[OSL | CCIE_Voice] MOH form flash
hi group, I need to know when i am making moh from flash at site B should i create a new device pool with region g711 to all and assign it to the moh server or it should be in the normal hq pool and put under ccm-fallback 239.1.1.3 instead of 239.1.1.1 so as to work g729
[OSL | CCIE_Voice] Qos Marking
hi, on making marking on the routers, do i make a policy match the protocols sip,h323..etc and set the dscp or do i make under the voip dialpeers and in the mgcp or do i have to have both Please advise
Re: [OSL | CCIE_Voice] ATA IVR not responding
Hi Michael,
Can u give more inputs reg turning IVR off via config-register? Haven't heard
abt this before, sounds interesting.
Yes the Red light does come on and goes off perfectly fine. I was able to
make/receive calls to/from ata phone.
ATA is not reachable now can't change the software image...:(
I did shake it, opened the box and tried pressing the little button but no
good.:(
--- On Sun, 5/3/09, Michael Ciarfello mciarfe...@iplogic.com wrote:
From: Michael Ciarfello mciarfe...@iplogic.com
Subject: RE: [OSL | CCIE_Voice] ATA IVR not responding
To: kapil atrish nice_cha...@yahoo.com, ccie_voice@onlinestudylist.com
ccie_voice@onlinestudylist.com
Date: Sunday, May 3, 2009, 9:00 AM
#yiv1591233982 P {
MARGIN-TOP:0px;MARGIN-BOTTOM:0px;}
Can you turn the IVR off via a config register? Might be disabled.
When you pick up the phone does the red light come on?
Try the factoryreset thingie.
Try re-loading the software image.
Try shaking it. (lol. just kidding)
From: ccie_voice-boun...@onlinestudylist.com
[ccie_voice-boun...@onlinestudylist.com] On Behalf Of kapil atrish
[nice_cha...@yahoo.com]
Sent: Saturday, May 02, 2009 11:31 PM
To: ccie_voice@onlinestudylist.com; Cliff McGlamry
Subject: Re: [OSL | CCIE_Voice] ATA IVR not responding
Yes, that's what I am trying to invoke the IVR but no response. I've changed
the cable, Phone, Phone port, removed lan cable and tried whole thing again but
no good.
--- On Sat, 5/2/09, Cliff McGlamry cl...@mcglamry.net wrote:
From: Cliff McGlamry cl...@mcglamry.net
Subject: Re: [OSL | CCIE_Voice] ATA IVR not responding
To: kapil atrish nice_cha...@yahoo.com, ccie_voice@onlinestudylist.com
Date: Saturday, May 2, 2009, 11:30 PM
Did you push the button on top of the ATA after picking up the phone on port
1? That's how you activate the IVR menu.
What exactly have you done?
- Original Message -
From:
kapil atrish
To:
ccie_voice@onlinestudylist.com
Sent: Saturday, May 02, 2009 8:25 AM
Subject: [OSL | CCIE_Voice] ATA IVR not responding
Hi list,
M not able to access the IVR menu on ATA.I know it should work on Phone 1 port,
no success on any of the port. Reboot didn't help. Removed lan cable and tried,
no success. Checked physical layer stuff. Has anyone had this issue earlier and
any troubleshooting
I can do??? M running SCCP image on it.
I was testing the ATA vlan stuff and put in a Vlan and OpFlag which made my ATA
unreachable. Now I don't have a switch to configure in the specific vlan and
access my ATA. Since IVR is not working I am not even able to revert the ATA
settings.
Thanks in advance...
Re: [OSL | CCIE_Voice] Qos Marking
Marwa, I think that we have to both because: - Policy-map marks the pkts that goes through the router - With ip qos dscp cs3 signaling and mgcp ip qos dscp cs3 signaling marks pkts thats are originated at router and are not inspect by the output policy-maps. Sergio From: marwa_ah...@seegypt.com To: ccie_voice@onlinestudylist.com Date: Sun, 3 May 2009 11:17:40 +0200 Subject: [OSL | CCIE_Voice] Qos Marking hi, on making marking on the routers, do i make a policy match the protocols sip,h323..etc and set the dscp or do i make under the voip dialpeers and in the mgcp or do i have to have both Please advise _ Deixe suas conversas mais divertidas. Baixe agora mesmo novos emoticons. É grátis! http://specials.br.msn.com/ilovemessenger/pacotes.aspx
Re: [OSL | CCIE_Voice] ATA IVR not responding
Are you sure you have the analog phone plugged into Port 1 on the ATA? If you have your analog phone plugged into Port 2, the IVR will not be heard. http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_configuration_example09186a00800c3a50.shtml#task3 Brian - Original Message - From: kapil atrish To: ccie_voice@onlinestudylist.com ; Michael Ciarfello Sent: Sunday, May 03, 2009 10:43 AM Subject: Re: [OSL | CCIE_Voice] ATA IVR not responding Hi Michael, Can u give more inputs reg turning IVR off via config-register? Haven't heard abt this before, sounds interesting. Yes the Red light does come on and goes off perfectly fine. I was able to make/receive calls to/from ata phone. ATA is not reachable now can't change the software image...:( I did shake it, opened the box and tried pressing the little button but no good.:( --- On Sun, 5/3/09, Michael Ciarfello mciarfe...@iplogic.com wrote: From: Michael Ciarfello mciarfe...@iplogic.com Subject: RE: [OSL | CCIE_Voice] ATA IVR not responding To: kapil atrish nice_cha...@yahoo.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Sunday, May 3, 2009, 9:00 AM Can you turn the IVR off via a config register? Might be disabled. When you pick up the phone does the red light come on? Try the factoryreset thingie. Try re-loading the software image. Try shaking it. (lol. just kidding) -- From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of kapil atrish [nice_cha...