Re: [cisco-voip] PER CALL BANDWIDTH CONSUMPTION OVER ETHERNET+802.1Q
Can you link us to the sources in question? I personally need a little more context to go with your question. In general, policing a single g711ulaw call is around 93kbps, and rounding it to 100kbps still achieves the goal of policing a single call. And yes, a class based policer would police media and signaling separately. Also, I saw something on medianet at last year Cisco Live, but other than that, I'm clueless about medianet. I can't say if and how things changed once medianet came in to the picture. I'm sure Vik wasn't considering that either, based on the fact that he teaches CCIE Collab boot camps, and medianet is not a part of the blueprint. On Wed, Apr 15, 2015 at 4:22 AM abbas Wali abba...@gmail.com wrote: hi all, Vik Malhi posted that for a successful g711 call HQSW(config-pmap-c)#police 90500 8000 exc drop ! 0 packet loss now, as per Ciso medianet 4 The VoIP and signaling traffic from the VVLAN can be policed to drop at 128 kbps and 32 kbps, respectively (as any excessive traffic matching this criteria would be indicative of network abuse) Question is 128 kbps supports 1 single voice stream of g711 OR if you go with Vik, you need to multiply 90500 with the number of calls you need on that port. I will assume that the sig is classified differently and handled by diff policer on that port. many thanks -- *Abbas Wali* ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] PER CALL BANDWIDTH CONSUMPTION OVER ETHERNET+802.1Q
medianet is http://www.cisco.com/c/en/us/td/docs/solutions/Enterprise/WAN_and_MAN/QoS_SRND_40/QoSCampus_40.html Vik's post http://www.collabcert.com/blog/qos/how-much-bandwidth-does-1-call-consume/ On 15 April 2015 at 13:44, Anthony Holloway avholloway+cisco-v...@gmail.com wrote: Can you link us to the sources in question? I personally need a little more context to go with your question. In general, policing a single g711ulaw call is around 93kbps, and rounding it to 100kbps still achieves the goal of policing a single call. And yes, a class based policer would police media and signaling separately. Also, I saw something on medianet at last year Cisco Live, but other than that, I'm clueless about medianet. I can't say if and how things changed once medianet came in to the picture. I'm sure Vik wasn't considering that either, based on the fact that he teaches CCIE Collab boot camps, and medianet is not a part of the blueprint. On Wed, Apr 15, 2015 at 4:22 AM abbas Wali abba...@gmail.com wrote: hi all, Vik Malhi posted that for a successful g711 call HQSW(config-pmap-c)#police 90500 8000 exc drop ! 0 packet loss now, as per Ciso medianet 4 The VoIP and signaling traffic from the VVLAN can be policed to drop at 128 kbps and 32 kbps, respectively (as any excessive traffic matching this criteria would be indicative of network abuse) Question is 128 kbps supports 1 single voice stream of g711 OR if you go with Vik, you need to multiply 90500 with the number of calls you need on that port. I will assume that the sig is classified differently and handled by diff policer on that port. many thanks -- *Abbas Wali* ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip -- *Abbas Wali* ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] PER CALL BANDWIDTH CONSUMPTION OVER ETHERNET+802.1Q
After reading the Medianet document, I'm certain they are just giving you an example, not a definitive answer nor the best practice. While 128kbps does police the port to a single g711ulaw call, it also allows for a little wiggle room, which I like. If you are looking for the absolute minimum bandwidth needed for a g711ulaw call, you could go lower than 128kbps, but you wouldn't gain anything. Don't forget that the BIB of the phone could cause more than a single call's worth of RTP to ingress the switch port, in which case your 128kbps would not be enough and you would have issues with things such as network recording or silent monitoring. On Wed, Apr 15, 2015 at 9:19 AM abbas Wali abba...@gmail.com wrote: medianet is http://www.cisco.com/c/en/us/td/docs/solutions/Enterprise/WAN_and_MAN/QoS_SRND_40/QoSCampus_40.html Vik's post http://www.collabcert.com/blog/qos/how-much-bandwidth-does-1-call-consume/ On 15 April 2015 at 13:44, Anthony Holloway avholloway+cisco-v...@gmail.com wrote: Can you link us to the sources in question? I personally need a little more context to go with your question. In general, policing a single g711ulaw call is around 93kbps, and rounding it to 100kbps still achieves the goal of policing a single call. And yes, a class based policer would police media and signaling separately. Also, I saw something on medianet at last year Cisco Live, but other than that, I'm clueless about medianet. I can't say if and how things changed once medianet came in to the picture. I'm sure Vik wasn't considering that either, based on the fact that he teaches CCIE Collab boot camps, and medianet is not a part of the blueprint. On Wed, Apr 15, 2015 at 4:22 AM abbas Wali abba...@gmail.com wrote: hi all, Vik Malhi posted that for a successful g711 call HQSW(config-pmap-c)#police 90500 8000 exc drop ! 0 packet loss now, as per Ciso medianet 4 The VoIP and signaling traffic from the VVLAN can be policed to drop at 128 kbps and 32 kbps, respectively (as any excessive traffic matching this criteria would be indicative of network abuse) Question is 128 kbps supports 1 single voice stream of g711 OR if you go with Vik, you need to multiply 90500 with the number of calls you need on that port. I will assume that the sig is classified differently and handled by diff policer on that port. many thanks -- *Abbas Wali* ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip -- *Abbas Wali* ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] DX80 Remote Access?
