[OSL | CCIE_Voice] CCIE VOICE LAB RACK for SALE IN Middle east
CCIE VOICE LAB RACK for SALE IN Middle east, I have all the boxes for the lab v3 requirement and will guarantee safe arrival of all the items: Core Switch: 1 x Cisco Catalyst WS-C3560G-24PS 24 Port PoE Terminal Server 1 x Cisco 2511 (1 Ethernet Port, 2 Serial Ports, 16 Terminal lines) HQ Site 1 x Cisco 2801 Router CISCO2801 2801 w/ AC PWR 1 x WIC2-2MFT-G703 RJ-48 Multiflex Trunk - T1 2 x PVDM2-16 16-Channel Packet Voice/Fax DSP Module 1 x WIC/VIC 2 Serial WAN Interface Card Site B 1 x Cisco 2811 Router CISCO2811 2811 w/ AC PWR 1 x VWIC2-1MFT-T1/E1 1-Port RJ-48 Multiflex Trunk - T1 1 x HWIC-4ESW (Inline Power) HWIC-4ESW Four port 10/100 Ethernet switch interface card 1 x WIC/VIC 2 Serial WAN Interface Card 1 x PVDM2-16 16-Channel Packet Voice/Fax DSP Module 1 x PVDM2-46 16-Channel Packet Voice/Fax DSP Module Site C 1 x Cisco 2811 Router CISCO2811 2811 w/ AC PWR 1 x NM-CUE NM-CUE Cisco Unity Express Network Module (includes SCUE-12-VM) 1 x VWIC-1MFT-E1 VWIC-1MFT-E1 1-Port RJ-48 Multiflex Trunk - E1 1 x WIC-2T WIC-2T 2-Port Serial WAN Interface Card 1 x PVDM2-64 64-Channel Packet Voice/Fax DSP Module PSTN Simulator 1 x Cisco 2801 Router AC PWR 1 x VWIC-2MFT-T1 VWIC-2MFT-T1 2-Port RJ-48 Multiflex Trunk - T1 1 x VWIC-1MFT-E1 VWIC-1MFT-E1 1-Port RJ-48 Multiflex Trunk - E1 2 x WIC-2T Serial WAN Interface Card 1 x PVDM2-64 64-Channel Packet Voice/Fax DSP Module 1 x PVDM2-16 64-Channel Packet Voice/Fax DSP Module IP Phones 8 x 7965 IP Phones CP-7965G Cisco IP Phone 7960G, FR Switch: 2610 WIC 4 port with Cables Cable: 3 x Serial back-to-back cable 8 x 10FTCat6 Patch cables 3 x Octal ASYNC Cable With RACK STAND ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] MOH Flash Problem
Hi Cliff, Could please let me know work around for this, If try upload second MOH file [ringback.wav] in flash, even it is not playing MOH files , If I am having two moh file in HQ, how to play the moh file in SITE-b [ site-b is in MOH flash ]. Thanks, Bala. --- On Sun, 26/4/09, Cliff McGlamry cl...@mcglamry.net wrote: From: Cliff McGlamry cl...@mcglamry.net Subject: Re: [OSL | CCIE_Voice] MOH Flash Problem To: mmailb...@yahoo.com, ccie_voice@onlinestudylist.com Date: Sunday, 26 April, 2009, 11:45 PM MOH from Flash allows only ONE MOH stream. You've defined two. The MOH audiot source for the second phone isn't on the flash (and can't be), so you get the dead air. - Original Message - From: Balamurugan Singaram To: ccie_voice@onlinestudylist.com Sent: Sunday, April 26, 2009 5:49 AM Subject: [OSL | CCIE_Voice] MOH Flash Problem In HQ Phone -1 [2001], I am using moh file source as sample audio source and HQ phone-2 [2002], I am using MOH file as ringback.wav. In Site-B I am using moh on flash, when I press hold in IPPhone from 2001 [audio source as sample source], I am hearning moh music in site-b, when I press hold in 2002 [audio source as ringback ] I am hearing dead air in site-b. In PSTN according to sample audio source I am hearing music it is normal, any suggestion to troubleshoot the above problem. Get your new Email address! Grab the Email name you've always wanted before someone else does! New Email addresses available on Yahoo! Get the Email name you#39;ve always wanted on the new @ymail and @rocketmail. Hurry before someone else does! http://mail.promotions.yahoo.com/newdomains/aa/
Re: [OSL | CCIE_Voice] VPIM - CME To Unity Not working...
Hi Clif,
The below can be done in CLI also, if it possible could you please update the
CLI steps to.
In General the CLI config for VPIM as follows:
!
network location id 100
name unity
email doamin unity.ccievoice.com
enable
end
!
network location id 200
name CUE
enable
email domain cue.ccievoice.com
end
!
network local location id 200
!
Could you please light me how to add the vpim-broadcast step also in CLI
itself.
Thank you.
Thanks,
Bala.
--- On Thu, 23/4/09, Cliff McGlamry cl...@mcglamry.net wrote:
From: Cliff McGlamry cl...@mcglamry.net
Subject: Re: [OSL | CCIE_Voice] VPIM - CME To Unity Not working...
To: Tech Guy tech...@oletu.com, ccie_voice@onlinestudylist.com
CCIE_Voice@onlinestudylist.com
Date: Thursday, 23 April, 2009, 9:58 AM
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#yiv1699913982 P.msoNormal {
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Oh...ok.
On CUE, go to the network locations setup. When you open the Unity location,
you'll see a blank that has vpim-broadcast in it by default. Blow that out and
put in the number (1001 or whatever number you assigned to your distribution
list on Unity). Then try it again and it will work (via the TUI).
Cliff
- Original Message -
From: Tech Guy
To: Cliff McGlamry ; ccie_voice@onlinestudylist.com
Sent: Thursday, April 23, 2009 12:14 AM
Subject: Re: [OSL | CCIE_Voice] VPIM - CME To Unity Not working...
I tried it; UNITY is rejecting the broadcast; CUE is sending the broadcaast
message to: vpim-broadc...@unity-lab.voip.lab; since that box does not exit on
UNITY, the mailserver rejected it. I then created a PDL on UNITY and added the
subscribers to the PDL; UNITY did not reject the message after that, but I only
got it as an attachment in Outlook Express and not on the phone.
I am still thinking Exchange Voice Connector is the problem; I did this on
oneof the IP Expert Pods; no time to troubleshoot Exchange Schema; but I did
check the Voice Mail connect to make such it has the correct SMTP policy.
Any suggestion?
see trace below.
Thanks.
Tech Guy
output trace-
4294 04/23 00:33:21.604 netw vpim 4 VPIM:
--=_NextPart_000_0041_01C9C3AE.AA9D4850--
4382 04/23 00:49:43.673 netw vpim 3 VPIM
4382 04/23 00:49:43.694 netw vpim 3 VPIM: To:
vpim-broadc...@unity-lab.voip.lab
4382 04/23 00:49:43.697 netw vpim 3 VPIM: From: Hel Bush4...@cue.voip.lab
4382 04/23 00:49:43.702 netw vpim 3 VPIM: Date: Thu, 23 Apr 2009 00:49:42 -0400
(EDT)
4382 04/23 00:49:43.702 netw vpim 3 VPIM: MIME-Version: 1.0 (Voice 2.0)
4382 04/23 00:49:43.702 netw vpim 3 VPIM: Content-Type:
Multipart/Voice-Message; Version=2.0;
4382 04/23 00:49:43.703 netw vpim 3 VPIM:
Boundary===VpimMsg==1240462183672
4382 04/23 00:49:43.703 netw vpim 3 VPIM: Content-Transfer-Encoding: 7bit
4382 04/23 00:49:43.704 netw vpim 3 VPIM: Message-ID:
FTX1030A4GU-AIM-FOC10120VQD-1240450494312-NBCM
4382 04/23 00:49:43.704 netw vpim 3 VPIM: Subject: Broadcast Message from CUE
Location 200
4382 04/23 00:49:43.705 netw vpim 3 VPIM: X-CISCO-SBM-ID:
FTX1030A4GU-AIM-FOC10120VQD-1240450494312-NBCM
4382 04/23 00:49:43.710 netw vpim 3 VPIM: X-CISCO-SBM-START-TIME: Thu, 23 Apr
2009 00:49:42 -0400 (EDT)
4382 04/23 00:49:43.714 netw vpim 3 VPIM: X-CISCO-SBM-END-TIME: Sat, 23 May
2009 00:49:18 -0400 (EDT)
4382 04/23 00:49:43.719 netw vpim 3 VPIM: X-CISCO-SBM-CUSTOM1:
437823F7B8173A212C8CF8B93E51FD60
4382 04/23 00:49:43.719 netw vpim 3 VPIM:
4382 04/23 00:49:43.720 netw vpim 3 VPIM: --==VpimMsg==1240462183672
4382 04/23 00:49:43.720 netw vpim 3 VPIM: Content-Type: Audio/32KADPCM
4382 04/23 00:49:43.720 netw vpim 3 VPIM: Content-Transfer-Encoding: Base64
4382 04/23 00:49:43.721 netw vpim 3 VPIM: Content-Description: VPIM Message
4382 04/23 00:49:43.722 netw vpim 3 VPIM: Content-Disposition: inline;
voice=Voice-Message
4382 04/23 00:49:43.722 netw vpim 3 VPIM: Content-ID:
FTX1030A4GU-AIM-FOC10120VQD-1240450494312-NBCM
4382 04/23 00:49:43.722 netw vpim 3 VPIM:
4382 04/23 00:49:43.729 netw vpim 7
4382 04/23 00:49:44.136 netw vpim 3 VPIMAUDIO:
4382 04/23 00:49:44.211 netw vpim 3 VPIM:
4382 04/23 00:49:44.212 netw vpim 3 VPIM: --==VpimMsg==1240462183672--
4383 04/23 00:49:45.087 netw vpim 4 VPIM: thread-index:
AcnDxekmR8XY/Uc9QQmjLgXFMpRI/w==
4383 04/23 00:49:45.088 netw vpim 4 VPIM: X-AvVPIMMessage: VOIP.LAB
4383 04/23 00:49:45.089 netw vpim 4 VPIM: From: postmas...@unity-lab.voip.lab
4383 04/23 00:49:45.090 netw vpim 4 VPIM: To: 4...@cue.voip.lab
Re: [OSL | CCIE_Voice] Auto QOS for LAN QOS
Auto QOS are allowed in real lab first ?, Basant can you explain how to enable auto qos in switch, I have never try this. --- On Sun, 15/3/09, basant yadav basant.ya...@gmail.com wrote: From: basant yadav basant.ya...@gmail.com Subject: [OSL | CCIE_Voice] Auto QOS for LAN QOS To: OSL Group ccie_voice@onlinestudylist.com Date: Sunday, 15 March, 2009, 5:40 PM Hi All Has anyone ever used/tried in practice or real lab to use Auto QOS for LAN/Campus QOS. I think its pretty useful if you don't want to mark anything on the switches and requirment is to only trust the COS coming from phones. It saves lot of time. I am about to start practicing it but would like some second opinions before I end up in a mess and has to change my mindset. - Basant Get your preferred Email name! Now you can @ymail.com and @rocketmail.com. http://mail.promotions.yahoo.com/newdomains/aa/
[OSL | CCIE_Voice] Transcoder and FXS problem
We are having 2811 for lab practice. In HQ site alone, I have configured IOS transcoder and using MRGL I have update where ever we need transcoder. Now my problem is from FXS 212x, I am not able to call SITE-B 300x number from 212x, if I configure transcoder in site-b I am able to make call from 212x to site-b Do you have any suggestion without configuring IOS transcoder in SITE-B, we can not call 3001 from HQ FXS. Thanks New Email names for you! Get the Email name you#39;ve always wanted on the new @ymail and @rocketmail. Hurry before someone else does! http://mail.promotions.yahoo.com/newdomains/aa/
Re: [OSL | CCIE_Voice] h.323 gateway config call preserve
IPExpert V3 Mock lab workbook, related to h.323 gateway I am able to see below command for call preservation: voice service voip h323 no h225 timeout keepalive even after update this command, my call are not preserved when the both ccm SUB and PUB goes down, I am missing something, could please some one light me. Thanks, --- On Tue, 10/3/09, Scott ODonnell scott.odonn...@gmail.com wrote: From: Scott ODonnell scott.odonn...@gmail.com Subject: Re: [OSL | CCIE_Voice] h.323 gateway config call preserve To: Cliff McGlamry cl...@mcglamry.net Cc: Kumar, Narinder narinder.ku...@uxcg.com.au, mmailb...@yahoo.com, ccie_voice@onlinestudylist.com Date: Tuesday, 10 March, 2009, 12:09 PM I think you need the no h225 timeout keepalive configured under the voice service voip / h323. This will allow the call to stay up after the GW stops receiving keepalives from the CM. http://www.cisco.com/en/US/products/sw/voicesw/ps2169/products_configuration_guide_chapter09186a0080558061.html H.323 Gateways and SRST On H.323 gateways, when the WAN link fails, active calls from Cisco IP phones to the PSTN are not maintained by default. Call preservation may work with the no h225 timeout keepalive command, but call preservation using the no h225 timeout keepalive command is not officially supported by Cisco Technical Support. Under default configuration, the H.323 gateway maintains a keepalive signal with Cisco CallManager and terminates H.323-to-PSTN calls if the keepalive signal fails, for example if the WAN link fails. To disable this behavior and help preserve existing calls from local IP phones, you can use the no h225 timeout keepalivecommand. Disabling the keepalive mechanism only affects calls that will be torn down as a result of the loss of the H.225 keepalive signal. For information regarding disconnecting a call when an inactive condition is detected. see the Media Inactive Call Detection document. On Tue, Mar 10, 2009 at 2:30 AM, Cliff McGlamry cl...