[OSL | CCIE_Voice] ipcc play prompt before agent hold off???
Hi all, Anyone can help me? I can't play prompt before agent take off the handset. My step: 1. set select resource connect to no. 2. place a play prompt under selected. 3. follow a connect. If the agent is ready when I make a call I can hear my prompt. But if the call put into queue, when the agent finish other call or change to ready status, I can't hear my prompt, just talk to agent straightway. Jeffrey
Re: [OSL | CCIE_Voice] ipcc play prompt before agent hold off???
Hi Michael, Thanks for your reply. I can do all your scenario you posted and more complicated than that. I am a programmer too, those script are too easy for me. My question is what's the steps or flows when call transfer from queue to agent. I check lots of documents about ipcc script on cisco.com, but it's no answer. I found When call is in the queue---agent change to ready---agent take the call---the program from the queue branch go to the connect branch inside the selected branch. I want to find what 's the step when call is out from queue to agent line. scenario 1. agent change to ready, caller still in the queue, send ring tone to agent. Can I send my prompt before send the ring tone to agent. scenario 2. agent change to ready, caller still in the queue, send ring tone to agent, agent take the phone. Can I send my prompt when agent take the phone immediately. I don't need a answer and follow to do this. I want to know what's the steps about this part or if the ipcc script can't do this. In many real system, like if you call to a company, when you in the queue (moh), before ring to agent you can hear For security reason, your call may be recorded I think IPCC can do that. Anyone can help me? Thanks. Jeffrey - Original Message From: Michael Ciarfello mciarfe...@iplogic.com To: jeffrey liujian jeffrey_liuj...@yahoo.com; Voice CCIE ccie_voice@onlinestudylist.com Sent: Thursday, April 30, 2009 1:11:22 AM Subject: RE: [OSL | CCIE_Voice] ipcc play prompt before agent hold off??? Is this from a workbook? If so, please cite the lab and question number. If it is not, *I'm* not going to answer this question. This same question has been posted all over the place recently by multiple people and is therefore suspect. It's an easy question to answer, a 1/10 on the IPCC configuration difficulty scale. If you've done the IPCC studying, you should be able to answer it. Keep working at it. Why did I just not post anything? Because I want other people to realize that braindumping is de-valuing the certification you are trying to achieve. Helping these people are not doing YOU any good. I'll tell you what. If you can answer this IPCC question, I'll help you on your question below. Others can use it for practice. Requirements: Create a prompt that says Call from NYC. Create a new prompt Call from San Jose Upload both to the newprompts folder (create it.) Create a script so that if the calling number is from the 212 area code it plays Call from NYC. Then transfer to a voicemail box that says Leave a message for the NYC staff. If from the 408 area code within M-F 8am to 5pm. Say Call from San Jose. Then queue the call to a queue called SJC_CSQ. Play hold music. If in queue for more than 22 seconds, forward to a voicemail box that says San Jose people are busy. Leave a message. If no one is in queue, the caller should hear ringback as the agent phone is ringing instead of music or nothing. If no agents are logged on, never queue the call. Go directly to the SJC voicemail. A variation on the above IPCC scenario. Instead of kicking the caller to voicemail after 22 seconds. Play the average hold time to the caller. Your average hold time is x minutes. If hold time is less than one minute say Hold time is less than one minute--we'll be right with you. You only need to play this ONCE. Hold music for rest of queue time. Repeat a reassurance message every 30 seconds that says Please continue to hold. Another variation I just did for a customer. Play the your position in queue is x. Just need to play it once. Hold music otherwise. Ringback when agent's phone answers. -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of jeffrey liujian Sent: Wednesday, April 29, 2009 8:17 AM To: Voice CCIE Subject: [OSL | CCIE_Voice] ipcc play prompt before agent hold off??? Hi all, Anyone can help me? I can't play prompt before agent take off the handset. My step: 1. set select resource connect to no. 2. place a play prompt under selected. 3. follow a connect. If the agent is ready when I make a call I can hear my prompt. But if the call put into queue, when the agent finish other call or change to ready status, I can't hear my prompt, just talk to agent straightway. Jeffrey
[OSL | CCIE_Voice] How to show isdn q931 on 6509 with T1 blade?
