Re: [OSL | CCIE_Voice] call-manager-fallback
not possible with call-manager-fallback. You still have your 3 party conference just that you can't configure hardware conferencing. Use telephony-service SRST On 13 April 2011 06:00, Erwan Erwan e_er...@yahoo.com wrote: hi all, why i can not register my IOS conf with call-manager-fallback ? it show TCP_CONN_ERROR here is config --- sccp local Vlan240 sccp ccm 10.10.201.1 identifier 2 sccp ccm 10.10.210.10 identifier 1 sccp sccp ccm group 1 associate ccm 1 priority 1 associate ccm 2 priority 2 associate profile 1 register sb-conf dspfarm profile 1 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 3 associate application SCCP call-manager-fallback max-conferences 8 gain -6 transfer-system full-consult ip source-address 10.10.201.1 port 2000 max-ephones 5 max-dn 10 no-reg Gateway IP Address: 10.10.201.1, Port Number: 2000 IP Precedence: 5 User Masked Codec list: None Call Manager: 10.10.201.1, Port Number: 2000 Priority: N/A, Version: 3.1, Identifier: 2 Trustpoint: N/A Call Manager: 10.10.210.10, Port Number: 2000 Priority: N/A, Version: 3.1, Identifier: 1 Trustpoint: N/A Conferencing Oper State: ACTIVE_IN_PROGRESS - Cause Code: TCP_CONN_ERROR ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] B-ACD Not working
Hi , We tested with B-ACD in CME . whenever we dial the pilot number call disconnect. Config application service aa flash:app-b-acd-aa-3.0.0.2.tcl paramspace english index 1 param menu-timeout 1 param dial-by-extension-option 1 param handoff-string aa paramspace english language en param max-time-vm-retry 2 param aa-pilot 7500 paramspace english location flash: param second-greeting-time 1 param welcome-prompt _bacd_welcome.au paramspace english prefix en param service-name ACD service ACD flash:app-b-acd-3.0.0.2.tcl paramspace english language en paramspace english index 0 param aa-hunt1 7001 param aa-hunt2 7002 param number-of-hunt-grps 4 param aa-hunt3 7003 Dial-peer dial-peer voice 7500 voip service aa destination-pattern 7500 session target ipv4:192.168.1.100 incoming called-number 7500 dtmf-relay h245-alphanumeric codec g711ulaw I have verified that all the audio files uploade in the flash. MoH working for IP Phones. When i check show call application session , it tries to establish session but it end session immediately___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] B-ACD Not working
param voice-mail is mandatory even if you are not sending the call to voice mail, you can configure a hunt pilot number or dn number On 13 April 2011 16:35, mgscip gpsvoiceexpe...@yahoo.com wrote: Hi , We tested with B-ACD in CME . whenever we dial the pilot number call disconnect. Config application service aa flash:app-b-acd-aa-3.0.0.2.tcl paramspace english index 1 param menu-timeout 1 param dial-by-extension-option 1 param handoff-string aa paramspace english language en param max-time-vm-retry 2 param aa-pilot 7500 paramspace english location flash: param second-greeting-time 1 param welcome-prompt _bacd_welcome.au paramspace english prefix en param service-name ACD service ACD flash:app-b-acd-3.0.0.2.tcl paramspace english language en paramspace english index 0 param aa-hunt1 7001 param aa-hunt2 7002 param number-of-hunt-grps 4 param aa-hunt3 7003 Dial-peer dial-peer voice 7500 voip service aa destination-pattern 7500 session target ipv4:192.168.1.100 incoming called-number 7500 dtmf-relay h245-alphanumeric codec g711ulaw I have verified that all the audio files uploade in the flash. MoH working for IP Phones. When i check show call application session , it tries to establish session but it end session immediately ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Need Advice
I was hoping you all could help me out with this problem. We had a power outage this past weekend and prior to this all was working fine. Yesterday we received reports from out branch offices that they could not make outbound calls. I had one of the users dial my cellphone number and my cellphone does ring, however the user does not receive any ringback, just dead air for about 7 seconds and then a fast busy. If I answer my cellphone they also receive a fast busy. The route groups for the remote sites have the GW here at the home office in their RG but it never goes to that gateway. I've had to remove the GWs at the branches from the route groups so that they can make outbound calls via the GW here. Here is the part that gets me, inbound dialing is still working fine via those gateways. I'm at a loss right now. TAC says its my phone company, but of course there is no trouble found on the lines according to them. We're running UCM 7.1.5.32900-2. The two gateways are both H323 gateways with FXO ports. One is a 2911 and the other is a 2801. Again I'm stumped because this was all working prior to the power outage at our home site. Any advice would be greatly appreciated. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] B-ACD Not working
