Re: [cisco-voip] CUCM PLAR SIP 8851
resolved it was when you add the SIp dial rule you have to "Add Plar" button and we were trying the Add Pattern instead. specify the button and thats that. thank you all. On Tue, 6 Jul 2021 at 00:41, Myron Young wrote: > Don’t think so, but not sure. I’m using 12-8-1-0001-455 > > On Jul 5, 2021, at 8:11 AM, Abbas Wali wrote: > > > could it be the phone firmware, i am using 12-5-1sr2-2 > > On Fri, 2 Jul 2021 at 19:28, Myron Young wrote: > >> I have it working with an 8841 and we have both the SIP dial rule as well >> as the blank translation pattern. >> >> Also there shouldn’t be any CSS on the phone, only on the line which has >> the Plar destination >> >> On Jul 2, 2021, at 2:10 PM, Anthony Holloway < >> avholloway+cisco-v...@gmail.com> wrote: >> >> >> It's been a while for me, but I think you need both: blank xlate and sip >> dial rule. Contrast this with a sccp phone, which only needs the xlate. >> >> Did you do both, or just one and then the other? >> >> On Fri, Jul 2, 2021 at 5:49 AM Abbas Wali wrote: >> >>> having issues with config. PLAR for a SIP 8851 phone. tried the blank >>> Part didnt work, tried the SIP dial rules with button or blank Pattern >>> still the same. any help? >>> >>> thank you >>> >>> --*AW* >>> ___ >>> 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 >> >> > > -- > *Abbas Wali* > > -- *Abbas Wali* ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] CUCM PLAR SIP 8851
could it be the phone firmware, i am using 12-5-1sr2-2 On Fri, 2 Jul 2021 at 19:28, Myron Young wrote: > I have it working with an 8841 and we have both the SIP dial rule as well > as the blank translation pattern. > > Also there shouldn’t be any CSS on the phone, only on the line which has > the Plar destination > > On Jul 2, 2021, at 2:10 PM, Anthony Holloway < > avholloway+cisco-v...@gmail.com> wrote: > > > It's been a while for me, but I think you need both: blank xlate and sip > dial rule. Contrast this with a sccp phone, which only needs the xlate. > > Did you do both, or just one and then the other? > > On Fri, Jul 2, 2021 at 5:49 AM Abbas Wali wrote: > >> having issues with config. PLAR for a SIP 8851 phone. tried the blank >> Part didnt work, tried the SIP dial rules with button or blank Pattern >> still the same. any help? >> >> thank you >> >> --*AW* >> ___ >> 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 > > -- *Abbas Wali* ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] cucm9 insert Local enduser via bat AD sync enable
thanks Anthony, 1 - disabling LDap for a little while will not affect the enduser profiles etc. I think it waits for the garbage collector 2- this one is interesting - never done that before? any link please On 19 February 2016 at 15:44, Anthony Holloway < avholloway+cisco-v...@gmail.com> wrote: > You might hit this defect, and not be able to BAT the local users in. > > https://tools.cisco.com/bugsearch/bug/CSCul81586 > > You have two options from here: > > 1) Disable LDAP Authentication briefly as you BAT import the local users > then enable it again > > 2) Use AXL to create the end users instead of BAT > > Otherwise, if you don't hit the issue, then you can just BAT them in like > normal. > > On Thu, Feb 18, 2016 at 12:37 PM, abbas Wali <abba...@gmail.com> wrote: > >> hi all, >> >> need to add around 100 local users in CUCM 9 which is sync with AD >> >> cant add them to AD as these are external accounts. >> >> any idea? >> >> -- >> thanks >> >> >> ___ >> 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] cucm9 insert Local enduser via bat AD sync enable