Hi all, I want to see what is the called number from hq pstn. Can I show isdn q931 or show voice dial-peer on 6509? Thanks. Jeffrey
[OSL | CCIE_Voice] h323 alias list
Hi all, I used no-reg for all ephone-dn number on sc cme, but I found in show gateway all the e164-id are there and register to gatekeeper. How can I clear the h323 alias list? Thanks. Jeffrey
[OSL | CCIE_Voice] site 1 g729 to hq sip g711 to site 2
Hi all, I can not make a call from site 1 2001 to site 2 3001. site 1 to hq g729, hq sip trunk g711 to site 2. But 3001 can call 1001,2001, 1 MTP and 1 xcode used. What's the reason? Thanks. Jeffrey
Re: [OSL | CCIE_Voice] h323 alias list
Hi Cliff, I try no gateway/gateway, shutdown gatekeeper/no shut, restart cme and gatekeeper router(hq-rtr). It's still there. Too weird. Jeffrey - Original Message From: Cliff McGlamry cl...@mcglamry.net To: jeffrey liujian jeffrey_liuj...@yahoo.com Sent: Wednesday, April 8, 2009 10:58:30 PM Subject: Re: [OSL | CCIE_Voice] h323 alias list Jeffrey, On CME do a no gateway to kill the gateway process and unregister from GK. If you have got all your no-reg in place, then you should be able to issue a gateway to restart the process. If on CME after doing this if you do a show gateway and still see the e164 numbers registering, reboot CME. That will clear it. Cliff - Original Message - From: jeffrey liujian jeffrey_liuj...@yahoo.com To: Voice CCIE ccie_voice@onlinestudylist.com Sent: Wednesday, April 08, 2009 8:41 AM Subject: [OSL | CCIE_Voice] h323 alias list Hi all, I used no-reg for all ephone-dn number on sc cme, but I found in show gateway all the e164-id are there and register to gatekeeper. How can I clear the h323 alias list? Thanks. Jeffrey
Re: [OSL | CCIE_Voice] h323 alias list
Hi all, I had mistake. It's my first time using telephony-service setup to setup DID dialplan, I forgot here. Thanks. - Original Message From: Cliff McGlamry cl...@mcglamry.net To: jeffrey liujian jeffrey_liuj...@yahoo.com Sent: Wednesday, April 8, 2009 10:58:30 PM Subject: Re: [OSL | CCIE_Voice] h323 alias list Jeffrey, On CME do a no gateway to kill the gateway process and unregister from GK. If you have got all your no-reg in place, then you should be able to issue a gateway to restart the process. If on CME after doing this if you do a show gateway and still see the e164 numbers registering, reboot CME. That will clear it. Cliff - Original Message - From: jeffrey liujian jeffrey_liuj...@yahoo.com To: Voice CCIE ccie_voice@onlinestudylist.com Sent: Wednesday, April 08, 2009 8:41 AM Subject: [OSL | CCIE_Voice] h323 alias list Hi all, I used no-reg for all ephone-dn number on sc cme, but I found in show gateway all the e164-id are there and register to gatekeeper. How can I clear the h323 alias list? Thanks. Jeffrey
Re: [OSL | CCIE_Voice] ccm to cme voicemail
Hi all, There is some trouble. ccm -- gk -- cme using g729, cme -- cue using g711. when ccm phone makes a call to cme phone or cue number: 1. call can be connected, but no sound at all. 2. if I change the code (between ccm and cme) to g711, everything is fine. Why does this happen? Thanks. Jeffrey From: Cliff McGlamry cl...@mcglamry.net To: Gughan Gug gughan...@yahoo.in Cc: ccie_voice@onlinestudylist.com Sent: Saturday, March 21, 2009 5:19:05 PM Subject: Re: [OSL | CCIE_Voice] ccm to cme voicemail Oh, this one bites lots of folks. In the telephony-service, make sure you have: call-forward pattern .T If you're missing that, it won't work. Cliff - Original Message - From: Gughan Gug To: ccie_voice@onlinestudylist.com Sent: Saturday, March 21, 2009 1:46 AM Subject: [OSL | CCIE_Voice] ccm to cme voicemail Hi, CCM phones call to CME phones through gatekeeper . phone from ccm site can call to cme voicemail but if the phone call to cme ip phone and get redirected the calls get disconnected. I have allowed the h323 to sip and sip to h323 coneectiion under voice service voip,transcoder at cme when a cme phone call the other one and at noan the calls get redirected properly to vociemail. Any help in this regard. anything i am missing here. since ccm phoes can call directly to cme voice mail using the voicemail number no issue, only when it gets redirectd the call get disconnected. Regards Gug Add more friends to your messenger and enjoy! Invite them now..