You need to use a user that has CTI control of the device. On Wed, Apr 15, 2015 at 12:56 PM, Scott Voll svoll.v...@gmail.com wrote: [image: Inline image 1] unless the admin user / pwd is something other than the CCM one then I guess it doesn't work. Scott On Wed, Apr 15, 2015 at 9:17 AM, Brian Meade bmead...@vt.edu wrote: You should be able to at least pull screenshots using http://x.x.x.x/CGI/Screenshot On Wed, Apr 15, 2015 at 12:04 PM, Scott Voll svoll.v...@gmail.com wrote: Anyone have any of these DX80's? How are you supporting it remotely? Log me in Rescue does not work. Webex does not work I'm deploying these to remove sites and need to support the users and don't want to drive 4 hours each way. TIA Scott ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] DX80 Remote Access?
It is an Android Tablet at the end of the day, so I did a google search for remotely supporting these guys and this was the first link. http://www.bomgar.com/solutions/mobile-device-support On Wed, Apr 15, 2015 at 11:05 AM Scott Voll svoll.v...@gmail.com wrote: Anyone have any of these DX80's? How are you supporting it remotely? Log me in Rescue does not work. Webex does not work I'm deploying these to remove sites and need to support the users and don't want to drive 4 hours each way. TIA Scott ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] IOS firewall with Expressway?
If you are using a subinterface you may be hitting this bug: https://tools.cisco.com/bugsearch/bug/CSCue70210/?reffering_site=dumpcr Conditions: PBR to forward the traffic to loopback address gets rejected Workaround: No Workaround Tommy Schlotterer | Systems Engineer CCNA, CCNA Voice 48325 Alpha Dr. Ste. 150 Wixom, MI 48393 p 248.468.0710 e tschlotte...@netechcorp.commailto:tschlotte...@netechcorp.com w netechcorp.comhttp://netechcorp.com/ [cid:image001.png@01D0.A02A6190] [cid:image002.png@01D0.A02A6190]https://www.linkedin.com/company/75843?trk=tyahtrkInfo=tarId%3A1397760375508%2Ctas%3Anetech%2Cidx%3A3-1-5 [cid:image003.png@01D0.A02A6190] https://www.facebook.com/NetechCorporation [cid:image004.png@01D0.A02A6190] https://twitter.com/netechcorp [cid:image005.png@01D0.A02A6190] https://www.youtube.com/user/NetechCorporation From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Erick Wellnitz Sent: Wednesday, April 15, 2015 12:26 PM To: cisco-voip@puck.nether.net Subject: [cisco-voip] IOS firewall with Expressway? Anyone ever used IOS firewall with Expressway? Having a difficult time with NAT configuration getting Expressway C to talk to the Expressway E public IP. ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] DX80 Remote Access?
[image: Inline image 1] unless the admin user / pwd is something other than the CCM one then I guess it doesn't work. Scott On Wed, Apr 15, 2015 at 9:17 AM, Brian Meade bmead...@vt.edu wrote: You should be able to at least pull screenshots using http://x.x.x.x/CGI/Screenshot On Wed, Apr 15, 2015 at 12:04 PM, Scott Voll svoll.v...@gmail.com wrote: Anyone have any of these DX80's? How are you supporting it remotely? Log me in Rescue does not work. Webex does not work I'm deploying these to remove sites and need to support the users and don't want to drive 4 hours each way. TIA Scott ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] SRST for UCCX CTI Route Points
Hello, Any ideas for an SRST configuration for a WAN outage that is preventing calls to a UCCX CTI Route Point? Calls to the RP are fine via the attached PRI's but calls via the WAN cannot see the route point. Michael Armstrong | Sr. Unified Communications Admin | Colorado Community College System | P:720.858.2882 | C:720.891.6010 ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] SRST for UCCX CTI Route Points
I think you should be able to add translations or number expansion from extension to E.164 in your SRST config. On Wed, Apr 15, 2015 at 9:53 AM, Armstrong, Michael michael.armstr...@cccs.edu wrote: Hello, Any ideas for an SRST configuration for a WAN outage that is preventing calls to a UCCX CTI Route Point? Calls to the RP are fine via the attached PRI’s but calls via the WAN cannot see the route point. *Michael Armstrong* | Sr. Unified Communications Admin | *Colorado Community College System* | P:720.858.2882 | C:720.891.6010 ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] DX80 Remote Access?