Hi, Add service aa flash:app-b-acd-aa-3.0.0.2.tcl param number-of-hunt-grps 3 and correct the number of hunt group in ACD service (I guess you have 3 not 4) Hope this will help Naoufal From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of mgscip Sent: Wednesday, April 13, 2011 5:36 PM To: ccie Subject: [OSL | CCIE_Voice] B-ACD Not working Hi , We tested with B-ACD in CME . whenever we dial the pilot number call disconnect. Config application service aa flash:app-b-acd-aa-3.0.0.2.tcl paramspace english index 1 param menu-timeout 1 param dial-by-extension-option 1 param handoff-string aa paramspace english language en param max-time-vm-retry 2 param aa-pilot 7500 paramspace english location flash: param second-greeting-time 1 param welcome-prompt _bacd_welcome.au paramspace english prefix en param service-name ACD service ACD flash:app-b-acd-3.0.0.2.tcl paramspace english language en paramspace english index 0 param aa-hunt1 7001 param aa-hunt2 7002 param number-of-hunt-grps 4 param aa-hunt3 7003 Dial-peer dial-peer voice 7500 voip service aa destination-pattern 7500 session target ipv4:192.168.1.100 incoming called-number 7500 dtmf-relay h245-alphanumeric codec g711ulaw I have verified that all the audio files uploade in the flash. MoH working for IP Phones. When i check show call application session , it tries to establish session but it end session immediately * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Need Advice
Figured it out. Thanks On Wed, Apr 13, 2011 at 12:04 PM, Bo Gao bga...@gmail.com wrote: A couple thing I would do: 1) Check if there are missing config on the Gateways, especially your outbound dial peers; Check the FXO port status; Check the physical line status; 2) Check CUCM DB Sync; 3) Are your phone lines patched on a patch panel or directly hooked on the routers? If you are using a patch panel, you can loop to either direction to troubleshoot further. Hope it helps, Bo On Wed, Apr 13, 2011 at 9:03 AM, Chevy chevy.man...@gmail.com wrote: I was hoping you all could help me out with this problem. We had a power outage this past weekend and prior to this all was working fine. Yesterday we received reports from out branch offices that they could not make outbound calls. I had one of the users dial my cellphone number and my cellphone does ring, however the user does not receive any ringback, just dead air for about 7 seconds and then a fast busy. If I answer my cellphone they also receive a fast busy. The route groups for the remote sites have the GW here at the home office in their RG but it never goes to that gateway. I've had to remove the GWs at the branches from the route groups so that they can make outbound calls via the GW here. Here is the part that gets me, inbound dialing is still working fine via those gateways. I'm at a loss right now. TAC says its my phone company, but of course there is no trouble found on the lines according to them. We're running UCM 7.1.5.32900-2. The two gateways are both H323 gateways with FXO ports. One is a 2911 and the other is a 2801. Again I'm stumped because this was all working prior to the power outage at our home site. Any advice would be greatly appreciated. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] difference between g729r8, g729ar8, g729br8, and g729abr8
Just wanted to share this with you guys: What is the difference between g729r8, g729ar8, g729br8, and g729abr8 ? The format of the codewords generated by the above 4 codecs are identical. The g729ar8 is a reduced complexity version of the G.729 speech codec. It is bit stream interoperable with the full version g729r8, i.e. a reduced complexity encoder may be used with a full implementation of the decoder, and vice versa. However, the performance of this codec defined in g729ar8 may not be as good as the full implementation of Recommendation G.729 in certain circumstances. The g729br8 is g729r8 with built-in VAD implemented. A SID (Silence Insertion Descriptor) frame will be generated when the line signal going from active to non-active. After that, no packets will be sent until the next talkspurt begins, or another SID frame will be sent if the characteristics of the line noise changes. The g729r8 and g729br8 are not fully interoperable because g729r8 does not recognize the SID frame. But the g729br8 can decode the codewords generatd by g729r8. g729ar8 = simplified verison of g729r8 g729br8 = g729r8 + built-in VAD g729abr8 = simplified g729r8 + built-in VAD g729r8 and g729ar8 are fully interoperable g729br8 and g729abr8 are fully interoperable g729(a)-- g729(a)b(OK) g729(a)b-- g729(a) (possibly ok, but not recommended) ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Help with SIP Firmware
Hi All I converted my 7961 phone to SIP Load for a Lab but am not able to revert the Phone to SCCP Load. For changing any network settings on the phone, like TFTP etc, the phone prompts for Authorization with Username Password. Has anyone faced this issue? Can anyone suggest how should I revert the phone back to factory defaults or procedure to edit network settings? A big thanks in advance. . Mann ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Need help configuring MVA in MGCP site