Re: [OSL | CCIE_Voice] ccm to cme voicemail
Hi Cliff and Tech, Thanks for your reply. I am not sure if the xcode works properly and I made some show to check it. The xcodes have been registered to ccm and cme successfully. On cme site: 1. show call active voice compact-- It shows the g729 to ccm ip phone address and g711 to cue ip address. 2. show dspfarm session-- It shows the g729 to cme loopback 0 and g711 to cme loopback 0. 3: show voip rtp connections-- It shows 2 active RTP connections, one from cme loopback 0 to ccm ip phone, the other from cme loopback 0 to cue. On ccm site: 1. There is no xcode involved in the process because IP phone can choose g729 to cme. 2. If I tick the outbound fast connection and MTP, it shows xcode works from the performance monitor interface when I made a call. I guess it may be an IPIPGW h323 to sip promble. When I use HDV in cme for xcode, it is a first generation dsp so it can only xcode from g729 to g711,but it can not translate from h323 dtmf to sip dtmf. The second generation dsp has an enhanced MTP funtion which can do xcode and translate dtmf together. Is that right? Someone can help me with this? Thanks. Jeffrey From: Cliff McGlamry cl...@mcglamry.net To: ccie_voice@onlinestudylist.com Sent: Sunday, April 5, 2009 12:49:10 AM Subject: Re: [OSL | CCIE_Voice] ccm to cme voicemail Dead air on a connected call means that the media stream is not in place. This can be for several reasons: 1. Transcoder at HQ site (if you're using the 6608 blade) is not registered, not in the correct MRG, MRGL, or location(this is a hidden killer), or using the WRONG DEFAULT GATEWAY. 2. Transcoder at CME site not registered. In addition to configuration of dspfarm profile, three sdspfarm commands are required in telephony-service. Make sure the g729r8 codec has been set up for transcoding. Use the show sccp to confirm the transcode resource is up, registered and operational. 3. SCCP is attempting to register to the incorrect address. The sccp ccm command should point at the same ip address being used as the source address in the telephony service section of your config. 4. Incoming dial-peer not explicitly defined correctly causing call to use default dial peer. Default dial peer cannot invoke transcoder. Use the show voice call status to see which dial peers are operational. This issue is the primary cause of this type problem, although anything in this list can cause it. 5. MRGL not correctly assigned to the GK Trunk. Cliff - Original Message - From: jeffrey liujian To: Cliff McGlamry ; Voice CCIE ; gughan...@yahoo.in ; rxlm...@gmail.com Sent: Saturday, April 04, 2009 8:44 AM Subject: Re: [OSL | CCIE_Voice] ccm to cme voicemail Hi all, There is some trouble. ccm -- gk -- cme using g729, cme -- cue using g711. when ccm phone makes a call to cme phone or cue number: 1. call can be connected, but no sound at all. 2. if I change the code (between ccm and cme) to g711, everything is fine. Why does this happen? Thanks. Jeffrey From: Cliff McGlamry cl...@mcglamry.net To: Gughan Gug gughan...