[OSL | CCIE_Voice] Lab exam CCM Features and services PDF
Hi, Would appreciate if anyone could let me know if we get the following guide in the exam. Cisco CallManager Features and Services Guide Cheers S
[OSL | CCIE_Voice] Fax on/off ramp with FXS into Unity
G'Day, Scenario - Microsoft Fax server, into and FXS port, via various different modems. Goal faxing into unity. Not the easiest set of items to troubleshoot. Going through tcl on-ramp script it looks like it should function, but should and working are oceans apart in this case. Out of interest has anyone ever got the above scenario working, with analog modem into FXS? Given it the one day limit, now moving on. Cheers S
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
Re: [OSL | CCIE_Voice] Lab exam CCM Features and services PDF
You can access that guide directly from the HELP of the Callmanager -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Stephen Collinson Enviado el: 10 June 2008 07:05 Para: OSL CCIE Voice Lab Exam Asunto: [OSL | CCIE_Voice] Lab exam CCM Features and services PDF Hi, Would appreciate if anyone could let me know if we get the following guide in the exam. Cisco CallManager Features and Services Guide Cheers S
Re: [OSL | CCIE_Voice] POTS Dial-peer
119002001 On Jun 10, 2008, at 7:13 AM, Balamurugan Singaram [EMAIL PROTECTED] wrote: 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] Problem Configuring IPIPGW
Hi, I am having problem configuring IPIPGW: *HQ phone (SCCP, connected to CM) *BR2 phone (SCCP, connected to CME) HQ phone can call BR2 phone. BR2 phone cannot call HQ phone.. Two Trunks are configured on CM are pointing to HQ router[IPIPGW]: *ICT trunk (H.323 non gatekeeper controlled) *SIP trunk I already configured a route pattern on CM to point to CME using ICT[H.323]. I am not sure if I need to configure same route pattern to use SIP trunk. Please advise, AH
Re: [OSL | CCIE_Voice] Gatekeeper 1st call g711 2nd call g729 - works but not as expected
You might want to uncheck the Media Transcoder required check box and try that way. What I found out that if you do not have hardware transcoder in your MRGL then the calls come in from CME over Gatekeeper to IPCC do not work. On Fri, May 30, 2008 at 10:50 AM, Rimon Vallavanatt Jr. [EMAIL PROTECTED] wrote: I have setup the following: Configured two regions on the CCM, one that talks G.711 to everything else and one that talks G.729 to everything else. Created two DP, GK-711 and GK-729 with their respective regions. I registered the GK in call manager and then created two trunks. One using the GK-G711-DP and another using GK-G729-DP. Then created one route group with both trunks using top down distribution with GK-711-Trunk first and GK-G729-Trunk second. Created a RL and RP to point to the Route Group. I set BRQ to true on the CUCMs. I've also tried it with two RGs. I've tried it with the voice class codec and with two different dial-peers, one with 711, one with 729. It works just fine. The gatekeeper shows one call 711 one call 729. The phones on the CME , if I hit the ? button show what I would expect. The problem is that at the HQ site the phones both show 711 when I hit the ? button. I verified that the 729 stream is being transcoded to g711. My question is why? Thanks, *Rimon Vallavanatt Jr.* *Director, Installations* Phone:713.881.7133 Fax:713.881.7233
[OSL | CCIE_Voice] Cisco CUE Module
Hi Guys Struggling on something, the Cue Module. I setup the CME with all the setting required including the web admin system name admin password cisco and have reset the cisco CUE to factory defaults setting the new account details to admin and cisco in CUE. When I get to the first setup page on the web page I get to login where it says first time only and then when trying to put in the username and password to integrate with CME I keep getting error to check username. I cant seem to get the two to synch up. Does anyone have any ideas on this? Thanks Paul
Re: [OSL | CCIE_Voice] calling out to FXS port via SIP trunk - no caller-id?