@yahoo.com] Sent: Saturday, May 02, 2009 11:31 PM To: ccie_voice@onlinestudylist.com; Cliff McGlamry Subject: Re: [OSL | CCIE_Voice] ATA IVR not responding Yes, that's what I am trying to invoke the IVR but no response. I've changed the cable, Phone, Phone port, removed lan cable and tried whole thing again but no good. --- On Sat, 5/2/09, Cliff McGlamry cl...@mcglamry.net wrote: From: Cliff McGlamry cl...@mcglamry.net Subject: Re: [OSL | CCIE_Voice] ATA IVR not responding To: kapil atrish nice_cha...@yahoo.com, ccie_voice@onlinestudylist.com Date: Saturday, May 2, 2009, 11:30 PM Did you push the button on top of the ATA after picking up the phone on port 1? That's how you activate the IVR menu. What exactly have you done? - Original Message - From: kapil atrish To: ccie_voice@onlinestudylist.com Sent: Saturday, May 02, 2009 8:25 AM Subject: [OSL | CCIE_Voice] ATA IVR not responding Hi list, M not able to access the IVR menu on ATA.I know it should work on Phone 1 port, no success on any of the port. Reboot didn't help. Removed lan cable and tried, no success. Checked physical layer stuff. Has anyone had this issue earlier and any troubleshooting I can do??? M running SCCP image on it. I was testing the ATA vlan stuff and put in a Vlan and OpFlag which made my ATA unreachable. Now I don't have a switch to configure in the specific vlan and access my ATA. Since IVR is not working I am not even able to revert the ATA settings. Thanks in advance...
[OSL | CCIE_Voice] ring policy is not working
Hi, I have a problem with ccm parameter Ring Setting of Busy Station Policy in Default mode when switching between hold and resume, it shouldn't get ring notification from incoming call but it gets. changing this parameter has no effect for me. CCM 4.1.3 (sr8a) anybody face this be4? Jeremy
[OSL | CCIE_Voice] Using Safari Browser (or alternative) on Mac for CCMAdmin Pages?
From the Voice BLS videos it seems that Mark Snow uses a Mac. I also have a Mac. The BLS videos are for CM4 so of course there is a Windows server which can be remoted to in order to run Internet Explorer to access the admin pages. Now with CUCM7 and other appliances there is no Windows server to remote to. When using Safari on my Mac to access the admin pages I have found that some pages don't load properly, to the point where it is unusable. My question is, what are Mac users doing to access the admin pages? I do have VMWare Fusion but it sucks battery life so I would like to find a native way to access the admin pages. I loaded IE for Mac but it also has issues with the admin pages. Firefox on Mac seems to work so far... but anyone using Safari without issue? http://slash128.com
[OSL | CCIE_Voice] 3rd day in the raw , any body know how none location interact with specific location ?
Hi, Third day in the raw, please discuss your idea about it. Is set up a simple scenario for call from HQ to 911 on pstn. HQ region : G711 within HQ Phone Location:48kbps HQ GW Location: none call will fail. What I thought is when just one location, it should treat none as unlimit. but it's not a case. CCM traces shows it deal with none as 80Kbps, can any one explain this behavior?
Re: [OSL | CCIE_Voice] 3rd day in the raw , any body know how none location interact with specific location ?
Jeremy: What you are seeing is that the call manager is seeing your call as a G711 call. I'm not sure what you are asking? Larry Hadrava CCIE #12203 CCNP CCNA Sr. Support Engineer – IPexpert, Inc. URL: http://www.IPexpert.com On Sun, May 3, 2009 at 1:56 PM, jeremy co jeremy.coo...@gmail.com wrote: Hi, Third day in the raw, please discuss your idea about it. Is set up a simple scenario for call from HQ to 911 on pstn. HQ region : G711 within HQ Phone Location:48kbps HQ GW Location: none call will fail. What I thought is when just one location, it should treat none as unlimit. but it's not a case. CCM traces shows it deal with none as 80Kbps, can any one explain this behavior?
Re: [OSL | CCIE_Voice] 3rd day in the raw , any body know how none location interact with specific location ?
Isn't this happening becasue of the region applied to phone is G711 and but location is only 48KPBS .but actually it needs 80KPBS to make the call. a) What about making a call to another phone which is none location. b) what if HQ region is set to G729 and try then... --- On Sun, 5/3/09, jeremy co jeremy.coo...@gmail.com wrote: From: jeremy co jeremy.coo...@gmail.com Subject: [OSL | CCIE_Voice] 3rd day in the raw , any body know how none location interact with specific location ? To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Sunday, May 3, 2009, 11:26 PM Hi, Third day in the raw, please discuss your idea about it. Is set up a simple scenario for call from HQ to 911 on pstn. HQ region : G711 within HQ Phone Location:48kbps HQ GW Location: none call will fail. What I thought is when just one location, it should treat none as unlimit. but it's not a case. CCM traces shows it deal with none as 80Kbps, can any one explain this behavior?
Re: [OSL | CCIE_Voice] 3rd day in the raw , any body know how none location interact with specific location ?
