Re: [OSL | CCIE_Voice] Dtmf problem with MVA
I believe you have not applied the *allow-connection h323 to h323* command under *voice service voip *on your h323 gateway. On Thu, Dec 8, 2011 at 11:57 AM, Ray jonha...@yahoo.com wrote: also make sure the mobile voice access is setup under media resources and also make sure the 4 digit ext on the RDP has a check in the check box. *From:* Mohammed Al Baqari baqari.voic...@gmail.com *To:* 'ccielabrat' ccielab...@gmail.com; ccie_voice@onlinestudylist.com *Sent:* Thursday, December 8, 2011 12:13 AM *Subject:* Re: [OSL | CCIE_Voice] Dtmf problem with MVA Use the command “debug voice application” and post the output after your enter the DTMF digits. Regards, Mohammed Al Baqari *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *ccielabrat *Sent:* Tuesday, December 06, 2011 11:59 PM *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] Dtmf problem with MVA I have MVA configured on my h323 router, with the appropriate dial peers as per the Cucm help pages. I am able to call into the piloting dn and I can hear the MVA application prompt me for my pin. When I press any digits, they are not recognized. I've tried to force a g711 codec and ensured Dtmf-relay is configured, but it doesn't change the problem. Has anyone run into this issue? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Multicast MOH with Branch IP Phones
Thank you all for reply. Mohd Baqari : All are 7961 Phones. Brajesh kumaR : MRGL has been assign to all phone's device pools ( not on individual phones) Julien Krieger : BR2 Ph-1 placing BR2 Ph-2 on hold *or* BR1 Ph-1 placing BR1 Ph-2 on hold *or* BR1 Phones placing BR2 Phones and vice versa. Ken Wyan : multicast moh 239.1.1.1 port 16384 route #fast ethernet IP # #Loop0 IP# - fast eth is a Voice sub Interface IP add. I tried adding ip pim dense-mode on voice sub interface but no luck. -SONU- On Mon, Nov 28, 2011 at 4:45 PM, Ken Wyan kew...@gmail.com wrote: Pls check ip pim dense-mode configured for Voice vlan interface as well multicast moh 239.1.1.1 port 16384 route #fast ethernet IP # #Loop0 IP#also add (br1 phones) vlan interface ip Thanks On Mon, Nov 28, 2011 at 2:11 PM, Julien Krieger krieger.jul...@gmail.comwrote: Hi, Who is putting who on hold? Are you using the same MRGLs for IPPhone that you used for the Gateway? Julien 2011/11/28 brajesh kumaR brjku...@gmail.com Can you please check if branch sites phones have MRGL assigned. What is the region setting between BR1 phone and moh server.If it is g729 then transcoder will be invoked to send spoofed moh stream to branch phone using flash moh file. Regards, Brajesh. On Mon, Nov 28, 2011 at 7:06 AM, Pradeep Kumar Sharma sonu.netwo...@gmail.com wrote: Hello Experts, I am working on Branch site Multicast MOH example and spoofing the moh from router flash memory. My MOH server inter-region setting is G711 for all branch sites. When placing a PSTN call on hold from a Branch phone, the Moh spoofing is working perfectly. When placing a internal branch to branch call on hold, moh is not working and i am getting a dead-air. At the same time:- - There is no debug output on the branch router. (debug ccm-manager music-on-hold all) - show perf query class Cisco MOH Device on call manager CLI showing MOHMulticastResourceActive = 1 - ccm-manager music-on-hold applied on the Branch router. - multicast moh 239.1.1.1 port 16384 route #fast ethernet IP # #Loop0 IP# applied on branch router. - Only 1 MOH server in MRGL with Multicast enabled. (max hop = 1) - Multicast routing is disabled on HQ and Branch Site for moh spoofing. Any thought ? -SONU- ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com http://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com http://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com http://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Uploading Promtps
Yes , Its possible. Once you click on Play/Record button to record a greeting, you will be able to see a Java recorder. Click on Option and Select Open file. -SONU- On Mon, Nov 28, 2011 at 7:46 PM, datucha123 datucha123 datucha...@gmail.com wrote: Hello, Is it possible to upload the custom prompts to Unity Connection? I know, that CUC has the integrated JAVA Applet, which allows you to record the prompts either through the PC or IP Phone. But what if I have a prompt file already located at my PC, and I want to use it in Unity Connection as Greeting ? How to upload the Prompt Files to CUC? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CUCM/UCCX Intergration