Anyone have any of these DX80's? How are you supporting it remotely? Log me in Rescue does not work. Webex does not work I'm deploying these to remove sites and need to support the users and don't want to drive 4 hours each way. TIA Scott ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] IOS firewall with Expressway?
Great thought but no subinterface. They added an Ethernet card to have enough interfaces. On Wed, Apr 15, 2015 at 10:28 AM, Tommy Schlotterer tschlotte...@netechcorp.com wrote: If you are using a subinterface you may be hitting this bug: https://tools.cisco.com/bugsearch/bug/CSCue70210/?reffering_site=dumpcr Conditions: PBR to forward the traffic to loopback address gets rejected Workaround: No Workaround *Tommy Schlotterer* | Systems Engineer CCNA, CCNA Voice 48325 Alpha Dr. Ste. 150 Wixom, MI 48393 *p* 248.468.0710 *e* tschlotte...@netechcorp.com *w **netechcorp.com* http://netechcorp.com/ [image: cid:DE00F175-A6C9-45A6-B3AC-D658551F1586] [image: cid:62EB95BF-20B4-4A1B-98FE-9E042594730A] https://www.linkedin.com/company/75843?trk=tyahtrkInfo=tarId%3A1397760375508%2Ctas%3Anetech%2Cidx%3A3-1-5 [image: cid:47C21B6C-578D-4D72-BFF4-8A482CE7A978] https://www.facebook.com/NetechCorporation [image: cid:A362BD4D-9EC8-47CC-96A8-8A17AF38C15C] https://twitter.com/netechcorp [image: cid:F52A69B8-DA49-4CD7-91E9-C057917D90C2] https://www.youtube.com/user/NetechCorporation *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf Of *Erick Wellnitz *Sent:* Wednesday, April 15, 2015 12:26 PM *To:* cisco-voip@puck.nether.net *Subject:* [cisco-voip] IOS firewall with Expressway? Anyone ever used IOS firewall with Expressway? Having a difficult time with NAT configuration getting Expressway C to talk to the Expressway E public IP. ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] DX80 Remote Access?
You should be able to at least pull screenshots using http://x.x.x.x/CGI/Screenshot On Wed, Apr 15, 2015 at 12:04 PM, Scott Voll svoll.v...@gmail.com wrote: Anyone have any of these DX80's? How are you supporting it remotely? Log me in Rescue does not work. Webex does not work I'm deploying these to remove sites and need to support the users and don't want to drive 4 hours each way. TIA Scott ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] DX80 Remote Access?
Same Error. On Wed, Apr 15, 2015 at 10:11 AM, Brian Meade bmead...@vt.edu wrote: You need to use a user that has CTI control of the device. On Wed, Apr 15, 2015 at 12:56 PM, Scott Voll svoll.v...@gmail.com wrote: [image: Inline image 1] unless the admin user / pwd is something other than the CCM one then I guess it doesn't work. Scott On Wed, Apr 15, 2015 at 9:17 AM, Brian Meade bmead...@vt.edu wrote: You should be able to at least pull screenshots using http://x.x.x.x/CGI/Screenshot On Wed, Apr 15, 2015 at 12:04 PM, Scott Voll svoll.v...@gmail.com wrote: Anyone have any of these DX80's? How are you supporting it remotely? Log me in Rescue does not work. Webex does not work I'm deploying these to remove sites and need to support the users and don't want to drive 4 hours each way. TIA Scott ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] DX80 Remote Access?