App user will be even more ideal but I dont think we can bat app users, can we? On 18 February 2016 at 19:00, Haas, Neal <nh...@co.fresno.ca.us> wrote: > How about an Application user account or an End User account? > > > > Thank you, > > Neal Haas > > > > *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf > Of *abbas Wali > *Sent:* Thursday, February 18, 2016 10:38 AM > *To:* cisco-voip@puck.nether.net > *Subject:* [cisco-voip] cucm9 insert Local enduser via bat AD sync enable > > > > hi all, > > > > need to add around 100 local users in CUCM 9 which is sync with AD > > > > cant add them to AD as these are external accounts. > > > > any idea? > > > > -- > > thanks > > > -- *Abbas Wali* ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] UCCX 9 prompts gibberish
Just found the old window sound recorder ( w2k3) it has the option to save in G729 voiceAge. On 17 February 2016 at 02:32, Anthony Holloway < avholloway+cisco-v...@gmail.com> wrote: > Correct. Do not change this setting unless you're ready with all newly > encoded G711ulaw prompts and region settings or transcoders which will > allow devices to talk to it at G711ulaw. > > On Tue, Feb 16, 2016 at 5:51 PM, abbas wali <abba...@gmail.com> wrote: > >> Yes indeed the system is set for G729 in the SParameters. >> >> >> >> But if changed all the existing prompts in 729 will not play? >> >> >> >> *From:* James Buchanan [mailto:james.buchan...@gmail.com] >> *Sent:* 16 February 2016 14:29 >> *To:* abbas wali <abba...@gmail.com> >> *Cc:* cisco-voip@puck.nether.net >> *Subject:* Re: [cisco-voip] UCCX 9 prompts gibberish >> >> >> >> Hello, >> >> In the System settings, are you set to use G729 for your prompts or G711? >> UCCX will not play one or the other. If you record G711 and upload to a >> system set to play G729, that'll be the result. >> >> Thanks, >> >> James >> >> >> >> On Tue, Feb 16, 2016 at 9:26 AM, abbas wali <abba...@gmail.com> wrote: >> >> Hi guys, >> >> >> >> Just need a quick help here. >> >> >> >> Every prompt I record ( via UnityC or Audacity etc ) upload and can only >> hear gibberish. >> >> >> >> But when I load an already saved file in G729 – it plays okay. >> >> >> >> I have checked an my regions for phone dpool and application trigger are >> in the same region set to g711. >> >> >> >> The only other thing is that I am calling from a softphone vpn’ed. ( but >> that shouldn’t make any difference ) >> >> >> >> Please help. >> >> Thanks in advance. >> >> >> ___ >> 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 >> >> > -- *Abbas Wali* ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] UCCX 9 prompts gibberish
Yes indeed the system is set for G729 in the SParameters. But if changed all the existing prompts in 729 will not play? From: James Buchanan [mailto:james.buchan...@gmail.com] Sent: 16 February 2016 14:29 To: abbas wali <abba...@gmail.com> Cc: cisco-voip@puck.nether.net Subject: Re: [cisco-voip] UCCX 9 prompts gibberish Hello, In the System settings, are you set to use G729 for your prompts or G711? UCCX will not play one or the other. If you record G711 and upload to a system set to play G729, that'll be the result. Thanks, James On Tue, Feb 16, 2016 at 9:26 AM, abbas wali <abba...@gmail.com <mailto:abba...@gmail.com> > wrote: Hi guys, Just need a quick help here. Every prompt I record ( via UnityC or Audacity etc ) upload and can only hear gibberish. But when I load an already saved file in G729 – it plays okay. I have checked an my regions for phone dpool and application trigger are in the same region set to g711. The only other thing is that I am calling from a softphone vpn’ed. ( but that shouldn’t make any difference ) Please help. Thanks in advance. ___ cisco-voip mailing list cisco-voip@puck.nether.net <mailto: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] UCCX 9 prompts gibberish
have got that in window xp sound recorder. but again all this is giving back G711. I think the last resort is upload the file as MoH file and then dloading again to get G729 :(( pain.. On 16 February 2016 at 16:42, Andreas Sikkema <asikk...@unet.nl> wrote: > James, > > > No, just for G711. I don't have a solution for G729. Hopefully someone > else > > does. > > G.729 is patent encumbered, so no "free" (beer or otherwise) solutions > there. > > Also, CCIT U-Law is G.711 u-law is PCM u-law is PCMU. There must be > one of these in Audacity. > > -- > Andreas Sikkema > ___ > 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] UCCX 9 prompts gibberish