Hi anil, a) it would fail b) call could be make successfully however , ig region G729 and location 23KB it would fail. what I'm not understanding here is why CCM invoke locations between none and specific location as it shouldn't Jeremy On Mon, May 4, 2009 at 5:07 AM, anil batra anil...@yahoo.com wrote: Isn't this happening becasue of the region applied to phone is G711 and but location is only 48KPBS .but actually it needs 80KPBS to make the call. a) What about making a call to another phone which is none location. b) what if HQ region is set to G729 and try then... --- On *Sun, 5/3/09, jeremy co jeremy.coo...@gmail.com* wrote: From: jeremy co jeremy.coo...@gmail.com Subject: [OSL | CCIE_Voice] 3rd day in the raw , any body know how none location interact with specific location ? To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Sunday, May 3, 2009, 11:26 PM Hi, Third day in the raw, please discuss your idea about it. Is set up a simple scenario for call from HQ to 911 on pstn. HQ region : G711 within HQ Phone Location:48kbps HQ GW Location: none call will fail. What I thought is when just one location, it should treat none as unlimit. but it's not a case. CCM traces shows it deal with none as 80Kbps, can any one explain this behavior?
[OSL | CCIE_Voice] Voicemail Issue in SRST mode
ON Branch 1 router in SRST mode, 2001 is a registeres phone and 2002 is not registered. When call comes to 2002 it forard to 2001. When a call comes to 2001 I am trying to Call Forward all to VM on 2001. Now if the calls comes to 3002 it shoudl still hit VM of 3002. I tried both methods - vm-intergationa and CTI RO on CCM-- with vm-integration it hits the VM of 2001 and with CTI RP method it hits the opening greeting. Any clue please..
Re: [OSL | CCIE_Voice] 3rd day in the raw , any body know how none location interact with specific location ?
Hey Jeremy, I think it's not becasue of the none location applied on Gateway or elsewhere it is becasue once you applu Location to Phone and that phone is in say G711 region so that and it is talking to another device say GW in same region G711. It wantes to reserve BW of 80KBPS from the Location but since you have only 48KBPS for that location which is not =80KPBS and the callit fails. I am sure the moment you apply same Location(48KPBS) as of phone on GW(assuming GW is also in same region as of phone) the will still fail as it not 80KBPS. But if you increase the Location BW to 80KPBS the call will go thru.. Did you try this please. HTH... --- On Mon, 5/4/09, jeremy co jeremy.coo...@gmail.com wrote: From: jeremy co jeremy.coo...@gmail.com Subject: Re: [OSL | CCIE_Voice] 3rd day in the raw , any body know how none location interact with specific location ? To: anil batra anil...@yahoo.com, lhadr...@ipexpert.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Monday, May 4, 2009, 12:48 AM Hi anil, a) it would fail b) call could be make successfully however , ig region G729 and location 23KB it would fail. what I'm not understanding here is why CCM invoke locations between none and specific location as it shouldn't Jeremy On Mon, May 4, 2009 at 5:07 AM, anil batra anil...@yahoo.com wrote: Isn't this happening becasue of the region applied to phone is G711 and but location is only 48KPBS .but actually it needs 80KPBS to make the call. a) What about making a call to another phone which is none location. b) what if HQ region is set to G729 and try then... --- On Sun, 5/3/09, jeremy co jeremy.coo...@gmail.com wrote: From: jeremy co jeremy.coo...@gmail.com Subject: [OSL | CCIE_Voice] 3rd day in the raw , any body know how none location interact with specific location ? To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Sunday, May 3, 2009, 11:26 PM Hi, Third day in the raw, please discuss your idea about it. Is set up a simple scenario for call from HQ to 911 on pstn. HQ region : G711 within HQ Phone Location:48kbps HQ GW Location: none call will fail. What I thought is when just one location, it should treat none as unlimit. but it's not a case. CCM traces shows it deal with none as 80Kbps, can any one explain this behavior?
Re: [OSL | CCIE_Voice] 3rd day in the raw , any body know how none location interact with specific location ?
You actually did receive a response two days ago. G711 uses 80k. You have limited it to 48. It won't work, and isn't going to with that setup. The bigger question is why you would do this in the first place, and you haven't posted the reason for that. - Original Message - From: jeremy co To: ccie_voice@onlinestudylist.com Sent: Sunday, May 03, 2009 1:56 PM Subject: [OSL | CCIE_Voice] 3rd day in the raw ,any body know how none location interact with specific location ? Hi, Third day in the raw, please discuss your idea about it. Is set up a simple scenario for call from HQ to 911 on pstn. HQ region : G711 within HQ Phone Location:48kbps HQ GW Location: none call will fail. What I thought is when just one location, it should treat none as unlimit. but it's not a case. CCM traces shows it deal with none as 80Kbps, can any one explain this behavior?
