I use call forward unregistered feature. When I call from HQ Phone-1 call routed through HQ Gateway. When I call from Site-C Phone-1 call routed through the GK first and then HQ Gateway. Below is the display I am getting on my Site-B phone display. Forward HQ Phone 1 (2001) For 3001 By3001 Forward Site-C Phone 1 (4001) For 3001 By3001 My question how can I achieve below display in FOR and BY field it should be E.164 number format and than 4 digits internal ID Forward (2001) For +19723033001 (3...) By+19723033001 (3...) Forward (4001) For +19723033001 (3...) By+19723033001 (3...) Thanking you in anticipation folks. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Call Forward Unregistered
I ran into this same issue. Supposedly it's a problem with CUCM 7.0 and VMWare. If you go to the Device Phone and click on the Site B phones Line and specifically assign the Voicemail Profile to the Line it might work. I had success a couple of times doing this, but then after resetting my rack the last time and assigning the VM profile to the Line I still had this issue. On Oct 17, 2010, at 3:28 PM, Afzal Bhutta wrote: Scenario: In SRST mode: HQ MGCP gateway and Site-B H322 gateway Site-C H322 gateway cme HQ and Site C phones are being able to call SiteB Phone 1 using 4 digits dialing in SRST.(Wan failure) I use call forward unregistered feature. When I call from HQ Phone-1 call routed through HQ Gateway. When I call from Site-C Phone-1 call routed through the GK first and then HQ Gateway. Below is the display I am getting on my Site-B phone display. Forward HQ Phone 1 (2001) For 3001 By3001 Forward Site-C Phone 1 (4001) For 3001 By3001 My question how can I achieve below display in FOR and BY field it should be E.164 number format and than 4 digits internal ID Forward (2001) For +19723033001 (3...) By+19723033001 (3...) Forward (4001) For +19723033001 (3...) By+19723033001 (3...) Thanking you in anticipation folks. ___ 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] CUC WMI issue
Fellows, MWI issue CUC integrated with CUCM. Users imported from CUCM MWI extension same (1998/1999) on both CUCM CUC. Null partition CSS Able to call both extensions for light on/off from all phones. One phone system in CUC Rebooted both CUC CUCM After all the tasks above, No MWI when message is left on any of the phones, though by pressing the message button you can retrieve these messages. Anyone with similar issue or an idea? Thanks in advance.___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Call Forward Unregistered
Hello guys If you want to manipulate this with CUCM the place to change the redirected number is the VM profile as indicated by Mark. Alternatively you could attach an additional rule to the translation-profile plugged inbound to the POTS call leg in the branch router in SRST mode and configure it to change the redirect-called number from to the e164 that you are after. Cheers On Mon, Oct 18, 2010 at 11:36 AM, Mark Holloway m...@markholloway.com wrote: I ran into this same issue. Supposedly it's a problem with CUCM 7.0 and VMWare. If you go to the Device Phone and click on the Site B phones Line and specifically assign the Voicemail Profile to the Line it might work. I had success a couple of times doing this, but then after resetting my rack the last time and assigning the VM profile to the Line I still had this issue. On Oct 17, 2010, at 3:28 PM, Afzal Bhutta wrote: Scenario: In SRST mode: HQ MGCP gateway and Site-B H322 gateway Site-C H322 gateway cme HQ and Site C phones are being able to call SiteB Phone 1 using 4 digits dialing in SRST.(Wan failure) I use call forward unregistered feature. When I call from HQ Phone-1 call routed through HQ Gateway. When I call from Site-C Phone-1 call routed through the GK first and then HQ Gateway. Below is the display I am getting on my Site-B phone display. Forward HQ Phone 1 (2001) For 3001 By3001 Forward Site-C Phone 1 (4001) For 3001 By3001 My question how can I achieve below display in FOR and BY field it should be E.164 number format and than 4 digits internal ID Forward (2001) For +19723033001 (3...) By+19723033001 (3...) Forward (4001) For +19723033001 (3...) By+19723033001 (3...) Thanking you in anticipation folks. ___ 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] Cme background image
Check... 1. Device Pool - Any Local route group selected? 2. CSS - Any patition in selected partitions? Date: Sun, 17 Oct 2010 18:01:19 + From: fatai_adeku...@yahoo.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Cme background image I worked on putting a background image on a cucme router. I uploaded the background image successfully n configure ''tftp server flash ..'' on the cme. I created cnf files in telephony service, reloaded d router and checked if d image is available for the phone but to know avail. Anybody with an idea of what i am doing wrong? Tks. Sent from my Nokia phone ___ 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] Call Forward Unregistered
I think the main thing to understand is that it should work using E164 in For/By under normal circumstances and everything else we are suggesting is a work around to a known bug with CUCM 7.0 and VMWare. On Oct 17, 2010, at 3:56 PM, Daniel Berlinski wrote: Hello guys If you want to manipulate this with CUCM the place to change the redirected number is the VM profile as indicated by Mark. Alternatively you could attach an additional rule to the translation-profile plugged inbound to the POTS call leg in the branch router in SRST mode and configure it to change the redirect-called number from to the e164 that you are after. Cheers On Mon, Oct 18, 2010 at 11:36 AM, Mark Holloway m...@markholloway.com wrote: I ran into this same issue. Supposedly it's a problem with CUCM 7.0 and VMWare. If you go to the Device Phone and click on the Site B phones Line and specifically assign the Voicemail Profile to the Line it might work. I had success a couple of times doing this, but then after resetting my rack the last time and assigning the VM profile to the Line I still had this issue. On Oct 17, 2010, at 3:28 PM, Afzal Bhutta wrote: Scenario: In SRST mode: HQ MGCP gateway and Site-B H322 gateway Site-C H322 gateway cme HQ and Site C phones are being able to call SiteB Phone 1 using 4 digits dialing in SRST.(Wan failure) I use call forward unregistered feature. When I call from HQ Phone-1 call routed through HQ Gateway. When I call from Site-C Phone-1 call routed through the GK first and then HQ Gateway. Below is the display I am getting on my Site-B phone display. Forward HQ Phone 1 (2001) For 3001 By3001 Forward Site-C Phone 1 (4001) For 3001 By3001 My question how can I achieve below display in FOR and BY field it should be E.164 number format and than 4 digits internal ID Forward (2001) For +19723033001 (3...) By+19723033001 (3...) Forward (4001) For +19723033001 (3...) By+19723033001 (3...) Thanking you in anticipation folks. ___ 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] Cme background image
I always use alias to avoid some path related issue Use debug tftp events to see what the phone is looking for when you are pressing around. Sent from my iPhone Pls pardon my fat fingers. On Oct 18, 2010, at 2:58 AM, Goran Selthofer seltho...@gmail.com wrote: what do you get on the phone when you try to click on background images selection? did you enable http server on your router? also, folder path is very important as for the phone types you are using. i recommend to use this document for that as it lists formats/folders for different phone types (and that is still valid for CME as well) http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_configuration_example09186a0080b3690c.shtml On Sun, Oct 17, 2010 at 8:04 PM, Prashant Patel prashantpatel...@gmail.com wrote: Make sure the filename is exactly as requested in the tftp request from phone. HTH Prashant On Sun, Oct 17, 2010 at 2:01 PM, fatai_adeku...@yahoo.com fatai_adeku...@yahoo.com wrote: I worked on putting a background image on a cucme router. I uploaded the background image successfully n configure ''tftp server flash ..'' on the cme. I created cnf files in telephony service, reloaded d router and checked if d image is available for the phone but to know avail. Anybody with an idea of what i am doing wrong? Tks. Sent from my Nokia phone ___ 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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