are we talking about converting them to G729? i took all the steps few times now and everytime its the same. even the file show it is been saved in 64kbps instead of 8. [image: Inline images 1] On 16 February 2016 at 16:03, James Buchanan <james.buchan...@gmail.com> wrote: > You can do it in Audacity under Other Formats when you export the file. > However, I've never seen an option for G729. > > On Tue, Feb 16, 2016 at 10:17 AM, abbas Wali <abba...@gmail.com> wrote: > >> yes seen them but again they save it in g711's. also in the new audacity >> there is no CCIT U-Law !! >> >> On 16 February 2016 at 15:12, James Buchanan <james.buchan...@gmail.com> >> wrote: >> >>> Even to g729? >>> >>> On 16 Feb 2016, at 10:11 AM, Haas, Neal <nh...@co.fresno.ca.us> wrote: >>> >>> I use audacity all of the time to convert to a “Cisco” format.. here are >>> two links for you >>> >>> >>> http://www.netcraftsmen.com/uc-toolkit-using-audacity-to-create-and-edit-cisco-uccx-prompts/ >>> >>> >>> >>> >>> http://xyfon.com/tech-tips/saving-wav-files-for-cisco-unified-call-centre-express-prompts-uccx-using-audacity/ >>> >>> >>> >>> >>> >>> *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net >>> <cisco-voip-boun...@puck.nether.net>] *On Behalf Of *abbas Wali >>> *Sent:* Tuesday, February 16, 2016 7:09 AM >>> *To:* James Buchanan <james.buchan...@gmail.com> >>> *Cc:* cisco-voip@puck.nether.net >>> *Subject:* Re: [cisco-voip] UCCX 9 prompts gibberish >>> >>> >>> >>> indeed that is the case. >>> >>> thanks alot. >>> >>> >>> >>> any free tool to record g729. Have tried Audacity but cant bring it as >>> low as 8kbps. >>> >>> >>> >>> On 16 February 2016 at 14:29, James Buchanan <james.buchan...@gmail.com> >>> wrote: >>> >>> Hello, >>> >>> In the System settings, are you set to use G729 for your prompts or >>> G711? UCCX will not play one or the other. If you record G711 and upload to >>> a system set to play G729, that'll be the result. >>> >>> Thanks, >>> >>> James >>> >>> >>> >>> On Tue, Feb 16, 2016 at 9:26 AM, abbas wali <abba...@gmail.com> wrote: >>> >>> Hi guys, >>> >>> >>> >>> Just need a quick help here. >>> >>> >>> >>> Every prompt I record ( via UnityC or Audacity etc ) upload and can only >>> hear gibberish. >>> >>> >>> >>> But when I load an already saved file in G729 – it plays okay. >>> >>> >>> >>> I have checked an my regions for phone dpool and application trigger >>> are in the same region set to g711. >>> >>> >>> >>> The only other thing is that I am calling from a softphone vpn’ed. ( but >>> that shouldn’t make any difference ) >>> >>> >>> >>> Please help. >>> >>> Thanks in advance. >>> >>> >>> >>> ___ >>> cisco-voip mailing list >>> cisco-voip@puck.nether.net >>> https://puck.nether.net/mailman/listinfo/cisco-voip >>> >>> >>> >>> >>> >>> >>> >>> -- >>> >>> *Abbas Wali* >>> >>> >> >> >> -- >> *Abbas Wali* >> > > -- *Abbas Wali* ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] UCCX 9 prompts gibberish
yes seen them but again they save it in g711's. also in the new audacity there is no CCIT U-Law !! On 16 February 2016 at 15:12, James Buchanan <james.buchan...@gmail.com> wrote: > Even to g729? > > On 16 Feb 2016, at 10:11 AM, Haas, Neal <nh...@co.fresno.ca.us> wrote: > > I use audacity all of the time to convert to a “Cisco” format.. here are > two links for you > > > http://www.netcraftsmen.com/uc-toolkit-using-audacity-to-create-and-edit-cisco-uccx-prompts/ > > > > > http://xyfon.com/tech-tips/saving-wav-files-for-cisco-unified-call-centre-express-prompts-uccx-using-audacity/ > > > > > > *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net > <cisco-voip-boun...@puck.nether.net>] *On Behalf Of *abbas Wali > *Sent:* Tuesday, February 16, 2016 7:09 AM > *To:* James Buchanan <james.buchan...