[OSL | CCIE_Voice] QoS command is not available
Hi, I am trying to configure polide dscp map on the BR1 router with ESW. The command mls qos map policed-dscp 24 to 10 is not available. Any ideas? Thanks, Onur.
Re: [OSL | CCIE_Voice] IOS Conf bdg at BR1
Hi are you using mtp and cfb to register? Also issue the command dspfarm in the config mode as well. Also do not forget to disable vad for codec g729 on the router. ( this has nothing to do with your stuation just missing) On Tue, May 12, 2009 at 11:18 AM, Nate Paschua nate...@gmail.com wrote: Hi, I am trying to setup IOS Conf bdg at BR1. However it is not registering to CCM status says unknown Do you have any suggestions how to fix this? Here is what i have done 1) Router config voice-card 2 dspfarm dsp services dspfarm ! ! sccp local Vlan107 sccp sccp ccm 142.7.64.12 priority 1 sccp ccm 142.7.64.11 priority 2 ! dspfarm transcoder maximum sessions 4 dspfarm confbridge maximum sessions 4 ! 2) ON CCM. use Mac add from vlan107 create IOS CONF BDG. Router is just NM-HDV Thanks Nate
[OSL | CCIE_Voice] QoS Policer!!!
One of the take it as you like questions in the lab probably is the aggrageta and microflow policers. In the OWLE lab3 question 40 is asking to police EACH SCCP phone should have the. each is the key word for microflow to me. However, in the solution guide eventhough microflow policer is created it is applied to the fast ethernet 0/1 in output direction. Isn't this supposed to be applied to individual port inward direction? Thank you for your help in advance. Onur.
Re: [OSL | CCIE_Voice] QoS Policer!!!
I think I found the answer for my own question: http://www.cisco.com/en/US/products/hw/switches/ps700/products_tech_note09186a00801c8c4b.shtml On Sun, May 10, 2009 at 12:58 PM, Onur Tufekci onurvc...@gmail.com wrote: One of the take it as you like questions in the lab probably is the aggrageta and microflow policers. In the OWLE lab3 question 40 is asking to police EACH SCCP phone should have the. each is the key word for microflow to me. However, in the solution guide eventhough microflow policer is created it is applied to the fast ethernet 0/1 in output direction. Isn't this supposed to be applied to individual port inward direction? Thank you for your help in advance. Onur.
[OSL | CCIE_Voice] SCCP FXS
I was looking around for an answer for a while. Only one that I was able to find is STCAPP to register FXS port as SCCP endpoint. This is only available IOS T train. I was not able to configure this on the IPEXPERT racks since the IOS is not T series plus there is no FXS ports accessable. On the cisco V2 IOS version it only says Main Tarin release. I am not sure if they have T Train. So I am looking for a way of configuring FXS port with SCCP. Any help will greatly be appreciated. Thanks, Onur.
Re: [OSL | CCIE_Voice] SCCP FXS
Hi! As you can see in my email I mention IPEXPERT labs so I do not believe it is work related. So I do not think I should be going to another web site (even I did). It is not required at the IPEXPERT study guides but it might be in the exam. So I am looking to see anybody has configured it since I do not have access to FXS por and CCM at the same time other then these labs. I hope this helps! On Sun, May 10, 2009 at 3:19 PM, ccieid1ot ccieid...@gmail.com wrote: If it's not requirer/supported why do you want to go through the hassle? If this is for work related, you should ask the other cisco voip email list. On Sun, May 10, 2009 at 1:29 PM, Onur Tufekci onurvc...@gmail.com wrote: I was looking around for an answer for a while. Only one that I was able to find is STCAPP to register FXS port as SCCP endpoint. This is only available IOS T train. I was not able to configure this on the IPEXPERT racks since the IOS is not T series plus there is no FXS ports accessable. On the cisco V2 IOS version it only says Main Tarin release. I am not sure if they have T Train. So I am looking for a way of configuring FXS port with SCCP. Any help will greatly be appreciated. Thanks, Onur.
Re: [OSL | CCIE_Voice] SCCP FXS
Ok! Thanks for the clarification. I knew the SIp one but just wanted make sure about the SCCP portion. On Sun, May 10, 2009 at 4:10 PM, Cliff McGlamry cl...@mcglamry.net wrote: You will not need to do this for the lab. As such, it doesn't matter whether or not it's possiblewhich it isn't with the IOS that you will have available. You will need to know how to set the FXS port up and connect it back to CCM via a SIP trunk though. *From:* Onur Tufekci onurvc...@gmail.com *Sent:* Sunday, May 10, 2009 2:29 PM *To:* OSL Group ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] SCCP FXS I was looking around for an answer for a while. Only one that I was able to find is STCAPP to register FXS port as SCCP endpoint. This is only available IOS T train. I was not able to configure this on the IPEXPERT racks since the IOS is not T series plus there is no FXS ports accessable. On the cisco V2 IOS version it only says Main Tarin release. I am not sure if they have T Train. So I am looking for a way of configuring FXS port with SCCP. Any help will greatly be appreciated. Thanks, Onur.
Re: [OSL | CCIE_Voice] Current lab IOS version + preserve PSTN calls when fallback to SRST on MGCP PRI GW = no way !!?
That is what I figured out! On Sun, May 10, 2009 at 4:18 PM, jeremy co jeremy.coo...@gmail.com wrote: Hi, I know h323 call preserve is possible, and also MGCP GW with CAS. With current lab IOS version and preserve PSTN calls when fallback to SRST on MGCP PRI GW scenario , it has no solution. So please correct me if I'm wrong. Jeremy
Re: [OSL | CCIE_Voice] Current lab IOS version + preserve PSTN calls when fallback to SRST on MGCP PRI GW = no way !!?
Here is an interesting link talking about call preservation! On Sun, May 10, 2009 at 4:18 PM, jeremy co jeremy.coo...@gmail.com wrote: Hi, I know h323 call preserve is possible, and also MGCP GW with CAS. With current lab IOS version and preserve PSTN calls when fallback to SRST on MGCP PRI GW scenario , it has no solution. So please correct me if I'm wrong. Jeremy
Re: [OSL | CCIE_Voice] Current lab IOS version + preserve PSTN calls when fallback to SRST on MGCP PRI GW = no way !!?
Uppss! http://www.networkworld.com/community/node/29909 On Sun, May 10, 2009 at 5:52 PM, Onur Tufekci onurvc...@gmail.com wrote: Here is an interesting link talking about call preservation! On Sun, May 10, 2009 at 4:18 PM, jeremy co jeremy.coo...@gmail.comwrote: Hi, I know h323 call preserve is possible, and also MGCP GW with CAS. With current lab IOS version and preserve PSTN calls when fallback to SRST on MGCP PRI GW scenario , it has no solution. So please correct me if I'm wrong. Jeremy
Re: [OSL | CCIE_Voice] moh multicast via h323 to PSTN not woorking via mgcp work fine
DId you put ccm-manager music-on-hold command? also make sure that you have the g711 assiged to MOH and under call-manager-fallback you got the routes with multicast moh command. On Sat, May 9, 2009 at 10:47 AM, zamuel del Toro sdelto...@hotmail.comwrote: I have configured the gateway via h323 to psnt and the pstn phone side not heard the moh audio. via mgcp work fine. any ideas? -- Get 5 GB of storage with Windows Live Hotmail. Sign up today.http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_5gb_112008
Re: [OSL | CCIE_Voice] moh multicast via h323 to PSTN not woorking via mgcp work fine
DO you see an output similar to this? You should do this when you put the call on hold that is coming from PSTN phone. PodX-BR1-RTR#show ccm-manager music-on-hold Current active multicast sessions : 1 Multicast RTP port Packets Call CodecIncoming Address number in/outid Interface === 239.1.1.1 16384 129/129 41 g711ulaw Lo0 On Sat, May 9, 2009 at 12:51 PM, Onur Tufekci onurvc...@gmail.com wrote: Create a regioun for g711 only Create a device pool and assign g711 region to this pool under services/media resource/ assign this region to one of the MOH server that is also set up for multicast. That is it. On Sat, May 9, 2009 at 12:47 PM, Sowmyashree Mahadevaiah sowmyashr...@gmail.com wrote: Hi Onur I am in a similar situation. How do we ensure g711u under call-manager-fallback? The one solution i heard was to have region in ccm with g711 put moh server under that. sowmya On Sat, May 9, 2009 at 9:40 AM, Onur Tufekci onurvc...@gmail.comwrote: DId you put ccm-manager music-on-hold command? also make sure that you have the g711 assiged to MOH and under call-manager-fallback you got the routes with multicast moh command. On Sat, May 9, 2009 at 10:47 AM, zamuel del Toro sdelto...@hotmail.com wrote: I have configured the gateway via h323 to psnt and the pstn phone side not heard the moh audio. via mgcp work fine. any ideas? -- Get 5 GB of storage with Windows Live Hotmail. Sign up today.http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_5gb_112008
Re: [OSL | CCIE_Voice] h323 Call Preserver
I got these settings and the h323 config attached! Allow TCP KeepAlives For H323True 10.8.200.20False 10.8.200.21False Allow Peer to Preserve H.323 CallsFalse 10.8.200.20True 10.8.200.21True H323-GW Description: Binary data
[OSL | CCIE_Voice] Fwd: h323 Call Preserver
-- Forwarded message -- From: Onur Tufekci onurvc...@gmail.com Date: Fri, May 8, 2009 at 10:30 AM Subject: Re: [OSL | CCIE_Voice] h323 Call Preserver To: kill mill jha...@gmail.com LOL!! You would hope! On Fri, May 8, 2009 at 10:25 AM, kill mill jha...@gmail.com wrote: How are u testing the failover. I hope you are not shutting the ccm service since that would send a reset to the gw and the gw will drop the call. On Fri, May 8, 2009 at 9:17 AM, Onur Tufekci onurvc...@gmail.com wrote: I got these settings and the h323 config attached! Allow TCP KeepAlives For H323True 10.8.200.20False 10.8.200.21False Allow Peer to Preserve H.323 CallsFalse 10.8.200.20True 10.8.200.21True
[OSL | CCIE_Voice] h323 Call Preserver
I been trying the call preserve trick everytime I set up H323 gateway. Out of 10 I got it to work only once. Is there any trick to this?
Re: [OSL | CCIE_Voice] how to change callwaiting duration?
Do you mean no answer duration? On Tue, May 5, 2009 at 4:33 PM, jeremy co jeremy.coo...@gmail.com wrote: Hi, anybody knows how to change callwaiting duration? I couldn't find it under CCM parameters Jeremy
Re: [OSL | CCIE_Voice] BACD Calls busy from CME
Thank you! It can not be easy can it? :) On Mon, May 4, 2009 at 2:39 PM, Cyrus cyrus@gmail.com wrote: Hi, CME will send an ARQ msg to GK even it wouldn't match RAS dialpeer (so if u have one zone and bandwidth restriction on that zone under 128K,u are in trouble!), so for internal phones works u need at least 128k BW on your GK zone. if calls arrive on G729 , u need extra 16K on Gk to provision. There is no known workaround for this yet. Cyrus On Tue, May 5, 2009 at 3:52 AM, Onur Tufekci onurvc...@gmail.com wrote: Hi, I am puzzeled with this scenario: BACD is set on CME. There is a CME phone dialing AA pilot hearing fast busy. If CME is unregistered from Gatekeeper then it works. I know you need to get bandwidth check if you are using your loopback interface. How is it possible to get it to work with when registered to Gatekeeper? Any ideas? Onur. -- Sirus Moghadasian CCIE #21862 (RS)
Re: [OSL | CCIE_Voice] CCM to CME calls keeps ringing
Looks like you just need to change it for the IPIPGW. On Wed, Apr 29, 2009 at 6:42 PM, Onur Tufekci onurvc...@gmail.com wrote: WOW That is scary. VIK you are the man. I never had that problem before. So do we have to uncheckWait for Far End H.245 Terminal Capability Set for all the GK trunks or just IPIPGW? On Wed, Apr 29, 2009 at 1:51 PM, Vik Malhi vma...@ipexpert.com wrote: Also look into the trunk settings- Wait for H245 TCS should not be checked. -- 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/communitieshttp://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: *Cliff McGlamry cl...@mcglamry.net *Date: *Tue, 28 Apr 2009 23:29:53 -0400 *To: *Onur Tufekci onurvc...@gmail.com, OSL Group ccie_voice@onlinestudylist.com *Subject: *Re: [OSL | CCIE_Voice] CCM to CME calls keeps ringing Your issue is probably not on the Gatekeeper. Your configuration looks okay as far as I can tell. Do a debug dial peer on the CME router and see which dial peer you are coming in on. My guess is that you're coming in on a dial peer that doesn't have the codec defined correctly, or possibly coming in on the default dial peer (which is always a not so good thing to have happen). I'm betting that the issue is the dial peer you're hitting inbound on CME is either the wrong dial peer, or it's misconfigured. - Original Message - *From:* Onur Tufekci mailto:onurvc...@gmail.com onurvc...@gmail.com *To:* ccie_voice@onlinestudylist.com *Sent:* Tuesday, April 28, 2009 11:13 PM *Subject:* [OSL | CCIE_Voice] CCM to CME calls keeps ringing I have the IPIPGW configured on the HQ Router. Calls from CME to CCM are successful but other way is not. CCM phone just keeps ringing even after picking up the call at CME phone. Is this even a valid configuration?
Re: [OSL | CCIE_Voice] CCM to CME calls keeps ringing
Umm! I will check that too. Everything was in default setting when I set the trunk up. On Wed, Apr 29, 2009 at 1:51 PM, Vik Malhi vma...@ipexpert.com wrote: Also look into the trunk settings- Wait for H245 TCS should not be checked. -- 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 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: *Cliff McGlamry cl...@mcglamry.net *Date: *Tue, 28 Apr 2009 23:29:53 -0400 *To: *Onur Tufekci onurvc...@gmail.com, OSL Group ccie_voice@onlinestudylist.com *Subject: *Re: [OSL | CCIE_Voice] CCM to CME calls keeps ringing Your issue is probably not on the Gatekeeper. Your configuration looks okay as far as I can tell. Do a debug dial peer on the CME router and see which dial peer you are coming in on. My guess is that you're coming in on a dial peer that doesn't have the codec defined correctly, or possibly coming in on the default dial peer (which is always a not so good thing to have happen). I'm betting that the issue is the dial peer you're hitting inbound on CME is either the wrong dial peer, or it's misconfigured. - Original Message - *From:* Onur Tufekci mailto:onurvc...@gmail.com onurvc...@gmail.com *To:* ccie_voice@onlinestudylist.com *Sent:* Tuesday, April 28, 2009 11:13 PM *Subject:* [OSL | CCIE_Voice] CCM to CME calls keeps ringing I have the IPIPGW configured on the HQ Router. Calls from CME to CCM are successful but other way is not. CCM phone just keeps ringing even after picking up the call at CME phone. Is this even a valid configuration?
Re: [OSL | CCIE_Voice] CCM to CME calls keeps ringing
WOW That is scary. VIK you are the man. I never had that problem before. So do we have to uncheckWait for Far End H.245 Terminal Capability Set for all the GK trunks or just IPIPGW? On Wed, Apr 29, 2009 at 1:51 PM, Vik Malhi vma...@ipexpert.com wrote: Also look into the trunk settings- Wait for H245 TCS should not be checked. -- 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 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: *Cliff McGlamry cl...@mcglamry.net *Date: *Tue, 28 Apr 2009 23:29:53 -0400 *To: *Onur Tufekci onurvc...@gmail.com, OSL Group ccie_voice@onlinestudylist.com *Subject: *Re: [OSL | CCIE_Voice] CCM to CME calls keeps ringing Your issue is probably not on the Gatekeeper. Your configuration looks okay as far as I can tell. Do a debug dial peer on the CME router and see which dial peer you are coming in on. My guess is that you're coming in on a dial peer that doesn't have the codec defined correctly, or possibly coming in on the default dial peer (which is always a not so good thing to have happen). I'm betting that the issue is the dial peer you're hitting inbound on CME is either the wrong dial peer, or it's misconfigured. - Original Message - *From:* Onur Tufekci mailto:onurvc...@gmail.com onurvc...@gmail.com *To:* ccie_voice@onlinestudylist.com *Sent:* Tuesday, April 28, 2009 11:13 PM *Subject:* [OSL | CCIE_Voice] CCM to CME calls keeps ringing I have the IPIPGW configured on the HQ Router. Calls from CME to CCM are successful but other way is not. CCM phone just keeps ringing even after picking up the call at CME phone. Is this even a valid configuration?