Re: [OSL | CCIE_Voice] Voicemail Issue in SRST mode
Hi anil, Unity see forwarding number of first called station as a forwarding number ,in your case 3002. if u use CTI approach it would work and it would transfer to 3002 mailbox. call-forward noan 39.. tim 6 CTI DN: 39XX , VM-Profile (30XX) and call forward all to unity. Regards, On Mon, May 4, 2009 at 5:19 AM, anil batra anil...@yahoo.com wrote: ON Branch 1 router in SRST mode, 2001 is a registeres phone and 2002 is not registered. When call comes to 2002 it forard to 2001. When a call comes to 2001 I am trying to Call Forward all to VM on 2001. Now if the calls comes to 3002 it shoudl still hit VM of 3002. I tried both methods - vm-intergationa and CTI RO on CCM-- with vm-integration it hits the VM of 2001 and with CTI RP method it hits the opening greeting. Any clue please.. -- Sirus Moghadasian CCIE #21862 (RS)
Re: [OSL | CCIE_Voice] Voicemail Issue in SRST mode
Hi Cyrus, I agree this part works when the call comes 3002 is forwaded to 3001and 3001 does not pick ..it goes to 3002 VM. But when you do CfwdAll on 3001 to Voicemail. Then it is not hitting 3001 VM rathrer it is hitting opening greetin...I checked debug in this case the calles number becomes 1600(VM pilot number) and I can't translate this to 3002 --- On Mon, 5/4/09, Cyrus cyrus@gmail.com wrote: From: Cyrus cyrus@gmail.com Subject: Re: [OSL | CCIE_Voice] Voicemail Issue in SRST mode To: anil batra anil...@yahoo.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Monday, May 4, 2009, 1:32 AM Hi anil, Unity see forwarding number of first called station as a forwarding number ,in your case 3002. if u use CTI approach it would work and it would transfer to 3002 mailbox. call-forward noan 39.. tim 6 CTI DN: 39XX , VM-Profile (30XX) and call forward all to unity. Regards, On Mon, May 4, 2009 at 5:19 AM, anil batra anil...@yahoo.com wrote: ON Branch 1 router in SRST mode, 2001 is a registeres phone and 2002 is not registered. When call comes to 2002 it forard to 2001. When a call comes to 2001 I am trying to Call Forward all to VM on 2001. Now if the calls comes to 3002 it shoudl still hit VM of 3002. I tried both methods - vm-intergationa and CTI RO on CCM-- with vm-integration it hits the VM of 2001 and with CTI RP method it hits the opening greeting. Any clue please.. -- Sirus Moghadasian CCIE #21862 (RS)
Re: [OSL | CCIE_Voice] 3rd day in the raw , any body know how none location interact with specific location ?
Cliff, question was not I have a location with 48K and wants to talk to another location with region G711 ,so why call fails it is obvious. The Question was ,why CCM wants to reserve bandwidth for other end while it's in none location. so call will fail. reason why I did this setup is understanding CCM behavior on locations. Anil, I guess your answer explains CCM's behavior so it doesn't related to other end location but instead it relates to source location. thanx On Mon, May 4, 2009 at 5:58 AM, Cliff McGlamry cl...@mcglamry.net wrote: You actually did receive a response two days ago. G711 uses 80k. You have limited it to 48. It won't work, and isn't going to with that setup. The bigger question is why you would do this in the first place, and you haven't posted the reason for that. - Original Message - *From:* jeremy co jeremy.coo...@gmail.com *To:* ccie_voice@onlinestudylist.com *Sent:* Sunday, May 03, 2009 1:56 PM *Subject:* [OSL | CCIE_Voice] 3rd day in the raw ,any body know how none location interact with specific location ? Hi, Third day in the raw, please discuss your idea about it. Is set up a simple scenario for call from HQ to 911 on pstn. HQ region : G711 within HQ Phone Location:48kbps HQ GW Location: none call will fail. What I thought is when just one location, it should treat none as unlimit. but it's not a case. CCM traces shows it deal with none as 80Kbps, can any one explain this behavior?
Re: [OSL | CCIE_Voice] Voicemail Issue in SRST mode
Hmm, I guess u set your cfwAll to your voice pilot number instead of CTI number (39..) or it's not a case? Cheers On Mon, May 4, 2009 at 6:07 AM, anil batra anil...@yahoo.com wrote: Hi Cyrus, I agree this part works when the call comes 3002 is forwaded to 3001and 3001 does not pick ..it goes to 3002 VM. But when you do CfwdAll on 3001 to Voicemail. Then it is not hitting 3001 VM rathrer it is hitting opening greetin...I checked debug in this case the calles number becomes 1600(VM pilot number) and I can't translate this to 3002 --- On *Mon, 5/4/09, Cyrus cyrus@gmail.com* wrote: From: Cyrus cyrus@gmail.com Subject: Re: [OSL | CCIE_Voice] Voicemail Issue in SRST mode To: anil batra anil...@yahoo.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Monday, May 4, 2009, 1:32 AM Hi anil, Unity see forwarding number of first called station as a forwarding number ,in your case 3002. if u use CTI approach it would work and it would transfer to 3002 mailbox. call-forward noan 39.. tim 6 CTI DN: 39XX , VM-Profile (30XX) and call forward all to unity. Regards, On Mon, May 4, 2009 at 5:19 AM, anil batra anil...@yahoo.comhttp://us.mc379.mail.yahoo.com/mc/compose?to=anil...@yahoo.com wrote: ON Branch 1 router in SRST mode, 2001 is a registeres phone and 2002 is not registered. When call comes to 2002 it forard to 2001. When a call comes to 2001 I am trying to Call Forward all to VM on 2001. Now if the calls comes to 3002 it shoudl still hit VM of 3002. I tried both methods - vm-intergationa and CTI RO on CCM-- with vm-integration it hits the VM of 2001 and with CTI RP method it hits the opening greeting. Any clue please.. -- Sirus Moghadasian CCIE #21862 (RS) -- Sirus Moghadasian CCIE #21862 (RS)