@mcglamry.net wrote: I believe there is also a service parameter command to allow call preservation in the event CCM goes down under h.323 in the CCM Service parameters. Seems like I've seen it there - Original Message - From: Kumar, Narinder To: mmailb...@yahoo.com ; ccie_voice@onlinestudylist.com Sent: Tuesday, March 10, 2009 2:18 AM Subject: Re: [OSL | CCIE_Voice] h.323 gateway config call preserve Need to add some more commands From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Balamurugan Singaram Sent: Tuesday, 10 March 2009 4:43 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] h.323 gateway config call preserve Hi, For h.323 gateway config call preserve in BR1 site the following config are ok, please some one comment on h.323 gateway config call preservation ? voice class h323 1 h225 timeout tcp establish 3 voice class codec 1---Need to add this codec preference 1 g711ulaw codec preference 2 g711alaw codec preference 3 g729r8 interface loopback 0 h323-gateway voip bind srcaddr 172.27.x.x h323-gateway voip interface ---Need to add this dial-peer voice 10 voip destination-pattern 2...$ session target ipv4:172.3.64.100 dtmf-relay h245-alphanumeric no huntstop voice-class h323 1 voice-class codec 1---Need to add this no vad ! dial-peer voice 11 voip destination-pattern 2...$ session target ipv4:172.3.64.101 dtmf-relay h245-alphanumeric preference 1 voice-class h323 1 voice-class codec 1---Need to add this no vad ! dial-peer voice 9 pots destination-pattern 9T port 0/0/0:23 ! on Service parameter: Stop routing on Unallocated number flag - false Thanks New Email addresses available on Yahoo! Get the Email name you've always wanted on the new @ymail and @rocketmail. Hurry before someone else does! CONFIDENTIALITY - The information contained in this electronic mail message is confidential and is intended solely for the addressee(s). If you are not an authorised recipient of this message please contact Getronics Australia immediately by reply email and destroy/delete this message from your computer. Any unauthorised form of reproduction of this message, or part thereof, is strictly prohibited. DISCLAIMER - Unless specifically indicated otherwise, the views and opinions expressed in this email are those of the sender and not Getronics Australia. While we endeavour to protect our network from computer viruses, Getronics Australia does not warrant that this email or any attachments are free of viruses or any other defects or errors. It is the duty of the recipient to virus scan and otherwise test any information contained in this email before loading onto any computer system. New Email addresses available on Yahoo! Get the Email name you#39;ve always
[OSL | CCIE_Voice] h.323 gateway config call preserve
Hi, For h.323 gateway config call preserve in BR1 site the following config are ok, please some one comment on h.323 gateway config call preservation ? voice class h323 1 h225 timeout tcp establish 3 interface loopback 0 h323-gateway voip bind srcaddr 172.27.x.x dial-peer voice 10 voip destination-pattern 2...$ session target ipv4:172.3.64.100 dtmf-relay h245-alphanumeric no huntstop voice-class h323 1 no vad ! dial-peer voice 11 voip destination-pattern 2...$ session target ipv4:172.3.64.101 dtmf-relay h245-alphanumeric preference 1 voice-class h323 1 no vad ! dial-peer voice 9 pots destination-pattern 9T port 0/0/0:23 ! on Service parameter: Stop routing on Unallocated number flag - false Thanks Get your preferred Email name! Now you can @ymail.com and @rocketmail.com. http://mail.promotions.yahoo.com/newdomains/aa/
[OSL | CCIE_Voice] Translation-profile outgoing
Hi, Volume -3, lab-5, task - 12: BR1 router we have the following config: voice translation-rule 25 rule 1 // /\0/ type any unknown plan any unknown voice translation-profile outbound-ani translate calling 20 translate called 25 vocie-port 0/0/0:23 translation-profile outgoing outbound-ani In general called number should be always associated with incoming number of voice translation-profile; but in solution it is associated with outgoing number, can some one light me on this called number can associate with outgoing number. Thanks. New Email addresses available on Yahoo! Get the Email name you#39;ve always wanted on the new @ymail and @rocketmail. Hurry before someone else does! http://mail.promotions.yahoo.com/newdomains/aa/
[OSL | CCIE_Voice] call group of DN number to conference at a time
Hi, IF I want to call group of DN number to conference at a time, I can dial one by one and call each one. There is any simple way like hunt group to call all of them at once for conference, please light on this. For ex in a company they have sales team, here manager want to call the whole team just by dialing a single number, any workaround for this ? Thanks, Bala. New Email addresses available on Yahoo! Get the Email name you#39;ve always wanted on the new @ymail and @rocketmail. Hurry before someone else does! http://mail.promotions.yahoo.com/newdomains/aa/
Re: [OSL | CCIE_Voice] Block incoming calls on H323 GW in SRST
working with COR list is the best solution for MGCP gateway I think so..when I try the same in h.323 gaeway the call is blocked all the time. Could please some one share the best solution for blocking calls in h.323 gateway to. Thanks --- On Thu, 19/2/09, Jose Gregorio Linero (jlinero) jlin...@cisco.com wrote: From: Jose Gregorio Linero (jlinero) jlin...@cisco.com Subject: Re: [OSL | CCIE_Voice] Block incoming calls on H323 GW in SRST To: DIEGO FERNANDO MACIAS SANCHEZ dmac...@javeriana.edu.co, ccie_voice@onlinestudylist.com Date: Thursday, 19 February, 2009, 4:15 AM Hi Try to use cor list. Regards, Jose From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of DIEGO FERNANDO MACIAS SANCHEZ Sent: Miércoles, Febrero 18, 2009 5:42 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Block incoming calls on H323 GW in SRST Hello all Does anybody know the way to block incoming calls from PSTN to an especific extension connected to an H323 GW. If i apply a blocking translation rule, this will be active in normal operation also. Regards DM AVISO LEGAL: El presente correo electronico no representa la opinion o el consentimiento oficial de la PONTIFICIA UNIVERSIDAD JAVERIANA. Este mensaje es confidencial y puede contener informacion privilegiada la cual no puede ser usada ni divulgada a personas distintas de su destinatario. Esta prohibida la retencion, grabacion, utilizacion, aprovechamiento o divulgacion con cualquier proposito. Si por error recibe este mensaje, por favor destruya su contenido y avise a su remitente. En este aviso legal se omiten intencionalmente las tildes. Get your new Email address! Grab the Email name you#39;ve always wanted before someone else does! http://mail.promotions.yahoo.com/newdomains/aa/
Re: [OSL | CCIE_Voice] Open Ended Questions
Now it is only for RS not for Voice, may be later they may ask for Voice track also. --- On Thu, 12/2/09, Mark Holloway m...@markholloway.com wrote: From: Mark Holloway m...@markholloway.com Subject: [OSL | CCIE_Voice] Open Ended Questions To: 'OSL Group' ccie_voice@onlinestudylist.com Date: Thursday, 12 February, 2009, 10:13 AM Does the CCIE Voice have open ended questions like the RS lab? Seems like a real stickler for some folks. Get your new Email address! Grab the Email name you#39;ve always wanted before someone else does! http://mail.promotions.yahoo.com/newdomains/aa/
Re: [OSL | CCIE_Voice] unity and SRST wired problem , not the classic bug question!
Hi Vik, The 4082032220 is CTI route point in CCM, the CTI route point solution works only if I am disabling isdn outgoing ie redirectin-number under serial interface 0/2/0:23, If I am enbaling isdn outgoing ie redirect-number, then the CTI solution is not working. Could please let me know the above solution is right or I am missing some thing. --- On Wed, 14/1/09, Vik Malhi vma...@ipexpert.com wrote: From: Vik Malhi vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] unity and SRST wired problem , not the classic bug question! To: mmailb...@yahoo.com, jeremy co jeremy.coo...@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Wednesday, 14 January, 2009, 12:34 PM In your lab what is 4082032220? It should be a RP with a VM Prof Mask = 3001 Call Fwd to VM. -- Vik Malhi – CCIE #13890, CCSI #31584 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: Balamurugan Singaram mmailb...@yahoo.com Reply-To: mmailb...@yahoo.com Date: Mon, 12 Jan 2009 21:33:19 -0800 (PST) To: jeremy co jeremy.coo...@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com, Vik Malhi vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] unity and SRST wired problem , not the classic bug question! Hi Vik, For SRST voice mail follow the CTI route point solution it, but till I am facing the redirect number problem. THE CTI debug is paste below, could you please let me know your suggestion please Best is solution to upgrade the IOS in home lab ? HQ# *Jan 11 04:14:13.417: ISDN Se0/3/0:23 Q931: TX - SETUP pd = 8 callref = 0x000F Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98383 Exclusive, Channel 3 Facility i = 0x9F8B0100A10F020101020100800748512D32303031 Protocol Profile = Networking Extensions 0xA10F020101020100800748512D32303031 Component = Invoke component Invoke Id = 1 Operation = CallingName Name presentation allowed Name = HQ-2001 Progress Ind i = 0x8083 - Origination address is non-ISDN Display i = 'HQ-2001' Calling Party Number i = 0x0081, '2001' Plan:Unknown, Type:Unknown Called Party Number i = 0x80, '19723033001' Plan:Unknown, Type:Unknown *Jan 11 04:14:13.457: ISDN Se0/3/0:23 Q931: RX - CALL_PROC pd = 8 callref = 0x 800F Channel ID i = 0xA98383 Exclusive, Channel 3 *Jan 11 04:14:13.517: ISDN Se0/3/0:23 Q931: RX - ALERTING pd = 8 callref = 0x8 00F Progress Ind i = 0x8188 - In-band info or appropriate now available *Jan 11 04:14:18.549: ISDN Se0/3/0:23 Q931: RX - SETUP pd = 8 callref = 0x0181 Bearer Capability i = 0x9090A2 Standard = CCITT Transfer Capability = 3.1kHz Audio Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Facility i = 0x9F8B0100A10F020101020100800748512D32303031 Protocol Profile = Networking Extensions 0xA10F020101020100800748512D32303031 Component = Invoke component Invoke Id = 1 Operation = CallingName Name presentation allowed Name = HQ-2001 Progress Ind i = 0x8083 - Origination address is non-ISDN Display i = 'HQ-2001' Calling Party Number i = 0x0081, '2001' Plan:Unknown, Type:Unknown Called Party Number i = 0xA1, '4082032220' Plan:ISDN, Type:National Redirecting Number i = 0x7FE0FF, '3001' Plan:Reserved, Type:Reserved *Jan 11 04:14:18.577: ISDN Se0/3/0:23 Q931: TX - CALL_PROC pd = 8 callref = 0x --- On Tue, 13/1/09, Vik Malhi vma...@ipexpert.com wrote: From: Vik Malhi vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] unity and SRST wired problem , not the classic bug question! To: jeremy co jeremy.coo...@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Tuesday, 13 January, 2009, 12:30 AM I don’t get any RDNIS so you are doing much better than me. I think this the RDNIS with SRST has bugs that are fixed in 12.4(7). What IOS are you using? -- Vik Malhi – CCIE #13890, CCSI #31584 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: jeremy co jeremy.coo...@gmail.com Date: Mon, 12 Jan 2009 21:51:57 +1100 To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] unity and SRST wired
Re: [OSL | CCIE_Voice] unity and SRST wired problem , not the classic bug question!