It seems your UCCX server is not integrated properly with CUCM. Use http://ccie-musketeers.blogspot.com/search/label/UCCX and fix it. -SONU- On Sun, Nov 27, 2011 at 12:37 PM, CCIEVoiceKP ccievoic...@gmail.com wrote: Have you re-syched the data yet? Subsystems -- Unified CM Telephony -- Data Resync KP On Sat, Nov 26, 2011 at 9:02 PM, Randall Crumm rrcr...@yahoo.com wrote: HI, I am working on the New Lab 1 UCCX portion. I am suppose to add an UCCX extension to each user page, but the field to add the UCCX exntension is not there. I also did not see a rmjtapi user in CUCM. This is the user in the intergration page on UCCX. Any thoughts? Randall ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com http://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Multicast MOH with Branch IP Phones
Hello Experts, I am working on Branch site Multicast MOH example and spoofing the moh from router flash memory. My MOH server inter-region setting is G711 for all branch sites. When placing a PSTN call on hold from a Branch phone, the Moh spoofing is * working* perfectly. When placing a internal branch to branch call on hold, moh is *not working*and i am getting a dead-air. At the same time:- - There is no debug output on the branch router. (debug ccm-manager music-on-hold all) - *show perf query class Cisco MOH Device *on call manager CLI showing *MOHMulticastResourceActive = 1* - ccm-manager music-on-hold applied on the Branch router. - multicast moh 239.1.1.1 port 16384 route #fast ethernet IP # #Loop0 IP# applied on branch router. - Only 1 MOH server in MRGL with Multicast enabled. (max hop = 1) - Multicast routing is disabled on HQ and Branch Site for moh spoofing. Any thought ? -SONU- ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Dialed Display Number.
Correct, i also apply this command globally. Well the point is, Its very easy to configure TEHO with Called Party Transformation Pattern. But If proctor is expecting to maintain the Called Display number in case of TEHO,i am not suppose to use Called Party Transformation Pattern. because it'll take preference over *no supplementary-service h225-notify cid-update* command or Route Pattern Digit manipulation and change the display number number on the calling phone. So if nothing is mentioned in the lab about display number , we have to take our own decision whether we want to maintain the display number or not. What you guys will recommend, should it go for CPTP Digit manipulation dial plan or RL/RG Digit manipulation Dial plan ? On Thu, Nov 24, 2011 at 6:16 PM, datucha123 datucha123 datucha...@gmail.com wrote: Can we also use the CUCM Service Parameter - Always display dialed digits or not? On Thu, Nov 24, 2011 at 4:39 PM, Gurpreet Singh Kukreja tycoononway1...@gmail.com wrote: Yes, I think he means to enter these commands on the gateway to disable sending of H.225 messages with caller-ID updates. You can enter the command on both, under the dial-peer under the global config mode. I do this under the global config mode; *voice service voip* *no supplementary-service h225-notify cid-update* Regards Gurpreet On Thu, Nov 24, 2011 at 3:53 AM, datucha123 datucha123 datucha...@gmail.com wrote: What do you mean under the No Supplementary service? So you mean *no supplementary-service h225-notify cid-update*? On Thu, Nov 24, 2011 at 4:00 AM, Gurpreet Singh Kukreja tycoononway1...@gmail.com wrote: Hi, This depends on what is asked in the Lab. If the lab tells you to display 10 digit number then you do it otherwise you don't. Logically, a user when dials a 7 digit number has no idea where the call is going from. For him, he is dialing a 7 digit #. So ideally, it should always display 7 digit called # in this situation. If nothing is mentioned in lab, i would leave it to 7 digit called #. HTH Regards Gurpreet On Mon, Nov 21, 2011 at 2:00 PM, Pradeep Kumar Sharma sonu.netwo...@gmail.com wrote: Hello Guys, Do we really have to take care of dialed display number on the phone while call is rerouting to the secondary path in CCIE Lab. For Example: HQ phone should be able make Local Calls. If HQ gateway is not available, the call should reroute via BR1 Gateway. HQ local number is : 394-2XXX (+1-212-394-2XXX). While rerouting this call via BR1, we have to prefix this number with 1212 in RL/RG . (91212 in case BR1 is H.323) At this point the dialed display number on the phone will automatically change from 393-2XXX to 12123942XXX or something like that. I know we can take care of it using no supplementary command and route pattern. But my question is , do we really have to worry about this ? -SONU- ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com http://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com http://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Dialed Display Number.