[OSL | CCIE_Voice] CCM to CME calls keeps ringing
I have the IPIPGW configured on the HQ Router. Calls from CME to CCM are successful but other way is not. CCM phone just keeps ringing even after picking up the call at CME phone. Is this even a valid configuration? HQ-RTR Description: Binary data
Re: [OSL | CCIE_Voice] IPPA Agent state ready auto
What is the paramater that you use other then ID Ext Pwd? Is it state? On Sun, Apr 5, 2009 at 3:14 AM, Arshad Dhunna arshad.dhu...@yahoo.co.inwrote: Yes it is possible in service parameter in IPCC And it is possible in CCM with the Url IPAgentLogin.jsp --- On *Sun, 5/4/09, Onur Tufekci onurvc...@gmail.com* wrote: From: Onur Tufekci onurvc...@gmail.com Subject: Re: [OSL | CCIE_Voice] IPPA Agent state ready auto To: Duy Nguyen ccieid...@gmail.com Cc: ccie_voice@onlinestudylist.com Date: Sunday, 5 April, 2009, 12:32 AM All paramaters are Auto work or Auto Available there, after an agent completes a call system puts the agent in Ready mode. There must be a file that has the parmater ready / not ready. Like the one in Extension mobility! doLogout-true !!! On Sat, Apr 4, 2009 at 2:55 PM, Duy Nguyen ccieid...@gmail.comhttp://in.mc951.mail.yahoo.com/mc/compose?to=ccieid...@gmail.com wrote: I believe you can. Go into Resource group in CRS if I'm not mistaken. On Sat, Apr 4, 2009 at 12:51 PM, Onur Tufekci onurvc...@gmail.comhttp://in.mc951.mail.yahoo.com/mc/compose?to=onurvc...@gmail.com wrote: Hi All, Is it possible to put the agents in the ready state automatically right after one button login? Thank you in advance, Onur. -- Get your own website and domain for just Rs.1,999/year.* Click here!http://in.rd.yahoo.com/tagline_ysb_3/*http://in.business.yahoo.com/
[OSL | CCIE_Voice] IPPA Agent state ready auto
Hi All, Is it possible to put the agents in the ready state automatically right after one button login? Thank you in advance, Onur.
Re: [OSL | CCIE_Voice] IPPA Agent state ready auto
All paramaters are Auto work or Auto Available there, after an agent completes a call system puts the agent in Ready mode. There must be a file that has the parmater ready / not ready. Like the one in Extension mobility! doLogout-true !!! On Sat, Apr 4, 2009 at 2:55 PM, Duy Nguyen ccieid...@gmail.com wrote: I believe you can. Go into Resource group in CRS if I'm not mistaken. On Sat, Apr 4, 2009 at 12:51 PM, Onur Tufekci onurvc...@gmail.com wrote: Hi All, Is it possible to put the agents in the ready state automatically right after one button login? Thank you in advance, Onur.
[OSL | CCIE_Voice] IPCCX prompt
Hi, I am not a regular here but I have a question that I can not find answer to. Anybody knows how to change default message duration for IPCCX prompts? Thank you in advance, Onur.
Re: [OSL | CCIE_Voice] Multiple Cisco IP Communicator
Yes that is correct you can not run multiple instances of IPC on your one machine unless you have virtual machines running. On Mon, Oct 20, 2008 at 10:53 AM, KIZILCABOLUK DENIZ [EMAIL PROTECTED] wrote: Hi, How can I open multiple IP Communicator on my laptop? Do you have an idea? Thanks, Deniz
Re: [OSL | CCIE_Voice] [OSL | CCIE_RS] a few special offers on Boot Camps, products and rack time
It was a long break for me too. I think it is a good time to get back to studying!! On Fri, Oct 10, 2008 at 2:46 PM, Matt Brooks at IPexpert [EMAIL PROTECTED] wrote: Get to work on your VOICE prep, Jo... no slacking! ;) - Matt On Fri, Oct 10, 2008 at 2:43 PM, Jo Knight [EMAIL PROTECTED] wrote: No study for me - passed on Monday :) A nice relaxing weekend for me - sorry guys! Jo #22262 Matt Brooks at IPexpert wrote: Here are some great offers, not advertised on our website, good for one week from today! http://www.ipexpert.com/index.cfm/a/p/social_networks Have a great weekend (of studying)! :) -- Matt Brooks Vice President - IPexpert, Inc. Telephone: +1.810.326.1444 x101 Cell: +1.810.434.7447 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Skype: IPexpert_Matt LinkedIn: http://www.linkedin.com/in/matthewbrooks -- Follow IPexpert at Twitter.com/IPexpert -- 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. -- -- Matt Brooks Vice President - IPexpert, Inc. Telephone: +1.810.326.1444 x101 Cell: +1.810.434.7447 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] Skype: IPexpert_Matt LinkedIn: http://www.linkedin.com/in/matthewbrooks -- Follow IPexpert at Twitter.com/IPexpert -- 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. --
Re: [OSL | CCIE_Voice] BACD
Also you have to have param voicemail otherwise script wont work. On Sat, Sep 27, 2008 at 1:13 AM, Paul and Bobs [EMAIL PROTECTED] wrote: Hi Here is my BACD config I am not getting it to load up. I have rebooted but still no luck. When I issue the command sho call application sessions i get nothing. When i issue the command call application voice load queue and the the one above still get nothing. application service queue flash:app-b-acd-2.0.0.0.tcl param queue-len 10 param aa-hunt3 4100 param aa-hunt4 4200 param number-of-hunt-grps 2 param aa-name aa param queue-manager-debugs 1 ! service aa flash:app-b-acd-aa-2.0.0.0.tcl paramspace english index 1 param number-of-hunt-grps 2 param dial-by-extension-option 5 paramspace english language en param max-time-vm-retry 2 param aa-pilot 3000 paramspace english location flash: param second-greeting-time 60 param call-retry-timer 15 paramspace english prefix en param max-time-call-retry 600 param service-name queue ! dial-peer voice 501 pots service aa incoming called-number 3000 direct-inward-dial port 0/3/0:15
Re: [OSL | CCIE_Voice] Nailed it down
Congratulations. On Mon, Jul 14, 2008 at 9:14 PM, ovais Iqbal [EMAIL PROTECTED] wrote: Dear All, Very glad to announce my Voice CCIE # 21482, got it today in 3rd attempt. Thanks every one for great support through out the study process, special thanks goes to Vik Malhi and ip Expert team. Once again thanks. -- Ovais Iqbal 416-294-7869
Re: [OSL | CCIE_Voice] Slow busy IPCCX
Thanks for the reply. It was hardware transcoder. On Sun, Jul 13, 2008 at 6:45 PM, Jonathan Charles [EMAIL PROTECTED] wrote: Well, first are your CTI RPs registered? Is JTAPI in service? What do the MIVR logs say? Jonathan On Sun, Jul 13, 2008 at 2:06 PM, Mehmet Tufekci [EMAIL PROTECTED] wrote: Anybody has any idea about why IPCCX would give slow busy signal. I can not see anything wrong. Any guidance will be appreciated.
Re: [OSL | CCIE_Voice] Conference tone
Thanks :) On Wed, Jul 9, 2008 at 5:20 AM, Ante Boras [EMAIL PROTECTED] wrote: I found the solution: *Party Entrance Tone* to False in CallManager service parameters Ante *Ante Boras [EMAIL PROTECTED]* Sent by: [EMAIL PROTECTED] 09.07.2008 11:10 To ccie_voice@onlinestudylist.com cc Subject Re: [OSL | CCIE_Voice] Conference tone Question: A tone must not be heard when users enter or leave a Callmanager controlled conference. answer? ante
[OSL | CCIE_Voice] Good read for Call Transfers / Forwards in CME
http://www.ciscopress.com/articles/article.asp?p=401648seqNum=9
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 29, Issue 9
What is your dial-peer for RAS has as far as codec setting? Do you have Requires Media Termination Point checked under trunk configuration? On Tue, Jul 8, 2008 at 10:25 AM, Kumar, Narinder [EMAIL PROTECTED] wrote: Setup CCM ---GK Controlled TR to Gatekeeper ---Gatekeeper - CME Both CME and CCM registered to Gatekeeper. Calls working both ways without any issues. When I call from CME to CCM the GK shows BW 128K which I was expecting When Call from CCM to CME the BW is 16K. I haven't configured g729 in my system at all. How I am seeing 16K ?? Any idea. Thanks NK -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, 7 July 2008 12:26 PM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 29, Issue 9 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. CUE QoS (Vol 1 Lab 15 task 5) (Abdalla Abdalla) 2. Lab 23.10 GK (Chuck) 3. IPPA Service Login (Nguyen Le) 4. Re: IPPA Service Login (Jonathan Charles) 5. Re: IPPA Service Login (Nguyen Le) 6. Re: IPPA Service Login (Jonathan Charles) 7. Re: 0 Conf max sessions (Vik Malhi) 8. Re: IPPA Service Login (Derrick Shumake) -- Message: 1 Date: Sun, 6 Jul 2008 14:06:43 -0700 (PDT) From: Abdalla Abdalla [EMAIL PROTECTED] Subject: [OSL | CCIE_Voice] CUE QoS (Vol 1 Lab 15 task 5) To: CCIE Voice StudyList ccie_voice@onlinestudylist.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Hi, I tried to apply? a service policy?to the service engine in the input direction but i get the error message that the service-ploicy can only be applied in the output direction. Any ideas why this is so?. Whereas the proctor guide solution shows that it can be applied in the input direction. regards AA -- next part -- An HTML attachment was scrubbed... URL: http://onlinestudylist.com/pipermail/ccie_voice/attachments/20080706/5d7 e6d17/attachment-0001.html -- Message: 2 Date: Sun, 6 Jul 2008 16:05:42 -0700 From: Chuck [EMAIL PROTECTED] Subject: [OSL | CCIE_Voice] Lab 23.10 GK To: OSL CCIE Voice Lab Exam ccie_voice@onlinestudylist.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 I don't understand how the call can be routed out the IPIPGW without a alias static command on the gatekeeper config. Shouldn't the HQ-RTR be registered to the GK (itself) as well? thanks! -- Message: 3 Date: Sun, 6 Jul 2008 21:20:59 -0500 From: Nguyen Le [EMAIL PROTECTED] Subject: [OSL | CCIE_Voice] IPPA Service Login To: CCIE Voice ccie_voice@onlinestudylist.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 On the IPPA Service. What would cause it to come up with this error. Unable to connecto to the IPPA service this is when you are trying to login via your phone. Thanks Nguyen -- next part -- An HTML attachment was scrubbed... URL: http://onlinestudylist.com/pipermail/ccie_voice/attachments/20080706/86c 11177/attachment-0001.html -- Message: 4 Date: Sun, 6 Jul 2008 21:49:47 -0500 From: Jonathan Charles [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] IPPA Service Login To: Nguyen Le [EMAIL PROTECTED] Cc: CCIE Voice ccie_voice@onlinestudylist.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 Check that you have communication to the IPCC server (aka the Pub...) Jonathan On Sun, Jul 6, 2008 at 9:20 PM, Nguyen Le [EMAIL PROTECTED] wrote: On the IPPA Service. What would cause it to come up with this error. Unable to connecto to the IPPA service this is when you are trying to login via your phone. Thanks Nguyen -- Message: 5 Date: Sun, 6 Jul 2008 21:54:32 -0500 From: Nguyen Le [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] IPPA Service Login To: Jonathan Charles [EMAIL PROTECTED] Cc: CCIE Voice ccie_voice@onlinestudylist.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 communication to the ipcc server is there. If i stop the IPPA service, I'll get a host not found error on the phones instead. going to this site connects, just gets the error. http://192.168.1.1:6293/ipphone/jsp/sciphonexml/IPAgentInitial.jsp
Re: [OSL | CCIE_Voice] 0 Conf max sessions
I actually configured all that right after I send this message but no luck. On Sun, Jul 6, 2008 at 11:23 PM, Vik Malhi [EMAIL PROTECTED] wrote: try configuring sccp ccm and sccp ccm group before you set the max sessions. Vik Malhi – CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] [EMAIL PROTECTED] Join our free online support and peer group communities: http://www.IPexpert.com/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 *Mehmet Tufekci *Sent:* Sunday, July 06, 2008 9:17 AM *To:* ccievoice1 *Cc:* OSL CCIE Voice Lab Exam *Subject:* Re: [OSL | CCIE_Voice] 0 Conf max sessions I have 1-3 configured under controller. I am able to see 1-8 available resources under my transcoder profile but no luck with conference. I know conference will not show up if you enable transcoder profile first. No matter what I tried I can not get it to run. On Jul 6, 2008, at 11:44 AM, ccievoice1 wrote: have you utilized all the dsp resources for your pri-group? On Sun, Jul 6, 2008 at 11:31 PM, Mehmet Tufekci [EMAIL PROTECTED] wrote: Hi All, I can not figure out why maximum session 0-0 is showing under conference profile. I did not enable transcoding profile yet. voice-card 0 dspfarm dsp services dspfarm ! ! ! interface Loopback0 ip address 172.3.102.1 255.255.255.255 ip ospf network point-to-point ! ! sccp local Loopback0 sccp ! dspfarm profile 1 transcode codec g711ulaw codec g729r8 shutdown ! dspfarm profile 2 conference codec g711ulaw codec g729r8 shutdown !