This is the problem... The Bomgar Android client allows you to securely support Android smartphones and tablets, including chatting with the end-user, transfering files, and viewing system information. No remote viewing. Scott On Wed, Apr 15, 2015 at 10:02 AM, Anthony Holloway avholloway+cisco-v...@gmail.com wrote: It is an Android Tablet at the end of the day, so I did a google search for remotely supporting these guys and this was the first link. http://www.bomgar.com/solutions/mobile-device-support On Wed, Apr 15, 2015 at 11:05 AM Scott Voll svoll.v...@gmail.com wrote: Anyone have any of these DX80's? How are you supporting it remotely? Log me in Rescue does not work. Webex does not work I'm deploying these to remove sites and need to support the users and don't want to drive 4 hours each way. TIA Scott ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] DX80 Remote Access?
Can you pull the console log? On Wed, Apr 15, 2015 at 2:05 PM, Scott Voll svoll.v...@gmail.com wrote: Same Error. On Wed, Apr 15, 2015 at 10:11 AM, Brian Meade bmead...@vt.edu wrote: You need to use a user that has CTI control of the device. On Wed, Apr 15, 2015 at 12:56 PM, Scott Voll svoll.v...@gmail.com wrote: [image: Inline image 1] unless the admin user / pwd is something other than the CCM one then I guess it doesn't work. Scott On Wed, Apr 15, 2015 at 9:17 AM, Brian Meade bmead...@vt.edu wrote: You should be able to at least pull screenshots using http://x.x.x.x/CGI/Screenshot On Wed, Apr 15, 2015 at 12:04 PM, Scott Voll svoll.v...@gmail.com wrote: Anyone have any of these DX80's? How are you supporting it remotely? Log me in Rescue does not work. Webex does not work I'm deploying these to remove sites and need to support the users and don't want to drive 4 hours each way. TIA Scott ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] DX80 Remote Access?
Associate a ccm end user to the device and user those credentials to authenticate. Dennis Heim | Emerging Technology Architect (Collaboration) World Wide Technology, Inc. | +1 314-212-1814 [twitter]https://twitter.com/CollabSensei [chat]xmpp:dennis.h...@wwt.com[Phone]tel:+13142121814[video]sip:dennis.h...@wwt.com Innovation happens on project squared -- http://www.projectsquared.comhttp://www.projectsquared.com/ Click here to join me in my Collaboration Meeting Roomhttps://wwt.webex.com/meet/dennis.heim From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Scott Voll Sent: Wednesday, April 15, 2015 12:57 PM To: Brian Meade; cisco-voip@puck.nether.net Subject: Re: [cisco-voip] DX80 Remote Access? [Inline image 1] unless the admin user / pwd is something other than the CCM one then I guess it doesn't work. Scott On Wed, Apr 15, 2015 at 9:17 AM, Brian Meade bmead...@vt.edumailto:bmead...@vt.edu wrote: You should be able to at least pull screenshots using http://x.x.x.x/CGI/Screenshot On Wed, Apr 15, 2015 at 12:04 PM, Scott Voll svoll.v...@gmail.commailto:svoll.v...@gmail.com wrote: Anyone have any of these DX80's? How are you supporting it remotely? Log me in Rescue does not work. Webex does not work I'm deploying these to remove sites and need to support the users and don't want to drive 4 hours each way. TIA Scott ___ cisco-voip mailing list cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] PER CALL BANDWIDTH CONSUMPTION OVER ETHERNET+802.1Q
Anthony, yes makes sense. but for the sake of argu. a single phone with even with BIB how many max g711 streams it can get to. 3? if so, for a safe figure can multiply by 3. moreover, I dont really understand this statement police 90500 8000 exc drop - as per docs, the actual transmission is 8k but on the avg. the max is 90k ( plz correct if wrong) On 15 April 2015 at 18:10, Anthony Holloway avholloway+cisco-v...@gmail.com wrote: After reading the Medianet document, I'm certain they are just giving you an example, not a definitive answer nor the best practice. While 128kbps does police the port to a single g711ulaw call, it also allows for a little wiggle room, which I like. If you are looking for the absolute minimum bandwidth needed for a g711ulaw call, you could go lower than 128kbps, but you wouldn't gain anything. Don't forget that the BIB of the phone could cause more than a single call's worth of RTP to ingress the switch port, in which case your 128kbps would not be enough and you would have issues with things such as network recording or silent monitoring. On Wed, Apr 15, 2015 at 9:19 AM abbas Wali abba...@gmail.com wrote: medianet is http://www.cisco.com/c/en/us/td/docs/solutions/Enterprise/WAN_and_MAN/QoS_SRND_40/QoSCampus_40.html Vik's post http://www.collabcert.com/blog/qos/how-much-bandwidth-does-1-call-consume/ On 15 April 2015 at 13:44, Anthony Holloway avholloway+cisco-v...