Re: [OSL | CCIE_Voice] Voicemail Issue in SRST mode
voicemail here is my config on SRST router - call-manager-fallback voicemail 912122211600 ( 1600 is VM pilot for Unity on HQ) call-forward no-ans 9121222117.. timeout 10 call-forward busy 9121222117.. on CCM I am using CTI RP 17XX fwdall to VM with VMMask 20xx With this when I press CfwdAll on 3001 the called number will be 912122211600 an dhence I can't do anything on CCM side. --- On Mon, 5/4/09, Cyrus cyrus@gmail.com wrote: From: Cyrus cyrus@gmail.com Subject: Re: [OSL | CCIE_Voice] Voicemail Issue in SRST mode To: anil batra anil...@yahoo.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Monday, May 4, 2009, 1:46 AM Hmm, I guess u set your cfwAll to your voice pilot number instead of CTI number (39..) or it's not a case? Cheers On Mon, May 4, 2009 at 6:07 AM, anil batra anil...@yahoo.com wrote: Hi Cyrus, I agree this part works when the call comes 3002 is forwaded to 3001and 3001 does not pick ..it goes to 3002 VM. But when you do CfwdAll on 3001 to Voicemail. Then it is not hitting 3001 VM rathrer it is hitting opening greetin...I checked debug in this case the calles number becomes 1600(VM pilot number) and I can't translate this to 3002 --- On Mon, 5/4/09, Cyrus cyrus@gmail.com wrote: From: Cyrus cyrus@gmail.com Subject: Re: [OSL | CCIE_Voice] Voicemail Issue in SRST mode To: anil batra anil...@yahoo.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Monday, May 4, 2009, 1:32 AM Hi anil, Unity see forwarding number of first called station as a forwarding number ,in your case 3002. if u use CTI approach it would work and it would transfer to 3002 mailbox. call-forward noan 39.. tim 6 CTI DN: 39XX , VM-Profile (30XX) and call forward all to unity. Regards, On Mon, May 4, 2009 at 5:19 AM, anil batra anil...@yahoo.com wrote: ON Branch 1 router in SRST mode, 2001 is a registeres phone and 2002 is not registered. When call comes to 2002 it forard to 2001. When a call comes to 2001 I am trying to Call Forward all to VM on 2001. Now if the calls comes to 3002 it shoudl still hit VM of 3002. I tried both methods - vm-intergationa and CTI RO on CCM-- with vm-integration it hits the VM of 2001 and with CTI RP method it hits the opening greeting. Any clue please.. -- Sirus Moghadasian CCIE #21862 (RS) -- Sirus Moghadasian CCIE #21862 (RS)
[OSL | CCIE_Voice] SIP Trunk G711
I think I understand this but just for clarification I have a few questions If I configured a SIP trunk between CM (version 4) and the HQ-RTR: 1) I should force G711 only in both directions across the trunk using inbound and outbound dial-peers (codec g711ulaw) on the HQ-RTR 2) I should put the SIP trunk in the HQ Device Pool, even though it speaks G729 to other regions as long as the DP contains an MRGL that contains the 6608 transcoder Regards, Mike Brooks CCIE#16027 (RS)
Re: [OSL | CCIE_Voice] SIP Trunk G711
This is how I have it set up. This allows the codec to remain G729 across the WAN. Also the HQ-MRGL in the HQ-DP contains a unicast MOH/g711only server. SIP Trunk: HQ device pool - HQ REGION: HQ-region to HQ-region = G711 HQ-region to BR1-region = G729 HQ-region to G729only-region = G729 HQ-region to G711only-region = G711 - HQ MRGL 6608 Transcoder Unicast MOH/g711-only region This should work... correct ? If the SIP trunk was in a G711only Device Pool then G711 would be used over the WAN. Regards, Mike Brooks CCIE#16027 (RS) On Mon, May 4, 2009 at 6:29 AM, Afatsum afat...@verizon.net wrote: A sip trunk needs an MTP, Unicast MOH as multicast MOH doesn't work and region setting as G711. You are better off making a separate region, DP, MRG, MRGL etc for SIP trunk. Faster and quicker if you preplan. - Original Message - *From:* Mike Brooks 2xcci...@gmail.com *To:* ccie_voice@onlinestudylist.com *Cc:* bryan.d.bro...@gmail.com *Sent:* Sunday, May 03, 2009 3:04 PM *Subject:* [OSL | CCIE_Voice] SIP Trunk G711 I think I understand this but just for clarification I have a few questions If I configured a SIP trunk between CM (version 4) and the HQ-RTR: 1) I should force G711 only in both directions across the trunk using inbound and outbound dial-peers (codec g711ulaw) on the HQ-RTR 2) I should put the SIP trunk in the HQ Device Pool, even though it speaks G729 to other regions as long as the DP contains an MRGL that contains the 6608 transcoder Regards, Mike Brooks CCIE#16027 (RS)
Re: [OSL | CCIE_Voice] SIP Trunk G711