Re: [OSL | CCIE_Voice] GDM Log-in
Hi, Here is a quota from someone about the solution but I was not able to figure out how to get it running. Please let me know if this makes sense to you. Regards, Onur. Yes, there is no need to create separate voice mail box for 2nd line, just use the same mailbox to access the GDM, the phone belongs to one person only. There is no way that you could access GDM directly by pressing 9 without logging in to the mailbox first. What u need to do is put 2nd lines DN as E.164 under first lines DN settings in CUE. After this when u take line 2 and press message key then it will ask for password only and then once u logged in you can press 9 to access GDM. On Mon, Jul 7, 2008 at 3:17 AM, Juan [EMAIL PROTECTED] wrote: Hi all, I tried to map the GDM ephone-dn to an existing button - but in that case the MWI light nor the envelope are lit. This holds for any ephone-dn overlayed too. Was anybody able to have the GDM message light lit without using an additional ephone-dn - is that even possible? Another question I have is whether it should be possible to generate local multicast MOH to PSTN phones (multicast moh 239.1.1.1 port ) on the BR1 router if the CCM instructs the MGCP endpoint on BR1 to join a G729 stream? Or does it need to be G711 MOH file no matter what. It looks that way in my case... Any help is much appreciated Kind regards, Juan -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Mehmet Tufekci *Sent:* Sunday, July 06, 2008 4:45 PM *To:* o Ninja *Cc:* ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] GDM Log-in I just learned this you can assign GDM number as you E164 number to your main line. On Jul 6, 2008, at 10:16 AM, o Ninja wrote: I know, but the case is that I dont want to spare a button on my phones just to know that GDM has messages, I wanted to receive these messages in lines I have configured previously. -- CC: ccie_voice@onlinestudylist.com From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] GDM Log-in Date: Sun, 6 Jul 2008 10:13:41 -0400 Users can see the voicemail light (if you use ephone-dn for the GDM) but from the discussion with others and reading little bit it does not seem possible to reach the VMs in GDM directly. No matter what you do you need to go thru your own mailbox and dial 9 to listen messages. On Jul 6, 2008, at 8:02 AM, o Ninja wrote: Hi Mehmet, I am trying to do that also, I want to leave a message to the GDM mailbox and then the members of this GDM receive to receive the messages. This is a simple solution but looks like CUE is not able to do it. -- Conheça já o Windows Live Spaces, o site de relacionamentos do Messenger! Crie já o seu! http://www.amigosdomessenger.com.br/ -- Conheça já o Windows Live Spaces, o site de relacionamentos do Messenger! Crie já o seu! http://www.amigosdomessenger.com.br
Re: [OSL | CCIE_Voice] 0 Conf max sessions
Here is the configuration and results: router2(config)#do show run Building configuration... Current configuration : 2938 bytes ! ! Last configuration change at 15:11:09 UTC Mon Jul 7 2008 ! NVRAM config last updated at 14:49:04 UTC Mon Jul 7 2008 ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec service password-encryption ! hostname router2 ! boot-start-marker boot-end-marker ! card type t1 0 0 no logging console enable secret 5 $1$7cJ4$U9gWuPCv.H1DPdXYIyWJD0 ! no aaa new-model ! resource policy ! network-clock-participate wic 0 ! ! ip cef ! ! no ip domain lookup ! isdn switch-type primary-ni ! voice-card 0 dspfarm dsp services dspfarm ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! controller T1 0/0/0 framing esf linecode b8zs ! controller T1 0/0/1 framing esf linecode b8zs ! ! ! ! interface FastEthernet0/0 ip address 192.168.2.1 255.255.255.0 duplex full speed 100 h323-gateway voip bind srcaddr 192.168.2.1 ! interface FastEthernet0/1 description Management_IP ip address 192.168.10.2 255.255.255.0 duplex full speed 100 ! interface Service-Engine1/0 no ip address shutdown ! ip route 192.168.1.0 255.255.255.0 192.168.10.1 ip route 192.168.3.0 255.255.255.0 192.168.10.3 ! ip http server no ip http secure-server ! ! ! ! control-plane ! ! ! ! ! sccp local FastEthernet0/0 sccp ccm 192.168.2.1 identifier 1 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 2 register xcoder associate profile 1 register conference ! dspfarm profile 2 transcode codec g711ulaw codec g729r8 maximum sessions 4 associate application SCCP shutdown ! dspfarm profile 1 conference codec g711ulaw codec g729r8 maximum sessions 1 associate application SCCP shutdown ! ! dial-peer voice 1000 voip answer-address 1... destination-pattern 1... session target ipv4:192.168.1.1 codec g711ulaw ! dial-peer voice 9 pots destination-pattern 9T incoming called-number 14345552... direct-inward-dial ! ! ! telephony-service max-ephones 2 max-dn 2 ip source-address 192.168.2.1 port 2000 auto assign 1 to 2 system message Your current options sdspfarm units 5 sdspfarm tag 1 xcoder sdspfarm tag 2 conference dialplan-pattern 1 14345552... extension-length 4 max-conferences 8 gain -6 call-forward pattern .T web admin system name cisco password cisco dn-webedit time-webedit transfer-system full-consult transfer-pattern .T secondary-dialtone 9 ! ! ephone-dn 1 dual-line number 2001 description 14345552001 name Gil Grissom ! ! router2(config)#dspfarm router2(config)# router2(config)# router2(config)# router2(config)#dspfarm pro 2 router2(config-dspfarm-profile)#no shut router2(config-dspfarm-profile)#dspfarm pro 1 router2(config-dspfarm-profile)#no shut Enabling profile failed due to insufficient CONFERENCING resources, resources available to support 0 sessions; please modify the configuration and retry router2(config-dspfarm-profile)#do show inv NAME: 2811 chassis, DESCR: 2811 chassis PID: CISCO2811 , VID: V03 , SN: FHK103671CU NAME: VWIC2-2MFT-T1/E1 - 2-Port RJ-48 Multiflex Trunk - T1/E1 on Slot 0 SubSlot 0, DESCR: VWIC2-2MFT-T1/E1 - 2-Port RJ-48 Multiflex Trunk - T1/E1 PID: VWIC2-2MFT-T1/E1 , VID: V01 , SN: FOC102540TY NAME: PVDMII DSP SIMM with one DSP on Slot 0 SubSlot 4, DESCR: PVDMII DSP SIMM with one DSP PID: PVDM2-16 , VID: V01 , SN: FOC1032054R NAME: NM-SE on Slot 1, DESCR: NM-SE PID: NM-CUE, VID: V03, SN: FOC10120B2E NAME: 40GB IDE Disc Daughter Card on Slot 1 SubSlot 0, DESCR: 40GB IDE Disc Daughter Card PID: , VID: 1.0, SN: FOC10170V2B On Mon, Jul 7, 2008 at 9:14 AM, Onur Tufekci [EMAIL PROTECTED] wrote: I actually configured all that right after I send this message but no luck. On Sun, Jul 6, 2008 at 11:23 PM, Vik Malhi [EMAIL PROTECTED] wrote: try configuring sccp ccm and sccp ccm group before you set the max sessions. Vik Malhi – CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] [EMAIL PROTECTED] Join our free online support and peer group communities: http://www.IPexpert.com/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 *Mehmet Tufekci *Sent:* Sunday, July 06, 2008 9:17 AM *To:* ccievoice1 *Cc:* OSL CCIE Voice Lab Exam *Subject:* Re: [OSL | CCIE_Voice] 0 Conf max sessions I have 1-3 configured under controller. I am able to see 1-8 available resources under my transcoder profile but no luck with conference. I know conference will not show up if you enable transcoder profile first. No matter what I tried I can not get it to run
Re: [OSL | CCIE_Voice] 0 Conf max sessions
And this is if I do it other way around: router2(config-dspfarm-profile)#dspfarm profile 2 router2(config-dspfarm-profile)#shut Disabling profile will disconnect active TRANSCODING calls, do you want to continue ? [yes/no]y router2(config-dspfarm-profile)# router2(config-dspfarm-profile)# router2(config-dspfarm-profile)#dspfarm profile 1 router2(config-dspfarm-profile)#no shut router2(config-dspfarm-profile)#dspfarm profile 2 router2(config-dspfarm-profile)#no shut Enabling profile failed due to insufficient TRANSCODING resources, resources available to support 0 sessions; please modify the configuration and retry router2(config-dspfarm-profile)# On Mon, Jul 7, 2008 at 10:39 AM, Onur Tufekci [EMAIL PROTECTED] wrote: Here is the configuration and results: router2(config)#do show run Building configuration... Current configuration : 2938 bytes ! ! Last configuration change at 15:11:09 UTC Mon Jul 7 2008 ! NVRAM config last updated at 14:49:04 UTC Mon Jul 7 2008 ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec service password-encryption ! hostname router2 ! boot-start-marker boot-end-marker ! card type t1 0 0 no logging console enable secret 5 $1$7cJ4$U9gWuPCv.H1DPdXYIyWJD0 ! no aaa new-model ! resource policy ! network-clock-participate wic 0 ! ! ip cef ! ! no ip domain lookup ! isdn switch-type primary-ni ! voice-card 0 dspfarm dsp services dspfarm ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! controller T1 0/0/0 framing esf linecode b8zs ! controller T1 0/0/1 framing esf linecode b8zs ! ! ! ! interface FastEthernet0/0 ip address 192.168.2.1 255.255.255.0 duplex full speed 100 h323-gateway voip bind srcaddr 192.168.2.1 ! interface FastEthernet0/1 description Management_IP ip address 192.168.10.2 255.255.255.0 duplex full speed 100 ! interface Service-Engine1/0 no ip address shutdown ! ip route 192.168.1.0 255.255.255.0 192.168.10.1 ip route 192.168.3.0 255.255.255.0 192.168.10.3 ! ip http server no ip http secure-server ! ! ! ! control-plane ! ! ! ! ! sccp local FastEthernet0/0 sccp ccm 192.168.2.1 identifier 1 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 2 register xcoder associate profile 1 register conference ! dspfarm profile 2 transcode codec g711ulaw codec g729r8 maximum sessions 4 associate application SCCP shutdown ! dspfarm profile 1 conference codec g711ulaw codec g729r8 maximum sessions 1 associate application SCCP shutdown ! ! dial-peer voice 1000 voip answer-address 1... destination-pattern 1... session target ipv4:192.168.1.1 codec g711ulaw ! dial-peer voice 9 pots destination-pattern 9T incoming called-number 14345552... direct-inward-dial ! ! ! telephony-service max-ephones 2 max-dn 2 ip source-address 192.168.2.1 port 2000 auto assign 1 to 2 system message Your current options sdspfarm units 5 sdspfarm tag 1 xcoder sdspfarm tag 2 conference dialplan-pattern 1 14345552... extension-length 4 max-conferences 8 gain -6 call-forward pattern .T web admin system name cisco password cisco dn-webedit time-webedit transfer-system full-consult transfer-pattern .T secondary-dialtone 9 ! ! ephone-dn 1 dual-line number 2001 description 14345552001 name Gil Grissom ! ! router2(config)#dspfarm router2(config)# router2(config)# router2(config)# router2(config)#dspfarm pro 2 router2(config-dspfarm-profile)#no shut router2(config-dspfarm-profile)#dspfarm pro 1 router2(config-dspfarm-profile)#no shut Enabling profile failed due to insufficient CONFERENCING resources, resources available to support 0 sessions; please modify the configuration and retry router2(config-dspfarm-profile)#do show inv NAME: 2811 chassis, DESCR: 2811 chassis PID: CISCO2811 , VID: V03 , SN: FHK103671CU NAME: VWIC2-2MFT-T1/E1 - 2-Port RJ-48 Multiflex Trunk - T1/E1 on Slot 0 SubSlot 0, DESCR: VWIC2-2MFT-T1/E1 - 2-Port RJ-48 Multiflex Trunk - T1/E1 PID: VWIC2-2MFT-T1/E1 , VID: V01 , SN: FOC102540TY NAME: PVDMII DSP SIMM with one DSP on Slot 0 SubSlot 4, DESCR: PVDMII DSP SIMM with one DSP PID: PVDM2-16 , VID: V01 , SN: FOC1032054R NAME: NM-SE on Slot 1, DESCR: NM-SE PID: NM-CUE, VID: V03, SN: FOC10120B2E NAME: 40GB IDE Disc Daughter Card on Slot 1 SubSlot 0, DESCR: 40GB IDE Disc Daughter Card PID: , VID: 1.0, SN: FOC10170V2B On Mon, Jul 7, 2008 at 9:14 AM, Onur Tufekci [EMAIL PROTECTED] wrote: I actually configured all that right after I send this message but no luck. On Sun, Jul 6, 2008 at 11:23 PM, Vik Malhi [EMAIL PROTECTED] wrote: try configuring sccp ccm and sccp ccm group before you set the max sessions. Vik Malhi – CCIE #13890 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] [EMAIL
Re: [OSL | CCIE_Voice] CUE QoS (Vol 1 Lab 15 task 5)
I am not sure what your configuration looks like but here is the config that I was able to place under servive-engine interface. class-map match-all sip match protocol sip class-map match-all rtp match access-group 110 ! ! policy-map sip class sip set ip dscp cs3 class rtp set ip dscp ef ! ! access-list 110 permit udp any range 16384 32767 any access-list 110 permit udp any any range 16384 32767 interface Service-Engine1/0 ip unnumbered FastEthernet0/0 service-module ip address 192.168.2.2 255.255.255.0 service-module ip default-gateway 192.168.2.1 service-policy input sip On Sun, Jul 6, 2008 at 5:06 PM, Abdalla Abdalla [EMAIL PROTECTED] wrote: Hi, I tried to apply a service policy to the service engine in the input direction but i get the error message that the service-ploicy can only be applied in the output direction. Any ideas why this is so?. Whereas the proctor guide solution shows that it can be applied in the input direction. regards AA
Re: [OSL | CCIE_Voice] CUE QoS (Vol 1 Lab 15 task 5)
Oh you are trying to queue the default packets but you can not do that. With applying the QoS policy to service-engine interface we are trinying to mark the packets not the queue them. If you take your fair queue under class class-default (or replace it with set ip dscp default) then you should be able to apply it to the interface. Onur. On Mon, Jul 7, 2008 at 11:57 AM, Abdalla Abdalla [EMAIL PROTECTED] wrote: Mine looks almost like yours. Here is the config I had on POD 18 BR2 router yesterday. class-map match-all SIP match access-group 102 class-map match-all RTP match access-group 101 ! policy-map CUE-MARK class RTP set dscp ef class SIP set dscp cs3 class class-default fair-queue ! access-list 101 permit udp any range 16384 32767 any access-list 101 permit udp any any range 16384 32767 access-list 102 permit udp any eq 5060 any access-list 102 permit udp any any eq 5060 ! ! ! interface FastEthernet0/0 no ip address duplex auto speed auto ! interface FastEthernet0/0.280 encapsulation dot1Q 280 ip address 10.8.202.1 255.255.255.0 no snmp trap link-status ! interface Service-Engine0/0 ip unnumbered FastEthernet0/0.280 service-module ip address 10.8.202.2 255.255.255.0 service-module ip default-gateway 10.8.202.1 ! Below is the output of the error message when i tried to attach the service policy in the input direction: P8-BR2-RTR(config-if)#service-policy ? history Keep history of QoS metrics input Assign policy-map to the input of an interface output Assign policy-map to the output of an interface P8-BR2-RTR(config-if)#service-policy inpu P8-BR2-RTR(config-if)#service-policy input CUE-MARK CBWFQ : Can be enabled as an output feature only - Original Message From: Onur Tufekci [EMAIL PROTECTED] To: Abdalla Abdalla [EMAIL PROTECTED] Cc: CCIE Voice StudyList ccie_voice@onlinestudylist.com Sent: Monday, July 7, 2008 3:54:59 PM Subject: Re: [OSL | CCIE_Voice] CUE QoS (Vol 1 Lab 15 task 5) I am not sure what your configuration looks like but here is the config that I was able to place under servive-engine interface. class-map match-all sip match protocol sip class-map match-all rtp match access-group 110 ! ! policy-map sip class sip set ip dscp cs3 class rtp set ip dscp ef ! ! access-list 110 permit udp any range 16384 32767 any access-list 110 permit udp any any range 16384 32767 interface Service-Engine1/0 ip unnumbered FastEthernet0/0 service-module ip address 192.168.2.2 255.255.255.0 service-module ip default-gateway 192.168.2.1 service-policy input sip On Sun, Jul 6, 2008 at 5:06 PM, Abdalla Abdalla [EMAIL PROTECTED] wrote: Hi, I tried to apply a service policy to the service engine in the input direction but i get the error message that the service-ploicy can only be applied in the output direction. Any ideas why this is so?. Whereas the proctor guide solution shows that it can be applied in the input direction. regards AA
Re: [OSL | CCIE_Voice] GDM Log-in
I used the notification future on the CUE just to see what happens. Main line rings for specified period of time and if you pick up the phone while it is ringing then you do not have to enter any passwords or IDs. On Mon, Jul 7, 2008 at 11:58 AM, Juan [EMAIL PROTECTED] wrote: Hi Ovais, Onur what I want to achieve is that instead of using a seperate button on the phone for GDM, that there's an indication of a mail in GDM using one of my existing buttons - thus not mapping my GDM ephone-dn on a seperate phone button. I found out that overlaying ephone-dn's (whether it be for GDM or any other DNs) does not the trick, so I wonder if there is another way to achieve this. kind regards, Juan -- *From:* Ovais Iqbal [mailto:[EMAIL PROTECTED] *Sent:* Monday, July 07, 2008 3:28 PM *To:* Onur Tufekci; [EMAIL PROTECTED]; Juan *Cc:* o Ninja; ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] GDM Log-in Yes I said this, what is your issue with GDM? What are u trying to achieve? Have you read about GDM's in the docs? Thanks Ovais Iqbal 416-294-7869 Sent from my BlackBerry device -- *From*: Onur Tufekci [EMAIL PROTECTED] *Date*: Mon, 7 Jul 2008 09:24:59 -0400 *To*: Juan[EMAIL PROTECTED] *CC*: o Ninja[EMAIL PROTECTED]; ccie_voice@onlinestudylist.com *Subject*: Re: [OSL | CCIE_Voice] GDM Log-in Hi, Here is a quota from someone about the solution but I was not able to figure out how to get it running. Please let me know if this makes sense to you. Regards, Onur. Yes, there is no need to create separate voice mail box for 2nd line, just use the same mailbox to access the GDM, the phone belongs to one person only. There is no way that you could access GDM directly by pressing 9 without logging in to the mailbox first. What u need to do is put 2nd lines DN as E.164 under first lines DN settings in CUE. After this when u take line 2 and press message key then it will ask for password only and then once u logged in you can press 9 to access GDM. On Mon, Jul 7, 2008 at 3:17 AM, Juan [EMAIL PROTECTED] wrote: Hi all, I tried to map the GDM ephone-dn to an existing button - but in that case the MWI light nor the envelope are lit. This holds for any ephone-dn overlayed too. Was anybody able to have the GDM message light lit without using an additional ephone-dn - is that even possible? Another question I have is whether it should be possible to generate local multicast MOH to PSTN phones (multicast moh 239.1.1.1 port ) on the BR1 router if the CCM instructs the MGCP endpoint on BR1 to join a G729 stream? Or does it need to be G711 MOH file no matter what. It looks that way in my case... Any help is much appreciated Kind regards, Juan -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Mehmet Tufekci *Sent:* Sunday, July 06, 2008 4:45 PM *To:* o Ninja *Cc:* ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] GDM Log-in I just learned this you can assign GDM number as you E164 number to your main line. On Jul 6, 2008, at 10:16 AM, o Ninja wrote: I know, but the case is that I dont want to spare a button on my phones just to know that GDM has messages, I wanted to receive these messages in lines I have configured previously. -- CC: ccie_voice@onlinestudylist.com From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [OSL | CCIE_Voice] GDM Log-in Date: Sun, 6 Jul 2008 10:13:41 -0400 Users can see the voicemail light (if you use ephone-dn for the GDM) but from the discussion with others and reading little bit it does not seem possible to reach the VMs in GDM directly. No matter what you do you need to go thru your own mailbox and dial 9 to listen messages. On Jul 6, 2008, at 8:02 AM, o Ninja wrote: Hi Mehmet, I am trying to do that also, I want to leave a message to the GDM mailbox and then the members of this GDM receive to receive the messages. This is a simple solution but looks like CUE is not able to do it. -- Conhe硠jᠯ Windows Live Spaces, o site de relacionamentos do Messenger! Crie jᠯ seu! http://www.amigosdomessenger.com.br/ -- Conhe硠jᠯ Windows Live Spaces, o site de relacionamentos do Messenger! Crie jᠯ seu! http://www.amigosdomessenger.com.br/
Re: [OSL | CCIE_Voice] param number-of-hunt-grps number
I believe that the sum of huntgroups that you can create is 10. so you can have up to 3 HG under aa and if you have 3 aas running then you have 9 hunt groups. You can add another aa with 1 hunt group in it. After that you need to specifiy how many Hun groups you have in total under your queue. On Mon, Jul 7, 2008 at 11:58 AM, [EMAIL PROTECTED] wrote: my understanding is under queue you can have 10 hunt group and under aa application you can have 3? Sara *Balamurugan Singaram [EMAIL PROTECTED]* wrote: 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 -- Stop! Global Warming ~ Yahoo! JAPAN Earth Projecthttp://pr.mail.yahoo.co.jp/earthproject/
[OSL | CCIE_Voice] Multicast MOH over WAN (To share)
Cisco Unified CallManager locations-based call admission control is capable of tracking unicast MoH streams traversing the WAN but not multicast MoH streams. Thus, even if WAN bandwidth has been fully subscribed, a multicast MoH stream will not be denied access to the WAN by call admission control. Instead, the stream will be sent across the WAN, likely resulting in poor audio stream quality and poor quality on all other calls traversing the WAN. To ensure that multicast MoH streams do not cause this over-subscription situation, you should over-provision the QoS configuration on all downstream WAN interfaces by configuring the low-latency queuing (LLQ) voice priority queue with additional bandwidth. Because MoH streams are uni-directional, only the voice priority queues of the downstream interfaces (from the central site to remote sites) must be over-provisioned. Add enough bandwidth for every unique multicast MoH stream that might traverse the WAN link. For example, if there are four unique multicast audio streams that could potentially traverse the WAN, then add 96 kbps to the voice priority queue (4 * 24 kbps per G.729 audio stream = 96 kbps). If you always want multicast MoH from the branch router flash, then you must configure the central-site server with an audio source that has the same multicast IP address and port number as configured on the branch router. In this scenario, because the multicast MoH audio stream is always coming from the router's flash, it is not necessary for the central site MoH server audio source to traverse the WAN. To prevent the central site audio stream(s) from traversing the WAN, use one of the following methods: •Configure a maximum hop count Configure the central-site MoH audio source with a maximum hop count (or TTL) low enough to ensure that it will not stream further than the central-site LAN. •Configure an access control list (ACL) on the WAN interface Configure an ACL on the central-site WAN interface to disallow packets destined to the multicast group address(es) from being sent out the interface. •Disable multicast routing on the WAN interface Do not configure multicast routing on the WAN interface, thus ensuring that multicast streams are not forwarded into the WAN. Figure 7-6http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/4x/42moh.html#wp1043924illustrates streaming multicast MoH from the flash of a remote router when it is not in SRST mode. After phone A places phone C on hold, phone C receives multicast MoH from the local SRST router. In this figure, the MoH server is streaming a multicast audio source to 239.192.240.1 (on RTP port 16384), however this stream has been limited to a maximum hop of one (1) to ensure that it will not travel off the local MoH server's subnet and across the WAN. At the same time, the branch office SRST router/gateway is multicasting an audio stream from flash. This stream is also using 239.192.240.1 as its multicast address and 16384 as the RTP port number. When phone A presses the Hold softkey, phone C receives the MoH audio stream sourced by the SRST router.