@gmail.com wrote: Can you link us to the sources in question? I personally need a little more context to go with your question. In general, policing a single g711ulaw call is around 93kbps, and rounding it to 100kbps still achieves the goal of policing a single call. And yes, a class based policer would police media and signaling separately. Also, I saw something on medianet at last year Cisco Live, but other than that, I'm clueless about medianet. I can't say if and how things changed once medianet came in to the picture. I'm sure Vik wasn't considering that either, based on the fact that he teaches CCIE Collab boot camps, and medianet is not a part of the blueprint. On Wed, Apr 15, 2015 at 4:22 AM abbas Wali abba...@gmail.com wrote: hi all, Vik Malhi posted that for a successful g711 call HQSW(config-pmap-c)#police 90500 8000 exc drop ! 0 packet loss now, as per Ciso medianet 4 The VoIP and signaling traffic from the VVLAN can be policed to drop at 128 kbps and 32 kbps, respectively (as any excessive traffic matching this criteria would be indicative of network abuse) Question is 128 kbps supports 1 single voice stream of g711 OR if you go with Vik, you need to multiply 90500 with the number of calls you need on that port. I will assume that the sig is classified differently and handled by diff policer on that port. many thanks -- *Abbas Wali* ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip -- *Abbas Wali* -- *Abbas Wali* ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] Rdirecting number over SIP PSTN,
Guys, I have an interesting situation. I am running calls through a sip profile to set the redirecting party on the call to be able to do TEHO out the SIP trunk that has virtual trunk groups to many Rate Centers. It works great. However, we have a few numbers that are called that we get the far-ends unity voicemail, and get a generic greeting. I believe what is happening is since we’re sending redirecting number (RDIS) to do Tail end hop off (TEHO) that is being passed all the way to the far end voicemail server. That server is trying to match the RDIS to a mailbox and it can’t so you get the generic greeting. RDNIS deliver through PRIs has been possible before, but I’ve never actually seen it done. Now with SIP it seems it is happening end to end. Any ideas on how to deal with this? For now I am just not setting redirecting party on the numbers we have identified as broken, but that is lame. Thanks, -Nate ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] PER CALL BANDWIDTH CONSUMPTION OVER ETHERNET+802.1Q
I read somewhere that a phone could generate up to 2.5x call traffic with its BIB. Multiplying by 3x would still be acceptable, I would think. The 8000 is a burst threshold over the policed rate. It's always been 8000 in my experience, but probably only because no one knows enough to adjust it You cannot have an average and a max rate with voice. It's constant (excluding VAD). Video on the other hand is variable. If you are studying for your CCIE, I can share with you that Cisco has publicly stated they have some percentage of forgiveness. I.e., If they say 3 g711ulaw calls worth of bandwidth, and I enter 90*3=270, but you enter 93*3=279 (or even round up to 280), we would both get the points. What the percentage is, I don't recall. I want to say it was like 10%. So for every 100kbps, you can be plus or minus 10kbps. On Wed, Apr 15, 2015 at 1:00 PM abbas Wali abba...@gmail.com wrote: Anthony, yes makes sense. but for the sake of argu. a single phone with even with BIB how many max g711 streams it can get to. 3? if so, for a safe figure can multiply by 3. moreover, I dont really understand this statement police 90500 8000 exc drop - as per docs, the actual transmission is 8k but on the avg. the max is 90k ( plz correct if wrong) On 15 April 2015 at 18:10, Anthony Holloway avholloway+cisco-v...@gmail.com wrote: After reading the Medianet document, I'm certain they are just giving you an example, not a definitive answer nor the best practice. While 128kbps does police the port to a single g711ulaw call, it also allows for a little wiggle room, which I like. If you are looking for the absolute minimum bandwidth needed for a g711ulaw call, you could go lower than 128kbps, but you wouldn't gain anything. Don't forget that the BIB of the phone could cause more than a single call's worth of RTP to ingress the switch port, in which case your 128kbps would not be enough and you would have issues with things such as network recording or silent monitoring. On Wed, Apr 15, 2015 at 9:19 AM abbas Wali abba...@gmail.com wrote: medianet is http://www.cisco.com/c/en/us/td/docs/solutions/Enterprise/WAN_and_MAN/QoS_SRND_40/QoSCampus_40.html Vik's post http://www.collabcert.com/blog/qos/how-much-bandwidth-does-1-call-consume/ On 15 April 2015 at 13:44, Anthony Holloway avholloway+cisco-v...@gmail.com wrote: Can you link us to the sources in question? I personally need a little more context to go with your question. In general, policing a single g711ulaw call is around 93kbps, and rounding it to 100kbps still achieves the goal of policing a single call. And yes, a class based policer would police media and signaling separately. Also, I saw something on medianet at last year Cisco Live, but other than that, I'm clueless about medianet. I can't say if and how things changed once medianet came in to the picture. I'm sure Vik wasn't considering that either, based on the fact that he teaches CCIE Collab boot camps, and medianet is not a part of the blueprint. On Wed, Apr 15, 2015 at 4:22 AM abbas Wali abba...