Hi Vik, For SRST voice mail follow the CTI route point solution it, but till I am facing the redirect number problem. THE CTI debug is paste below, could you please let me know your suggestion please Best is solution to upgrade the IOS in home lab ? HQ# *Jan 11 04:14:13.417: ISDN Se0/3/0:23 Q931: TX - SETUP pd = 8 callref = 0x000F Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98383 Exclusive, Channel 3 Facility i = 0x9F8B0100A10F020101020100800748512D32303031 Protocol Profile = Networking Extensions 0xA10F020101020100800748512D32303031 Component = Invoke component Invoke Id = 1 Operation = CallingName Name presentation allowed Name = HQ-2001 Progress Ind i = 0x8083 - Origination address is non-ISDN Display i = 'HQ-2001' Calling Party Number i = 0x0081, '2001' Plan:Unknown, Type:Unknown Called Party Number i = 0x80, '19723033001' Plan:Unknown, Type:Unknown *Jan 11 04:14:13.457: ISDN Se0/3/0:23 Q931: RX - CALL_PROC pd = 8 callref = 0x 800F Channel ID i = 0xA98383 Exclusive, Channel 3 *Jan 11 04:14:13.517: ISDN Se0/3/0:23 Q931: RX - ALERTING pd = 8 callref = 0x8 00F Progress Ind i = 0x8188 - In-band info or appropriate now available *Jan 11 04:14:18.549: ISDN Se0/3/0:23 Q931: RX - SETUP pd = 8 callref = 0x0181 Bearer Capability i = 0x9090A2 Standard = CCITT Transfer Capability = 3.1kHz Audio Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Facility i = 0x9F8B0100A10F020101020100800748512D32303031 Protocol Profile = Networking Extensions 0xA10F020101020100800748512D32303031 Component = Invoke component Invoke Id = 1 Operation = CallingName Name presentation allowed Name = HQ-2001 Progress Ind i = 0x8083 - Origination address is non-ISDN Display i = 'HQ-2001' Calling Party Number i = 0x0081, '2001' Plan:Unknown, Type:Unknown Called Party Number i = 0xA1, '4082032220' Plan:ISDN, Type:National Redirecting Number i = 0x7FE0FF, '3001' Plan:Reserved, Type:Reserved *Jan 11 04:14:18.577: ISDN Se0/3/0:23 Q931: TX - CALL_PROC pd = 8 callref = 0x --- On Tue, 13/1/09, Vik Malhi vma...@ipexpert.com wrote: From: Vik Malhi vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] unity and SRST wired problem , not the classic bug question! To: jeremy co jeremy.coo...@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Tuesday, 13 January, 2009, 12:30 AM I don’t get any RDNIS so you are doing much better than me. I think this the RDNIS with SRST has bugs that are fixed in 12.4(7). What IOS are you using? -- Vik Malhi – CCIE #13890, CCSI #31584 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: jeremy co jeremy.coo...@gmail.com Date: Mon, 12 Jan 2009 21:51:57 +1100 To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] unity and SRST wired problem , not the classic bug question! Hi, I tried to set up SRST and unity scenario. Here is the problem unity--HQ ---pstn-BR1 (SRST) 2001 3001 2001 call 3001 and CFNA redirect call to unity via pstn , redirecting number works fine but only 8 digits passed to unity Here is the out put of debug isdn on HQ when call forwarded to unity. HQ :499-202-2 BR1 :899-303-3XXX voice pilot number : 2229 Mar 11 20:01:40.060: ISDN Se0/0:23 Q931: RX - SETUP pd = 8 callref = 0x008E Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Calling Party Number i = 0x2181, '4992022002' Plan:ISDN, Type:National Called Party Number i = 0xA1, '499209' Plan:ISDN, Type:National Redirecting Number i = 0xFF, '8993033001' Plan:Reserved, Type:Reserved I can see from call viewer in unity : dialed number calling number forwarding 93033001 4992022002 93033001 So why only 8 digits pass to ccm? Jeremy Get your preferred Email name! Now you can @ymail.com and @rocketmail.com. http://mail.promotions.yahoo.com/newdomains/aa/
Re: [OSL | CCIE_Voice] unity and SRST wired problem , not the classic bug question!
I have the same problem in IOS 12.13 and later, could please let me know the workaround for this. --- On Tue, 13/1/09, Vik Malhi vma...@ipexpert.com wrote: From: Vik Malhi vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] unity and SRST wired problem , not the classic bug question! To: jeremy co jeremy.coo...@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Tuesday, 13 January, 2009, 12:30 AM I don’t get any RDNIS so you are doing much better than me. I think this the RDNIS with SRST has bugs that are fixed in 12.4(7). What IOS are you using? -- Vik Malhi – CCIE #13890, CCSI #31584 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: jeremy co jeremy.coo...@gmail.com Date: Mon, 12 Jan 2009 21:51:57 +1100 To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] unity and SRST wired problem , not the classic bug question! Hi, I tried to set up SRST and unity scenario. Here is the problem unity--HQ ---pstn-BR1 (SRST) 2001 3001 2001 call 3001 and CFNA redirect call to unity via pstn , redirecting number works fine but only 8 digits passed to unity Here is the out put of debug isdn on HQ when call forwarded to unity. HQ :499-202-2 BR1 :899-303-3XXX voice pilot number : 2229 Mar 11 20:01:40.060: ISDN Se0/0:23 Q931: RX - SETUP pd = 8 callref = 0x008E Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Calling Party Number i = 0x2181, '4992022002' Plan:ISDN, Type:National Called Party Number i = 0xA1, '499209' Plan:ISDN, Type:National Redirecting Number i = 0xFF, '8993033001' Plan:Reserved, Type:Reserved I can see from call viewer in unity : dialed number calling number forwarding 93033001 4992022002 93033001 So why only 8 digits pass to ccm? Jeremy New Email names for you! Get the Email name you#39;ve always wanted on the new @ymail and @rocketmail. Hurry before someone else does! http://mail.promotions.yahoo.com/newdomains/aa/
Re: [OSL | CCIE_Voice] TEHO backup calls to BR2 H323 GW - Failing
If your route list is configured as explained, set both the Stop Routing on User Busy Flag and Stop Routing on Unallocated Number Flag service parameters to False. In order to do this, go to Cisco CallManager Admin Service Service Parameters Select a Server Cisco CallManager and set the parameters to False. --- On Sat, 22/11/08, Shadab Abbasi (moabbasi) [EMAIL PROTECTED] wrote: From: Shadab Abbasi (moabbasi) [EMAIL PROTECTED] Subject: [OSL | CCIE_Voice] TEHO backup calls to BR2 H323 GW - Failing To: ccie_voice@onlinestudylist.com Date: Saturday, 22 November, 2008, 8:03 PM Hello All, I am facing some strange issue here. I have BR2 setup as H323 Gateway and added into CallManager as H323 Gateway. My HQ DID is 575-212-2000 ; my BR2 DID is 608-323-3000 During TEHO calls: PRIMARY: When I call from HQ to BR2 PSTN Local calls: 91608555 || Call is going though H323 HW as 555 (this works gr8) BACK-UP: However when I shutdown the voice-ports in H323 GW to test my Local HQ T1 GW as backup (as 1608555), I am hearing a ANN message as “Your call cannot be completed as dialed, please check…..” Until yesterday, it was a success, today no luck. (I tried to remove H323 as Primary just use my local HQ T1 GW as primary for this LD call), this works! Anyone faced similar issue? Thanks! Regards, Shadab Abbasi TSN SE - Unified Communications Technology Solutions Network (TSN) [EMAIL PROTECTED] Ph: +91.80.4103.6436 (off) +91.974.009.0334(Mob) TSN-WiKi: Home Page Get your new Email address! Grab the Email name you#39;ve always wanted before someone else does! http://mail.promotions.yahoo.com/newdomains/aa/
[OSL | CCIE_Voice] preserve call in h323 gateway in SRST mode
Hi, Do some one know how to preserve call in h323 gateway in SRST mode ? when the call is in process when we shut the CCM services, the call get disconnected it does not goes to SRST mode. voice service voip h323 no h225 timeout keepalive In MGCP gateway, the following below command will preserve call, but this is not working for H.323 gateway and above command is also is not working ? any suggestion please. application global service alternate default Thanks, Get your preferred Email name! Now you can @ymail.com and @rocketmail.com. http://mail.promotions.yahoo.com/newdomains/aa/
[OSL | CCIE_Voice] cme phones to two different unity systems
Hi, We have two cme phones in BR2 two different unity systems: 1st phone press messages button and go to unity 4.0.5 greetings 2nd phone press messages button and go to CUE greetings How to make it work? Thanks, Get your preferred Email name! Now you can @ymail.com and @rocketmail.com. http://mail.promotions.yahoo.com/newdomains/aa/
[OSL | CCIE_Voice] BR1 h323 gateway + preserve call in h323
Hi, It the right fashion to configure voip dialpeers from BR1 h323 gateway + preserve call in h323 gateway in SRST mode or I am missing something ? voice class h323 1 h225 timeout tcp establish 3 h225 timeout setup 5 voice service voip h323 no h225 timeout keepalive dial-peer voice 10 voip destination-pattern [23]... voice-class codec 1 voice-class h323 1 session target ipv4:1.1.1.1 ip qos dscp cs3 sig codec g711ulaw ! dial-peer voice 11 voip preference 1 destination-pattern [23]... voice-class codec 1 voice-class h323 1 session target ipv4:1.1.1.2 codec g711ulaw ip qos dscp cs3 sig! CCM system parameters: Allow TCP keepalive for H323: False Allow peer to preserve H323 Call: True Thanks, Get your new Email address! Grab the Email name you#39;ve always wanted before someone else does! http://mail.promotions.yahoo.com/newdomains/aa/
Re: [OSL | CCIE_Voice] IPCCX ring back tone to caller
Then try mucic on hold.. file as ring back tone... --- On Mon, 10/11/08, Erick Pineda [EMAIL PROTECTED] wrote: From: Erick Pineda [EMAIL PROTECTED] Subject: [OSL | CCIE_Voice] IPCCX ring back tone to caller To: OSL CCIE Voice Lab Exam ccie_voice@onlinestudylist.com Date: Monday, 10 November, 2008, 6:57 PM an ipccx agent gets a call, when he makes the tranfer the callers hears a ring back tone. does any boby has an idea how to do it, because right now when i make the tranfer the caller hear mucic on hold.. Regards Erick Get your preferred Email name! Now you can @ymail.com and @rocketmail.com. http://mail.promotions.yahoo.com/newdomains/aa/
Re: [OSL | CCIE_Voice] Gateway Channel selection control ???