Re: [OSL | CCIE_Voice] CCM-GK call routing problem
Is your Route list registered? On Fri, Jun 27, 2008 at 2:40 PM, Diego Macias [EMAIL PROTECTED] wrote: Hello All Actually I am working on POD 13 and I have some problem In CCM to GK call routing, im getting fast busy tone with calls routed to GK. CCM-GK trunk is registered: CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags --- - --- - - - 10.3.200.21 54434 10.3.200.21 54067 GKVOIP-GW ENDPOINT-ID: 460CC6840002 VERSION: 2 AGE: 27 secs SupportsAnnexE: FALSE g_supp_prots: 0x0050 H323-ID: GK_HQ_1 Voice Capacity Max.= Avail.= Current.= 0 In order to isolate the problem I only configured one route pattern with no partition and one phone with no partition neither CSS. Route pattern is routing directly to GK_Trunk (no RL, RG configured), and I still. got fast busy tone. debug gatekeeper main 10 doestn give any output, so CM is not sending digits to it. I have reset Trunk many times. Anyone knows what the problem could be?
Re: [OSL | CCIE_Voice] BACD Problem
You right. I missed the part of configuration. On Wed, Jun 25, 2008 at 6:07 AM, Nguyen Le [EMAIL PROTECTED] wrote: Transcoder can be invoked for BACD. You just have to do it through the loopback interface. On Tue, Jun 24, 2008 at 11:50 PM, Mehmet Tufekci [EMAIL PROTECTED] wrote: If you are using g729 from the gateway then the answer might be transcoder can not be invoked for BACD. On Jun 24, 2008, at 9:38 PM, Nguyen Le wrote: Try unchecking wait for h.245 terminal capabilities in your GK trunk configuration On Tue, Jun 24, 2008 at 6:41 PM, Jose Linero Welcker [EMAIL PROTECTED] wrote: Hi: I am testing the B-ACD TCL in BR2, the connection between BR2 (CME) and the CCM is trough a gatekeeper and when I called to the pilot number of the script is not working. The local calls from the IP Phones registered to the CME are working and the script is ok, the calls coming from the PSTN to the BACD are working too. The configuration I have specifically to BACD is: application service queue flash:app-b-acd-2.1.0.0.tcl param queue-len 15 param queue-manager-debugs 1 param aa-hunt2 4210 param number-of-hunt-grps 1 ! service aa flash:app-b-acd-aa-2.1.0.0.tcl paramspace english index 1 param number-of-hunt-grps 1 param handoff-string aa param dial-by-extension-option 1 paramspace english language en param max-time-vm-retry 2 param aa-pilot 4500 paramspace english location flash: param second-greeting-time 60 param welcome-prompt _bacd_welcome.au param call-retry-timer 15 param voice-mail 4600 param max-time-call-retry 700 param service-name queue dial-peer voice 10 voip destination-pattern 4500 session target ipv4:172.1.102.1 dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 11 voip service aa incoming called-number 4500 dtmf-relay h245-alphanumeric codec g711ulaw The dial peer to receive the calls from CCM is: dial-peer voice 5 voip translation-profile incoming DNIS destination-pattern [23]... session target ras incoming called-number . tech-prefix 1# dtmf-relay h245-alphanumeric I have configured the transcodec: BR2-RTR-2821#sh sccp SCCP Admin State: UP Gateway IP Address: 142.101.66.1, Port Number: 2000 IP Precedence: 5 User Masked Codec list: None Call Manager: 142.101.66.1, Port Number: 2000 Priority: N/A, Version: 3.1, Identifier: 1 Transcoding Oper State: ACTIVE - Cause Code: NONE Active Call Manager: 142.101.66.1, Port Number: 2000 TCP Link Status: CONNECTED, Profile Identifier: 1 Reported Max Streams: 8, Reported Max OOS Streams: 0 Supported Codec: g711ulaw, Maximum Packetization Period: 30 Supported Codec: g711alaw, Maximum Packetization Period: 30 Supported Codec: g729r8, Maximum Packetization Period: 60 Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30 and the BR2 router is also been configured as IPIPGW: voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip ! Doing a debug voice application I have this error: BR2-RTR-2821# Jun 24 23:32:09.308: //-1//AFW_:/C_ServiceSession_Event_Handler: Jun 24 23:32:09.308: //-1//AFW_:/AFW_Session_New: Jun 24 23:32:09.308: //146//AFW_:/C_PackageSession_NewCall: Session module listened by TclModule_45F87050_0_99922156 Jun 24 23:32:09.308: //146//AFW_:/Open_SetupIndication: Calling #(3001), Called #(852#4500), peer_tag(5) Jun 24 23:32:09.308: //-1//AFW_:/C_PackageSession_GetSigPeer: Jun 24 23:32:19.396: //146//AFW_:/AnyState_Disconnected: Jun 24 23:32:19.396: //146//AFW_:/Session_Close: lastFailureCause 47 Jun 24 23:32:19.396: //146//AFW_:/AFW_M_Session_Terminate: Jun 24 23:32:19.396: //146//AFW_:/AFW_M_Session_Terminate: lastFailureCause 47 Looking for the meaning of this error is: Last Disconnect Cause is 2F , Last Disconnect Text is no resource (47), I am stuck with this problem, any idea of what is the cause? Regards, Jose -- Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! Try it!http://spaces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends.aspxmkt=en-us
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
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] Fast busy TEHO from HQ
Thank you Ahmet. Cheers, Onur. On Sat, Jun 7, 2008 at 3:47 PM, ahmet can [EMAIL PROTECTED] wrote: Hi, May be this is helpful for you, ISDN Disconnect Cause 47 47 - Resource unavailable/New Destination This cause is used to report a resource unavailable event only when no other cause in the resource unavailable class applies. or This cause is used to indicate that the original destination is unavailable and to invoke redirection to a new destination. Regards, Ahmet Can -- Date: Sat, 7 Jun 2008 15:06:41 -0400 From: [EMAIL PROTECTED] To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Fast busy TEHO from HQ Hello everyone, Anybody knows what is going on? I was able to make a call couple of times then I started to get fast busy Any ideas will be greatly appreciated. Regards, Onur. Jun 7 19:04:38.939: ISDN Se0/0/0:15 Q931: Applying typeplan for sw-type 0x12 is 0x 0 0x0, Calling num 211003 Jun 7 19:04:38.943: ISDN Se0/0/0:15 Q931: Applying typeplan for sw-type 0x12 is 0x 0 0x0, Called num 331322 Jun 7 19:04:38.943: ISDN Se0/0/0:15 Q931: TX - SETUP pd = 8 callref = 0x008D Bearer Capability i = 0x8090A3 Standard = CCITT nbs p;Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98383 Exclusive, Channel 3 Display i = 'HQ IP Comm' Calling Party Number i = 0x0081, '211003' Plan:Unknown, Type:Unknown Called Party Number i = 0x80, '331322' Plan:Unknown, Type:Unknown Jun 7 19:04:3 8.967: ISDN Se0/0/0:15 Q931: RX - CALL_PROC pd = 8 callref = 0x808D Channel ID i = 0xA98383 Exclusive, Channel 3 Jun 7 19:04:39.191: ISDN Se0/0/0:15 Q931: RX - ALERTING pd = 8 callref = 0x808D Progress Ind i = 0x8188 - In-band info or appropriate now available Jun 7 19:04:39.287: ISDN Se0/0/0:15 Q931: TX - DISCONNECT pd = 8 callref = 0x008 D *Cause i = 0x80AF - Resource unavailable, unspecified* Jun 7 19:04:39.295: ISDN Se0/0/0:15 Q931: RX - RELEASE pd = 8 callref = 0x808D Jun 7 19:04:39.299: ISDN Se0/0/0:15 Q931: TX - RELEASE_COMP pd = 8 callref = 0x0 08D -- Explore the seven wonders of the world Learn more!http://search.msn%0A+.com/results.aspx?q=7+wonders+worldmkt=en-USform=QBRE
Re: [OSL | CCIE_Voice] H323 with SRST
Hi Jacob, I think you are thinking about the MGCP gateway SRST. I was more curious about the H323 gateway with SRST. Cheers, Onur. On Fri, Jun 6, 2008 at 9:34 AM, Jacob Owen [EMAIL PROTECTED] wrote: Onur, Normally, if you are using PRI Backhaul to the CCM during normal operation you should have no dial-peers that point to the CCM's. The H.323 dial-peers you create will only be invoked when the connection to the CCM's are down and SRST kicks in. The call application alternate default (yes, I know it's the old version of the command) tells the router if you lose connection to the CCM fall back to using H323 (default) as the alternate. That works in conjunction with the ccm-manager fallback-mgcp command to tell the router how to handle calls once SRST mode is ON. --- Onur Tufekci [EMAIL PROTECTED] wrote: I was going over the lab 7 and got a question in mind! We are using h323 dial-peers already for normal times. When we go to SRST dial-peers that are pointing to CallManagers are not functional anymore so for users at BR1 site should we set up another dial-peer and give it a lower (i mean higher number) preference? When SRST is not functional will this break the requirement? Cheers, Onur. Jacob Owen CCIE #14063 (RS, Service Provider), CCVP, CCDP
Re: [OSL | CCIE_Voice] H323 with SRST
Thanks Jacop. I think I got the answer I was looking for. So 3rd dial-peer should be in place for the fail-over time. Thank you again, Onur. On Fri, Jun 6, 2008 at 10:24 AM, Jacob Owen [EMAIL PROTECTED] wrote: Onur, I believe your dial-peer 1 would catch any PSTN calls destined for BR1 phones, but unless you are using the dialplan-pattern command under call-manager-fallback I think you'll need to drop 10 digit calls to 4 with a voice translation rule/profile (you can use others, I just like that). I think a 3rd dial-peer for destination-pattern 1... would be necessary most likely with a preference of 2 (main ccm dial-peer pref 0, 2nd ccm dial-peer pref 1). The other solution would be to set up a Num-exp that turned 1... into say 9121222x1... which would then match your long distance dial-peer. 6 of one, 1/2 dozen of another. I like the seperate dial-peer since I can fine tune cor since some users needing to reach HQ are not allowed to call Long Distance but we'd want them to be able to reach the HQ Long Distance Numbers. Hope this helps. --- Onur Tufekci [EMAIL PROTECTED] wrote: You right but if the connection is broken to UCM then these dialpeers are no good so we need another one??? dial-peer voice 1 pots incoming called-number . direct-inward-dial port 0/0/0:23 ! dial-peer voice 1000 voip destination-pattern [12]... voice-class codec 1 voice-class h323 1 session target ipv4:10.X.200.20 dtmf-relay h245-alphanumeric ip qos dscp cs3 signaling no vad ! dial-peer voice 2000 voip preference 1 destination-pattern [12]... voice-class codec 1 voice-class h323 1 session target ipv4:10.X.200.21 dtmf-relay h245-alphanumeric ip qos dscp cs3 signaling no vad On Fri, Jun 6, 2008 at 10:07 AM, Jacob Owen [EMAIL PROTECTED] wrote: Onur, DOH! You are totally correct, I see BR1 and instantly think of MGCP. Can you post your config, I am trying to visualize what your dial-peers pointing to CCM's are for, I would think they were for inbound calls only but I could be wrong. I am just thinking if you are running H323 on the BR1 gateway it should already have the dial-peers created since the CCM would just point to BR1 as an H323 Gateway. --- Onur Tufekci [EMAIL PROTECTED] wrote: Hi Jacob, I think you are thinking about the MGCP gateway SRST. I was more curious about the H323 gateway with SRST. Cheers, Onur. On Fri, Jun 6, 2008 at 9:34 AM, Jacob Owen [EMAIL PROTECTED] wrote: Onur, Normally, if you are using PRI Backhaul to the CCM during normal operation you should have no dial-peers that point to the CCM's. The H.323 dial-peers you create will only be invoked when the connection to the CCM's are down and SRST kicks in. The call application alternate default (yes, I know it's the old version of the command) tells the router if you lose connection to the CCM fall back to using H323 (default) as the alternate. That works in conjunction with the ccm-manager fallback-mgcp command to tell the router how to handle calls once SRST mode is ON. --- Onur Tufekci [EMAIL PROTECTED] wrote: I was going over the lab 7 and got a question in mind! We are using h323 dial-peers already for normal times. When we go to SRST dial-peers that are pointing to CallManagers are not functional anymore so for users at BR1 site should we set up another dial-peer and give it a lower (i mean higher number) preference? When SRST is not functional will this break the requirement? Cheers, Onur. Jacob Owen CCIE #14063 (RS, Service Provider), CCVP, CCDP Jacob Owen CCIE #14063 (RS, Service Provider), CCVP, CCDP Jacob Owen CCIE #14063 (RS, Service Provider), CCVP, CCDP
Re: [OSL | CCIE_Voice] Forwarded calls from CME phones to CUE are cleared if call coming from CCM
:* Juan [mailto:[EMAIL PROTECTED] *Sent:* Thursday, May 29, 2008 11:44 AM *To:* 'OSL CCIE Voice Lab Exam' *Subject:* RE: [OSL | CCIE_Voice] Forwarded calls from CME phones to CUE are cleared if call coming from CCM Hi Gustavo, have you found the issue with forwards to CUE? I see the same here, but in my setup the incoming leg on CME is SIP, using g729. As transcoding does not engage if 'launched' by SIP, CUE can't take the call as it only speaks g711. cheers, Juan -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Mark Snow *Sent:* Friday, May 23, 2008 4:26 PM *To:* OSL CCIE Voice Lab Exam *Subject:* Re: [OSL | CCIE_Voice] Forwarded calls from CME phones to CUE are cleared if call coming from CCM Yes - please post your config! :) -- 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 May 22, 2008, at 2:36 PM, Onur Tufekci wrote: Can you please post all the config on R2? On Thu, May 22, 2008 at 4:15 PM, Sanchez Galarza, Gustavo - (Col) [EMAIL PROTECTED] wrote: Hi: I have a very estrange issue: I make a call from a CCM phone to CME phone and the call is forwarded by no answer to CUE Voicemail but immediately I receive a busy tone in CCM. I see that the CME invoke the transcoder but the call doesn't proceed. If I make a direct call from CCM to CUE Pilot, this proceeds correctly and I hear the prompts and the transcoder operates good. I have configured transcoder, allow-connections h t s allow-connections s t h Anybody could provides me some feedback, idea? Thanks ** *Gustavo Sánchez*
Re: [OSL | CCIE_Voice] Forwarded calls from CME phones to CUE are cleared if call coming from CCM