@yahoo.in Cc: ccie_voice@onlinestudylist.com Sent: Saturday, March 21, 2009 5:19:05 PM Subject: Re: [OSL | CCIE_Voice] ccm to cme voicemail Oh, this one bites lots of folks. In the telephony-service, make sure you have: call-forward pattern .T If you're missing that, it won't work. Cliff - Original Message - From: Gughan Gug To: ccie_voice@onlinestudylist.com Sent: Saturday, March 21, 2009 1:46 AM Subject: [OSL | CCIE_Voice] ccm to cme voicemail Hi, CCM phones call to CME phones through gatekeeper . phone from ccm site can call to cme voicemail but if the phone call to cme ip phone and get redirected the calls get disconnected. I have allowed the h323 to sip and sip to h323 coneectiion under voice service voip,transcoder at cme when a cme phone call the other one and at noan the calls get redirected properly to vociemail. Any help in this regard. anything i am missing here. since ccm phoes can call directly to cme voice mail using the voicemail number no issue, only when it gets redirectd the call get disconnected. Regards Gug Add more friends to your messenger and enjoy! Invite them now..
[OSL | CCIE_Voice] h323 caller name
Hi All, I just test call between ccm and cme. Both side ip phones can see caller name when call is made from cme to ccm, but if call is made from ccm to cme, olny cme ip phone can see caller name. Is it a ios bug or something? Thanks. Jeffrey
[OSL | CCIE_Voice] MOH to PSTN
Hi all, I found something weird about MOH. I have a sub moh server for multicast and a pub for unicast. I made a MRGL which put pub first and then sub, and bind it to HQ device pool. I sent moh inside HQ phones it choose unicast(pub) but when I sent moh to pstn it choose multicast(sub). The sub is the primary CCM. What't the reason? I miss somewhere? Thanks. Jeffrey
Re: [OSL | CCIE_Voice] MOH to PSTN
Hi kapil, Thanks for your reply. I put hq-mrgl(first pub unicast moh, second sub multicast moh) to hq-device-pool, and put hq-device-pool to HQ IP phone HQ pstn gateway. When the sub and pub moh server work at the same time, I found it play sub if the mgcp pstn gateway registers to sub and it play pub if the mgcp pstn gateway registers to pub. It has nothing to do with the regisration if there is olny one moh server. Thanks. Jeffrey From: kapil atrish nice_cha...@yahoo.com To: jeffrey liujian jeffrey_liuj...@yahoo.com Sent: Monday, March 30, 2009 8:58:18 PM Subject: Re: [OSL | CCIE_Voice] MOH to PSTN What's the MRGL on Gateway? I can recall, the holder decides which file to play and the holdee decides from which server to take that file from. So if your Gateway MRGL doesn't have access to Pub, it will always try to access MOH from Sub. --- On Mon, 3/30/09, jeffrey liujian jeffrey_liuj...@yahoo.com wrote: From: jeffrey liujian jeffrey_liuj...@yahoo.com Subject: [OSL | CCIE_Voice] MOH to PSTN To: Voice CCIE ccie_voice@onlinestudylist.com Date: Monday, March 30, 2009, 12:38 PM Hi all, I found something weird about MOH. I have a sub moh server for multicast and a pub for unicast. I made a MRGL which put pub first and then sub, and bind it to HQ device pool. I sent moh inside HQ phones it choose unicast(pub) but when I sent moh to pstn it choose multicast(sub). The sub is the primary CCM. What't the reason? I miss somewhere? Thanks. Jeffrey
[OSL | CCIE_Voice] srst forwarding to unity
Hi all, I have an issue about srst to voicemail. The isdn q931 from HQ PSTN to CCM include calling number, called number and redirecting number, but in unity call viewer I just saw Reason= direct, calling number, called number and no forwarding station. What is the reason? Thanks. Jeffrey
[OSL | CCIE_Voice] Unity forwarding
Hi all, I dial hq 1001, it no answer or busy to unity and how can I configure unity so it can forward to hq pstn number? No AA in unity, but I found all call transfer in unity are refer to AA. Anyone can help me? Thanks. Jeffrey