@gmail.com> > *Cc:* cisco-voip@puck.nether.net > *Subject:* Re: [cisco-voip] UCCX 9 prompts gibberish > > > > indeed that is the case. > > thanks alot. > > > > any free tool to record g729. Have tried Audacity but cant bring it as low > as 8kbps. > > > > On 16 February 2016 at 14:29, James Buchanan <james.buchan...@gmail.com> > wrote: > > Hello, > > In the System settings, are you set to use G729 for your prompts or G711? > UCCX will not play one or the other. If you record G711 and upload to a > system set to play G729, that'll be the result. > > Thanks, > > James > > > > On Tue, Feb 16, 2016 at 9:26 AM, abbas wali <abba...@gmail.com> wrote: > > Hi guys, > > > > Just need a quick help here. > > > > Every prompt I record ( via UnityC or Audacity etc ) upload and can only > hear gibberish. > > > > But when I load an already saved file in G729 – it plays okay. > > > > I have checked an my regions for phone dpool and application trigger are > in the same region set to g711. > > > > The only other thing is that I am calling from a softphone vpn’ed. ( but > that shouldn’t make any difference ) > > > > Please help. > > Thanks in advance. > > > > ___ > 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] UCCX 9 prompts gibberish
Hi guys, Just need a quick help here. Every prompt I record ( via UnityC or Audacity etc ) upload and can only hear gibberish. But when I load an already saved file in G729 - it plays okay. I have checked an my regions for phone dpool and application trigger are in the same region set to g711. The only other thing is that I am calling from a softphone vpn'ed. ( but that shouldn't make any difference ) Please help. Thanks in advance. ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] uccx 9 simple queueing
Hi all, I am a newish bee in the UCCX world. currently need to write a script which will - ring agent phone - if not picked then put agent not ready mode. ( cant change the cluster wide as other need that ) - call queues and plays inqueue mesg - after certain time it goes to VM etc. any help will be appreciated thanks -- ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Jabber phone mode outbound calls issue
Hi Ryan, Thanks for the detailed response. Yes the issue is with Jabber clients and not the IP phones. The line itself which is shared with many devices, can make calls on any other device but fails when made from Jabber. I ran all the below Utils and all came out without any significant alarms The NTP, though is at stratum 4. But again that's for both the clusters and one of them can make calls with jabber. Have ran some traces as below These are multiple failed calls. Not sure why there are so many REFER messages !! Thank s From: Ryan Huff [mailto:ryanh...@outlook.com] Sent: 24 November 2015 15:32 To: abbas wali <abba...@gmail.com>; cisco-voip@puck.nether.net Subject: Re: [cisco-voip] Jabber phone mode outbound calls issue I assume this is only with the Jabber clients and not IP phones as well? The annunciation message you're getting from Call Manager is typically reserved for when the calling device does not have access to the called device (or pattern). If you're confident that you're CSS/Partitions are correct you may need to look at OS level items. I recently assisted someone who presented with similar symptoms; everything worked fine except Jabber client egress and the solution there was NTP (incorrect/unsupported NTP can cause very, very strange behavior in UCOS). I would give the cluster a quick health check (performed from the CLI of the publisher); * utils dbreplication runtimestate * Looking for everything to come back with a (2) Setup Completedi message in the Replication Setup column * utils diagnose module validate_network * Looking for it to come back with Passed (anything fails like reverse DNS ... etc and it will explain) * utils ntp status * Looking for it to show synchronized and a stratum 3 (or lower) * Windows servers (SNTP) are unsupported for NTP and may cause issues even if it shows synchronized * utils ntp server list * Looking for any ntp servers referenced by hostname/FQDN rather than IP address (you should reference ntp servers by IP address) If everything comes back healthy, I would setup a test call scenario and pull traces off of CCM and follow the call flow. If one of the health checks fail, I would resolve that and then you may have to schedule a cluster restart (if possible). = Ryan = Email: <mailto:ryanthomash...