Hi Mark, I have check BugNavigator, it hits 1000 bugs..for a search it is to difficult to search for specific bugs, in past days we have TAC case collection which is very useful and easy to search, I think now TAC case collection it is not updated. Could you please someone suggest how to narrow the bug tool kit search. Thanks, Bala. --- On Wed, 29/10/08, Mark Snow [EMAIL PROTECTED] wrote: From: Mark Snow [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] Gateway Channel selection control ??? To: Paul and Bobs [EMAIL PROTECTED] Cc: ccie_voice@onlinestudylist.com Date: Wednesday, 29 October, 2008, 8:49 AM BTW - that's not to say that I recommend it - but for lab purposes should be all good. There could be bugs associated with it - and I would definitely check BugNavigator before putting it into production :) cheers, -- Mark Snow CCIE #14073 (Voice, Security) Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: [EMAIL PROTECTED] -- 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. -- On Oct 28, 2008, at 10:20 PM, Paul and Bobs wrote: Hi All I was wandering if anyone know of a way using both MGCP and H.323 to control the channells on an E1/T1 circuit. For example - If I have a single E1 service with only 20 channels and I want to say reserve 5 for outgoing and reserve 15 for incoming, is there a way on both protocols to do this. Thanks Paul Get your preferred Email name! Now you can @ymail.com and @rocketmail.com. http://mail.promotions.yahoo.com/newdomains/aa/
[OSL | CCIE_Voice] MGCP Gateway POTS dial-peer
Hi, For Cisco IOS Software Release 12.3(7)T or later the Pots dial-peer configuration for MGCP gateway should like below or even service mgcpapp is not needed ? Could you please correct me if I am wrong ? dial-peer voice 10 pots service mgcpapp incoming called-number . direct-inward-dial port 1/0:15 Thanks, Bala. Get your preferred Email name! Now you can @ymail.com and @rocketmail.com. http://mail.promotions.yahoo.com/newdomains/aa/
[OSL | CCIE_Voice] Urgent Priority for Internal numbers
Hi, How to set Urgent Priority for Internal numbers, could please let me know the workaround. Thanks, Bala. New Email addresses available on Yahoo! Get the Email name you#39;ve always wanted on the new @ymail and @rocketmail. Hurry before someone else does! http://mail.promotions.yahoo.com/newdomains/aa/
[OSL | CCIE_Voice] Urgent Priority for Internal numbers
Hi, Any Workaround to set Urgent Priority for Internal numbers. Thanks, Bala. New Email names for you! Get the Email name you#39;ve always wanted on the new @ymail and @rocketmail. Hurry before someone else does! http://mail.promotions.yahoo.com/newdomains/aa/
Re: [OSL | CCIE_Voice] Configuring Voice-Card for DSPFarm
Hi Mark, If we are using PVDM 1 then we should used only the following config: Voice-card 0 dspfarm dsp services dspfarm Could you please explain. Thanks, Bala. --- On Thu, 23/10/08, Mark Snow [EMAIL PROTECTED] wrote: From: Mark Snow [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] Configuring Voice-Card for DSPFarm To: Kevin Porter [EMAIL PROTECTED] Cc: ccie_voice@onlinestudylist.com Date: Thursday, 23 October, 2008, 2:03 AM dspfarm is for PVDM (orig) dsp services dspfarm is for PVDM2 and the config you pointed out Must be used when bridging or sharing PVDM2 resources as you mentioned. Cheers, -- Mark Snow CCIE #14073 (Voice, Security) Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: [EMAIL PROTECTED] -- 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. -- On Oct 22, 2008, at 3:19 PM, Kevin Porter wrote: Quick question…when configuring the Voice-Card to act as a dspfarm, I have seen almost all the IPExpert config’s to look like this.. Voice-card 0 dspfarm dsp services dspfarm but, in the real world, I can not do this with a “shared” PVDM that is terminating voice circuits (PRI, FXS, FXO, etc…). I have had to use the following config: Voice-card 0 no dspfarm dsp services dspfarm So, the question is this, is there anything wrong with my config that would cause issues? Thanks, Kevin Kevin Porter Systems Engineer L4 Netelligent Corporation 400 South Woods Mill Drive, Suite 105 St. Louis , MO 63017 Office: (314) 392-6921 Cell: (314) 852-1252 Fax: (314) 392-9760 [EMAIL PROTECTED] www.netelligent.com Bridging The Gap Between Good and GREAT IP Communications! Get your preferred Email name! Now you can @ymail.com and @rocketmail.com. http://mail.promotions.yahoo.com/newdomains/aa/
Re: [OSL | CCIE_Voice] FW: PSTN Gateway Backbone GK configuration
I am using the below config, it is working for me, try it: - isdn switch-type primary-ni ! voice-card 0 no dspfarm ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! controller E1 0/2/0 pri-group timeslots 1-3,16 ! controller T1 0/3/0 framing esf linecode b8zs pri-group timeslots 1-3,24 ! controller T1 0/3/1 framing esf linecode b8zs pri-group timeslots 1-3,24 ! ! ! ! ! ! interface Serial0/2/0:15 no ip address isdn switch-type primary-ni isdn protocol-emulate network isdn incoming-voice voice no cdp enable ! interface Serial0/3/0:23 no ip address isdn switch-type primary-ni isdn protocol-emulate network isdn incoming-voice voice no cdp enable ! interface Serial0/3/1:23 no ip address isdn switch-type primary-ni isdn protocol-emulate network isdn incoming-voice voice no cdp enable ! ! voice-port 0/2/0:15 ! voice-port 0/3/0:23 ! voice-port 0/3/1:23 ! ! ! ! ! dial-peer voice 10 pots destination-pattern 915535 incoming called-number . port 0/3/0:23 ! dial-peer voice 11 pots destination-pattern 902773 port 0/3/1:23 ! dial-peer voice 12 pots destination-pattern 01176 incoming called-number . port 0/2/0:15 ! ! ! ! telephony-service load 7960-7940 P00403020214 max-ephones 4 max-dn 12 ip source-address 192.168.41.1 port 2000 create cnf-files version-stamp 7960 Feb 19 2007 15:52:07 max-conferences 8 gain -6 ! ! ephone-dn 1 number 911 ! ! ephone-dn 2 number 0119876543210 ! ! ephone-dn 3 number 902463 ! ! ephone-dn 4 number 916435 ! ! ephone-dn 5 number 212555 ! ! ephone 1 mac-address button 1:1 2:2 3:3 4:4 5:5 For the conection beteen PSTN router and your gw you need to create a crossover cable to connect the E1 / T1 make the connection as follows: 14 25 41 52 Regards, Bala. --- On Wed, 22/10/08, Shadab Abbasi (moabbasi) [EMAIL PROTECTED] wrote: From: Shadab Abbasi (moabbasi) [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] FW: PSTN Gateway Backbone GK configuration To: marwa [EMAIL PROTECTED], ccie_voice@onlinestudylist.com Date: Wednesday, 22 October, 2008, 4:06 PM Thanks Marwa, I got the same at the below mentioned location. Seeing the configuration, I found that translation-profile is applied under voice-ports, however NO translation-profile is created; just seeing translation-rules??? Any advice! -Shadab From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of marwa Sent: Wednesday, October 22, 2008 3:51 PM To: Shadab Abbasi (moabbasi); ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] FW: PSTN Gateway Backbone GK configuration Hello, You can find the initial configurations for all the devices in the Proctor Labs support forum: http://proctorlabs.com/forum/ To access the forum, simply create a Proctor Labs account, which is free of charge, and use those credentials to enter the support forum. You'll find the configs you're looking for in the Voice FAQ section. Marwa - Original Message - From: Shadab Abbasi (moabbasi) To: ccie_voice@onlinestudylist.com Sent: Wednesday, October 22, 2008 10:54 AM Subject: [OSL | CCIE_Voice] FW: PSTN Gateway Backbone GK configuration Hello Experts, I am setting up my LAB planning to run PSTN GW Backbone (PSTN) – GK on the same router (2821) Can someone help me out with the configuration or sample? Regards, Shadab Get your preferred Email name! Now you can @ymail.com and @rocketmail.com. http://mail.promotions.yahoo.com/newdomains/aa/
Re: [OSL | CCIE_Voice] Block International calls inbound H.323 / SRST
Hi, try the following config: dial-peer voice 499 voip destination-pattern 4083332002 session target ipv4:10.1.200.20 [CCM] ! dial-peer voice 500 voip preference 1 destination-pattern 4083332002 session target ipv4:172.1.101.1 [Loop Back] ! dial-peer voice 501 voip translation-profile incoming loop call-block translation-profile incoming block incoming called-number 4083332002 ! voice translation-rule 1 rule 1 /^4083332002/ /2002/ ! voice translation-rule 2 rule 1 reject /3001/ ! ! voice translation-profile block translate calling 2 ! voice translation-profile loop translate called 1 Regards, Bala. --- On Wed, 15/10/08, jonny vegas [EMAIL PROTECTED] wrote: From: jonny vegas [EMAIL PROTECTED] Subject: [OSL | CCIE_Voice] Block International calls inbound H.323 / SRST To: ccie_voice@onlinestudylist.com Date: Wednesday, 15 October, 2008, 10:09 PM Goal is to block calls from international type numbers when in SRST / H.323. I have a method of doing this but it is a little long winded and requires visibility of the ANI. Wondering if any one else has worked out a quick way, based on voice translation profile / COR / ANO. The approach where one blocks it on the inbound Dial Peer with a VTP does not work for H.323. The call must be allowed into the router so the normal H.323 dial peers to CCM function. Happy thinking. New Email addresses available on Yahoo! Get the Email name you#39;ve always wanted on the new @ymail and @rocketmail. Hurry before someone else does! http://mail.promotions.yahoo.com/newdomains/aa/
Re: [OSL | CCIE_Voice] H323 COR ?
Hi Vik, When gw is H323, these restrictions will apply all the time and ''not only under SRST as desired', Could you please correct me If I am wrong ? Thanks, Bala. --- On Tue, 7/10/08, Vikram Malhi [EMAIL PROTECTED] wrote: From: Vikram Malhi [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] H323 COR ? To: [EMAIL PROTECTED] Cc: Jacob Owen [EMAIL PROTECTED], ccie_voice@onlinestudylist.com Date: Tuesday, 7 October, 2008, 12:12 PM voice translation-rule 11 rule 1 reject // type international voice translation-profil test translate calling 11 dial-peer voice 3001 pots call-block translation-profile incoming test incoming called-num 3001 Vik Malhi – CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] 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. On Oct 6, 2008, at 11:35 PM, Balamurugan Singaram wrote: my question is to block only incoming International number to particular Ephone number when it is in SRST mode as H.323 gateway ? Get your preferred Email name! Now you can @ymail.com and @rocketmail.com. http://mail.promotions.yahoo.com/newdomains/aa/
Re: [OSL | CCIE_Voice] SRST Voicemail Integration
Hi All, How can we block international (or specific calling number) call ONLY in SRST mode? in H323 gateway. Thanks, Bala. --- On Mon, 6/10/08, Edi Hamlet [EMAIL PROTECTED] wrote: From: Edi Hamlet [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] SRST Voicemail Integration To: Chris Parker [EMAIL PROTECTED], Vikram Malhi [EMAIL PROTECTED] Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Monday, 6 October, 2008, 7:13 AM Hi Parker, i think the cfw in alias will work if the PSTN accept 1212225 and pass it to HQ gateway. if the PSTN only accept 12122251xxx, then the cfw in alias will not work. i think the workaround if still want using alias is put cfw 912122251xxx which is not already used in HQ, then use translation pattern to translate 1xxx to 200x. i haven't try this, but i think it's gonna work. cmiiw.. cheers, edi - Original Message From: Chris Parker [EMAIL PROTECTED] To: Vikram Malhi [EMAIL PROTECTED] Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Sunday, October 5, 2008 3:50:50 AM Subject: Re: [OSL | CCIE_Voice] SRST Voicemail Integration The other method I've seen posted that has caught my interest involves setting the call forward noanswer/busy to a DID on the HQ PRI for each SRST phone using the alias command under call-manager-fallback. Then in UCM putting those DIDs on a CTI route point with call forward all to voicemail. So basically on BR1: call-manager-fallback alias 1 2001 to 2001 cfw 912122252001 timeout 4 alias 1 2002 to 2002 cfw 912122252002 timeout 4 alias 1 2003 to 2003 cfw 912122252003 timeout 4 So for the extensions 2001-2003 at BR1 calls get forwarded to 12122252001-3. You have a pots peer than puts those back out to the PSTN. They ring in on the 6608 PRI and if signifcant digits are set to four in the gateway config UCM will try and send the call to 2001-2002 respectively. UCM sees the extension as OOS and sends it on to voicemail. In the case where you dont have a DID on the HQ PRI that matches the BR1 number on the last 4 digits you can do the same thing but you have to set up a CTI route point for each number that is forwarded to voice mail and then transform the number on UCM before it goes to Unity or use an alternate extension in Unity. Chris Vikram Malhi wrote: Know all possible workarounds...I personally don't like the vm-integration method. Do you know any other methods? Vik Malhi – CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] 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. On Oct 4, 2008, at 6:46 AM, Chris Parker wrote: Hello, I have been reviewing the methods of voice mail fall back with SRST, and I am wondering which method will actually work in the Lab? It seems that success relies on the behavior of PSTN. The vm-integration method seems to work fine on the Proctor Labs gear, but will that translate to the real lab? What is the safest / best way to do this? Chris Parker New Email addresses available on Yahoo! Get the Email name you#39;ve always wanted on the new @ymail and @rocketmail. Hurry before someone else does! http://mail.promotions.yahoo.com/newdomains/aa/
Re: [OSL | CCIE_Voice] H323 COR ?