Here is a link that some other group talking about the same issue: http://www.voiceie.com/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=8;t=000599 On Tue, Jun 3, 2008 at 8:16 AM, Onur Tufekci [EMAIL PROTECTED] wrote: same here! I got these configured voice service voip allow connections h323 to sip allow connections sip to h323 h323 sip Transcoder is registered Under telephony service call-forward pattern .T transfer-system full-consult transfer-pattern .T ephone-dn 3 dual-line call-forward busy 3600 call-forward noan 3600 timeout 10 On Thu, May 29, 2008 at 7:55 AM, Juan [EMAIL PROTECTED] wrote: Please disregard my previous mail : it seems Xcoding does indeed engage, even if the call comes from SIP g729 and it gets xcoded to g711 (direct call to CUE from CCM) In the past I think I overlooked this, as I was under the impression transcoding from SIP was not supported. Hence I thought to only have forwards to CUE work if the incoming dialpeer on CME would be h323. So, I have the same problem as you did now: no forwards to CUE work by means of the command: 'call-forward noan 3600 timeout 10' :-S When I set the DN manually to forward all to 3600, it works however... The outbound trunk on CCM is h323 (MTP checked, not waiting on h245 call capabilties, outbound fast start enabled or disabled- it doesn't matter: same as above (?) - I'd think of faststart outbound if h323-SIP...) Any help is greatly appreciated - I'm looking into it for some hours now.. I attached the ccapi output and dialpeer info from CME: BR2-RTR# May 29 2008 13:32:50.525 CEST: //209//CCAPI/cc_api_caps_ind: Call Entry Is Not Found May 29 2008 13:32:50.525 CEST: //-1/00409C510200/CCAPI/cc_api_display_ie_subfields: cc_api_call_setup_ind_common: cisco-username=2122251003 - ccCallInfo IE subfields - cisco-ani=2122251003 cisco-anitype=0 cisco-aniplan=0 cisco-anipi=0 cisco-anisi=1 dest=3001 cisco-desttype=0 cisco-destplan=0 cisco-rdie= cisco-rdn= cisco-rdntype=0 cisco-rdnplan=0 cisco-rdnpi=0 cisco-rdnsi=0 cisco-redirectreason=-1 May 29 2008 13:32:50.525 CEST: //-1/00409C510200/CCAPI/cc_api_call_setup_ind_common: Interface=0x66847600, Call Info( Calling Number=2122251003(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed), BR2-RTR#Called Number=3001(TON=Unknown, NPI=Unknown), Calling Translated=FALSE, Subsriber Type Str=Unknown, FinalDestinationFlag=TRUE, Incoming Dial-peer=2, Progress Indication=NULL(0), Calling IE Present=TRUE, Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=209 May 29 2008 13:32:50.525 CEST: //-1/00409C510200/CCAPI/ccCheckClipClir: In: Calling Number=2122251003(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed) May 29 2008 13:32:50.525 CEST: //-1/00409C510200/CCAPI/ccCheckClipClir: Out: Calling Number=2122251003(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed) May 29 2008 13:32:50.525 CEST: //209/00409C510200/CCAPI/cc_api_call_setup_ind_common: Set Up Event Sent; Call Info(Calling Number=2122251003(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed), Called Number=3001(TON=Unknown, NPI=Unknown)) May 29 2008 13:32:50.525 CEST: //209/00409C510200/CCAPI/cc_process_call_setup_ind: Event=0x66CE82F8 May 29 2008 13:32:50.525 CEST: //209/00409C510200/CCAPI/ccCallSetContext: Context=0x719A2F34 May 29 2008 13:32:50.525 CEST: //209/00409C510200/CCAPI/cc_process_call_setup_ind: CCAPI handed cid 209 with tag 2 to app _ManagedAppProcess_Default May 29 2008 13:32:50.525 CEST: //209/00409C510200/CCAPI/ccCallProceeding: Progress Indication=NULL(0) May 29 2008 13:32:50.529 CEST: //209/00409C510200/CCAPI/ccCallSetupRequest: Destination=, Calling IE Present=TRUE, Mode=0, Outgoing Dial-peer=20008, Params=0x719A6CC4, Progress Indication=O BR2-RTR#RIGINATING SIDE IS NON ISDN(3) May 29 2008 13:32:50.529 CEST: //209/00409C510200/CCAPI/ccCheckClipClir: In: Calling Number=2122251003(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed) May 29 2008 13:32:50.529 CEST: //209/00409C510200/CCAPI/ccCheckClipClir: Out: Calling Number=2122251003(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed) May 29 2008 13:32:50.529 CEST: //209/00409C510200/CCAPI/ccCallSetupRequest: Destination Pattern=3001$, Called Number=3001, Digit Strip=TRUE May 29 2008 13:32:50.529 CEST: //209/00409C510200/CCAPI/ccCallSetupRequest: Calling Number=2122251003(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed), Called Number=3001(TON=Unknown, NPI=Unknown), Redirect Number=, Display Info=HQ-phn3 Account Number=2122251003, Final Destination Flag=TRUE, Guid=00409C51-CC85-D11D-0200-0A3D8169, Outgoing Dial-peer=20008 May
[OSL | CCIE_Voice] Prevent looping in UNITY
Does any one has an answer for this? To prevent looping in Unity do we use port numbers only (160* etc..) or we use them with masks (2122211600* etc...)?
[OSL | CCIE_Voice] VRACK WEB Page is not accessible!!!
Online vrack web page is not accessible it keeps loading and nothing happens.
[OSL | CCIE_Voice] Per flow Policing
Hi guys, I started to have headaches after two hours of internet research and banging my head to walls. Is PER FLOW policing as same as the PER PORT policing which is MICROFLOW??? Thanks, Onur.
Re: [OSL | CCIE_Voice] Extension Mobility on IP Blue
what is the error message you are getting if there is any? On Thu, May 29, 2008 at 12:15 PM, Ahmed Hamed [EMAIL PROTECTED] wrote: Hi, Any idea how to implement Extension Mobility on IP Blue? I am trying to configure SERVICES button in the IP Blue but with no luck! Please advise, AH
Re: [OSL | CCIE_Voice] CAD installation PDF
Thank you. On Wed, May 28, 2008 at 6:12 PM, Randy Banaria [EMAIL PROTECTED] wrote: Onur, CAD is available on the desktop. On Wed, May 28, 2008 at 11:08 PM, Onur Tufekci [EMAIL PROTECTED] wrote: Does anybody know if the CAD installation pdf guide is on the desktop now? -- Kind Regards, Randy Banaria 07752839106
[OSL | CCIE_Voice] Host not found
If the phone registered to subscriber and the services URL is pointing to publisher then I get Host Not Found Error. If the phone is registered to pub then everything is alright. Is this normal?
Re: [OSL | CCIE_Voice] Host not found
Thank you for your reply, Yes I did change the parameter to IP address of the publisher. This only happens when phone is registered to sub. Thanks, Onur On Wed, May 28, 2008 at 6:20 PM, Jacob Owen [EMAIL PROTECTED] wrote: Onur, Did you change in the Enterprise Parameters to use the IP of the Publisher and not the name in the Services section? It is using the name by default but has to be changed to the IP for it to work http://10.x.200.21/ insetad of http://CCMPUBLISHER. --- Onur Tufekci [EMAIL PROTECTED] wrote: If the phone registered to subscriber and the services URL is pointing to publisher then I get Host Not Found Error. If the phone is registered to pub then everything is alright. Is this normal? Jacob Owen CCIE #14063 (RS, Service Provider), CCVP, CCDP
Re: [OSL | CCIE_Voice] Host not found
I took a break for an hour and now looks like it works!! On Wed, May 28, 2008 at 6:31 PM, Jacob Owen [EMAIL PROTECTED] wrote: Onur, Did you fix the SQL issue between the Pub/Sub? To ensure your phones are getting the correct URL go to the Settings on your phone then 3 for Network Configuration then to 29 which should show you your Services URL. If it is incorrect try resetting the phone and see if it changes to the correct URL. --- Onur Tufekci [EMAIL PROTECTED] wrote: Thank you for your reply, Yes I did change the parameter to IP address of the publisher. This only happens when phone is registered to sub. Thanks, Onur On Wed, May 28, 2008 at 6:20 PM, Jacob Owen [EMAIL PROTECTED] wrote: Onur, Did you change in the Enterprise Parameters to use the IP of the Publisher and not the name in the Services section? It is using the name by default but has to be changed to the IP for it to work http://10.x.200.21/ insetad of http://CCMPUBLISHER. --- Onur Tufekci [EMAIL PROTECTED] wrote: If the phone registered to subscriber and the services URL is pointing to publisher then I get Host Not Found Error. If the phone is registered to pub then everything is alright. Is this normal? Jacob Owen CCIE #14063 (RS, Service Provider), CCVP, CCDP Jacob Owen CCIE #14063 (RS, Service Provider), CCVP, CCDP
[OSL | CCIE_Voice] Can not access to VRACK web site
Hi, I been trying to open the web site for the VRACK but it keeps loading. I have session in 3 minutes! Regards, Onur.
Re: [OSL | CCIE_Voice] Proctorlabs Outage
Thank you, Onur. On Sun, May 25, 2008 at 10:01 AM, Drew lePla [EMAIL PROTECTED] wrote: Dear Proctor Labs customer, Proctor Labs support would like to apologize for the recent Proctor Labs server downtime. We have identified and resolved the issue so you should now be able to login at www.proctorlabs.com . We will be sending out two voucher codes for those students who's sessions were affected by this outage. Again we apologize for the inconvenience and thank you for your patience and understanding. If I can be of further assistance, please let me know. Drew LePla - Comp TIA A+, CCNA Technical Support Engineer – Ipexpert, Inc. Telephone: +1.810.326.1444 x204 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] IPexpert - The Global Leader in Self-Study, Classroom-Based, Video Class-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.
[OSL | CCIE_Voice] CME -- GK -- 6608 Calls fail
Hi, I was working on the Volume 3 workbook (I think it is great) first lab and I could not get the calls to successfully complete to PSTN phone from CME phone. I have the translation pattern set up in UCM for 1#.! with predot / prefix 9 Route list in the CallManager includes 6608 mgcp BR1 in this order I hear one ring and fast busy right after. If I change the order of my gateways in the Route List then it works fine. Any ideas? Onur.
Re: [OSL | CCIE_Voice] Dial-peer match
Thank you Christopher. I have those commands under my dial-peers. Am I making any mistake with it? Inbound is any different then the method I am using? On Fri, May 23, 2008 at 3:46 PM, Ellington, Chris [EMAIL PROTECTED] wrote: Most sip providers can either listen to inband (dtmf-relay h245-alpha) or RFC2833 (dtmf-relay rtp-nte) – but from the config below it doesn't appear that you are using either one. Chris *Christopher Ellington* | VoIP/SIP Engineer phone fax +1.317.715.8578 | [EMAIL PROTECTED] CCIE #6814 *Interactive Intelligence Inc.* Deliberately Innovative www.inin.com *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Onur Tufekci *Sent:* Friday, May 23, 2008 3:42 PM *To:* OSL CCIE Voice Lab Exam *Subject:* [OSL | CCIE_Voice] Dial-peer match Hi All, I have two dial-peers set up when the call comes in from CALLMANAGER it matches dial-peer 10 showing incoming dial-peer in debug voip ccapi inout. Then dial-peer 20 is matched for out going dial-peer. Call works fine but the DTMF does not work correctly. Sometimes DTMF is generated twice. --- Why calling number gets matched with destination pattern? How to avoid generating DTMF twice? dial-peer voice 10 voip destination-pattern 1703123 session target ipv4:10.x.x.x--- CALLMANAGER H323 Gateway dtmf-relay h245-alphanumeric dial-peer voice 20 voip destination-pattern 1[2-9]..[2-9].. session target ipv4:10.x.x.x SIP Server dtmf-relay rtf-nte session protocol sipv2
Re: [OSL | CCIE_Voice] Thanks. I finally passed.
Congrads. Did you pass at you first try? On Fri, May 23, 2008 at 3:49 PM, Ovais Iqbal [EMAIL PROTECTED] wrote: Congratulations, enjoy the summer-08 Ovais Iqbal 416-294-7869 Sent from my BlackBerry device -Original Message- From: IPheaders [EMAIL PROTECTED] Date: Fri, 23 May 2008 14:46:11 To:CCIE Voice ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Thanks. I finally passed. I just wanted to take a moment to says thanks to Mark, Vik, and everyone else at IPExpert for the countless hours they put in to develop a quality product and going the extra mile to help moderate this forum and to help keep us straight. I also wanted to thank everyone that participates in this forum as well. I have never engaged in any type of forum in the past but found myself thorougly enjoying my experience with this one. I did just recently pass my voice IE lab last week. I'm sharing this not to boast, but rather to encourage everyone to keep working hard and to promote the affectiveness of this forum and IPExpert's product. I wish the best of luck to everyone and I will continue to check in on this forum and help contribute whenever I can. Cheers, Scott - CCIE #20903
Re: [OSL | CCIE_Voice] Dial-peer match
Thank you Chris. I will double check. On Fri, May 23, 2008 at 3:57 PM, Ellington, Chris [EMAIL PROTECTED] wrote: Each method is different – h245-alpha is different than rtp-nte is different than sip-notify – the SIP provider will have to tell you which you are using. Also, it appears that you are using a CUBE/SBC (IPIPGW) so there may be other issues with DTMF. chris *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Onur Tufekci *Sent:* Friday, May 23, 2008 3:55 PM *To:* OSL CCIE Voice Lab Exam *Subject:* Re: [OSL | CCIE_Voice] Dial-peer match Thank you Christopher. I have those commands under my dial-peers. Am I making any mistake with it? Inbound is any different then the method I am using? On Fri, May 23, 2008 at 3:46 PM, Ellington, Chris [EMAIL PROTECTED] wrote: Most sip providers can either listen to inband (dtmf-relay h245-alpha) or RFC2833 (dtmf-relay rtp-nte) – but from the config below it doesn't appear that you are using either one. Chris *Christopher Ellington* | VoIP/SIP Engineer phone fax +1.317.715.8578 | [EMAIL PROTECTED] CCIE #6814 *Interactive Intelligence Inc.* Deliberately Innovative www.inin.com *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Onur Tufekci *Sent:* Friday, May 23, 2008 3:42 PM *To:* OSL CCIE Voice Lab Exam *Subject:* [OSL | CCIE_Voice] Dial-peer match Hi All, I have two dial-peers set up when the call comes in from CALLMANAGER it matches dial-peer 10 showing incoming dial-peer in debug voip ccapi inout. Then dial-peer 20 is matched for out going dial-peer. Call works fine but the DTMF does not work correctly. Sometimes DTMF is generated twice. --- Why calling number gets matched with destination pattern? How to avoid generating DTMF twice? dial-peer voice 10 voip destination-pattern 1703123 session target ipv4:10.x.x.x--- CALLMANAGER H323 Gateway dtmf-relay h245-alphanumeric dial-peer voice 20 voip destination-pattern 1[2-9]..[2-9].. session target ipv4:10.x.x.x SIP Server dtmf-relay rtf-nte session protocol sipv2
Re: [OSL | CCIE_Voice] Thanks. I finally passed.
Great achievement. Congrads again. On Fri, May 23, 2008 at 4:10 PM, IPheaders [EMAIL PROTECTED] wrote: No, sir. It took me 7 tries. However, I was very aggressive and took the exam every 30-40 days and started taking the exam before I was ready. On Fri, May 23, 2008 at 2:55 PM, Onur Tufekci [EMAIL PROTECTED] wrote: Congrads. Did you pass at you first try? On Fri, May 23, 2008 at 3:49 PM, Ovais Iqbal [EMAIL PROTECTED] wrote: Congratulations, enjoy the summer-08 Ovais Iqbal 416-294-7869 Sent from my BlackBerry device -Original Message- From: IPheaders [EMAIL PROTECTED] Date: Fri, 23 May 2008 14:46:11 To:CCIE Voice ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Thanks. I finally passed. I just wanted to take a moment to says thanks to Mark, Vik, and everyone else at IPExpert for the countless hours they put in to develop a quality product and going the extra mile to help moderate this forum and to help keep us straight. I also wanted to thank everyone that participates in this forum as well. I have never engaged in any type of forum in the past but found myself thorougly enjoying my experience with this one. I did just recently pass my voice IE lab last week. I'm sharing this not to boast, but rather to encourage everyone to keep working hard and to promote the affectiveness of this forum and IPExpert's product. I wish the best of luck to everyone and I will continue to check in on this forum and help contribute whenever I can. Cheers, Scott - CCIE #20903 -- There are only 10 types of people in the world: Those who understand binary, and those who don't
Re: [OSL | CCIE_Voice] GK Bandwidth use ?