Hello Guys, Do we really have to take care of dialed display number on the phone while call is rerouting to the secondary path in CCIE Lab. For Example: HQ phone should be able make Local Calls. If HQ gateway is not available, the call should reroute via BR1 Gateway. HQ local number is : 394-2XXX(+1-212-394-2XXX). While rerouting this call via BR1, we have to prefix this number with 1212 in RL/RG . (91212 in case BR1 is H.323) At this point the dialed display number on the phone will automatically change from 393-2XXX to 12123942XXX or something like that. I know we can take care of it using no supplementary command and route pattern. But my question is , do we really have to worry about this ? -SONU- ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Why my outbound calls (using RGs) to pstn doesn't work?
The Problem is , your Route List is not getting registered with any of cucm (pub or sub) The call will never reach to the gateway until Route List get registered. There are few things you need to verify. 1- You have atleast one RG added in the Router List. 2- Try to reset it and see if its getting registered. 3- If it doesnt work. Delete the Route List and recreate it with some different name. On Wed, Nov 16, 2011 at 8:14 AM, Shirley, Kris C. kcshir...@tmhs.orgwrote: What is your IOS version? *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Voiper *Sent:* Tuesday, November 15, 2011 1:43 PM *To:* Rrcrumm *Cc:* ccie_voice@onlinestudylist.com; kew...@gmail.com *Subject:* Re: [OSL | CCIE_Voice] Why my outbound calls (using RGs) to pstn doesn't work? Unfortunately, I am still stuck. I took two days off from work (Mon, Tue) to work on it and still no success Will keep trying. Thanks, Voiper On Tue, Nov 15, 2011 at 2:35 PM, Rrcrumm rrcr...@yahoo.com wrote: We're u able to resolve your issue? Sent from my iPhone On Nov 15, 2011, at 10:32 AM, Voiper datapack...@gmail.com wrote: Hi Friends: Thank you all for you quick and helpful suggestions. It show you are all very focused and have the determination to get the coveted number. I need to step up my efforts after seeing how you all quickly helped with feedback. All your help is truly appreciated. Voiper @RRCRUMM: Did you reset the h323 gw in cucm? Do you see an ip address towards the upper portion of the page above the name/ ip address field? - The h323 gw is configured and I see the ip address outbound calls to pstn using h323 or mcgp gw directly is working @Ashwani ash_r...@hotmail.com Did you bind source interface on HQ router for H323? And same interface's IP assigned on CUCM? Also did you have incoming dial-peer from CUCM? Paste your HQ config here. - interface FastEthernet0/0.20 description VOICE encapsulation dot1Q 20 ip address 10.10.200.3 255.255.255.0 ip helper-address 10.10.210.10 h323-gateway voip bind srcaddr 10.10.200.3 HQ router config is attached. @Ken Wyan kew...@gmail.com Pls check your route list rl-test config digit manipulations etc.. in rg-test (under rl-test). If you have done any digit manipulations in rg-test (under rl-test) , your RP digit-manipulations will not work Add rg-test as Local Route Group for Device Pool In addition to debug isdn --- debug ccapi inout , dialpeer may help to isolate issue - No digit manipulations on any of the above. I tried to keep it as simple as possible ran above debug and nothing gets generated. @Pradeep Kumar Sharma sonu.netwo...@gmail.com Verify the registration status of your Route List. Try to reset it once. Run the debug voice ccapi inout on H323 Gw and see if the outbound calls really hitting to the gateway on not ? - route list status unknown. Reset many times. What is your registration status? CCAPI: debug voip ccapi inout is ON (filter is OFF) CCH323 SPI: H225 State Machine tracing is enabled (filter is OFF) - above debug show nothing happening. Call just not reaching the h323 gw @Mohd Baqari baqari.voic...@gmail.com On you gateway, do the following: 1. Debug cch323 h225 messages. Check if any h323 signaling is reaching the gateway. 