Hi, How can we block international (or specific calling number) call ONLY in SRST mode? in H323 gateway. Thanks, Bala. --- On Mon, 6/10/08, Mike Brooks [EMAIL PROTECTED] wrote: From: Mike Brooks [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] H323 COR ? To: kapil atrish [EMAIL PROTECTED], ccie_voice@onlinestudylist.com Date: Monday, 6 October, 2008, 1:35 AM Kapil, I am referring to an H323 gateway not an MGCP gateway. Therefore L3 info is not backhauled to CM. Regards, Mike Brooks CCIE#16027 (RS) On Sun, Oct 5, 2008 at 3:58 PM, kapil atrish [EMAIL PROTECTED] wrote: When not in SRST mode, all layer-3 information (DNIS, ANI) are back-hauled to CCM directly and COR won't trigger. Jacob Owen [EMAIL PROTECTED] wrote: Mike, I was under the impression since the call came into the H323 gateway from UCM (GW isn't in SRST) it wasn't tagged with an incoming corlist and therefore could reach all remote PSTN numbers. When the router drops back to SRST the phones would register with a corlist incoming and therefore be limited to where they could call. Hopefully someone will let me know if I am incorrect. You could also test this by adding a corlist incoming to the inbound voip dial-peer and see if you can call. On Sun, Oct 5, 2008 at 12:35 PM, Mike Brooks [EMAIL PROTECTED] wrote: If COR is configured on H323 dial-peers on an H323 gateway, is the dial-peer COR only in affect when in SRST mode ? If not, wouldn't you be performing COR twice once on the CallManager and also on the H323-GW ? for example: phones/CSS h323-gw inbound voip dial-peer (KEY) --- h323gw outbound pots dial-peer (LOCK) or h323-gw inbound pots dial-peer (KEY) -- h323-gw outbound voip dial-peer (LOCK) -- h323-gw/CSS (on CM) If COR is in affect regardless of if it the site is in SRST mode (which I assume it would be) should you just not configure COR (keys/locks) on the inbound/outbound VOIP dial-peer to/from CM ? Regards, Mike Brooks CCIE# 16027 (RS) -- Jacob Owen CCIE #14063 (RS, Service Provider), CCDP, CCVP Get your preferred Email name! Now you can @ymail.com and @rocketmail.com. http://mail.promotions.yahoo.com/newdomains/aa/
Re: [OSL | CCIE_Voice] BACD Issue
I have upload call BACD in CME, when I call BACD from PSTN and CME IPPhone it works fine, but when I call the BACD script from HQ the BACD is not working, even the CODEC is g711ulaw between HQ and CME. Could please let me know what I am missing or any workaround to make BACD to work from HQ to CME. --- On Fri, 3/10/08, Cardwell, Mark [EMAIL PROTECTED] wrote: From: Cardwell, Mark [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] BACD Issue To: [EMAIL PROTECTED], [EMAIL PROTECTED], ccie_voice@onlinestudylist.com Date: Friday, 3 October, 2008, 5:12 PM Did you reload to application? Or just reload the router Mark Cardwell | Systems Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | [EMAIL PROTECTED] D: 571.225.0132 | www.presidio.com From: [EMAIL PROTECTED] To: Edi Hamlet ; ccie_voice@onlinestudylist.com Sent: Thu Oct 02 23:49:56 2008 Subject: Re: [OSL | CCIE_Voice] BACD Issue I had both POTS and VOIP dial-peer . Same results on both pots and VOIP. Cheers Narinder From: Edi Hamlet [mailto:[EMAIL PROTECTED] Sent: Friday, 3 October 2008 1:44 PM To: Kumar, Narinder; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] BACD Issue where did you attach the aa application? on POTS or VOIP dial peer? Try to verify the AA service by calling from IP Phone registered to CME directly to AA pilot number. Create VOIP dial peer in order for IP Phone to be able to call to AA pilot number ! dial-peer voice 4000 voip service aa incoming called-number 4000 destination-pattern 4000 session target ipv4:loopback or CME ip address codec g711ulaw no vad dtmf-relay h245-alphanumeric ! I think AA service will work fine and you will hear welcome prompt if all AA parameters are set correctly with correct spelling. - Original Message From: Kumar, Narinder [EMAIL PROTECTED] To: Edi Hamlet [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Friday, October 3, 2008 10:16:48 AM Subject: RE: [OSL | CCIE_Voice] BACD Issue Initially I had “_bacd_welcome.au” instead of “en_bacd_welcome.au”, I had the same issue with en_bacd_welcome.au As well. Cheers Narinder From: Edi Hamlet [mailto:[EMAIL PROTECTED] Sent: Friday, 3 October 2008 1:13 PM To: Kumar, Narinder; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] BACD Issue you should strip you welcome prompt name in param welcome-prompt to _bacd_welcome.au instead of en_bacd_welcome.au - Original Message From: Kumar, Narinder [EMAIL PROTECTED] To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Friday, October 3, 2008 8:51:06 AM Subject: [OSL | CCIE_Voice] BACD Issue I don't hear any welcome message after 20 sec the IVR start playing.. 1. for sales 2. for customer service 3. for Dial by Extn I have configured 1 for HG 1, 2 for HG 2 and 4 to dial by extn But my BACD IVR plays 3 for Dial by Extn... Below is the config application service aa flash:app-b-acd-aa-2.1.2.2.tcl paramspace english language en paramspace english index 1 paramspace english location flash: param service-name queue param handoff-string aa param aa-pilot 4000 param number-of-hunt-grps 2 param second-greeting-time 30 param call-retry-timer 15 param max-time-call-retry 600 param voice-mail 4600 param max-time-vm-retry 2 param dial-by-extension-option 4 param max-extension-length 4 param welcome-prompt en_bacd_welcome.au param menu-timeout 6 service queue flash:app-b-acd-2.1.2.2.tcl param queue-len 10 param number-of-hunt-grps 2 param aa-hunt1 4111 param aa-hunt2 4222 param queue-manager-debugs 1 Flash File : 22 18836 app-b-acd-2.1.2.2-ReadMe.txt 23 24985 app-b-acd-2.1.2.2.tcl 24 35485 app-b-acd-aa-2.1.2.2.tcl 25 75650 en_bacd_allagentsbusy.au 26 83291 en_bacd_disconnect.au 27 63055 en_bacd_enter_dest.au 28 37952 en_bacd_invalidoption.au 29 496521 en_bacd_music_on_hold.au 30 123446 en_bacd_options_menu.au 31 42978 en_bacd_welcome.au 32 34794 en_bacd_xferto_operator.au CONFIDENTIALITY - The information contained in this electronic mail message is confidential and is intended solely for the addressee(s). If you are not an authorised recipient of this message please contact Getronics Australia immediately by reply email and destroy/delete this message from your computer. Any unauthorised form of reproduction of this message, or part thereof, is strictly prohibited. DISCLAIMER - Unless specifically indicated otherwise, the views and opinions expressed in this email are those of the sender and not Getronics Australia. While we endeavour to protect our network from computer viruses, Getronics Australia does not warrant that this email or any attachments are free of viruses or any other defects or errors. It is the duty of the recipient
Re: [OSL | CCIE_Voice] BACD Issue
--- On Fri, 3/10/08, Cardwell, Mark [EMAIL PROTECTED] wrote: From: Cardwell, Mark [EMAIL PROTECTED] Subject: RE: [OSL | CCIE_Voice] BACD Issue To: [EMAIL PROTECTED] Date: Friday, 3 October, 2008, 7:26 PM What happens when you call from HQ? Also break out you dial-p into 2 not income called and destination-pattern on the same. Cheers! Mark Cardwell | Systems Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | [EMAIL PROTECTED] D: 571.225.0132 | www.presidio.com From: Balamurugan Singaram [mailto:[EMAIL PROTECTED] Sent: Friday, October 03, 2008 9:54 AM To: [EMAIL PROTECTED]; [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com; Cardwell, Mark Subject: Re: [OSL | CCIE_Voice] BACD Issue I have upload call BACD in CME, when I call BACD from PSTN and CME IPPhone it works fine, but when I call the BACD script from HQ the BACD is not working, even the CODEC is g711ulaw between HQ and CME. Could please let me know what I am missing or any workaround to make BACD to work from HQ to CME. --- On Fri, 3/10/08, Cardwell, Mark [EMAIL PROTECTED] wrote: From: Cardwell, Mark [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] BACD Issue To: [EMAIL PROTECTED], [EMAIL PROTECTED], ccie_voice@onlinestudylist.com Date: Friday, 3 October, 2008, 5:12 PM Did you reload to application? Or just reload the router Mark Cardwell | Systems Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | [EMAIL PROTECTED] D: 571.225.0132 | www.presidio.com From: [EMAIL PROTECTED] To: Edi Hamlet ; ccie_voice@onlinestudylist.com Sent: Thu Oct 02 23:49:56 2008 Subject: Re: [OSL | CCIE_Voice] BACD Issue I had both POTS and VOIP dial-peer . Same results on both pots and VOIP. Cheers Narinder From: Edi Hamlet [mailto:[EMAIL PROTECTED] Sent: Friday, 3 October 2008 1:44 PM To: Kumar, Narinder; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] BACD Issue where did you attach the aa application? on POTS or VOIP dial peer? Try to verify the AA service by calling from IP Phone registered to CME directly to AA pilot number. Create VOIP dial peer in order for IP Phone to be able to call to AA pilot number ! dial-peer voice 4000 voip service aa incoming called-number 4000 destination-pattern 4000 session target ipv4:loopback or CME ip address codec g711ulaw no vad dtmf-relay h245-alphanumeric ! I think AA service will work fine and you will hear welcome prompt if all AA parameters are set correctly with correct spelling. - Original Message From: Kumar, Narinder [EMAIL PROTECTED] To: Edi Hamlet [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Friday, October 3, 2008 10:16:48 AM Subject: RE: [OSL | CCIE_Voice] BACD Issue Initially I had “_bacd_welcome.au” instead of “en_bacd_welcome.au”, I had the same issue with en_bacd_welcome.au As well. Cheers Narinder From: Edi Hamlet [mailto:[EMAIL PROTECTED] Sent: Friday, 3 October 2008 1:13 PM To: Kumar, Narinder; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] BACD Issue you should strip you welcome prompt name in param welcome-prompt to _bacd_welcome.au instead of en_bacd_welcome.au - Original Message From: Kumar, Narinder [EMAIL PROTECTED] To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Friday, October 3, 2008 8:51:06 AM Subject: [OSL | CCIE_Voice] BACD Issue I don't hear any welcome message after 20 sec the IVR start playing.. 1. for sales 2. for customer service 3. for Dial by Extn I have configured 1 for HG 1, 2 for HG 2 and 4 to dial by extn But my BACD IVR plays 3 for Dial by Extn... Below is the config application service aa flash:app-b-acd-aa-2.1.2.2.tcl paramspace english language en paramspace english index 1 paramspace english location flash: param service-name queue param handoff-string aa param aa-pilot 4000 param number-of-hunt-grps 2 param second-greeting-time 30 param call-retry-timer 15 param max-time-call-retry 600 param voice-mail 4600 param max-time-vm-retry 2 param dial-by-extension-option 4 param max-extension-length 4 param welcome-prompt en_bacd_welcome.au param menu-timeout 6 service queue flash:app-b-acd-2.1.2.2.tcl param queue-len 10 param number-of-hunt-grps 2 param aa-hunt1 4111 param aa-hunt2 4222 param queue-manager-debugs 1 Flash File : 22 18836 app-b-acd-2.1.2.2-ReadMe.txt 23 24985 app-b-acd-2.1.2.2.tcl 24 35485 app-b-acd-aa-2.1.2.2.tcl 25 75650 en_bacd_allagentsbusy.au 26 83291 en_bacd_disconnect.au 27 63055 en_bacd_enter_dest.au 28 37952 en_bacd_invalidoption.au 29 496521 en_bacd_music_on_hold.au 30 123446 en_bacd_options_menu.au 31 42978 en_bacd_welcome.au 32 34794 en_bacd_xferto_operator.au
Re: [OSL | CCIE_Voice] G729 kbps with cRTP
G.729 CRTP FRF.12 call - 13 KBPS MLPP call - 15 KBPS --- On Fri, 3/10/08, Carter, Bill [EMAIL PROTECTED] wrote: From: Carter, Bill [EMAIL PROTECTED] Subject: [OSL | CCIE_Voice] G729 kbps with cRTP To: ccie_voice@onlinestudylist.com Date: Friday, 3 October, 2008, 8:04 PM what is the per call bandwidth requirement for a G.729 call using FRF.12 and cRTP ? New Email names for you! Get the Email name you#39;ve always wanted on the new @ymail and @rocketmail. Hurry before someone else does! http://mail.promotions.yahoo.com/newdomains/aa/
Re: [OSL | CCIE_Voice] Unity VPIM
You can get a demo license from your local Cisco-AM --- On Wed, 24/9/08, Paul and Bobs [EMAIL PROTECTED] wrote: From: Paul and Bobs [EMAIL PROTECTED] Subject: [OSL | CCIE_Voice] Unity VPIM To: ccie_voice@onlinestudylist.com Date: Wednesday, 24 September, 2008, 3:25 PM HI All Does anyone have a demo license for Unity with VPIM that I could use to get it working in the lab. Cheers Paul New Email addresses available on Yahoo! Get the Email name you#39;ve always wanted on the new @ymail and @rocketmail. Hurry before someone else does! http://mail.promotions.yahoo.com/newdomains/aa/
[OSL | CCIE_Voice] How to synchronize NTP with Unity ?