@gmail.com wrote: hi all, Vik Malhi posted that for a successful g711 call HQSW(config-pmap-c)#police 90500 8000 exc drop ! 0 packet loss now, as per Ciso medianet 4 The VoIP and signaling traffic from the VVLAN can be policed to drop at 128 kbps and 32 kbps, respectively (as any excessive traffic matching this criteria would be indicative of network abuse) Question is 128 kbps supports 1 single voice stream of g711 OR if you go with Vik, you need to multiply 90500 with the number of calls you need on that port. I will assume that the sig is classified differently and handled by diff policer on that port. many thanks -- *Abbas Wali* ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip -- *Abbas Wali* -- *Abbas Wali* ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] License requirements to upgrade from 12.4 to 15
I have a customer with 18 2800 voice gateways in production and 5 cold spares. Are they required to purchase a license in order to upgrade from 12.4 to 15 for the base and/or voice features? For the cold spares would they need to have those under SmartNET in order to swap them out with a failed router or can they be upgraded and license transferred in the event of a failure? Hank ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Rdirecting number over SIP PSTN,
Nate, I am pretty sure your diagnosis is correct. I had the exact same issue happen to me. In my case, I controlled both ends so I made the redirecting number an alternate extension in the cuc voicemail account and all was fine (it was only one number so it was a simple solution). Couldn't you fixup the RDINS with a transformation mask? Thanks, Ryan Original Message From: NateCCIE . natec...@gmail.com Sent: Wednesday, April 15, 2015 01:48 PM To: cisco-voip (cisco-voip@puck.nether.net) cisco-voip@puck.nether.net Subject: [cisco-voip] Rdirecting number over SIP PSTN, Guys, I have an interesting situation. I am running calls through a sip profile to set the redirecting party on the call to be able to do TEHO out the SIP trunk that has virtual trunk groups to many Rate Centers. It works great. However, we have a few numbers that are called that we get the far-ends unity voicemail, and get a generic greeting. I believe what is happening is since we’re sending redirecting number (RDIS) to do Tail end hop off (TEHO) that is being passed all the way to the far end voicemail server. That server is trying to match the RDIS to a mailbox and it can’t so you get the generic greeting. RDNIS deliver through PRIs has been possible before, but I’ve never actually seen it done. Now with SIP it seems it is happening end to end. Any ideas on how to deal with this? For now I am just not setting redirecting party on the numbers we have identified as broken, but that is lame. Thanks, -Nate ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] SRST for UCCX CTI Route Points
You can make SRST configurations for calls to be sent to the DIDs of the CTI Route Points. But what would make this fail is the call from the CTI RP to the CTI Port is a Call redirect. This means a call redirect request is sent new leg is created with calling number same as the original calling number but the called number is now the CTI port and the previous call leg which reached the CTI RP via SRST is disconnected. What happens to the call redirect request? How would you reach the CTI port? that’s the question. Regards, Abhiram Kramadhati Technical Solutions Manager, CBABU CCIE Voice # 40065 From: Armstrong, Michael michael.armstr...@cccs.edumailto:michael.armstr...@cccs.edu Date: Thursday, 16 April 2015 1:53 am To: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net Subject: [cisco-voip] SRST for UCCX CTI Route Points Hello, Any ideas for an SRST configuration for a WAN outage that is preventing calls to a UCCX CTI Route Point? Calls to the RP are fine via the attached PRI’s but calls via the WAN cannot see the route point. Michael Armstrong | Sr. Unified Communications Admin| Colorado Community College System | P:720.858.2882 | C:720.891.6010 ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] Please remove me from this email distribution list
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Re: [cisco-voip] SRST for UCCX CTI Route Points