Hi Juan, How did you set your Gatekeeper up? Gatekeeper should show 16 K I believe but some of my previous attempts were not successful either. I was reading 128 on the gatekeeper instead of 16. Cheers, Onur. On Mon, May 19, 2008 at 3:52 PM, Juan [EMAIL PROTECTED] wrote: Hi, can someone give me some help with the following quite basic issue I have: I try to hook up a h323 ATA to CCM via a GK. I thought I'd use a trunk for this on the CCM, in device pool HQ that speaks g711 with HQ phones, and g729 with BR1 phones. Making a call from HQ ATA shows g711 on the phone, and a call from BR1ATA shows g729 on the phone. Is it normal the gatekeeper shows 64K of bandwidth for both calls? First, I would have thought 128K for G711 calls and 16K for g729 calls. many thanks for the feedback - I appreciate it, Juan
Re: [OSL | CCIE_Voice] qos marking ideas !!!
Hi, How are you? I been trying to figure this out as well. One thing that I can point out in your configuration is that the HQ router incoming from 6500. If you mark all the traffic with DSCP 0 for the class-default then you will be marking the Voice Payloads as well I think. So just leaving the class class-default blank should not touch the voice payload everything else will have their original DSCP values. Cheers, Onur. On Fri, May 9, 2008 at 3:42 AM, Djokic Sinisa [EMAIL PROTECTED] wrote: hi team.. i'm new on this list and have some concerns about QoS.. so, maybe someone can help.. so, the thing is that i want to mark signalig traffic ( h323, sccp, mgcp, ras, sip ) on HQ-RTR, RSB-RTR and RSC-RTR, and NOT trust markings on the switches or to remark on them.. so, this is idea how to do it, but i have some concerns as you would see and doubts about it.. so, if anyone has idea how to do it i'd appreciated it.. so here it is.. *for HQ-RTR* ! ip access-list extended CONTROL-HQ permit tcp any range 2000 2002 any ccm-to-phones we need cover both directions i think - it all goes over the same subinf on HQ-RTR permit tcp any any range 2000 2002 phones-to-ccm we need cover both directions i think - it all goes over the same subinf on HQ-RTR permit tcp any eq 2428 any ccm-to-mgcp-gw permit tcp any any eq 2428 mgcp-gw-to-ccm6608-to-ccm, for RSB-RTR-to-ccm it goes on RSB-RTR permit udp any eq 2427 any ccm-to-mgcp-gw permit udp any any eq 2427 mgcp-gw-to-ccm6608-to-ccm, for RSB-RTR-to-ccm it goes on RSB-RTR permit tcp any any eq 1720 permit udp any eq 1719 any ccm-to-gk vice-versa gk-to-ccm i'm not shure should i do it and how to do it permit tcp any any eq 1718 ccm-to-gk permit udp any any eq 5060 ccm-to-sip-gwvice versa handles ip qos command under dial-peer permit tcp any any eq 5060 ccm-to-sip-gwvice versa handles ip qos command under dial-peer ! class-map match-any CONTROL-HQ match access-group name CONTROL-HQ ! policy-map MARK class CONTROL-HQ set dscp cs3 class class-default *I SUPPOSE IT MUST be default one as well, beacuse if not we may have undesired data traffic with wrong marking traversing our WAN* set dscp default *but it so, there must be class for RTP as well or it would be remarked and that's bad* ! interface FastEthernet0/0.XY - VOICE ONE service-policy input MARK ! interface FastEthernet0/0.XY - DATA ONE *SHOULD i put in on data subinterface as well - the same reason as above, should i take care of potentially uneamted traffic form data vlan* service-policy input MARK *this makes sence to put only in HQ ethernet ingress subinterfaces - input..* *i'm not shure should it be put on data as well..i think yes..* *for RSB-RTR* ip access-list extended CONTROL-RSB permit tcp any any range 2000 2002 phones-to-ccm ! class-map match-any CONTROL-RSB match access-group name CONTROL-RSB ! policy-map MARK class CONTROL-RSB set dscp cs3 class class-default *I SUPPOSE IT MUST be default one as well, beacuse if not we may have undesired data traffic with wrong marking traversing our link* set dscp default *the only place i can think of this have sense to put is in input direction on interface Vlan XY '( voice ) as well as in input direction on interface Vlan XY ( data )..* *mgcp ip qos dscp cs3 control handles mgcp originating from router* *no H323 from-and-to-RSB when srst is working, then wan is down* *no RAS from-and-to RSB* *no SIP from-and-to RSB* ! *for RSB-RTR* ip access-list extended CONTROL-RSB permit tcp any any range 2000 2002 phones-to-ccm ALTHOUGH i don't se point to mark SCCP on RSC since doesn't traverse WAN ! class-map match-any CONTROL-RSB match access-group name CONTROL-RSB ! policy-map MARK class CONTROL-RSB set dscp cs3 class class-default *I SUPPOSE IT MUST be default one as well, beacuse if not we may have undesired data traffic with wrong marking traversing our link* set dscp default *ip qos dscp cs3 signaling handles h323 and ras originating from router* *no mgcp from-and-to-RSC when srst is working, then wan is down* *no RAS from-and-to RSB* *SIP from-and-to RSB we
Re: [OSL | CCIE_Voice] DSP Issues on BR2
I think this is NM-HDV2 module so your configuration should change accordingly. Can you please check that is the module? On Mon, May 19, 2008 at 10:51 PM, Paul and Bobs [EMAIL PROTECTED] wrote: My config is pasted below. Fro some reson when I enter dspfarm this command disappears. and when i enter dspfarm transcode maximum session ? i get 0-0. Once I get these configured how can i check to see that the dsp are correctly configured and is there a command to see how many dsp are left nad how many are confgiure for different services. controller E1 0/0/0 pri-group timeslots 1-3,16 voice-card 0 dspfarm dsp services dspfarm sccp local FastEthernet0/0.240 sccp ccm 10.4.202.1 priority 1 sccp telephony-service max-ephones 30 max-dn 30 ip source-address 10.4.202.1 port 2000 system message Your current options sdspfarm units 2 sdspfarm transcode sessions 1 sdspfarm tag 1 mtp00128031cca8 time-zone 42 time-format 24 voicemail 4111 max-conferences 8 gain -6 call-forward pattern .T dn-webedit time-webedit secondary-dialtone 9
[OSL | CCIE_Voice] Another QoS question
Hello Everyone, I know that access-groups can be applied to either directions on a interface. We create access-lists and associate them with the service policies. After that we apply the service policy to an interface. Does this work the same way as access-groups? If so one of the ACL entry will not work depending on the direction we apply the service policy to. class-map sig match access-group 100 policy-map sig class sig set ip dscp af31 int range fast 0/23 - 24-- interface for phone so callmanager should be destination also there is signaling coming from CallManager!! service-policy *input* sig *access-list 100 permit tcp any range 2000 2002 any -- CallManager Source port* *access-list 100 permit tcp any any range 2000 2002 -- CallManager Destination port* Is that correct? Cheers, Onur.
[OSL | CCIE_Voice] QoS Question
Hi All, Does anyone know how to reserve %5 to a type of traffic but not use bandwidth percent command? Cheers, Onur.
[OSL | CCIE_Voice] Can not establish communication to BACD
Hi All, I am getting 3 ring backs and then fast busy with this configuration over gatekeeper. I can dial internally fine and Over the Gatekeeper i can reach to voice-mail and can get transfered to VM. When I try to go to BACD then it does not work. I looked at the Call Legs and it seems like they are all g729r. I can not get system to use g711u internally. num-exp 2#3... 3... dial-peer voice 3600 voip destination-pattern 3[6].. session protocol sipv2 session target ipv4:10.20.202.2 dtmf-relay sip-notify codec g711ulaw ! dial-peer voice 1000 pots service aa incoming called-number 3010 port 0/0/0:15 ! dial-peer voice 3010 voip service aa destination-pattern 3010 session target ipv4:172.20.102.1 incoming called-number 3010 dtmf-relay h245-alphanumeric codec g711ulaw ! dial-peer voice 2000 voip session target ras incoming called-number . codec g729r
[OSL | CCIE_Voice] Gatekeeper Bandwith
I am looking at the bandwidth utilizations from CCM to CME and CME to CCM. Calls that are going out from CCM are showing as using 16 K and calls that are originated from CME are showing up as 128 K. My interzone limit 64 K so I am really confused about how this might be happening. Any ideas from any one? I checked my regions and codec setting on the dial-peers at least 4 times. GATEKEEPER ZONES GK name Domain Name RAS Address PORT FLAGS --- --- --- - - HQ-RTRipexpert.com 10.X.200.3 1719 LS BANDWIDTH INFORMATION (kbps) : Maximum total bandwidth : unlimited *Current total bandwidth : 128.0* *Maximum interzone bandwidth : 64* Current interzone bandwidth : 0.0 Maximum session bandwidth : unlimited * Total number of concurrent calls : 1* SUBNET ATTRIBUTES : All Other Subnets : (Enabled) PROXY USAGE CONFIGURATION : Inbound Calls from all other zones : to terminals in local zone HQ-RTR : use proxy to gateways in local zone HQ-RTR : do not use proxy to MCUs in local zone HQ-RTR : do not use proxy Outbound Calls to all other zones : from terminals in local zone HQ-RTR : use proxy from gateways in local zone HQ-RTR : do not use proxy
Re: [OSL | CCIE_Voice] CME B-ACD
instead of translation rule you can just add E164 id in CUE as 23215 to the user's mailbox that you want to send the call out to and create an ephone-dn 23215 with Call Forward All to VM. On Sun, May 11, 2008 at 2:11 PM, Mike O [EMAIL PROTECTED] wrote: Jason, I tried the following with no luck here is my config. What I am trying to do is give the under the option to goto voice mail while on hold inside of a hunt group, can it be done? application service queue flash:app-b-acd-2.1.2.2.tcl param queue-len 15 param aa-hunt5 23215 param aa-hunt1 3000 param aa-hunt2 3001 param number-of-hunt-grps 3 param queue-manager-debugs 1 voice translation-rule 10 rule 10 /23215/ /3215/ ephone-dn 160 number 23215 call-forward all 5000 translate called 10 - Original Message - *From:* jason sung [EMAIL PROTECTED] *To:* Mike O [EMAIL PROTECTED] *Cc:* ccie_voice@onlinestudylist.com *Sent:* Sunday, May 11, 2008 10:50 AM *Subject:* Re: [OSL | CCIE_Voice] CME B-ACD I have not tried what I am about to say but technically it should work. Under service queue, assign a hunt pilot for voicemail. For example as follows application service queue param aa-hunt5 23215 ephone-dn 5 number 23215 call-forward all 3600 (3600 is your VM pilot). Don't forget to assign the translation that converts 23215 to 3215... So now if a caller wants to go to voicemail press 5 On Sat, May 10, 2008 at 1:46 PM, Mike O [EMAIL PROTECTED] wrote: I setup ACD with CME and it seems to be working pretty good. I was wondering if thier is away to have a user while on hold in a call queue, press a button and be dropped in voice mail instead of staying on hold. Is this possible? Thanks, Mike
Re: [OSL | CCIE_Voice] IPMA shared mode
I set it up in the office and it looks same as normal softkey template. On May 4, 2008, at 7:45 PM, Gregory Jost (grjost) [EMAIL PROTECTED] wrote: In shared line mode, is DND the only feature available to the manager? All I see is the bell on the display. Greg Jost Network Consulting Engineer Unified Communications Practice Cisco Systems, Inc. 214-274-1922 From: Onur Tufekci [mailto:[EMAIL PROTECTED] Sent: Sunday, May 04, 2008 6:27 PM To: Gregory Jost (grjost) Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] IPMA shared mode For me it was service settings for the ipma and service restart. Sent from my iPhone. On May 4, 2008, at 6:08 PM, Gregory Jost (grjost) [EMAIL PROTECTED] wrote: For some reason, I’m not able to login to the Assistant Console wi th an attendant using shared line mode. Proxy mode assistants can login. Any ideas? image001.jpg Greg Jost Network Consulting Engineer Unified Communications Practice Cisco Systems, Inc.
Re: [OSL | CCIE_Voice] Cisco Agent Desktop
Depends on version I guess. 8080 did not work for me. Whatever the doc says for that version. On May 1, 2008, at 10:23 PM, Paul and Bobs [EMAIL PROTECTED] wrote: When trying to enable the IP phone service for IPCC so a user can login in and out of the queue, do you use port 6293 or 8080. I am finding these two ports in defferent documentation. Thanks P
[OSL | CCIE_Voice] New study guide
Hi All, When is the new study guide coming out? I slowed down my studies since do not want to work with older material and keep repeating the same thing. Cheers, Onur.
Re: [OSL | CCIE_Voice] one-way audio from cm to cme
Can you post the cme configuration please? On Wed, Apr 16, 2008 at 11:59 AM, [EMAIL PROTECTED] wrote: i am testing cm/cme integration, i configured non-gk-control trunk for cme from cm. when cm phone (1001)call cme phone(3001), i hear one way audio. 3001 can hear 1001 but not the other way around. what could be the problem? Sara -- GANBARE! NIPPON! Win your ticket to Olympic Games 2008.http://pr.mail.yahoo.co.jp/ganbare-nippon/
Re: [OSL | CCIE_Voice] Dial Plan design question, AAR with vmail...
Are you trying to configure only AAR or only SRST? On Wed, Apr 16, 2008 at 2:03 PM, Jonathan Charles [EMAIL PROTECTED] wrote: OK, so AAR kicks in and I forward the call to the PSTN, the user I was dialing does not answer and the call should forward to vmail... how would we make this work? voicemail under srst config? Jonathan
Re: [OSL | CCIE_Voice] Just started workbook and nagging question has popped up again...
I think it is up to you which one to use for 3550. On the ether switch module you have to use trunking. On Wed, Apr 16, 2008 at 8:11 PM, Jonathan Charles [EMAIL PROTECTED] wrote: OK, one last time, for the 3550 config, I am seeing one way to do it from IPExpert and another from Cisco: From IPExpert: int fa0/1 switchport trunk encap dot1q switchport voice vlan 250 switchport native vlan 150 switchport mode trunk From Cisco CCM SRND: int fa0/1 switchport access 150 switchport voice vlan 250 Which way are they looking for on the exam? Jonathan
[OSL | CCIE_Voice] 6608 Incoming call troubleshooting
Hello everyone, Is there a way to debug the incoming call on 6608 T1 module? Thank you, Onur.
Re: [OSL | CCIE_Voice] 6608 Incoming call troubleshooting
So I figured that the 6608 registered to sub and the HQ phone registered to pub does not work or vice versa. I get fast busy. If i register both to same server then all good. I could not understand much about the Tracy tool output what was going on. On Fri, Apr 11, 2008 at 9:26 AM, Jacob Owen [EMAIL PROTECTED] wrote: Onur, What issues are you having exactly or are you just wondering for future reference? I have made about every mistake you can on the 6608 T1 Module so I figured I might be able to help should you be having issues. --- Onur Tufekci [EMAIL PROTECTED] wrote: Hello everyone, Is there a way to debug the incoming call on 6608 T1 module? Thank you, Onur. Jacob Owen CCIE #14063 (RS, Service Provider), CCVP, CCDP __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
Re: [OSL | CCIE_Voice] VOICE Passed !!!!!!!!!!!!