Hi, How to synchronize NTP with Unity ? Thanks, Get your preferred Email name! Now you can @ymail.com and @rocketmail.com. http://mail.promotions.yahoo.com/newdomains/aa/
[OSL | CCIE_Voice] param number-of-hunt-grps number
Hi, Could please explain what is the difference between param number-of-hunt grps number under aap-b-acd and under aap-b-acd-aa Under app-b-acd: Router(config-app)# service queue flash:app-b-acd-2.1.0.0.tcl param number-of-hunt-grps number It range is 1 - 10 the same command under Router(config-app)# service aa flash:app-b-acd-aa-2.1.0.0.tcl param number-of-hunt-grps number It range is 1 - 3 please explain in detail about the range of 1-10 and 1-3 of this commnad. Thanks, Bala. Send instant messages to your online friends http://uk.messenger.yahoo.com
Re: [OSL | CCIE_Voice] POTS Dial-peer
Hi, If I dial 9002001, then the outgoing number will be 900112001 or 112001 ? dial-peer voice 10 pots destination-pattern 900T forward digit all prefix 11 Thanks, Bala. dschulz [EMAIL PROTECTED] wrote: To get around this, you can set the how many digits to forward by using the forward-digits command. HTH Dave - From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Balamurugan Singaram Sent: Monday, June 09, 2008 1:39 AM To: OSL CCIE Voice Lab Exam Subject: Re: [OSL | CCIE_Voice] POTS Dial-peer By default pots dial peer will strip the wildcard, so I think 2[015] will be get stripped, thanks for your reply Chand. --Bala. Chad Stachowicz [EMAIL PROTECTED] wrote: 0014152001 On Sun, Jun 8, 2008 at 9:42 PM, Balamurugan Singaram [EMAIL PROTECTED] wrote: Hi, In the following dial peer ; If I dail 2001, ougoing will be 001415201 or 0014152001, Could please let me know. dial-peer voice 31 pots destination-pattern 2[015].. port 0/0:15 prefix 0014152 Thanks, Bala. Send instant messages to your online friends http://uk.messenger.yahoo.com Send instant messages to your online friends http://uk.messenger.yahoo.com Send instant messages to your online friends http://uk.messenger.yahoo.com
[OSL | CCIE_Voice] POTS Dial-peer
Hi, In the following dial peer ; If I dail 2001, ougoing will be 001415201 or 0014152001, Could please let me know. dial-peer voice 31 pots destination-pattern 2[015].. port 0/0:15 prefix 0014152 Thanks, Bala. Send instant messages to your online friends http://uk.messenger.yahoo.com
[OSL | CCIE_Voice] univercd URL
Hi, Could we able to open the Cisco IOS Voice Configuration Library in lab, since the URL is en/us/docs/ http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/cisco_ios_voice_configuratio n_library_glossary/vcl.htm. Thank you, Bala. Send instant messages to your online friends http://uk.messenger.yahoo.com
[OSL | CCIE_Voice] In B-ACD script and operator group
Hi, In B-ACD script the hunt group with the highest aa-hunt number is the operator group and allows this group to be reached when a caller dials 0. param aa-hunt1 1001 param-aa-hunt2 2002., In the above example I need to complete the following, when I press 0, the call should routed to 2002, when I press 2 it should not route the call to 2002, how to attain this. Could please explain this. Bala. Send instant messages to your online friends http://uk.messenger.yahoo.com
Re: [OSL | CCIE_Voice] In B-ACD script and operator group
Thanks a lot John..it is working.. John [EMAIL PROTECTED] wrote: What you should probably do is remove the aa-hunt2 statement and make it an aa-hunt10 That way, you can address the need for the operator and the operator option is only 0. John. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Balamurugan Singaram Sent: Thursday, April 10, 2008 8:57 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] In B-ACD script and operator group Hi, In B-ACD script the hunt group with the highest aa-hunt number is the operator group and allows this group to be reached when a caller dials 0. param aa-hunt1 1001 param-aa-hunt2 2002., In the above example I need to complete the following, when I press 0, the call should routed to 2002, when I press 2 it should not route the call to 2002, how to attain this. Could please explain this. Bala. Send instant messages to your online friends http://uk.messenger.yahoo.com Send instant messages to your online friends http://uk.messenger.yahoo.com
[OSL | CCIE_Voice] B-acd script in Dynamips
Hi, When I try b-acd script in Dynamips, I getting the following error message %CALL_CONTROL-6-APP_NOT_FOUND:. Could please let me know can we run b-acd scripts in Dynamips, it will work or I am missing. Thanks, Bala Send instant messages to your online friends http://uk.messenger.yahoo.com
Re: [OSL | CCIE_Voice] B-acd script in Dynamips
Hi, I have upload the b-acd script in Image folder in dynamips, and following is my config, could please let me know what I am missing: voice service voip allow-connections h323 to h323 allow-connections h323 to sip no supplementary-service h450.2 no supplementary-service h450.3 ephone-hunt 1 longest-idle pilot list 4001, 4002 timeout 10 ephone-hunt 2 longest-idle pilot list 4101, 4102 timeout 10 application service queue flash:app-b-acd-2.1.0.0.tcl param number-of-hunt-grps 2 param aa-hunt2 param aa-hunt3 param queue-len 15 param queue-manager-debugs 1 ! service aa flash:app-b-acd-aa-2.1.0.0.tcl paramspace english index 1 paramspace english language en paramspace english location flash: param service-name queue param handoff-string aa param aa-pilot 8005550123 param welcome-prompt _bacd_welcome.au param number-of-hunt-grps 2 param dial-by-extension-option 1 param second-greeting-time 60 param call-retry-timer 15 param max-time-call-retry 700 param max-time-vm-retry 2 param voice-mail 5003 ! dial-peer voice 222 voip service aa destination-pattern 8005550123 session target ipv4:192.168.1.1 incoming called-number 8005550123 dtmf-relay h245-alphanumeric codec g711ulaw no vad int loopback0 ip address 192.168.1.1 Thanks, ccievoice1 [EMAIL PROTECTED] wrote: Yes, BACD is able to work in dynamips. HTH On Wed, Apr 9, 2008 at 3:40 PM, Balamurugan Singaram [EMAIL PROTECTED] wrote: Hi, When I try b-acd script in Dynamips, I getting the following error message %CALL_CONTROL-6-APP_NOT_FOUND:. Could please let me know can we run b-acd scripts in Dynamips, it will work or I am missing. Thanks, Bala Send instant messages to your online friends http://uk.messenger.yahoo.com Send instant messages to your online friends http://uk.messenger.yahoo.com
Re: [OSL | CCIE_Voice] B-acd script in Dynamips
no I am not able to tcl files in flash, but I have copy all the tcl files in dynamips image folder, could please let me know how to upload tcl files in flash [dynamips] --- PSTN#sh flash System CompactFlash directory: File Length Name/status 1 187715 crashinfo_20020301-012431 [16777212 bytes used, 0 available, 16777212 total] 16384K bytes of ATA System CompactFlash (Read/Write) PSTN#dir flash: Directory of flash:/ 1 -rw- 187715no date crashinfo_20020301-012431 16777212 bytes total (0 bytes free) -- ccievoice1 [EMAIL PROTECTED] wrote: When you do dir flash: Can you see all the tcl files in the flash: ?? HTH On Wed, Apr 9, 2008 at 4:00 PM, Balamurugan Singaram [EMAIL PROTECTED] wrote: Hi, I have upload the b-acd script in Image folder in dynamips, and following is my config, could please let me know what I am missing: voice service voip allow-connections h323 to h323 allow-connections h323 to sip no supplementary-service h450.2 no supplementary-service h450.3 ephone-hunt 1 longest-idle pilot list 4001, 4002 timeout 10 ephone-hunt 2 longest-idle pilot list 4101, 4102 timeout 10 application service queue flash:app-b-acd-2.1.0.0.tcl param number-of-hunt-grps 2 param aa-hunt2 param aa-hunt3 param queue-len 15 param queue-manager-debugs 1 ! service aa flash:app-b-acd-aa-2.1.0.0.tcl paramspace english index 1 paramspace english language en paramspace english location flash: param service-name queue param handoff-string aa param aa-pilot 8005550123 param welcome-prompt _bacd_welcome.au param number-of-hunt-grps 2 param dial-by-extension-option 1 param second-greeting-time 60 param call-retry-timer 15 param max-time-call-retry 700 param max-time-vm-retry 2 param voice-mail 5003 ! dial-peer voice 222 voip service aa destination-pattern 8005550123 session target ipv4:192.168.1.1 incoming called-number 8005550123 dtmf-relay h245-alphanumeric codec g711ulaw no vad int loopback0 ip address 192.168.1.1 Thanks, ccievoice1 [EMAIL PROTECTED] wrote: Yes, BACD is able to work in dynamips. HTH On Wed, Apr 9, 2008 at 3:40 PM, Balamurugan Singaram [EMAIL PROTECTED] wrote: Hi, When I try b-acd script in Dynamips, I getting the following error message %CALL_CONTROL-6-APP_NOT_FOUND:. Could please let me know can we run b-acd scripts in Dynamips, it will work or I am missing. Thanks, Bala Send instant messages to your online friends http://uk.messenger.yahoo.com Send instant messages to your online friends http://uk.messenger.yahoo.com Send instant messages to your online friends http://uk.messenger.yahoo.com
Re: [OSL | CCIE_Voice] B-acd script in Dynamips
Thanks it working now :)) ccievoice1 [EMAIL PROTECTED] wrote: Definitely you need the tcl files to be located in your router flash: You can use tftp to upload the tcl files. On Wed, Apr 9, 2008 at 4:28 PM, Balamurugan Singaram [EMAIL PROTECTED] wrote: no I am not able to tcl files in flash, but I have copy all the tcl files in dynamips image folder, could please let me know how to upload tcl files in flash [dynamips] --- PSTN#sh flash System CompactFlash directory: File Length Name/status 1 187715 crashinfo_20020301-012431 [16777212 bytes used, 0 available, 16777212 total] 16384K bytes of ATA System CompactFlash (Read/Write) PSTN#dir flash: Directory of flash:/ 1 -rw- 187715no date crashinfo_20020301-012431 16777212 bytes total (0 bytes free) -- ccievoice1 [EMAIL PROTECTED] wrote: When you do dir flash: Can you see all the tcl files in the flash: ?? HTH On Wed, Apr 9, 2008 at 4:00 PM, Balamurugan Singaram [EMAIL PROTECTED] wrote: Hi, I have upload the b-acd script in Image folder in dynamips, and following is my config, could please let me know what I am missing: voice service voip allow-connections h323 to h323 allow-connections h323 to sip no supplementary-service h450.2 no supplementary-service h450.3 ephone-hunt 1 longest-idle pilot list 4001, 4002 timeout 10 ephone-hunt 2 longest-idle pilot list 4101, 4102 timeout 10 application service queue flash:app-b-acd-2.1.0.0.tcl param number-of-hunt-grps 2 param aa-hunt2 param aa-hunt3 param queue-len 15 param queue-manager-debugs 1 ! service aa flash:app-b-acd-aa-2.1.0.0.tcl paramspace english index 1 paramspace english language en paramspace english location flash: param service-name queue param handoff-string aa param aa-pilot 8005550123 param welcome-prompt _bacd_welcome.au param number-of-hunt-grps 2 param dial-by-extension-option 1 param second-greeting-time 60 param call-retry-timer 15 param max-time-call-retry 700 param max-time-vm-retry 2 param voice-mail 5003 ! dial-peer voice 222 voip service aa destination-pattern 8005550123 session target ipv4:192.168.1.1 incoming called-number 8005550123 dtmf-relay h245-alphanumeric codec g711ulaw no vad int loopback0 ip address 192.168.1.1 Thanks, ccievoice1 [EMAIL PROTECTED] wrote: Yes, BACD is able to work in dynamips. HTH On Wed, Apr 9, 2008 at 3:40 PM, Balamurugan Singaram [EMAIL PROTECTED] wrote: Hi, When I try b-acd script in Dynamips, I getting the following error message %CALL_CONTROL-6-APP_NOT_FOUND:. Could please let me know can we run b-acd scripts in Dynamips, it will work or I am missing. Thanks, Bala Send instant messages to your online friends http://uk.messenger.yahoo.com Send instant messages to your online friends http://uk.messenger.yahoo.com Send instant messages to your online friends http://uk.messenger.yahoo.com Send instant messages to your online friends http://uk.messenger.yahoo.com
Re: [OSL | CCIE_Voice] With Dynamips, CCME GUI Page is possible.
Hi Abdul,
I have done the following:
ip http server
no ip http secure-server
ip http path flash:
and since it is Dynamips, I have updated all my GUI files in Image folder,
Could Please let me know what I am missing.
Note - In Dynamips B-ACD script will work ?
Thanks,
Bala.
abdulminim rizk [EMAIL PROTECTED] wrote:
.hmmessage P { margin:0px; padding:0px } body.hmmessage {
FONT-SIZE: 10pt; FONT-FAMILY:Tahoma }Can you check if
p http server
no ip http secure-server
ip http path flash:
!
is configured [ of course u are making sure that GUI files are located in the
Flash memory ]
thanks
-
Date: Tue, 8 Apr 2008 04:49:42 +0100
From: [EMAIL PROTECTED]
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] With Dynamips, CCME GUI Page is possible.
Hi,
I have upload all CMEGUI files in Image folder, but the ccmegui file is not
open in Dynamips [http page], I am able to ping the ip source address of
telephony services.
Could please let me know with dynamips cmegui is possible.
Regards,
Bala.
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[OSL | CCIE_Voice] With Dynamips, CCME GUI Page is possible.