From: akramadh akram...@cisco.commailto:akram...@cisco.com Date: Thursday, 16 April 2015 9:42 am To: Armstrong, Michael michael.armstr...@cccs.edumailto:michael.armstr...@cccs.edu, cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net Subject: Re: [cisco-voip] SRST for UCCX CTI Route Points You can make SRST configurations for calls to be sent to the DIDs of the CTI Route Points. But what would make this fail is the call from the CTI RP to the CTI Port is a Call redirect. This means a call redirect request is sent with the redirect destination as the CTI port the UCCX has chose and then a new leg has to be created with calling number same as the original calling number but the called number is now the CTI port and the previous call leg which reached the CTI RP via SRST disconnected. What happens to the call redirect request? How would you reach the CTI port? that’s the problem Regards, Abhiram Kramadhati Technical Solutions Manager, CBABU CCIE Voice # 40065 From: Armstrong, Michael michael.armstr...@cccs.edumailto:michael.armstr...@cccs.edu Date: Thursday, 16 April 2015 1:53 am To: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net Subject: [cisco-voip] SRST for UCCX CTI Route Points Hello, Any ideas for an SRST configuration for a WAN outage that is preventing calls to a UCCX CTI Route Point? Calls to the RP are fine via the attached PRI’s but calls via the WAN cannot see the route point. Michael Armstrong | Sr. Unified Communications Admin| Colorado Community College System | P:720.858.2882 | C:720.891.6010 ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Finesse customisations - feedback
Abhiram, Thanks for soliciting the feedback. I've found a few items on my list that I'll share. All my experience is from UCCX by the way. - UCCX Finesse API doesn't support passing call variables in the MAKE_CALL request. This has been a huge problem for me, since we are using this API to place calls from a CRM. We want to pass the account numbers into Finesse but it won't let us. This is documented in the Finesse dev guide as a limitation/not supported. But I wish the BU would reconsider this to provide more capabilities into UCCX/Finesse API. - UCCX Finesse API for 'Update Call Variable' data doesn't allow an admin user to submit requests on behalf of agents. We tried to use this to have the CRM update the call variable after the fact (due to #1 above) but the CRM wasn't able to do it because the admin credentials on Finesse don't have access to this API. It has to be the agent themselves who update their own call, which is a problem since the CRM doesn't have the agent's credentials. - Finesse gadgets that are placed in a 'tab' in the desktop layout have frames added around them. There is no way to prevent Finesse from adding this frame/border.However, if you add the gadget to the global 'page' inside Finesse, then it does not add a frame.I think that Finesse should allow the admin to choose whether or not a frame/border is added to a gadget, regardless of whether the gadget is added to 'page' or 'tab'. - Finesse default Team Performance gadget. It would be nice to have the system reason codes added into this gadget. Today, if the user is 'not ready' due to a custom code, it is displayed in the gadget. However, if they are not ready due to an inbound non-acd call or outbound call, the gadget just shows as 'not ready'. This has let to supervisors not being able to determine whether an agent is slacking off, or on an outbound call. - Functionality to alert/remind a user that they are in 'not ready' state. I've had problems in small call centers where an agent will forgot their in not ready, or will RONA a call and not realize. In any case, they end up not ready and don't realize it. If Finesse could have an option to remind users they are not ready, this would help in certain environments This should definitely be an option that could be enabled, since some environments would not want to have this feature. - Access to source code for the callcontrol teamperformance gadget (or ability to modify those gadgets) - 'Finesse Lite' that can be added into a Jabber custom tab. There are some third party solutions for this today, but should be something that could be included by default. Justin On Tue, Apr 14, 2015 at 4:15 AM, Abhiram Kramadhati (akramadh) akram...@cisco.com wrote: Hi Erik, This is actually a follow up of that and also a follow-up of the Ask-the-Expert session I did on Finesse. I noticed that there are quite a few partners on this group who might not have had a chance to provide feedback on those two events and hence the question. Cheers, Abhiram Kramadhati Sent from my iPhone On 14 Apr 2015, at 5:46 pm, Erik Goppel egop...@gmail.com wrote: Abhiram, Please have a look at the discussion on the community. https://communities.cisco.com/thread/51226 this list was set up ate the UCCE Tech Summit in Amsterdam, and has been discussed with Ted Phipps with the whole group there. Also please ask your question in that thread or the community, as that is the place where ATP Partners, will look, as not all of them are subscribed to a list like this. Thanks, Erik Goppel Technical Consultant Unified Communications Dimension Data Netherlands On Tue, Apr 14, 2015 at 4:57 AM, Abhiram Kramadhati (akramadh) akram...