Re: [OSL | CCIE_Voice] Unity forwarding
Hi Sara, Thanks for your reply, It can't forward to pstn directly, no input or menu permitted. From: saralilin2...@yahoo.co.jp saralilin2...@yahoo.co.jp To: liujian jeffrey jeffrey_liuj...@yahoo.com; CCIE Voice ccie_voice@onlinestudylist.com Sent: Monday, March 23, 2009 12:30:30 AM Subject: Re: [OSL | CCIE_Voice] Unity forwarding call transfer to subscriber then select ring this number, - input the number 91234567 remember to add call retriction table to allow this string. jeffrey liujian jeffrey_liuj...@yahoo.com wrote: Hi all, I dial hq 1001, it no answer or busy to unity and how can I configure unity so it can forward to hq pstn number? No AA in unity, but I found all call transfer in unity are refer to AA. Anyone can help me? Thanks. Jeffrey Power up the Internet with Yahoo! Toolbar.
[OSL | CCIE_Voice] cue ccn application voicemail cannot start
Hi guys, Is there anyone can help me? I am wondering why cue-2.3.4 ccn application voicemail cannot start? I just run from factory default. Thanks. Jeffrey
Re: [OSL | CCIE_Voice] cue ccn application voicemail cannot start
I already configure it and can show ccn application in CUE module, but just found mwi and messagenotify two applications running in it. - Original Message From: Cliff McGlamry cl...@mcglamry.net To: jeffrey liujian jeffrey_liuj...@yahoo.com Sent: Monday, March 9, 2009 3:40:58 AM Subject: Re: [OSL | CCIE_Voice] cue ccn application voicemail cannot start Do you have the interface for the service engine 0/0 configured and no shut? - Original Message - From: jeffrey liujian jeffrey_liuj...@yahoo.com To: Voice CCIE ccie_voice@onlinestudylist.com Sent: Sunday, March 08, 2009 8:46 AM Subject: [OSL | CCIE_Voice] cue ccn application voicemail cannot start Hi guys, Is there anyone can help me? I am wondering why cue-2.3.4 ccn application voicemail cannot start? I just run from factory default. Thanks. Jeffrey
[OSL | CCIE_Voice] (no subject)
Hi guys, Is there anyone can help me? I am wondering which Cisco Unity Express version I need to use for prepare the voice lab? Thanks. Jeffrey
Re: [OSL | CCIE_Voice] No Voice Labs in US until August?
Hi Steve, I had the same situation two weeks ago when I passed the voice written test and tried to book a lab position. I found the only seat before August I can get is in Tokyo. And I tried many times a day to refresh the screen and finally get a seat in May. If you can't find the position, I would suggest you to keep refreshing the screen everyday and you will probably find some seat dropped by other people or unpaid within 90 days. Hopefully it can help. Jeffrey - Original Message From: Steve Sarrick ssarr...@drsllc.net To: ccie_voice@onlinestudylist.com Sent: Saturday, February 28, 2009 4:30:28 AM Subject: [OSL | CCIE_Voice] No Voice Labs in US until August? So I have been studying away for months now and was finally ready to schedule an exam for my voice CCIE. To my surprise, I went online only to find no available dates until August 26th in the US. At which time, I assume will be the new version. Did I just waste a bunch of my time? I know a lot of info is transferable, but certainly a lot of my time has been wasted. Very disappointing. Steve Sarrick CCVP, CCNP, MCSE DRS LLC 412-480-3861 (m) 877-527-3423 (o) No virus found in this outgoing message. Checked by AVG. Version: 7.5.557 / Virus Database: 270.11.3/1975 - Release Date: 2/27/2009 7:05 AM