Hi, I have upload all CMEGUI files in Image folder, but the ccmegui file is not open in Dynamips [http page], I am able to ping the ip source address of telephony services. Could please let me know with dynamips cmegui is possible. Regards, Bala. Send instant messages to your online friends http://uk.messenger.yahoo.com
Re: [OSL | CCIE_Voice] CCM 6.x Virtual Machine
Hi, I have successfully run CCM in VMware register the IP Phone thanks a lot for your guidance. I have one NIC card [but it is not connected to switch] and one windows loop back ipaddress can I able to run PUB, SUB and unity at the same time with one loopback IPAddress or Could you please guide me to run 3 instance at a time in VMware. Thanks, Vovan L [EMAIL PROTECTED] wrote: sounds like you ethernet interface which is in bridge group are not connected to network. you can greate loopback interface and bind VMnet0 to it instead of physical ethernet interface. google loopback in Windows assuming that your VM runs on Microsoft OS. cheers - Original Message - From: Balamurugan Singaram To: ccie_voice@onlinestudylist.com Sent: Wednesday, March 05, 2008 12:03 AM Subject: [OSL | CCIE_Voice] CCM 6.x Virtual Machine Hi, I have tried accessing CCM 6.x Pre-configured Virtual Machine, However when I power on the image it gives the following error: The network bridge on device VMnet 0 is temporarily down because the bridged ethernet interface is down. Could please let me know how to assign the IPaddress to Pre-configured Virtual CCM 6.x image. Thanks, Bala. Send instant messages to your online friends http://uk.messenger.yahoo.com Send instant messages to your online friends http://uk.messenger.yahoo.com
Re: [OSL | CCIE_Voice] CCM 6.x Virtual Machine
Hi Vovan, I have create window loopback interface, how bind VMnet0 to loopback interface. Thanks, Bala. Vovan L [EMAIL PROTECTED] wrote: sounds like you ethernet interface which is in bridge group are not connected to network. you can greate loopback interface and bind VMnet0 to it instead of physical ethernet interface. google loopback in Windows assuming that your VM runs on Microsoft OS. cheers - Original Message - From: Balamurugan Singaram To: ccie_voice@onlinestudylist.com Sent: Wednesday, March 05, 2008 12:03 AM Subject: [OSL | CCIE_Voice] CCM 6.x Virtual Machine Hi, I have tried accessing CCM 6.x Pre-configured Virtual Machine, However when I power on the image it gives the following error: The network bridge on device VMnet 0 is temporarily down because the bridged ethernet interface is down. Could please let me know how to assign the IPaddress to Pre-configured Virtual CCM 6.x image. Thanks, Bala. Send instant messages to your online friends http://uk.messenger.yahoo.com Send instant messages to your online friends http://uk.messenger.yahoo.com
Re: [OSL | CCIE_Voice] CCM 6.x Virtual Machine
Hi, VM-Ware server [CCM 4.x] IPAddress 172.168.x.x, I have created windows loopback address and assign the address in VMConsole as: HOST - Virtual Network Editor - VMNET0: Microsoft loopback adapter, but from windows server I am not able to ping VM-Server 172.168.x.x, could please suggest me the solution. Thanks, Bala. Vovan L [EMAIL PROTECTED] wrote: sounds like you ethernet interface which is in bridge group are not connected to network. you can greate loopback interface and bind VMnet0 to it instead of physical ethernet interface. google loopback in Windows assuming that your VM runs on Microsoft OS. cheers - Original Message - From: Balamurugan Singaram To: ccie_voice@onlinestudylist.com Sent: Wednesday, March 05, 2008 12:03 AM Subject: [OSL | CCIE_Voice] CCM 6.x Virtual Machine Hi, I have tried accessing CCM 6.x Pre-configured Virtual Machine, However when I power on the image it gives the following error: The network bridge on device VMnet 0 is temporarily down because the bridged ethernet interface is down. Could please let me know how to assign the IPaddress to Pre-configured Virtual CCM 6.x image. Thanks, Bala. Send instant messages to your online friends http://uk.messenger.yahoo.com Send instant messages to your online friends http://uk.messenger.yahoo.com
Re: [OSL | CCIE_Voice] CCM 6.x Virtual Machine
Hi, VMServer IP Address - 172.16.130.86/24 default gateway 172.16.130.1, I am having windows loop back ipaddress 172.16.130.87/24 in the following subnet, the VMneto is mapped to window loop back [172.16.130.87/24] address, reboot both vmserver and windows server 2003 but I am not able to ping the VM-CUCM server 172.16.130.86. Note - when the VMware server run, two virtual networks [vmnet1 and vmnet8] are coming up, they are in different subnet 192.168.x.x. Could please let me know what I am missing. Thanks, Vovan L [EMAIL PROTECTED] wrote: please provide ipconfig /all - Original Message - From: Balamurugan Singaram To: ccie_voice@onlinestudylist.com Sent: Wednesday, March 05, 2008 8:39 AM Subject: Re: [OSL | CCIE_Voice] CCM 6.x Virtual Machine Hi, VM-Ware server [CCM 4.x] IPAddress 172.168.x.x, I have created windows loopback address and assign the address in VMConsole as: HOST - Virtual Network Editor - VMNET0: Microsoft loopback adapter, but from windows server I am not able to ping VM-Server 172.168.x.x, could please suggest me the solution. Thanks, Bala. Vovan L [EMAIL PROTECTED] wrote: sounds like you ethernet interface which is in bridge group are not connected to network. you can greate loopback interface and bind VMnet0 to it instead of physical ethernet interface. google loopback in Windows assuming that your VM runs on Microsoft OS. cheers - Original Message - From: Balamurugan Singaram To: ccie_voice@onlinestudylist.com Sent: Wednesday, March 05, 2008 12:03 AM Subject: [OSL | CCIE_Voice] CCM 6.x Virtual Machine Hi, I have tried accessing CCM 6.x Pre-configured Virtual Machine, However when I power on the image it gives the following error: The network bridge on device VMnet 0 is temporarily down because the bridged ethernet interface is down. Could please let me know how to assign the IPaddress to Pre-configured Virtual CCM 6.x image. Thanks, Bala. Send instant messages to your online friends http://uk.messenger.yahoo.com Send instant messages to your online friends http://uk.messenger.yahoo.com Send instant messages to your online friends http://uk.messenger.yahoo.com
[OSL | CCIE_Voice] CCM 6.x Virtual Machine
Hi, I have tried accessing CCM 6.x Pre-configured Virtual Machine, However when I power on the image it gives the following error: The network bridge on device VMnet 0 is temporarily down because the bridged ethernet interface is down. Could please let me know how to assign the IPaddress to Pre-configured Virtual CCM 6.x image. Thanks, Bala. Send instant messages to your online friends http://uk.messenger.yahoo.com
[OSL | CCIE_Voice] Access Callmanager Corporate directory
To Access Callmanager Corporate directory from CME router, which one is the right method? http:/PUB_IP/CCMCIP/xmldirectory.asp or http://PUB_IP/localdirectory Could you please suggest me? Thanks, Bala Send instant messages to your online friends http://uk.messenger.yahoo.com
[OSL | CCIE_Voice] Mark signalling traffic in CCM.
Hi, To mark signalling traffic in CCM, we can accomplish this only in Enterprise Parameter as follows: Dscp for SCCP phone-based services = default(0) Dscp for SCCP phone configuration = CS3 Dscp for CCM to device interface = CS3 Or could please let me know; there is any other place in CCM to mark signalling traffic. Thanks, Bala. Send instant messages to your online friends http://uk.messenger.yahoo.com
[OSL | CCIE_Voice] Layer 3 QOS Configuration
Hi, For g.711 g.729 call in CCM it takes 80 kbps 24 kbps, if it is Gatekeeper it takes 128 16 kbps. Now if I want to pass only 2 g.729 call in WAN QOS config how should I set the priority value and bandwidth for signal? Any suggested voice priority signal bandwidth for layer three specific to g.711 and g.729 calls. Thanks, Bala. Send instant messages to your online friends http://uk.messenger.yahoo.com
[OSL | CCIE_Voice] FRTS QOS percent
Hi, FRTS question as follows: Configure FRTS between HQ and CME site, bandwidth of 128 kbps. Voice media traffic is strict-priority que guaranteed 33% of available bandwidth. Guarantee voice control traffic 5% of available bandwidth. The solution as follows: Method -1: Policy-map LLQ Class voice-media Priority percent 33 Class voice-sig Bandwidth percent 5 Method 2 Policy-map LLQ Class voice-media Priority 42 Class voice-sig Bandwidth 6 The above two config will accomplish the same or not? Can please let me know which one is the best method to use. Thanks, Bala. Send instant messages to your online friends http://uk.messenger.yahoo.com
[OSL | CCIE_Voice] SIP trunk between CME and CCM 4.1
Hi, I am trying a topology sip trunk, with ccm 4.1 and CME, CME source ip address - 172.16.2.100, from CME IPPhone I am able to reach CCM IPPhone, but from CCM IPPhone to CME I am not able to reach CME IPPhone. I am having route pattern in CCM to CME as siptrunk as gateway, In SIP Trunk is configured as follows in CCM: Sip trunk destination address - 172.16.2.100, Enable the media termination point required, but media resource group list is [none], in ccm side. I am not configured any transcoder in CCM side. Region is g711ulaw. In CME side transcoder is up. Could please let me know what I missing from CCM to CME call routing via SIP trunk. Thank you, Bala Send instant messages to your online friends http://uk.messenger.yahoo.com
Re: [OSL | CCIE_Voice] CUE, call forward busy is not working via PSTN Line.
Hi Vik, I am using the following command as below, so the 10 digit number hit CUE as 4 digit number. dialplan-pattern 1 4085551... extension-length 4 I think it should hit CUE voicemail. Could please let me know your suggestion. Thanks, Bala. Vik Malhi [EMAIL PROTECTED] wrote: I am pretty sure you are using dialplan pattern- the called-number will be expanded from 4 digits to 10 digits when the call coming in from the PSTN is forwarded. Ensure you have a dial-peer with the expanded number (all other settings are the same as the current dial-peer pointing to CUE). Use the debug voip dialpeer command to help troubleshoot. Vik Malhi CCIE Voice Instructor / Developer - IPexpert, Inc. CCIE Voice #13890 CCSI #31584 URL: http://www.IPexpert.com Toll Free: +1.866.225.8064 International: +1.810.326.1444 - From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Balamurugan Singaram Sent: Saturday, December 01, 2007 7:11 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CUE,call forward busy is not working via PSTN Line. Hi, The call-forward busy is working when call is called from CCM to CME, codec g729. When I try call from PSTN to CME, call-forward is not working, I get busy tone after 3 ring. Note - I have configured trancoder in CME. Thanks, Bala. Send instant messages to your online friends http://uk.messenger.yahoo.com Send instant messages to your online friends http://uk.messenger.yahoo.com
[OSL | CCIE_Voice] CUE, call forward busy is not working via PSTN Line.