I agree with you all. Who is this person any ways! On Fri, Apr 11, 2008 at 9:08 PM, Chad Stachowicz [EMAIL PROTECTED] wrote: Yeah, the commitment that IPExpert makes to their candidates is amazing. Unmatched in the industry, and highly appreciated by all its students. Thanks guys!! Cheers, Chad On Fri, Apr 11, 2008 at 5:40 PM, Jacob Owen [EMAIL PROTECTED] wrote: Yeah, Plus the so called Practical Labs from Internetwork expert haven't even been released yet. Nothing like coming on a mailing list that is supported by IPExpert who does a fantastic job supporting voice ie candidates and talking up some other vendor that has been talking about releasing a workbook since July '07. What a T-R-O-L-L --- jason sung [EMAIL PROTECTED] wrote: Dude, A monkey can pass the test given the questions. Please keep your tips and ideas to yourself. I am sure you will pass your next CCIE using cciecert.net On Fri, Apr 11, 2008 at 7:26 PM, ccie2007 [EMAIL PROTECTED] wrote: I just passed yesterday on Tokyo I am really pleasure with this achievement First my recommendation for all guys to understand all topic of the blue print from Cisco site and documentation CD as a main resource Second I use Internetwork Expert's as practical Labs which contain a lot of the real LAB concepts, great explanation for various topics and cover almost all topics in the blue print. thanks Brain Also i really recommand that you go to cciecert.net then you will get a real ccie LAB information from this site My advice to all to go through this certificate because I have now a lot of understanding of network technology My next attempt may be CCIE SP Regards #14867 CCIE Security, R/S, VOICE Hiroyasu Kato Jacob Owen CCIE #14063 (RS, Service Provider), CCVP, CCDP __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
Re: [OSL | CCIE_Voice] MGCP and SRST
check this out ccm-manager fall-back call application alternate DEFAULT http://www.cisco.com/warp/public/788/AVVID/mgcpfallback.html On Fri, Apr 11, 2008 at 11:06 PM, Paul and Bobs [EMAIL PROTECTED] wrote: BR1 Config BR1-RTR#sho run Building configuration... Current configuration : 6844 bytes ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname BR1-RTR ! boot-start-marker boot system flash:c2801-adventerprisek9_ivs-mz.124-15.T3.bin boot-end-marker ! logging buffered 51200 warnings ! no aaa new-model clock timezone AEST 10 clock summer-time AEDT recurring last Sun Oct 2:00 last Sun Mar 2:00 network-clock-participate wic 1 ip cef ! ! ! ! ip domain name iptlab.local ip auth-proxy max-nodata-conns 3 ip admission max-nodata-conns 3 ! multilink bundle-name authenticated ! isdn switch-type primary-qsig ! voice-card 0 dsp services dspfarm ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! application global service alternate DEFAULT ! ! ! crypto pki trustpoint TP-self-signed-3566742966 enrollment selfsigned subject-name cn=IOS-Self-Signed-Certificate-3566742966 revocation-check none rsakeypair TP-self-signed-3566742966 ! ! crypto pki certificate chain TP-self-signed-3566742966 certificate self-signed 01 3082024C 308201B5 A0030201 02020101 300D0609 2A864886 F70D0101 04050030 31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274 69666963 6174652D 33353636 37343239 3636301E 170D3038 30343131 30363234 31305A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649 4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D33 35363637 34323936 3630819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281 8100DFA5 C8BF2A0D 6FF5F6F4 7D50FE3D 44386FAD 7884AC3D 845C472D A70AD441 7646F9A4 B92AC281 D1FD75F4 20AE3963 01AA0B20 98CD7801 339CBB46 D55A9B88 7EF00720 5384C2E5 C197C70E 11BDE619 796E4C3D 842C5CD7 8744A436 6BEC79A1 1B1B7603 2F97C7A7 B4785F92 FA4C054C 550FCCE8 7E5F5B79 32D6E0B8 56F33AA9 9DF50203 010001A3 74307230 0F060355 1D130101 FF040530 030101FF 301F0603 551D1104 18301682 14425231 2D525452 2E697074 6C61622E 6C6F6361 6C301F06 03551D23 04183016 8014E188 94733001 2A686D55 575893B8 81DD7266 2F85301D 0603551D 0E041604 14E18894 7330012A 686D5557 5893B881 DD72662F 85300D06 092A8648 86F70D01 01040500 03818100 8775320B D78C0C5D 20E6773C 6F95A384 3ADEE764 AA82FA54 543BB4BD 7451816A 248C685F BB93E382 9F66642A 275B9A8B CC9D215B EF4EA650 74B7B945 1F398A8D D0DE53C6 D3FA0F03 966F0359 54FE3AE2 215364B6 1F5C6DFC 254D8EC4 D3FA6BE5 6B2EC3C9 3B9F7DB7 0A3C47A5 6FC9BA8E D237C971 E40FBC39 514D2CD6 9A8286AB quit ! ! ! ! ! ! controller E1 0/1/0 pri-group timeslots 1-3,16 service mgcp ! ! class-map match-any RTP match dscp ef match access-group 101 class-map match-any SIG match dscp af31 match dscp cs3 match access-group 102 ! ! policy-map LLQ class RTP priority percent 33 set dscp ef class SIG bandwidth 8 set dscp cs3 class class-default fair-queue set dscp default ! ! ! ! ! interface Loopback0 ip address 255.255.255.255 ! interface FastEthernet0/0 description BR1 LAN no ip address duplex auto speed auto ! interface FastEthernet0/0.112 encapsulation dot1Q 112 ip address 1 255.255.255.0 ip helper-address ip pim sparse-dense-mode ! interface FastEthernet0/0.113 encapsulation dot1Q 113 ip address 255.255.255.0 ip helper-address ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! interface Serial0/1/0:15 no ip address encapsulation hdlc isdn switch-type primary-qsig isdn incoming-voice voice isdn bind-l3 ccm-manager isdn outgoing display-ie isdn outgoing ie redirecting-number no cdp enable ! interface Serial0/2/0 no ip address encapsulation frame-relay no fair-queue frame-relay traffic-shaping ! interface Serial0/2/0.16 point-to-point ip address 255.255.255.252 ip pim sparse-dense-mode shutdown snmp trap link-status frame-relay interface-dlci 16 ! ! router eigrp 100 network network no auto-summary ! ip forward-protocol nd ip route 0.0.0.0 0.0.0.0 ! ! ip http server ip http secure-server ! access-list 100 deny tcp any any range 2000 2002 access-list 100 deny tcp any any range 2427 2428 access-list 100 deny udp any any range 2427 2428 access-list 100 permit ip any any access-list 101 permit udp any any range 16384 32767 access-list 102 permit tcp any any eq 1719 access-list 102 permit tcp any any eq 1718 access-list 102 permit tcp any any eq 1720 access-list 102 permit udp any any eq 1718 access-list 102 permit udp any any eq 1719 access-list 102 permit udp any any eq 1720 access-list 102 permit tcp any any eq 5060 access-list
Re: [OSL | CCIE_Voice] MGCP and SRST
I am glad it works if can keep me updated about the west AA I will really appreciate. I could not get it to run. It only says your call will be disconnected and disconnects it. Cheers, Onur Sent from my iPhone. On Apr 11, 2008, at 11:28 PM, Paul and Bobs [EMAIL PROTECTED] wrote: That command is in the config above. Wheh i enter it , it changes to application global service alternate DEFAULT I dont have cm-manager fallback-mgcp in the config either. On Sat, Apr 12, 2008 at 1:23 PM, ccievoice1 [EMAIL PROTECTED] wrote: Yes, Your config has ccm-manager fallback-mgcp and call application alternate default missing ... On Sat, Apr 12, 2008 at 11:18 AM, Onur Tufekci [EMAIL PROTECTED] wrote: it should be ccm-manager fallback-mgcp On Fri, Apr 11, 2008 at 11:16 PM, Onur Tufekci [EMAIL PROTECTED] wrote: check this out ccm-manager fall-back call application alternate DEFAULT http://www.cisco.com/warp/public/788/AVVID/mgcpfallback.html On Fri, Apr 11, 2008 at 11:06 PM, Paul and Bobs [EMAIL PROTECTED] wrote: BR1 Config BR1-RTR#sho run Building configuration... Current configuration : 6844 bytes ! version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname BR1-RTR ! boot-start-marker boot system flash:c2801-adventerprisek9_ivs-mz.124-15.T3.bin boot-end-marker ! logging buffered 51200 warnings ! no aaa new-model clock timezone AEST 10 clock summer-time AEDT recurring last Sun Oct 2:00 last Sun Mar 2:00 network-clock-participate wic 1 ip cef ! ! ! ! ip domain name iptlab.local ip auth-proxy max-nodata-conns 3 ip admission max-nodata-conns 3 ! multilink bundle-name authenticated ! isdn switch-type primary-qsig ! voice-card 0 dsp services dspfarm ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! application global service alternate DEFAULT ! ! ! crypto pki trustpoint TP-self-signed-3566742966 enrollment selfsigned subject-name cn=IOS-Self-Signed-Certificate-3566742966 revocation-check none rsakeypair TP-self-signed-3566742966 ! ! crypto pki certificate chain TP-self-signed-3566742966 certificate self-signed 01 3082024C 308201B5 A0030201 02020101 300D0609 2A864886 F70D0101 04050030 31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274 69666963 6174652D 33353636 37343239 3636301E 170D3038 30343131 30363234 31305A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649 4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D33 35363637 34323936 3630819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281 8100DFA5 C8BF2A0D 6FF5F6F4 7D50FE3D 44386FAD 7884AC3D 845C472D A70AD441 7646F9A4 B92AC281 D1FD75F4 20AE3963 01AA0B20 98CD7801 339CBB46 D55A9B88 7EF00720 5384C2E5 C197C70E 11BDE619 796E4C3D 842C5CD7 8744A436 6BEC79A1 1B1B7603 2F97C7A7 B4785F92 FA4C054C 550FCCE8 7E5F5B79 32D6E0B8 56F33AA9 9DF50203 010001A3 74307230 0F060355 1D130101 FF040530 030101FF 301F0603 551D1104 18301682 14425231 2D525452 2E697074 6C61622E 6C6F6361 6C301F06 03551D23 04183016 8014E188 94733001 2A686D55 575893B8 81DD7266 2F85301D 0603551D 0E041604 14E18894 7330012A 686D5557 5893B881 DD72662F 85300D06 092A8648 86F70D01 01040500 03818100 8775320B D78C0C5D 20E6773C 6F95A384 3ADEE764 AA82FA54 543BB4BD 7451816A 248C685F BB93E382 9F66642A 275B9A8B CC9D215B EF4EA650 74B7B945 1F398A8D D0DE53C6 D3FA0F03 966F0359 54FE3AE2 215364B6 1F5C6DFC 254D8EC4 D3FA6BE5 6B2EC3C9 3B9F7DB7 0A3C47A5 6FC9BA8E D237C971 E40FBC39 514D2CD6 9A8286AB quit ! ! ! ! ! ! controller E1 0/1/0 pri-group timeslots 1-3,16 service mgcp ! ! class-map match-any RTP match dscp ef match access-group 101 class-map match-any SIG match dscp af31 match dscp cs3 match access-group 102 ! ! policy-map LLQ class RTP priority percent 33 set dscp ef class SIG bandwidth 8 set dscp cs3 class class-default fair-queue set dscp default ! ! ! ! ! interface Loopback0 ip address 255.255.255.255 ! interface FastEthernet0/0 description BR1 LAN no ip address duplex auto speed auto ! interface FastEthernet0/0.112 encapsulation dot1Q 112 ip address 1 255.255.255.0 ip helper-address ip pim sparse-dense-mode ! interface FastEthernet0/0.113 encapsulation dot1Q 113 ip address 255.255.255.0 ip helper-address ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! interface Serial0/1/0:15 no ip address encapsulation hdlc isdn switch-type primary-qsig isdn incoming-voice voice isdn bind-l3 ccm-manager isdn outgoing display-ie isdn outgoing ie redirecting-number no cdp enable ! interface Serial0/2/0 no ip address encapsulation frame-relay no fair-queue frame-relay traffic-shaping ! interface Serial0/2/0.16 point-to-point ip address 255.255.255.252 ip pim sparse-dense-mode shutdown snmp trap link-status frame-relay interface-dlci 16 ! ! router eigrp 100 network
Re: [OSL | CCIE_Voice] Fwd: bandwidth usage
I did not get any answers for my couple of questions either. I tried fixing the SRST AA question that is in the study material but no luck. It still does not work I just gave up on it until I see someone is trying it. On Fri, Mar 28, 2008 at 4:49 PM, jason sung [EMAIL PROTECTED] wrote: Mark, can you please shed some light on this question. Either I am asking someting so stupid nobody wants to answer OR I am asking something impossible? Basically I am trying to send few g711 calls and check the bandwidth and than compare it with few g729 calls. -- Forwarded message -- From: jason sung [EMAIL PROTECTED] Date: Thu, Mar 27, 2008 at 9:25 PM Subject: bandwidth usage To: CCIE Maillist ccie_voice@onlinestudylist.com I have been trying different commands, but none of them give me a definative answer on HOW TO CHECK BANDWIDTH USAGE on the router? Does anybody have any ideas? I tried the show policy-map interface command but that does not show me what I want.
Re: [OSL | CCIE_Voice] Correct Partitions and CSS
I would not sperate the Internal and mwi partitions unless it is necessary. It is just personal pref but the reason you have mwi partition is to block users from dialing that extension same with the others. It is just time consuming depending on how fast you are. onur. On Wed, Mar 26, 2008 at 6:41 AM, Victor Esperanza [EMAIL PROTECTED] wrote: In regards to Partitions and Calling Search spaces for a real word deployment (single site in this example). Can someone verify that my PT and CSS's are correct. Partitions: 911_pt local_pt ld_pt int_pt Phone_pt Internal_pt (For VM Pilot and vm ports) PSTN_PT (for PSTN_CSS) MWI_PT local_css (has 911_pt and local_pt and internal_pt) ld_css (911_pt, local_pt, ld_pt and internal_pt) int_pt ((911_pt, local_pt, ld_pt, int_pt and internal_pt) internal_css (has phones_pt, internal_pt) PSTN_CSS (has pstn_pt, and is the css I put on gateway page for incoming calls) mwi_css (phones_pt and mwi_pt) For Voicemail. Do the users_pt need to have access to both vm pilot and vm ports? Or just vm Pilot? And I assume that the internal_css needs the phones_pt in it's css. For PSTN, if the incoming CSS on gateway page has pstn_css which only has pstn_pt, is this ok? What access does that internal CSS need? Just admitance into the system? Or do we need to put phones_pt in that CSS, so calls from the pstn can reach the phones? thanks to all in advance, -- In a rush? Get real-time answers with Windows Live Messenger.http://www.windowslive.com/messenger/overview.html?ocid=TXT_TAGLM_WL_Refresh_realtime_042008
Re: [OSL | CCIE_Voice] bandwidth usage
show policy-map interface fast0/0 should can you confirm? On Wed, Mar 26, 2008 at 10:55 AM, jason sung [EMAIL PROTECTED] wrote: How can I check bandwidth usage on a policy map? For example I want to send a regular g729 call and then I want to send a compressed g729 call. Basically I want to compare the bandwidth usage between the two. TIA.
[OSL | CCIE_Voice] Fwd: Question 22 from Lab 5 on Proctorlabs Web site
-- Forwarded message -- From: Onur Tufekci [EMAIL PROTECTED] Date: Wed, Mar 26, 2008 at 11:21 AM Subject: Re: [OSL | CCIE_Voice] Question 22 from Lab 5 on Proctorlabs Web site To: Allen Rounsavell [EMAIL PROTECTED] Please see if this helps. Byt the way they changed the search engine on Cisco website so I could not search for anything. That sucks. Also can you let us know if it works? http://www.cisco.com/en/US/docs/ios/11_3/feature/guide/ftpserve.html On Wed, Mar 26, 2008 at 11:02 AM, Allen Rounsavell [EMAIL PROTECTED] wrote: Cannot use TFTP to push license file to CUE have to use FTP. Had to use my own FTP client to get the file over. The guide suggests U can setup FTP locally on the CME. -- *From:* Onur Tufekci [mailto:[EMAIL PROTECTED] *Sent:* Wednesday, March 26, 2008 10:43 AM *To:* Allen Rounsavell *Subject:* Re: [OSL | CCIE_Voice] Question 22 from Lab 5 on Proctorlabs Web site it is tftp server that is for CallManager I think. You can find it under c drive/ prog flies/ cisco/tftp root. TFTP is enabled by default on CCM as you can imagine. On Wed, Mar 26, 2008 at 10:38 AM, Allen Rounsavell [EMAIL PROTECTED] wrote: Question 22 from Lab 5 on Proctorlabs Web site suggests using FTP locally from CME/CUE router. The license file is there however I cannot do the ftp-server enable cmd suggested from the solution guide on the router. Generally used my ftp software that I had locally on my laptop. Which will not be available on the lab:( What am I missing in using the ftp-server enable cmd from the CME/CUE router?? Would love to know what I am messing up so I may be able to do this in the real lab if the problem is presented? Tks, Allen
[OSL | CCIE_Voice] Fwd: Question 22 from Lab 5 on Proctorlabs Web site
-- Forwarded message -- From: Allen Rounsavell [EMAIL PROTECTED] Date: Wed, Mar 26, 2008 at 11:38 AM Subject: RE: [OSL | CCIE_Voice] Question 22 from Lab 5 on Proctorlabs Web site To: Onur Tufekci [EMAIL PROTECTED] Yep that's what I attempted. However it would not take the cmd ftp-server enable or ftp-server topdir disk0:/syslogd.dir in conf t. I was wondering if I had something else to enable globally before this would work? Did not recognize the cmd at all. -- *From:* Onur Tufekci [mailto:[EMAIL PROTECTED] *Sent:* Wednesday, March 26, 2008 11:21 AM *To:* Allen Rounsavell *Subject:* Re: [OSL | CCIE_Voice] Question 22 from Lab 5 on Proctorlabs Web site Please see if this helps. Byt the way they changed the search engine on Cisco website so I could not search for anything. That sucks. Also can you let us know if it works? http://www.cisco.com/en/US/docs/ios/11_3/feature/guide/ftpserve.html On Wed, Mar 26, 2008 at 11:02 AM, Allen Rounsavell [EMAIL PROTECTED] wrote: Cannot use TFTP to push license file to CUE have to use FTP. Had to use my own FTP client to get the file over. The guide suggests U can setup FTP locally on the CME. -- *From:* Onur Tufekci [mailto:[EMAIL PROTECTED] *Sent:* Wednesday, March 26, 2008 10:43 AM *To:* Allen Rounsavell *Subject:* Re: [OSL | CCIE_Voice] Question 22 from Lab 5 on Proctorlabs Web site it is tftp server that is for CallManager I think. You can find it under c drive/ prog flies/ cisco/tftp root. TFTP is enabled by default on CCM as you can imagine. On Wed, Mar 26, 2008 at 10:38 AM, Allen Rounsavell [EMAIL PROTECTED] wrote: Question 22 from Lab 5 on Proctorlabs Web site suggests using FTP locally from CME/CUE router. The license file is there however I cannot do the ftp-server enable cmd suggested from the solution guide on the router. Generally used my ftp software that I had locally on my laptop. Which will not be available on the lab:( What am I missing in using the ftp-server enable cmd from the CME/CUE router?? Would love to know what I am messing up so I may be able to do this in the real lab if the problem is presented? Tks, Allen
[OSL | CCIE_Voice] SRST AA
I can not get this script to work it just prompts your call will be disconnected and call gets disconnected. Is there any one with any information regarding how to troubleshoot this? I can only imagine either no had this problem or no one cares to answer. This is my second try asking. application service aa flash:its-CISCO.2.0.1.0.tcl param operator 2001 paramspace english language en paramspace english index 0 paramspace english location flash: paramspace english prefix en param aa-pilot 2000 !