Can you please try: Sip-ua Remote-party-id I am curious if it works. On Jun 9, 2008, at 11:11 PM, Jane Ryer (jryer) [EMAIL PROTECTED] wrote: I set up a SIP trunk from Call Manager to a router with an FXS port. When I call from the analog phone attached to the FXS port to an IP Blue phone registered to Call Manager, I do see the name and number for the FXS port (as set via station-id commands on the voice- port for the FXS port). However, if I call out from the IP Blue phone to the analog phone, all I see on the IP Blue phone is the number I dialed (4001) – no name. Is this to be expected with SIP t runks? Here is the relevant portion of my router config: voice-port 0/2/1 station-id name Analog Phone station-id number 2122214001 caller-id enable (not sure whether this accomplished anything or not – didn’t work differently with or without it) ! dial-peer voice 4000 voip session protocol sipv2 session target ipv4:10.x.x.x (IP address of my CCM) incoming called-number 4... dtmf-relay rtp-nte (just realized that I put this command on this dial peer but not the one to CCM) codec g711ulaw no vad ! dial-peer voice 4001 pots destination-pattern 4001 port 0/2/1 ! dial-peer voice 1000 voip destination-pattern 1... session protocol sipv2 session target ipv4:10.x.x.x (IP address of my CCM) codec g711ulaw no vad ! Any insight would be appreciated. Is this supposed to work or not? Is it just a limitation of SIP? Or am I missing some configuration that is needed to pass the called name back? Thanks,
Re: [OSL | CCIE_Voice] calling out to FXS port via SIP trunk - nocaller-id?
Hey, Onur, I will try it tomorrow when I go in to the office and post results here. Jane From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Onur Tufekci Sent: Tuesday, June 10, 2008 4:08 PM To: OSL CCIE Voice Lab Exam Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] calling out to FXS port via SIP trunk - nocaller-id? Can you please try: Sip-ua Remote-party-id I am curious if it works. On Jun 9, 2008, at 11:11 PM, Jane Ryer (jryer) [EMAIL PROTECTED] wrote: I set up a SIP trunk from Call Manager to a router with an FXS port. When I call from the analog phone attached to the FXS port to an IP Blue phone registered to Call Manager, I do see the name and number for the FXS port (as set via station-id commands on the voice-port for the FXS port). However, if I call out from the IP Blue phone to the analog phone, all I see on the IP Blue phone is the number I dialed (4001) - no name. Is this to be expected with SIP trunks? Here is the relevant portion of my router config: voice-port 0/2/1 station-id name Analog Phone station-id number 2122214001 caller-id enable (not sure whether this accomplished anything or not - didn't work differently with or without it) ! dial-peer voice 4000 voip session protocol sipv2 session target ipv4:10.x.x.x (IP address of my CCM) incoming called-number 4... dtmf-relay rtp-nte (just realized that I put this command on this dial peer but not the one to CCM) codec g711ulaw no vad ! dial-peer voice 4001 pots destination-pattern 4001 port 0/2/1 ! dial-peer voice 1000 voip destination-pattern 1... session protocol sipv2 session target ipv4:10.x.x.x (IP address of my CCM) codec g711ulaw no vad ! Any insight would be appreciated. Is this supposed to work or not? Is it just a limitation of SIP? Or am I missing some configuration that is needed to pass the called name back? Thanks,
[OSL | CCIE_Voice] gatekeeper routing
hi all, If the gatekeeper finds the destination zone for a call is in zone A, does that mean that - using default tech-prefix routing - it has to find a GW in the same destination zone that's registered with the default tech-prefix? Or can it be any gateway, even in another zone, say zone B, as long as it's registered with the default tech prefix? cheers, Juan
Re: [OSL | CCIE_Voice] gatekeeper routing
It has to be in zone A. Vik Malhi - CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: mailto:[EMAIL PROTECTED] [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. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Juan Sent: Tuesday, June 10, 2008 3:50 PM To: 'OSL CCIE Voice Lab Exam' Subject: [OSL | CCIE_Voice] gatekeeper routing hi all, If the gatekeeper finds the destination zone for a call is in zone A, does that mean that - using default tech-prefix routing - it has to find a GW in the same destination zone that's registered with the default tech-prefix? Or can it be any gateway, even in another zone, say zone B, as long as it's registered with the default tech prefix? cheers, Juan
Re: [OSL | CCIE_Voice] calling out to FXS port via SIP trunk -nocaller-id?