Hi, The call-forward busy is working when call is called from CCM to CME, codec g729. When I try call from PSTN to CME, call-forward is not working, I get busy tone after 3 ring. Note - I have configured trancoder in CME. Thanks, Bala. Send instant messages to your online friends http://uk.messenger.yahoo.com
[OSL | CCIE_Voice] SRST mode
Hi, BR2, phone1-3001 and phone2-3002 and HQ phone1-1001 and phone2-1002. When the BR2 is in SRST mode, can I call BR2 from HQ with normal 4 digit number [3001] or 10 digit pstn numbers to reach BR2. Normal user from HQ will dial only the 4 digit number to reach BR2 [3001, 3002], but it is not hitting the BR2 phone at SRST mode. But I can call BR2 with 10 digit PSTN number it hitting BR2, It is normal or I should reach the BR2 with 4 digit number from HQ in SRST mode also. Please let me know your suggestion. Thanks, Bala. Send instant messages to your online friends http://uk.messenger.yahoo.com
[OSL | CCIE_Voice] B-ACD script error
Hi, I am getting the following error in B-ACD script. %CALL_CONTROL-6-APP_NOT_FOUND: Applic 104 not found. Handing callid 22 to the alternate app. The running-config as follows: --- version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname cme ! boot-start-marker boot-end-marker ! ! no aaa new-model ! resource policy ! network-clock-participate wic 2 ip cef ! ! isdn switch-type primary-ni voice-card 0 no dspfarm ! ! ! ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip no supplementary-service h450.2 no supplementary-service h450.3 ! ! ! application service queue flash:app-b-acd-2.1.2.2.tcl param queue-len 15 param aa-hunt1 param number-of-hunt-grps 1 param queue-manager-debugs 1 ! service aa flash: ! service aaa builtin:app-b-acd-aa-2.1.2.2.tcl paramspace english index 1 param aa-pilot param number-of-hunt-grps 1 param handoff-string aaa param dial-by-extension-option 1 paramspace english language en param welcome-prompt _bacd_welcome.au param call-retry-timer 15 param service-name queue paramspace english location flash: param second-greeting-time 60 param max-time-vm-retry 2 param voice-mail 411 param max-time-call-retry 700 ! ! ! ! ! ! controller T1 0/2/0 framing esf linecode b8zs pri-group timeslots 1-3,24 ! ! ! ! interface Loopback2 ip address 192.168.1.1 255.255.255.0 ! interface GigabitEthernet0/0 no ip address shutdown duplex auto speed auto media-type rj45 ! interface GigabitEthernet0/1 ip address 172.16.2.100 255.255.255.0 duplex auto speed auto media-type rj45 ! interface FastEthernet0/0/0 ! interface FastEthernet0/0/1 ! interface FastEthernet0/0/2 ! interface FastEthernet0/0/3 ! interface Dot11Radio0/1/0 no ip address shutdown speed basic-1.0 basic-2.0 basic-5.5 6.0 9.0 basic-11.0 12.0 18.0 24.0 36.0 48.0 54.0 station-role root ! interface Serial0/2/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice no cdp enable ! interface Serial0/3/0 no ip address shutdown no fair-queue ! interface GigabitEthernet1/0 no ip address ! interface Service-Engine2/0 no ip address shutdown ! interface Vlan1 no ip address ! ! ip http server no ip http secure-server ! ! ! tftp-server flash:P00307020200.bin tftp-server flash:P00307020200.loads tftp-server flash:P00307020200.sb2 tftp-server flash:P00307020200.sbn tftp-server flash:P00403020214.bin ! control-plane ! ! ! voice-port 0/2/0:23 ! ! ! ! ! dial-peer voice 104 voip service aaa destination-pattern session target ipv4:192.168.1.1 incoming called-number dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 500 pots service aaa destination-pattern incoming called-number port 0/2/0:23 ! ! ! telephony-service load 7910 P00403020214 load 7960-7940 P00307020200 max-ephones 4 max-dn 4 ip source-address 172.16.2.100 port 2000 auto assign 1 to 4 dialplan-pattern 1 12341... extension-length 4 voicemail 4211 max-conferences 12 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Nov 26 2007 14:26:18 ! ! ephone-dn 1 dual-line number 1001 call-forward busy 411 call-forward noan 411 timeout 10 ! ! ephone-dn 2 dual-line number 1002 call-forward busy 411 call-forward noan 411 timeout 10 ! ! ephone-dn 3 dual-line number 1003 call-forward busy 411 call-forward noan 411 timeout 10 ! ! ephone-dn 4 dual-line number 1004 call-forward busy 411 call-forward noan 411 timeout 10 ! ! ephone 1 no multicast-moh mac-address 0007.EB26.DE79 type 7940 button 1:1 ! ! ! ephone 2 no multicast-moh mac-address 000A.8A93.E0AB type 7960 button 1:2 ! ! ! ephone 3 no multicast-moh ! ! ! ephone 4 no multicast-moh ! ! ephone-hunt 1 longest-idle pilot list 1001, 1002, 1003, 1004 timeout 10, 10, 10, 10 Thanks, Bala. Send instant messages to your online friends http://uk.messenger.yahoo.com
Re: [OSL | CCIE_Voice] B-ACD script error
Hi, Thanks a lot, now the B-ACD Service Started, but when I press number one [1] it should hit huntgroup as follows but it fails. Could you please let me know what I am missing in this config. application service queue flash:app-b-acd-2.1.2.2.tcl param queue-len 15 param aa-hunt1 param number-of-hunt-grps 1 param queue-manager-debugs 1 ephone-hunt 1 longest-idle pilot list 1001, 1002, 1003, 1004 timeout 10, 10, 10, 10 Belicza Zsolt [EMAIL PROTECTED] wrote: Hi! I think the following line is incorrect: service aaa builtin:app-b-acd-aa-2.1.2.2.tcl If the file app-b-acd-aa-2.1.2.2.tcl is in your flash, then the correct command is (and subcommand params and paramspaces): service aaa flash:app-b-acd-aa-2.1.2.2.tcl Regards, Zsolt - From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Balamurugan Singaram Sent: Monday, November 26, 2007 3:53 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] B-ACD script error Hi, I am getting the following error in B-ACD script. %CALL_CONTROL-6-APP_NOT_FOUND: Applic 104 not found. Handing callid 22 to the alternate app. The running-config as follows: --- version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname cme ! boot-start-marker boot-end-marker ! ! no aaa new-model ! resource policy ! network-clock-participate wic 2 ip cef ! ! isdn switch-type primary-ni voice-card 0 no dspfarm ! ! ! ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip no supplementary-service h450.2 no supplementary-service h450.3 ! ! ! application service queue flash:app-b-acd-2.1.2.2.tcl param queue-len 15 param aa-hunt1 param number-of-hunt-grps 1 param queue-manager-debugs 1 ! service aa flash: ! service aaa builtin:app-b-acd-aa-2.1.2.2.tcl paramspace english index 1 param aa-pilot param number-of-hunt-grps 1 param handoff-string aaa param dial-by-extension-option 1 paramspace english language en param welcome-prompt _bacd_welcome.au param call-retry-timer 15 param service-name queue paramspace english location flash: param second-greeting-time 60 param max-time-vm-retry 2 param voice-mail 411 param max-time-call-retry 700 ! ! ! ! ! ! controller T1 0/2/0 framing esf linecode b8zs pri-group timeslots 1-3,24 ! ! ! ! interface Loopback2 ip address 192.168.1.1 255.255.255.0 ! interface GigabitEthernet0/0 no ip address shutdown duplex auto speed auto media-type rj45 ! interface GigabitEthernet0/1 ip address 172.16.2.100 255.255.255.0 duplex auto speed auto media-type rj45 ! interface FastEthernet0/0/0 ! interface FastEthernet0/0/1 ! interface FastEthernet0/0/2 ! interface FastEthernet0/0/3 ! interface Dot11Radio0/1/0 no ip address shutdown speed basic-1.0 basic-2.0 basic-5.5 6.0 9.0 basic-11.0 12.0 18.0 24.0 36.0 48.0 54.0 station-role root ! interface Serial0/2/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice no cdp enable ! interface Serial0/3/0 no ip address shutdown no fair-queue ! interface GigabitEthernet1/0 no ip address ! interface Service-Engine2/0 no ip address shutdown ! interface Vlan1 no ip address ! ! ip http server no ip http secure-server ! ! ! tftp-server flash:P00307020200.bin tftp-server flash:P00307020200.loads tftp-server flash:P00307020200.sb2 tftp-server flash:P00307020200.sbn tftp-server flash:P00403020214.bin ! control-plane ! ! ! voice-port 0/2/0:23 ! ! ! ! ! dial-peer voice 104 voip service aaa destination-pattern session target ipv4:192.168.1.1 incoming called-number dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 500 pots service aaa destination-pattern incoming called-number port 0/2/0:23 ! ! ! telephony-service load 7910 P00403020214 load 7960-7940 P00307020200 max-ephones 4 max-dn 4 ip source-address 172.16.2.100 port 2000 auto assign 1 to 4 dialplan-pattern 1 12341... extension-length 4 voicemail 4211 max-conferences 12 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Nov 26 2007 14:26:18 ! ! ephone-dn 1 dual-line number 1001 call-forward busy 411 call-forward noan 411 timeout 10 ! ! ephone-dn 2 dual-line number 1002 call-forward busy 411 call-forward noan 411 timeout 10 ! ! ephone-dn 3 dual-line number 1003 call-forward busy 411 call-forward noan 411 timeout 10 ! ! ephone-dn 4 dual-line number 1004 call-forward busy 411 call-forward noan 411 timeout 10 ! ! ephone 1 no multicast-moh mac-address 0007.EB26.DE79 type 7940 button 1:1 ! ! ! ephone 2 no multicast-moh mac-address 000A.8A93.E0AB type 7960 button 1:2 ! ! ! ephone 3 no multicast-moh
[OSL | CCIE_Voice] Music On Hold Server
Hi, The MOH is not getting registered with CCM. I had try the following troubleshooting method update Default TFTP IP Address under IP voice media streaming application and rebooting the CCM server, but MOH is not get registered with CCM. Could please let me know the troubleshooting steps to make register MOH with CCM? Music On Hold Server: MOH_ccmpub (MOH_ccmpub) Registration: Not Registered IP Address: x.11.100.y Thanks, Bala Send instant messages to your online friends http://uk.messenger.yahoo.com
[OSL | CCIE_Voice] CME/CUE error
Hi, I am getting the following error when try to integrate CME/CUE, could please let me know the detail about the error and how to come back to config t, mode from this error display. WARNING:: IOS communication appears delayed! WARNING:: WARNING:: Please verify the Service Engine IP Address WARNING:: and Default Gateway are configured correctly WARNING:: on the service engine interface in IOS. Thanks, Bala Send instant messages to your online friends http://uk.messenger.yahoo.com
Re: [OSL | CCIE_Voice] CME/CUE error
Hi, Thanks a lot, now I am not able to reach config mode...the error is going on display. When I try Crtl+Shift+6 and x it is going back to terminal server and back telnet to CUE router, it display the same following error message. Any commands to break from the following error mode back to Config mode. Thank you, Bala. Belicza Zsolt [EMAIL PROTECTED] wrote: Hi! Here is a good configuration for nm-cue: interface service-engine 1/0 no shut ip unnumbered vlan 400 - source should be the voice interface service-module ip address 20.0.0.0 255.255.255.0 service-module default-gateway 20.0.0.1 -- IP address of vlan 400 ! ip route 20.0.0.5 255.255.255.255 service-engine 1/0 required for routing after these configuration, you should be able to ping 20.0.0.5 from the router. To return back to the router: type: ctrl+shift+6 then release and press x now u are back from telnet. You can type disc [enter] to not return back automatically. Regard, Zsolt - From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Balamurugan Singaram Sent: Saturday, November 24, 2007 3:33 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME/CUE error Hi, I am getting the following error when try to integrate CME/CUE, could please let me know the detail about the error and how to come back to config t, mode from this error display. WARNING:: IOS communication appears delayed! WARNING:: WARNING:: Please verify the Service Engine IP Address WARNING:: and Default Gateway are configured correctly WARNING:: on the service engine interface in IOS. Thanks, Bala Send instant messages to your online friends http://uk.messenger.yahoo.com Send instant messages to your online friends http://uk.messenger.yahoo.com
Re: [OSL | CCIE_Voice] Trunk Configuration
Hi Mark/David, In CCIE Lab I think we should use h.225 GK Trunk only. Could you please correct me if I am wrong? Thanks, Bala. Mark Snow [EMAIL PROTECTED] wrote: Thanks David - exactly correct. Also keep in mind what is on the other end of the GK or NonGK Trunk. If it is a CCM or a CME - then best in real life to use a ICT (GK or nonGK controlled) as this will provide you with the best integration. If it is a non-Cisco H323 GW - then obviously a H323 if no GK is involved - or if a GK is involved - a H225 GK Trunk. As David said - for the lab - do whatever it says or if it doesn't specify - then use what we outlined above to make your decision. Cheers, Mark Snow CCIE #14073 (Voice, Security) CCSI #31583 Senior Technical Instructor - IPexpert, Inc. A Cisco Learning Partner - We Accept Learning Credits! Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: [EMAIL PROTECTED] 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. On Nov 14, 2007, at 6:56 AM, David Blair wrote: For the CCIE Voice Lab or Real Life? CCIE Voice Lab - Would be whatever the lab guide says. Real Life - Do you have say over 20 H.323 gateways? You might think about implementing a GateKeeper (GK). Gatekeeper is basically a trafiic cop for H.323 gateways. So if you have a Gatekeeper select #1 or #3 below depending on the type of trunk you need to setup. Oerwise select #2 below. David L. Blair - Date: Wed, 14 Nov 2007 09:54:57 + From: [EMAIL PROTECTED] To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Trunk Configuration Hi, The following types of h.323 trunks can be configured: 1) H.225 trunk (GK controlled). 2 Inter-cluster trunk (non-GK Controlled) 3) Inter-cluster trunk (GK Controlled) In IPexpert page 84, [Inter-cluster trunk (non-GK controlled)] is used and in page 98, H.225 trunk (GK controlled) is used. Could you please help me which method is the best method for h.323 trunk? Thanks, Bala. Send instant messages to your online friends http://uk.messenger.yahoo.com - Peek-a-boo FREE Tricks Treats for You! Get 'em! Send instant messages to your online friends http://uk.messenger.yahoo.com
[OSL | CCIE_Voice] busy-out the unused B-Channels
Hi, We should busy-out the unused B-Channels for 6608 gateway, could you please confirm that we should do the same for MGCP gateway also [busy-out the unused B-Channels] or it is not necessary for MGCP gateway. Thanks, Bala. Send instant messages to your online friends http://uk.messenger.yahoo.com
[OSL | CCIE_Voice] debug the incoming digit in 6608 gateway.
Hi, In 6608 gateway, there is any command equal to [debug isdn q931] or how we can find the find the incoming digit number in 6608 T1 gateway. Thanks, Bala. Send instant messages to your online friends http://uk.messenger.yahoo.com