Re: [OSL | CCIE_Voice] SRST AA
Thank you Justin, I was about to give up trying. I got the translation rule running and aa answering the call but it says your call will be disconnected and disconnects the call. Di you ever get it to work? I also tried to look at debug voice application but no luck! Onur. P4-BR1-RTR#debug dialpeer This CLI command is now 'debug voip dialpeer all' P4-BR1-RTR# *Mar 25 01:06:06.483: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now connected to 911 N/A *Mar 25 01:06:08.287: %ISDN-6-DISCONNECT: Interface Serial0/0/0:0 disconnected from 911 , call lasted 1 seconds P4-BR1-RTR#debug isdn q931 debug isdn q931 is ON. P4-BR1-RTR# P4-BR1-RTR# P4-BR1-RTR# P4-BR1-RTR# P4-BR1-RTR# *Mar 25 01:07:42.127: ISDN Se0/0/0:23 Q931: RX - SETUP pd = 8 callref = 0x0011 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 = 0x0081, '911' Plan:Unknown, Type:Unknown Called Party Number i = 0xA1, '6175242000' Plan:ISDN, Type:National *Mar 25 01:07:42.143: ISDN Se0/0/0:23 Q931: TX - CALL_PROC pd = 8 callref = 0x8011 Channel ID i = 0xA98381 Exclusive, Channel 1 *Mar 25 01:07:42.143: ISDN Se0/0/0:23 Q931: TX - CONNECT pd = 8 callref = 0x8011 *Mar 25 01:07:42.159: ISDN Se0/0/0:23 Q931: RX - CONNECT_ACK pd = 8 callref = 0x0011 *Mar 25 01:07:42.159: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now connected to 911 N/A *Mar 25 01:07:43.959: %ISDN-6-DISCONNECT: Interface Serial0/0/0:0 disconnected from 911 , call lasted 1 seconds *Mar 25 01:07:43.963: ISDN Se0/0/0:23 Q931: TX - DISCONNECT pd = 8 callref = 0x8011 Cause i = 0x8090 - Normal call clearing *Mar 25 01:07:43.971: ISDN Se0/0/0:23 Q931: RX - RELEASE pd = 8 callref = 0x0011 *Mar 25 01:07:43.975: ISDN Se0/0/0:23 Q931: TX - RELEASE_COMP pd = 8 callref = 0x8011 voice translation-rule 1 rule 1 /617524\(2...\)/ /\1/ voice translation-profile strip translate called 1 ! dial-peer voice 2 pots translation-profile incoming strip service aa incoming called-number . port 0/0/0:23 ! On Mon, Mar 24, 2008 at 7:17 PM, Justin Steinberg [EMAIL PROTECTED] wrote: try this... change param aa-pilot 2000 to param aa-pilot 61752X2000 where X is your pod number alternatively, use a translation rule/profile to convert 10 digits to 4 digits On Mon, Mar 24, 2008 at 8:14 PM, Onur Tufekci [EMAIL PROTECTED] wrote: I can not get this script to work it just prompts your call will be disconnected and call gets disconnected. Is there any one with any information regarding how to troubleshoot this? I can only imagine either no had this problem or no one cares to answer. This is my second try asking. application service aa flash:its-CISCO.2.0.1.0.tcl param operator 2001 paramspace english language en paramspace english index 0 paramspace english location flash: paramspace english prefix en param aa-pilot 2000 !
Re: [OSL | CCIE_Voice] Section 8.10 SRST AA
I might have found the problem!
Does anyone know what this command signifies and how it should be used?
*paramspace english index *
Onur.
On Fri, Mar 21, 2008 at 11:22 AM, Onur Tufekci [EMAIL PROTECTED]
wrote:
I configured the section 8.10 as recommended. My calls are going to script
but the message stating that it will be disconnected and disconnects my
call. I tried to debug it but do not understand what is going on. Is there
anyone having the same problem? I researched all internet there is one post
about the same problem from 2006 but no solution. Also I am not able to
insert command param cm-pilot 2000.
application
service aa flash:its-CISCO.2.0.1.0.tcl
param operator 2001
paramspace english index 1
paramspace english language en
paramspace english location flash:
paramspace english prefix en
param aa-pilot 6173202000
dial-peer voice 3000 pots
service aa
incoming called-number 6175202000
port 0/0/0:23
forward-digits all
***
*
*Mar 21 03:51:56.823: //-1//AFW_:/AFW_DataList_GetFirst: Elem =
0x47DBD278, with Instance = 0x47D7C8C4
P20-BR1-RTR#Received
//-1//AFW_:/AFW_Process_GetPriorityQEvent:
Event[APP_EV_TCLMODULE_DONE(165)] {
//-1//AFW_:/AFW_Process_GetPriorityQEvent: EXECENV[0x47D7C8C4][Default]
//-1//AFW_:/AFW_Process_GetPriorityQEvent:
MOD[TclModule_460C77A8_0_5846924] (
//-1//AFW_:/AFW_Process_GetPriorityQEvent: )
//-1//AFW_:/AFW_Process_GetPriorityQEvent: }
//29//AFW_:/AFW_M_TclModule_EventPreProcess:
//29//AFW_:/AFW_Object_WalkListeners:
//29//AFW_:/AFW_M_Object_ShowListeners: START
//29//AFW_:/AFW_M_Object_ShowListeners: END
//-1//AFW_:/AFW_Instance_DecrRefCount: Object: 0x47D64F98, Type: Event,
RefCount: 0
//29//AFW_:/AFW_M_Event_Free:
//29//AFW_:/AFW_M_Event_Free: MODULEDONEEVENT for a Module:
TclModule_460C77A8_0_5846924
//-1//AFW_:/AFW_M_Object_UnSetExecEnv: ObjCount: 0, CmdPending 0
//-1//AFW_:HN0059378C:/AFW_M_Event_Free: ExecEnv objCount: 0
//-1//SERV:/AFW_Service_ReleaseExecEnv: Script Name = Default cache = true
calls = 0
//29//AFW_:/AFW_ExecEnv_UnSetRoot: Execenv = 0x47D7C8C4
//-1//AFW_:/AFW_Process_UnLock: pProcess(0x46F1BB08)=0
//-1//AFW_:/AFW_Instance_DecrRefCount: Object: 0x46F1BB08, Type: Process,
RefCount: 1
//-1//AFW_:/AFW_Instance_DecrRefCount: Object: 0x470A0E0C, Type:
DataArray, RefCount: 0
//-1//AFW_:/AFW_Event_New:
//-1//AFW_:/AFW_Class_Allocate: Malloc Data Space: Event(Size=2072)
//29//AFW_:/AFW_Process_GetCcqEvent: Received
//-1//AFW_:/AFW_Process_GetCcqEvent: Event[CC_EV_CALL_HANDOFF_RETURN(21)]
{
//-1//AFW_:/AFW_Process_GetCcqEvent: EXECENV[0x47D7C7EC][aa]
//-1//AFW_:/AFW_Process_GetCcqEvent: MOD[Handoff_47D6CDB4_0_5846916] (
//-1//AFW_:/AFW_Process_GetCcqEvent: )
//-1//AFW_:/AFW_Process_GetCcqEvent: }
//29//Hand:/AFW_M_Handoff_EventPreProcess:
//29//AFW_:/AFW_ExecEnv_SetModuleScope: NULL ---
Handoff_47D6CDB4_0_5846916
//29//Hand:/AFW_M_Handoff_Action:
//29//Hand:/AFW_Handoff_Action:
//29//Hand:/act_return:
//29/FD3F3E92800E/AFW_:/AFW_M_Leg_SetExecEnv:
//29//AFW_:/AFW_ExecEnv_IncrPendingCmd: PendingCmdCount: 2
//-1/FD3F3E92800E/AFW_:LP:EE47D7C7EC000:LG29:/AFW_M_Object_SetExecEnv:
ObjCount: 3, CmdPending 2
//-1//AFW_:/AFW_DataArray_ElementSet: Adding param: LEG_29, type: Leg
//-1//AFW_:/AFW_Instance_IncrRefCount: Object: 0x460BBF78, Type: Leg,
RefCount: 2
//29//Hand:/ah_transfer_inc_return: CallID 29 returned to xfer
//29//Hand:/ah_transfer_complete: Returning 1 objects
//-1//AFW_:/AFW_Event_New:
//-1//AFW_:/AFW_Class_Allocate: Malloc Data Space: Event(Size=2072)
//29//AFW_:/AFW_Module_ReturnArgEv:
//-1//AFW_:/AFW_DataList_New:
//-1//AFW_:/AFW_Class_Allocate: Malloc Data Space: DataList(Size=40)
//-1//AFW_:/AFW_DataList_Enqueue: Trying to add element to a list
//-1//AFW_:/AFW_DataList_GetWrapper: Looking for list element 0x460BBF78
//-1//AFW_:/AFW_DataList_Enqueue: Adding element:
//-1//AFW_:/AFW_Instance_IncrRefCount: Object: 0x460BBF78, Type: Leg,
RefCount: 3
//-1//AFW_:/AFW_DataList_GetFirst: Elem = 0x483544EC, with Instance =
0x460BBF78
//29//AFW_:/AFW_Module_ReturnArgEv: Return List (remove=TRUE){LEG[29
][LEG_INCACKED(2)][Cause(0)]}
//-1//AFW_:/AFW_DataArray_ElementDelete: param name LEG_
%SYS-3-CPUHOG: Task is running for (2004)msecs, more than (2000)msecs
(1/0),process = AFW_application_process.
-Traceback= 0x403030EC 0x413D82D4 0x413DA5D8 0x413D8E78 0x413C71F8
0x413C7300 0x41CDBE30 0x41CDA960 0x41CD8A6C 0x41CDA130 0x41CDA1C8 0x41D00250
0x41D00370 0x41D004A4 0x41D00674 0x41CD9938 29
//-1//AFW_:/AFW_Instance_DecrRefCount: Object: 0x460BBF78, Type: Leg,
RefCount: 2
//29//AFW_:/AFW_Module_UnListen: NumObjects: 0
//29//AFW_:/AFW_ExecEnv_SetModuleScope: Handoff_47D6CDB4_0_5846916 ---
NULL
//-1//AFW_:/AFW_Instance_DecrRefCount: Object: 0x47D657B0, Type: Event,
RefCount: 0
//29//AFW_
[OSL | CCIE_Voice] FRTS and/or MLPPP
Is there any way figuring out what to use between the sites? If you can not use FRF.12 with the low speed link to one site and you are free to use FRF.12 to another low speed site connection. What is the approach that we need to take? Is it ok to use FRTS without FRF.12 on low speed links? According to QoS SRND we need to use it. MLPPP is another option to use but mixture of it with FRTS is not useful on the main router if you are running FRTS to one site and MLPP to other site. Is there a general approach for all this? Thank you in advance, Onur.
Re: [OSL | CCIE_Voice] FRTS and/or MLPPP
Thank you so much for the reply. I just started using this forum and doing the labs provided by IPExpert with rental Vracks. So is it true that when MLPPP used for one link the other link also has to be MLPPP. The dilemma comes from the command that I tried to enter under serial interface 0/0/0 frame-relay traffic-shaping. For the MLPPP we need to enter this command but if we already are using the FRTS then error message states that we need to remove other traffic shaping commands. Regards, Onur. On Thu, Mar 20, 2008 at 12:16 PM, ccievoice1 [EMAIL PROTECTED] wrote: well, if you need to do interleaving, then MLPPP. On Fri, Mar 21, 2008 at 12:05 AM, Edward French [EMAIL PROTECTED] wrote: My feeling is that if you can not use FRF.12 everywhere use MLPPP Ed - Original Message From: Onur Tufekci [EMAIL PROTECTED] To: ccie_voice@onlinestudylist.com Sent: Thursday, March 20, 2008 10:15:58 AM Subject: [OSL | CCIE_Voice] FRTS and/or MLPPP Is there any way figuring out what to use between the sites? If you can not use FRF.12 with the low speed link to one site and you are free to use FRF.12 to another low speed site connection. What is the approach that we need to take? Is it ok to use FRTS without FRF.12 on low speed links? According to QoS SRND we need to use it. MLPPP is another option to use but mixture of it with FRTS is not useful on the main router if you are running FRTS to one site and MLPP to other site. Is there a general approach for all this? Thank you in advance, Onur.
Re: [OSL | CCIE_Voice] R: tech-prefix on gatekeeper
Hi, Can you also send show gatekeeper endpo.. output and show gatekeeper gw... output. Looks like it is matching to first available local zone. Under your GK-RL / RG you are adding the prefix 2#. Gatekeeper is not receiving it looks like. Thanks, Onur. On Thu, Mar 20, 2008 at 12:19 PM, Fernando Ferraioli [EMAIL PROTECTED] wrote: This is the debug: Mar 20 16:15:00.134: gk_handle_timers: managed timer expired 0x45A9C9D0 Mar 20 16:15:03.270: gk_process: QUEUE_EVENT (minor 0) wakeup Mar 20 16:15:03.678: gk_process: QUEUE_EVENT (minor 0) wakeup Mar 20 16:15:03.682: gk_rassrv_arq: arqp=0x4607B50C, crv=0x2, answerCall=0 Mar 20 16:15:03.682: gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC Mar 20 16:15:03.682: gk_dns_query: No Name servers Mar 20 16:15:03.682: rassrv_get_addrinfo: (3003) Tech-prefix match failed. Mar 20 16:15:03.682: rassrv_get_addrinfo: (3003) Matched zone prefix 3 and remainder 003 Mar 20 16:15:03.682: gk_rassrv_get_ingress_network: returning default ingress network = 1 Mar 20 16:15:03.682: rassrv_arq_select_viazone: about to check the source side, src_zonep=0x471A44CC Mar 20 16:15:03.682: rassrv_arq_select_viazone: matched zone is HQ-RTR1, and z_invianamelen=0 Mar 20 16:15:03.682: rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x471A44CC Mar 20 16:15:03.682: rassrv_arq_select_viazone: matched zone is HQ-RTR1, and z_outvianamelen=0 Mar 20 16:15:03.682: rassrv_get_addrinfo: No tech prefix Mar 20 16:15:03.682: rassrv_get_addrinfo: Alias not found Mar 20 16:15:03.682: rassrv_get_addrinfo: (3003) unknown address and no default technology defined. Mar 20 16:15:03.682: gk_rassrv_sep_arq: rassrv_get_addrinfo() failed (return code = 0x103) Mar 20 16:15:15.134: gk_process: got a TIMER event Mar 20 16:15:15.134: gk_handle_timers Mar 20 16:15:15.134: gk_handle_timers: managed timer expired 0x45A9C9D0 Thanks FF -- *Da:* Devildoc [mailto:[EMAIL PROTECTED] *Inviato:* gio 20/03/2008 13.27 *A:* Fernando Ferraioli; ccie_voice@onlinestudylist.com *Oggetto:* RE: [OSL | CCIE_Voice] tech-prefix on gatekeeper On the H323 GK-controlled trunk, make sure you uncheck the Wait for far end capability exchange and uncheck MTP requirement. Also, do a debug gatekeepr main 10 on the HQ-RTR and post your output here. JD -- Date: Thu, 20 Mar 2008 08:33:38 +0100 From: [EMAIL PROTECTED] To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] tech-prefix on gatekeeper I'm configuring GK the question required that I don't have to use a default tech prefix. If I call from CME to CM (DN 1001) it's ok, from CM to CME it doesn't work. My conf: HQ gatekeeper zone local CCM-GK ipexpert.com 172.21.100.1 zone remote PSTN ipexpert.com 10.21.200.2 1719 zone local BR2-GK ipexpert.com zone prefix PSTN 011* zone prefix CCM-GK 1... zone prefix BR2-GK 3... no shutdown CME: interface Loopback0 ip address 172.21.102.1 255.255.255.255 ip ospf network point-to-point h323-gateway voip interface h323-gateway voip id BR2-GK ipaddr 172.21.100.1 1719 h323-gateway voip h323-id CME h323-gateway voip tech-prefix 2# h323-gateway voip bind srcaddr 172.21.102.1 CCM: Tech-prefix 2# On CME I tried num-exp 2#3… 3… and many others! Any Ideas? Thanks Fernando -- Need to know the score, the latest news, or you need your Hotmail(R)-get your fix. Check it out. http://www.msnmobilefix.com/Default.aspx