2. If signaling is received successfully, run the command debug voice dialpeer. Check which dial peer is matched and whats the configuration of it 3. If no h323 signaling is received, post cucm traces after making a call - above debug show nothing happening. Call just not reaching the h323 gw Do I need to run RTMT for trace? I installed it on my desktop and ran trace 'Collect Files' (attached). First time using, so I don't know if I got the right one? ccm0007.txt ccm0007.txt SDL002_100_10.txt HQ-RTR.txt Methodist. Leading Medicine. Recognized by *U.S.News World Report* as one of America's Best Hospitals in 13 specialties. Named to *FORTUNE* magazine's 100 Best Companies to Work For list six years in a row. Designated as a Magnet hospital for excellence in nursing. Visit us at methodisthealth.com. Follow us at twitter.com/MethodistHosp and www.facebook.com/MethodistHospitalhttp://www.facebook.com/methodisthospital. ***CONFIDENTIALITY NOTICE*** This e-mail is the property of The Methodist Hospital and/or its relevant affiliates and may contain restricted and privileged material for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender and delete all copies of the message. Thank you. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job
Re: [OSL | CCIE_Voice] Voice Class Codec Question
Thanks Everyone. Thanks Kshitij for this gr8 explanation. :) On Mon, Nov 14, 2011 at 6:06 PM, Kshitij Singhi martinian.ksin...@gmail.com wrote: Hi Sonu, So for a call on the Branch site, I can think of a few scenarios: 1. Call coming in for an IP Phone at the branch site from the PSTN. 2. Call being made from a phone at Branch Site 1 to another phone at Branch Site 1. 3. Call being made from a phone at Branch Site X to a phone at Branch Site 1. 4. Call coming in for an IP Phone at the branch site from the PSTN rolling over to Unity. 5. Call coming in via a GK to a phone at Branch Site 1 For 2. and 3. we don't need to worry about the dial peers. This is SCCP signalling within CUCM itself and the codec selected is going to be governed by the Region settings. I am assuming that we have the following regions created (all sites are CUCM sites - if there is a GK involved, there might be a CME site in which case one of the Regions will not be there): Reg-SiteA Reg-SiteB Reg-SiteC Reg-GK Reg-MOH Within the same region, the relationship is G.711. Inter-region relationships are G.729. The only exception to this rule is the MOH region which is G.711 throughout. For 5. the dial-peer with a session target of ras shouldn't have any codec defined on it. That would invoke G.729r8 on such calls. For 1 and 2 we have the dial peer set up as you have described. In such a case, the Destination phone will be in Reg-SiteB and the ingress GW will also be in the same Region. So it doesn't really matter how we specify the voice class codec since this is not a call between sites. For 4, Unity should be in Reg-SiteA and the IP Phone/Ingress GW in Reg-SiteB. Thus, even though G711ulaw will be advertised in the TCS to CUCM, only G.729 will be negotiated due to the Inter-region relationship defined. What we should be looking at are calls from Site A TEHO to a Site B PSTN phone (Or something similar). According to me, this is a call between sites and once again, we needn't worry about the preference of codecs in the voice class command since: Site A IP Phone (Reg-SiteA) will be calling the egress GW at Site B (Reg-SiteB). If the incoming dial peer at Site B has both codecs defined, the GW will send an H.245 TCS to CUCM advertising both codecs. However, CUCM will enforce the region relationship(s) mentioned previously and will thus negotiate only G.729. Note that the preference of the codecs in the voice class codec becomes a matter of concern only when something like this happens: 1. An H.323 endpoint advertises G.711 as first preference and then G.729 2. GW advertises G729 as first preference and then G.711. In this scenario, the MSD will be the tie breaker. In most cases, for a CUCM scenario, CUCM becomes the master and wins, so to speak although I don't know of any way to define a codec preference on CUCM as such. On Mon, Nov 14, 2011 at 12:18 PM, ccie_voice-requ...