From the looks of it, it should work. Except that you don't have an MTP in your MRG, or you haven't mentioned it here. The reason behind creating separate Region, DP etc for SIP trunk is that if something goes wrong with the SIP trunk setup, you are not troubleshooting all of the HQ regions, DP, locations and MRGL etc. Your setup is isloated for SIP trunk. And it doens't consume so much time either to do it. your region will still follow the restrictions over WAN etc and will xcode to move between codecs. Just matter of personal preference and speed. - Original Message - From: Mike Brooks To: Afatsum ; OSL Group Cc: bryan.d.bro...@gmail.com Sent: Sunday, May 03, 2009 3:40 PM Subject: Re: [OSL | CCIE_Voice] SIP Trunk G711 This is how I have it set up. This allows the codec to remain G729 across the WAN. Also the HQ-MRGL in the HQ-DP contains a unicast MOH/g711only server. SIP Trunk: HQ device pool - HQ REGION: HQ-region to HQ-region = G711 HQ-region to BR1-region = G729 HQ-region to G729only-region = G729 HQ-region to G711only-region = G711 - HQ MRGL 6608 Transcoder Unicast MOH/g711-only region This should work... correct ? If the SIP trunk was in a G711only Device Pool then G711 would be used over the WAN. Regards, Mike Brooks CCIE#16027 (RS) On Mon, May 4, 2009 at 6:29 AM, Afatsum afat...@verizon.net wrote: A sip trunk needs an MTP, Unicast MOH as multicast MOH doesn't work and region setting as G711. You are better off making a separate region, DP, MRG, MRGL etc for SIP trunk. Faster and quicker if you preplan. - Original Message - From: Mike Brooks To: ccie_voice@onlinestudylist.com Cc: bryan.d.bro...@gmail.com Sent: Sunday, May 03, 2009 3:04 PM Subject: [OSL | CCIE_Voice] SIP Trunk G711 I think I understand this but just for clarification I have a few questions If I configured a SIP trunk between CM (version 4) and the HQ-RTR: 1) I should force G711 only in both directions across the trunk using inbound and outbound dial-peers (codec g711ulaw) on the HQ-RTR 2) I should put the SIP trunk in the HQ Device Pool, even though it speaks G729 to other regions as long as the DP contains an MRGL that contains the 6608 transcoder Regards, Mike Brooks CCIE#16027 (RS)
Re: [OSL | CCIE_Voice] SIP Trunk G711
Oh, I see your point. So the SIPTRUNK-MRGL shoud have ONLY the unicast-moh server, software MTP, and 6608 transcoder ? What about the other resources such as 6608 conference or software conferenceor annunciator ? Shouldn't these be included as well ??? If not..why not ? Regards, Mike Brooks CCIE#16027 (RS) On Mon, May 4, 2009 at 6:57 AM, Afatsum afat...@verizon.net wrote: From the looks of it, it should work. Except that you don't have an MTP in your MRG, or you haven't mentioned it here. The reason behind creating separate Region, DP etc for SIP trunk is that if something goes wrong with the SIP trunk setup, you are not troubleshooting all of the HQ regions, DP, locations and MRGL etc. Your setup is isloated for SIP trunk. And it doens't consume so much time either to do it. your region will still follow the restrictions over WAN etc and will xcode to move between codecs. Just matter of personal preference and speed. - Original Message - *From:* Mike Brooks 2xcci...@gmail.com *To:* Afatsum afat...@verizon.net ; OSL Groupccie_voice@onlinestudylist.com *Cc:* bryan.d.bro...@gmail.com *Sent:* Sunday, May 03, 2009 3:40 PM *Subject:* Re: [OSL | CCIE_Voice] SIP Trunk G711 This is how I have it set up. This allows the codec to remain G729 across the WAN. Also the HQ-MRGL in the HQ-DP contains a unicast MOH/g711only server. SIP Trunk: HQ device pool - HQ REGION: HQ-region to HQ-region = G711 HQ-region to BR1-region = G729 HQ-region to G729only-region = G729 HQ-region to G711only-region = G711 - HQ MRGL 6608 Transcoder Unicast MOH/g711-only region This should work... correct ? If the SIP trunk was in a G711only Device Pool then G711 would be used over the WAN. Regards, Mike Brooks CCIE#16027 (RS) On Mon, May 4, 2009 at 6:29 AM, Afatsum afat...@verizon.net wrote: A sip trunk needs an MTP, Unicast MOH as multicast MOH doesn't work and region setting as G711. You are better off making a separate region, DP, MRG, MRGL etc for SIP trunk. Faster and quicker if you preplan. - Original Message - *From:* Mike Brooks 2xcci...@gmail.com *To:* ccie_voice@onlinestudylist.com *Cc:* bryan.d.bro...@gmail.com *Sent:* Sunday, May 03, 2009 3:04 PM *Subject:* [OSL | CCIE_Voice] SIP Trunk G711 I think I understand this but just for clarification I have a few questions If I configured a SIP trunk between CM (version 4) and the HQ-RTR: 1) I should force G711 only in both directions across the trunk using inbound and outbound dial-peers (codec g711ulaw) on the HQ-RTR 2) I should put the SIP trunk in the HQ Device Pool, even though it speaks G729 to other regions as long as the DP contains an MRGL that contains the 6608 transcoder Regards, Mike Brooks CCIE#16027 (RS)
Re: [OSL | CCIE_Voice] 3rd day in the raw , any body know how none location interact with specific location ?
do you setup, then once the call fails, reload your gateway. The call should go through. I had run into a similiar situation while testing and my deduction is that it has something to do with once you change location setting in different place, it doens't sync very well. So check if reboot the gateway or rebooting both CM and gateway in your situation help. Avoid using none anywhere, it may be quick but its harder to troubleshoot. - Original Message - From: jeremy co To: ccie_voice@onlinestudylist.com Sent: Sunday, May 03, 2009 10:56 AM Subject: [OSL | CCIE_Voice] 3rd day in the raw ,any body know how none location interact with specific location ? Hi, Third day in the raw, please discuss your idea about it. Is set up a simple scenario for call from HQ to 911 on pstn. HQ region : G711 within HQ Phone Location:48kbps HQ GW Location: none call will fail. What I thought is when just one location, it should treat none as unlimit. but it's not a case. CCM traces shows it deal with none as 80Kbps, can any one explain this behavior?