I persevered a while back and never got Calling Name to function from CCM to CME via the SIP trunk. So at the moment it appears that this is the way it works unless somebody has any further insight. BTW- I expect the CME phone to function in the same way as an FXS portsee below for the reason why. P27-BR2-RTR#sh telephony-service voice-port voice-port 50/0/1 station-id number 3001 station-id name br2 phn2 timeout ringing 12 ! P27-BR2-RTR#sh telephony-service dial-p dial-peer voice 20001 pots destination-pattern 3001$ huntstop progress_ind setup enable 3 port 50/0/1 Vik Malhi - CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: mailto:[EMAIL PROTECTED] [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. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jane Ryer (jryer) Sent: Tuesday, June 10, 2008 3:26 PM To: OSL CCIE Voice Lab Exam Subject: Re: [OSL | CCIE_Voice] calling out to FXS port via SIP trunk -nocaller-id? Hey, Onur, I will try it tomorrow when I go in to the office and post results here. Jane _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Onur Tufekci Sent: Tuesday, June 10, 2008 4:08 PM To: OSL CCIE Voice Lab Exam Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] calling out to FXS port via SIP trunk - nocaller-id? Can you please try: Sip-ua Remote-party-id I am curious if it works. On Jun 9, 2008, at 11:11 PM, Jane Ryer (jryer) [EMAIL PROTECTED] wrote: I set up a SIP trunk from Call Manager to a router with an FXS port. When I call from the analog phone attached to the FXS port to an IP Blue phone registered to Call Manager, I do see the name and number for the FXS port (as set via station-id commands on the voice-port for the FXS port). However, if I call out from the IP Blue phone to the analog phone, all I see on the IP Blue phone is the number I dialed (4001) - no name. Is this to be expected with SIP trunks? Here is the relevant portion of my router config: voice-port 0/2/1 station-id name Analog Phone station-id number 2122214001 caller-id enable (not sure whether this accomplished anything or not - didn't work differently with or without it) ! dial-peer voice 4000 voip session protocol sipv2 session target ipv4:10.x.x.x (IP address of my CCM) incoming called-number 4... dtmf-relay rtp-nte (just realized that I put this command on this dial peer but not the one to CCM) codec g711ulaw no vad ! dial-peer voice 4001 pots destination-pattern 4001 port 0/2/1 ! dial-peer voice 1000 voip destination-pattern 1... session protocol sipv2 session target ipv4:10.x.x.x (IP address of my CCM) codec g711ulaw no vad ! Any insight would be appreciated. Is this supposed to work or not? Is it just a limitation of SIP? Or am I missing some configuration that is needed to pass the called name back? Thanks,
Re: [OSL | CCIE_Voice] PRI into 6500 in real world
Where does one get the XXTracy tool? On Mon, Jun 9, 2008 at 3:20 PM, Rick Grimes [EMAIL PROTECTED] wrote: You can also just browse to the ip address of the registered gateway and look at the port statistics. Make a call inbound from the pstn and make an outbound call. The statistics with increment on the line the call was made from or coming into. -Rick From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David L. Blair Sent: Sunday, June 08, 2008 8:15 AM To: OSL CCIE Voice Lab Exam Subject: Re: [OSL | CCIE_Voice] PRI into 6500 in real world You can also use the Dick Tracy tool or newer XXTracy tool. David - Original Message - From: Nguyen Le To: OSL CCIE Voice Lab Exam Sent: Saturday, June 07, 2008 10:08 PM Subject: Re: [OSL | CCIE_Voice] PRI into 6500 in real world You can use RTMT and monitor the PRI. It'll show you real time which channel Telco is sending on. Nguyen On Sat, Jun 7, 2008 at 8:56 PM, Daniel Dellinger [EMAIL PROTECTED] wrote: In a real word scenario on a 6608 blade, how do you determine if telco is sending calls down PRI in bottom up or top down channel fashion? I.e. what is the command to determine this? thanks E-mail Disclaimer: Any views expressed in this message may be those of the individual sender, if the sender has not specifically stated that he/she is acting within his/her authority on behalf of or specifically states them to be the views of CSI Technology Outfitters. The information in this email may be confidential, and is intended only for the use of or dissemination by the recipient named above. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution, or copying of this communication, or any of its contents, is strictly prohibited. If you have received this communication in error, please re-send this communication to the sender and delete the original message and any copy of it from your computer system. To report any abuse, please forward to [EMAIL PROTECTED]