@onlinestudylist.comwrote: Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: [VM Ware Question] (Adam Thompson) 2. Voice Class Codec Question. (Pradeep Kumar Sharma) 3. Re: can not save script in script repository (=?gbk?B?YnJ1bm8=?=) 4. Re: Number of IP Phones in the lab (Google) 5. uccx Unified CM Telephony Subsystem gray out (=?gbk?B?YnJ1bm8=?=) -- Message: 1 Date: Sun, 13 Nov 2011 21:15:36 -0500 From: Adam Thompson arthomp...@gmail.com To: michael.se...@compucom.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] [VM Ware Question] Message-ID: cabvs7z0bs+7hsfnf8t8o6yuyg88c713bkk_35jft2e0ngor...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 ESXi 5 has pretty much the same look and feel of 4.1. The biggest difference is how they do the licensing. In 4.1, the licensing was based on the number of CPUs. In 5, it is based on RAM, and another thing I believe. Oh, and it does include vSphere. HTH -Adam On Sun, Nov 13, 2011 at 8:21 PM, michael.se...@compucom.com wrote: Anyone out there in CCIE Voice LAND using ESXi 5 yet. Looking for some feedback before installing. Does it come with the VSphere Client like 4.1? Does it have pretty much the look and feel of 4.1? ** ** ** ** ** ** ** ** ** ** ** ** ** ** ___ For more information regarding industry leading CCIE Lab training
Re: [OSL | CCIE_Voice] Why my outbound calls (using RGs) to pstn doesn't work?
Verify the registration status of your Route List. Try to reset it once. Run the debug voice ccapi inout on H323 Gw and see if the outboud calls really hitting to the gateway on not ? On Tue, Nov 15, 2011 at 9:54 AM, Ken Wyan kew...@gmail.com wrote: Pls check your route list rl-test configuration digit manipulations etc.. in rg-test (under rl-test). If you have done any digit manipulations in rg-test (under rl-test) , your RP digit-manipulations will not work. Add rg-test as Local Route Group for Device Pool In addition to debug isdn --- debug ccapi inout , dialpeer may help to isolate issue. Thanks On Tue, Nov 15, 2011 at 8:58 AM, Voiper datapack...@gmail.com wrote: Hi Follow Labbers: I am writing to get help with my lab. I have been struggling with something as simple as outbound calls to the PSTN! I had got stuck with Vol 1 lab 5.2 (Standard local route) and despite many suggestions I just couldn't get it working. So, I back tracked and tried the simplest step to make an outbound call and still not able to do so. What gives? This is my failed attempt: Created: H323 gateway on HQ MGPC gateway on BR1 911 route pattern * As long as I don't create route groups and point directly to h323 or mgcp gateway, my outbound to pstn works. Then I created: rg-test -- 10.10.200.3 (h323 gw) rl-test -- rg-test 911 -- rl-test partitions -- none HQ dial-peer voice 911 pots desti-pa 911 port 0/0/0:23 forward digits 3 dial 911 from HQ phones and get busy tone debug isdn 931 shows nothing but, csim start 911 working, so dial peer is correct did the above test (rg, rl, rp) for BR1 phones too with mgcp and get busy tone. Again, if I point rp straight to mgcp gw, then it works. started everything from Lab 1, just in case I did something wrong, reloaded routers, went over walk-through video step by step, consulted srnd, looked into other vendor videos. I have not been able to get past Vol 1 Lab 5... Any feedback or suggestion will be very much appreciated. Regards, VOIPER ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com http://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Voice Class Codec Question.
Hello Guys, If we have been asked to use G729 codec between sites, then what should be my first preference of codec in voice class codec profile for incoming calls to call manager on branch sites. Generally we use :- on Brance 1 Router: voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 dial-peer voice 1000 voip destination-pattern 1...$ voice-class codec 1 session target ipv4:X.X.X.X (Pub or Subs) dtmf-relay h245-alphanumeric no vad but in this case we are using g711 over the wan and i think this is not the requirement. Is it the correct configuration Or do we need to use g729 codec as the first preference ? -SONU- ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