Re: [OSL | CCIE_Voice] SIP Trunk G711
There are limitation on what SIP trunk and FXS port can do. Try conferencing from your phone on FXS port using the SIP trunk for example. - Original Message - From: Mike Brooks To: Afatsum Cc: OSL Group ; bryan.d.bro...@gmail.com Sent: Sunday, May 03, 2009 4:15 PM Subject: Re: [OSL | CCIE_Voice] SIP Trunk G711 Oh, I see your point. So the SIPTRUNK-MRGL shoud have ONLY the unicast-moh server, software MTP, and 6608 transcoder ? What about the other resources such as 6608 conference or software conferenceor annunciator ? Shouldn't these be included as well ??? If not..why not ? Regards, Mike Brooks CCIE#16027 (RS) On Mon, May 4, 2009 at 6:57 AM, Afatsum afat...@verizon.net wrote: From the looks of it, it should work. Except that you don't have an MTP in your MRG, or you haven't mentioned it here. The reason behind creating separate Region, DP etc for SIP trunk is that if something goes wrong with the SIP trunk setup, you are not troubleshooting all of the HQ regions, DP, locations and MRGL etc. Your setup is isloated for SIP trunk. And it doens't consume so much time either to do it. your region will still follow the restrictions over WAN etc and will xcode to move between codecs. Just matter of personal preference and speed. - Original Message - From: Mike Brooks To: Afatsum ; OSL Group Cc: bryan.d.bro...@gmail.com Sent: Sunday, May 03, 2009 3:40 PM Subject: Re: [OSL | CCIE_Voice] SIP Trunk G711 This is how I have it set up. This allows the codec to remain G729 across the WAN. Also the HQ-MRGL in the HQ-DP contains a unicast MOH/g711only server. SIP Trunk: HQ device pool - HQ REGION: HQ-region to HQ-region = G711 HQ-region to BR1-region = G729 HQ-region to G729only-region = G729 HQ-region to G711only-region = G711 - HQ MRGL 6608 Transcoder Unicast MOH/g711-only region This should work... correct ? If the SIP trunk was in a G711only Device Pool then G711 would be used over the WAN. Regards, Mike Brooks CCIE#16027 (RS) On Mon, May 4, 2009 at 6:29 AM, Afatsum afat...@verizon.net wrote: A sip trunk needs an MTP, Unicast MOH as multicast MOH doesn't work and region setting as G711. You are better off making a separate region, DP, MRG, MRGL etc for SIP trunk. Faster and quicker if you preplan. - Original Message - From: Mike Brooks To: ccie_voice@onlinestudylist.com Cc: bryan.d.bro...@gmail.com Sent: Sunday, May 03, 2009 3:04 PM Subject: [OSL | CCIE_Voice] SIP Trunk G711 I think I understand this but just for clarification I have a few questions If I configured a SIP trunk between CM (version 4) and the HQ-RTR: 1) I should force G711 only in both directions across the trunk using inbound and outbound dial-peers (codec g711ulaw) on the HQ-RTR 2) I should put the SIP trunk in the HQ Device Pool, even though it speaks G729 to other regions as long as the DP contains an MRGL that contains the 6608 transcoder Regards, Mike Brooks CCIE#16027 (RS)
Re: [OSL | CCIE_Voice] ATA IVR not responding
Check the ATA configuration guide for H.323. Not sure if it's there or not. Maybe you are using the IVR wrong or something. There is also an ATA 186 FAQ document. From: kapil atrish [nice_cha...@yahoo.com] Sent: Sunday, May 03, 2009 10:43 AM To: ccie_voice@onlinestudylist.com; Michael Ciarfello Subject: RE: [OSL | CCIE_Voice] ATA IVR not responding Hi Michael, Can u give more inputs reg turning IVR off via config-register? Haven't heard abt this before, sounds interesting. Yes the Red light does come on and goes off perfectly fine. I was able to make/receive calls to/from ata phone. ATA is not reachable now can't change the software image...:( I did shake it, opened the box and tried pressing the little button but no good.:( --- On Sun, 5/3/09, Michael Ciarfello mciarfe...@iplogic.com wrote: From: Michael Ciarfello mciarfe...@iplogic.com Subject: RE: [OSL | CCIE_Voice] ATA IVR not responding To: kapil atrish nice_cha...@yahoo.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Sunday, May 3, 2009, 9:00 AM Can you turn the IVR off via a config register? Might be disabled. When you pick up the phone does the red light come on? Try the factoryreset thingie. Try re-loading the software image. Try shaking it. (lol. just kidding) From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of kapil atrish [nice_cha...@yahoo.com] Sent: Saturday, May 02, 2009 11:31 PM To: ccie_voice@onlinestudylist.com; Cliff McGlamry Subject: Re: [OSL | CCIE_Voice] ATA IVR not responding Yes, that's what I am trying to invoke the IVR but no response. I've changed the cable, Phone, Phone port, removed lan cable and tried whole thing again but no good. --- On Sat, 5/2/09, Cliff McGlamry cl...@mcglamry.net wrote: From: Cliff McGlamry cl...@mcglamry.net Subject: Re: [OSL | CCIE_Voice] ATA IVR not responding To: kapil atrish nice_cha...@yahoo.com, ccie_voice@onlinestudylist.com Date: Saturday, May 2, 2009, 11:30 PM Did you push the button on top of the ATA after picking up the phone on port 1? That's how you activate the IVR menu. What exactly have you done? - Original Message - From: kapil atrish To: ccie_voice@onlinestudylist.com Sent: Saturday, May 02, 2009 8:25 AM Subject: [OSL | CCIE_Voice] ATA IVR not responding Hi list, M not able to access the IVR menu on ATA.I know it should work on Phone 1 port, no success on any of the port. Reboot didn't help. Removed lan cable and tried, no success. Checked physical layer stuff. Has anyone had this issue earlier and any troubleshooting I can do??? M running SCCP image on it. I was testing the ATA vlan stuff and put in a Vlan and OpFlag which made my ATA unreachable. Now I don't have a switch to configure in the specific vlan and access my ATA. Since IVR is not working I am not even able to revert the ATA settings. Thanks in advance...