Hi Experts, I am able to configure MVA in h323 site but in MGCP I create similar voip dial-peers and application but it is not working. In other words how to configure MGCP hairpin in order to MVA to work. Thanks in advance Shrini ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Help with SIP Firmware
Hi All I got it working. Had to reconfigure SIP Registrar register the phone ot it. Then with username password assigned to the phone, I could change the network settings. Though the phone kept on showing the administrator username only. Good day. Mann ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] use 26xm to setup ccie voice lab
They are modular router but will work. 26xxXM NM-HD-2VE ( it has builtin PVDMs) VWIC-2MFT-T1 or 1MFT based on your site requirement. VWIC-2MFT-E1 Or also you can use VWIC-1MFT-E1/T1 For framerelay: NM-4T (on PSTN router) NM-1T on Site routers Serial cables - 3 VWIC-4ESW - 2 3750 Switch On 4/12/2011 10:24 PM, bruno wrote: do some guys have any experience to use 26XX xm to setup ccie voice lab ? could u kindly give the hareware and software list. include e1 t1 card /pvdm stuff. as detail as possible.thanks in advance. BR, bruno ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Help with SIP Firmware
Auto register it to CUCM. On Wed, Apr 13, 2011 at 9:24 PM, Mann Chaddha mann.chad...@gmail.comwrote: Hi All I converted my 7961 phone to SIP Load for a Lab but am not able to revert the Phone to SCCP Load. For changing any network settings on the phone, like TFTP etc, the phone prompts for Authorization with Username Password. Has anyone faced this issue? Can anyone suggest how should I revert the phone back to factory defaults or procedure to edit network settings? A big thanks in advance. . Mann ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Help with SIP Firmware
Select unlock key use password cisco OR Connect to Callmanager with dhcp enabled (router dhcp is also ok) by pressing # key when you see orange row lights on phone button dial 123456789*0# in sequence and relax. On 4/13/2011 9:24 PM, Mann Chaddha wrote: Hi All I converted my 7961 phone to SIP Load for a Lab but am not able to revert the Phone to SCCP Load. For changing any network settings on the phone, like TFTP etc, the phone prompts for Authorization with Username Password. Has anyone faced this issue? Can anyone suggest how should I revert the phone back to factory defaults or procedure to edit network settings? A big thanks in advance. . Mann ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Help with SIP Firmware
The simplest way is to create the phones in CUCM, choose SCCP as the protocol, have your dhcp tftp settings set as the CUCM IP. Go to settings and type **# and erase the configs to force the phone to acquire new settings. On 14 April 2011 07:24, Mann Chaddha mann.chad...@gmail.com wrote: Hi All I converted my 7961 phone to SIP Load for a Lab but am not able to revert the Phone to SCCP Load. For changing any network settings on the phone, like TFTP etc, the phone prompts for Authorization with Username Password. Has anyone faced this issue? Can anyone suggest how should I revert the phone back to factory defaults or procedure to edit network settings? A big thanks in advance. . Mann ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Help with SIP Firmware
http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a00800941bb.shtml Complete these steps: 1. Unplug the power cable from the phone, and then plug in the cable again. The phone begins its power up cycle. 2. Immediately press and hold *#* and while the Headset, Mute, and Speaker buttons begin to flash in sequence, release #. The line buttons flash in sequence in order to indicate that the phone waits for you to enter the key sequence for the reset. 3. Press *123456789*0#* within 60 seconds after the Headset, Mute, and Speaker buttons begin to flash. If you repeat a key within the sequence, for example, if you press 1*22*3456789*0#, the sequence is still accepted and the phone resets. If you do not complete this key sequence or do not press any keys, after 60 seconds the Headset, Mute, and Speaker buttons no longer flash, and the phone continues with its normal startup process. The phone does not reset. If you enter an invalid key sequence, the buttons no longer flash, and the phone continues with its normal startup process. The phone does not reset. If you enter this key sequence correctly, the phone displays this prompt: upgrading On Wed, Apr 13, 2011 at 11:54 PM, Rogers Ochieng rogersochi...@gmail.comwrote: The simplest way is to create the phones in CUCM, choose SCCP as the protocol, have your dhcp tftp settings set as the CUCM IP. Go to settings and type **# and erase the configs to force the phone to acquire new settings. On 14 April 2011 07:24, Mann Chaddha mann.chad...@gmail.com wrote: Hi All I converted my 7961 phone to SIP Load for a Lab but am not able to revert the Phone to SCCP Load. For changing any network settings on the phone, like TFTP etc, the phone prompts for Authorization with Username Password. Has anyone faced this issue? Can anyone suggest how should I revert the phone back to factory defaults or procedure to edit network settings? A big thanks in advance. . Mann ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