@cisco.com wrote: Hi all, Have received good amount of feedback about Finesse and some of the features. We are currently working on this and in that regard, would appreciate some feedback on this from your experience from the field: *Which are the most common customisations you have been asked from customers when you are trying to deploy Finesse? - a customisation need not be a new gadget integrating to another server. It could just be modifying out-of-the-box Finesse behaviour. * If you have got some time and if you have something to share about this, please drop me a line at *akram...@cisco.com akram...@cisco.com* Thanks in advance! *Abhiram Kramadhati* Technical Solutions Manager Customer Solutions Success team, CBABU akram...@cisco.com Phone: *+61 2 8446 6257 %2B61%202%208446%206257* CCIE Voice - 40065 *Cisco Systems Australia Pty Limited* The Forum 201 Pacific Highway 2065 St Leonards Australia Cisco.com http://www.cisco.com/web/AU/ http://wwwin.cisco.com/marketing/corporate/brand/intelbrand/brandstrat/signature/Insert%20your%20LinkedIn%20link
Re: [cisco-voip] License requirements to upgrade from 12.4 to 15
2016 is last day of tac support for the 2800’s. Technically any code upgrade requires proper support. Ironically, the 1800/2800/3800 series lacked the ability to lock the ios by feature set other than say gatekeeper. When the G2’s came out we got feature locking. Then by around 15.2 we went back to right to use licenses because it was such a pain to administrator for all parties involved. Customers are still required legally to purchase the licenses. In your scnerio.. they need to have purchased the voice feature set. Then the devices need to be under support to be entitled to any version of IOS. That is my take on this scenario. Dennis Heim | Emerging Technology Architect (Collaboration) World Wide Technology, Inc. | +1 314-212-1814 [twitter]https://twitter.com/CollabSensei [chat]xmpp:dennis.h...@wwt.com[Phone]tel:+13142121814[video]sip:dennis.h...@wwt.com Innovation happens on project squared -- http://www.projectsquared.comhttp://www.projectsquared.com/ Click here to join me in my Collaboration Meeting Roomhttps://wwt.webex.com/meet/dennis.heim From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Hank Keleher (AM) Sent: Wednesday, April 15, 2015 4:54 PM To: cisco-voip@puck.nether.net Subject: [cisco-voip] License requirements to upgrade from 12.4 to 15 I have a customer with 18 2800 voice gateways in production and 5 cold spares. Are they required to purchase a license in order to upgrade from 12.4 to 15 for the base and/or voice features? For the cold spares would they need to have those under SmartNET in order to swap them out with a failed router or can they be upgraded and license transferred in the event of a failure? Hank ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Cisco Finesse vs CAD
We have depolyed it for one customer so far. It better and worse than CAD at the same time. It's like when Cisco moved from CUPC to Jabber. As it's based on new tech it has potential but there's a huge backlog for feature parity with CAD. What do you say to customers who were used to basic functions like chat, blind transfer and recent call list. The agent cant even see a call history list! I have given up hope with Cisco... maybe in UCCX 11. On Tue, Apr 14, 2015 at 5:40 PM, Mathew Miller miller.mat...@gmail.com wrote: I would assume that is a fair assessment that CAD will be phased out. There is a decent white paper on the differences between CAD and Finesse here http://www.cisco.com/c/en/us/products/collateral/customer-collaboration/unified-contact-center-express/white-paper-c11-730883.pdf (It's slightly out of date for 10.6). On Mon, Apr 13, 2015 at 11:58 AM, JASON BURWELL jason.burw...@foundersfcu.com wrote: I am planning a UCCX upgrade and based on recent posts, am I correct to understand that CAD is now being phased out in favor of Finesse? Also, can anyone currently on Finesse share any information on how well it works? I initially heard some negative feedback about it but that was earlier in the lifecycle so I am keeping my fingers crossed the stability has improved. Can anyone point me to a link that has any screen shots, user training or any other information giving me a preview of what the end user experience will be like? For those who have migrated from CAD to Finesse, what was the experience with training and user acceptance on Finesse? Thanks Jason ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] PER CALL BANDWIDTH CONSUMPTION OVER ETHERNET+802.1Q
hi all, Vik Malhi posted that for a successful g711 call HQSW(config-pmap-c)#police 90500 8000 exc drop ! 0 packet loss now, as per Ciso medianet 4 The VoIP and signaling traffic from the VVLAN can be policed to drop at 128 kbps and 32 kbps, respectively (as any excessive traffic matching this criteria would be indicative of network abuse) Question is 128 kbps supports 1 single voice stream of g711 OR if you go with Vik, you need to multiply 90500 with the number of calls you need on that port. I will assume that the sig is classified differently and handled by diff policer on that port. many thanks -- *Abbas Wali* ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