Re: [OSL | CCIE_Voice] ring policy is not working
If the call is on hold, then the station is idle. Jonathan On Sun, May 3, 2009 at 12:29 PM, jeremy co jeremy.coo...@gmail.com wrote: Hi, I have a problem with ccm parameter Ring Setting of Busy Station Policy in Default mode when switching between hold and resume, it shouldn't get ring notification from incoming call but it gets. changing this parameter has no effect for me. CCM 4.1.3 (sr8a) anybody face this be4? Jeremy
Re: [OSL | CCIE_Voice] V2 Blueprint - CME Version
3.3 for v2, 7.0 for v3 Jonathan On Sat, May 2, 2009 at 7:37 PM, Alex Hannah alex.han...@gmail.com wrote: Correct me if I’m wrong but the CME version on Blueprint v2 is 3.3 correct? Thanks, Alex
Re: [OSL | CCIE_Voice] Blueprint v3 -- UCCX -- 12.03 Redundancy
The blue print speiified 5 servers: CUCM 7 Pub CUCM 7 Sub CUC 7 CUPS 7 CUCCX 7 Jonathan On Sat, May 2, 2009 at 9:14 PM, Norma Exel normae...@writeme.com wrote: earlier versions of ipccx did not have redundancy (two ipccx servers) so my guess is multiple uccx running in a cluster. -NE - Original Message - From: Brian Valentine To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Blueprint v3 -- UCCX -- 12.03 Redundancy Date: Sat, 2 May 2009 12:00:21 -0400 Anyone have an idea what is required here? Is this referring to High Availability or to CTIManager Redundancy? Brian -- Be Yourself @ mail.com! Choose From 200+ Email Addresses Get a Free Account at www.mail.com!
Re: [OSL | CCIE_Voice] SIP Trunk G711
And I think SW MTP shd be in G711 only DP too. --- On Mon, 5/4/09, Afatsum afat...@verizon.net wrote: From: Afatsum afat...@verizon.net Subject: Re: [OSL | CCIE_Voice] SIP Trunk G711 To: Mike Brooks 2xcci...@gmail.com Cc: OSL Group ccie_voice@onlinestudylist.com, bryan.d.bro...@gmail.com Date: Monday, May 4, 2009, 6:15 AM There are limitation on what SIP trunk and FXS port can do. Try conferencing from your phone on FXS port using the SIP trunk for example. - Original Message - From: Mike Brooks To: Afatsum Cc: OSL Group ; bryan.d.bro...@gmail.com Sent: Sunday, May 03, 2009 4:15 PM Subject: Re: [OSL | CCIE_Voice] SIP Trunk G711 Oh, I see your point. So the SIPTRUNK-MRGL shoud have ONLY the unicast-moh server, software MTP, and 6608 transcoder ? What about the other resources such as 6608 conference or software conferenceor annunciator ? Shouldn't these be included as well ??? If not..why not ? Regards, Mike Brooks CCIE#16027 (RS) On Mon, May 4, 2009 at 6:57 AM, Afatsum afat...@verizon.net wrote: From the looks of it, it should work. Except that you don't have an MTP in your MRG, or you haven't mentioned it here. The reason behind creating separate Region, DP etc for SIP trunk is that if something goes wrong with the SIP trunk setup, you are not troubleshooting all of the HQ regions, DP, locations and MRGL etc. Your setup is isloated for SIP trunk. And it doens't consume so much time either to do it. your region will still follow the restrictions over WAN etc and will xcode to move between codecs. Just matter of personal preference and speed. - Original Message - From: Mike Brooks To: Afatsum ; OSL Group Cc: bryan.d.bro...@gmail.com Sent: Sunday, May 03, 2009 3:40 PM Subject: Re: [OSL | CCIE_Voice] SIP Trunk G711 This is how I have it set up. This allows the codec to remain G729 across the WAN. Also the HQ-MRGL in the HQ-DP contains a unicast MOH/g711only server. SIP Trunk: HQ device pool - HQ REGION: HQ-region to HQ-region = G711 HQ-region to BR1-region = G729 HQ-region to G729only-region = G729 HQ-region to G711only-region = G711 - HQ MRGL 6608 Transcoder Unicast MOH/g711-only region This should work... correct ? If the SIP trunk was in a G711only Device Pool then G711 would be used over the WAN. Regards, Mike Brooks CCIE#16027 (RS) On Mon, May 4, 2009 at 6:29 AM, Afatsum afat...@verizon.net wrote: A sip trunk needs an MTP, Unicast MOH as multicast MOH doesn't work and region setting as G711. You are better off making a separate region, DP, MRG, MRGL etc for SIP trunk. Faster and quicker if you preplan. - Original Message - From: Mike Brooks To: ccie_voice@onlinestudylist.com Cc: bryan.d.bro...@gmail.com Sent: Sunday, May 03, 2009 3:04 PM Subject: [OSL | CCIE_Voice] SIP Trunk G711 I think I understand this but just for clarification I have a few questions If I configured a SIP trunk between CM (version 4) and the HQ-RTR: 1) I should force G711 only in both directions across the trunk using inbound and outbound dial-peers (codec g711ulaw) on the HQ-RTR 2) I should put the SIP trunk in the HQ Device Pool, even though it speaks G729 to other regions as long as the DP contains an MRGL that contains the 6608 transcoder Regards, Mike Brooks CCIE#16027 (RS)
[OSL | CCIE_Voice] Suggestion for version stamping posts?
This is probably out of line, but some of us have given up on v2 and want to deal only with v3 material... So, if you are posting for v2 only (so a Unity 4, CCM 4.1, IPCC 4.0, SRST/CCME 3.3, CUE 2.3)... label it V2 If you are posting for v3 only (CUCM 7, CUCX 7, CUC 7, CUPS 7, CUE 7, CCME/SRST 7...)... label it V3 If you are posting for Gatekeeper, Dial Peers, or anything else that would apply to either (call routing, etc...), then no need to post a version number... Jonathan