@outlook.com> ryanthomash...@outlook.com Spark: <mailto:ryanthomash...@outlook.com> ryanthomash...@outlook.com Twitter: <http://twitter.com/ryanthomashuff> @ryanthomashuff LinkedIn: <http://linkedin.com/in/ryanthomashuff> ryanthomashuff Web <http://ryanthomashuff.com> ryanthomashuff.com _ From: cisco-voip < <mailto:cisco-voip-boun...@puck.nether.net> cisco-voip-boun...@puck.nether.net> on behalf of abbas wali < <mailto:abba...@gmail.com> abba...@gmail.com> Sent: Tuesday, November 24, 2015 9:58 AM To: <mailto:cisco-voip@puck.nether.net> cisco-voip@puck.nether.net Subject: [cisco-voip] Jabber phone mode outbound calls issue Hi all, Jabber phone only mode (10.5.2) is unable to make any outbound calls including any internal calls even to reach the voicemail. Inbound calls are working. This is happening in CUCM 9.1 When dial anything , I get the "your call cannt be completed as dialled please consult." I have checked via the DNA and the line settings are okay and calls permitted. Hence the CSSs\DP are okay. Strangely, we have another cluster CM 9.1 with the same jabber version and setting and it has no issues making any calls. Any suggestions. Thanks ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] Jabber phone mode outbound calls issue
Its set to Standard Analysis. Something else I have noticed. There are application dial rules defined. On top (with top priority ) there was Default rule beginning with blank, 0 digit strip and append 8. Some of the traces I found all my dials were appended by 8. Okay I have now moved the default dial rule to the bottom and all the correct one are on top. Now I can dial internally across cluster which is good. But cant dial external If that's the case and have to define full dial plan in the app dial rule that will become quiet messy. From: Ryan Huff [mailto:ryanh...@outlook.com] Sent: 24 November 2015 18:30 To: abbas wali <abba...@gmail.com>; cisco-voip@puck.nether.net Subject: Re: [cisco-voip] Jabber phone mode outbound calls issue Also and if it isn't too late, before you pull the SDL trace on the test call, can you verify that the Digit Complexity Analysis is set to TranslationAndAlternatePatternAnalysis under Service Parameters->Cisco Call Manager? = Ryan = Email: ryanthomash...@outlook.com <mailto:ryanthomash...@outlook.com> Spark: ryanthomash...@outlook.com <mailto:ryanthomash...@outlook.com> Twitter: @ryanthomashuff <http://twitter.com/ryanthomashuff> LinkedIn: ryanthomashuff <http://linkedin.com/in/ryanthomashuff> Web ryanthomashuff.com <http://ryanthomashuff.com> _ From: cisco-voip <cisco-voip-boun...@puck.nether.net <mailto:cisco-voip-boun...@puck.nether.net> > on behalf of Ryan Huff <ryanh...@outlook.com <mailto:ryanh...@outlook.com> > Sent: Tuesday, November 24, 2015 1:00 PM To: abbas wali; cisco-voip@puck.nether.net <mailto:cisco-voip@puck.nether.net> Subject: Re: [cisco-voip] Jabber phone mode outbound calls issue I would deffinatley look into getting the clock sync to a strata 3 on the pub and then restart ntp services. Can you do a test call on one of the Jabber clients and pull of the SDL traces for the call? Sent from my T-Mobile 4G LTE Device Original message From: abbas wali Date:11/24/2015 12:13 PM (GMT-05:00) To: 'Ryan Huff' ,cisco-voip@puck.nether.net Subject: RE: [cisco-voip] Jabber phone mode outbound calls issue Hi Ryan, Thanks for the detailed response. Yes the issue is with Jabber clients and not the IP phones. The line itself which is shared with many devices, can make calls on any other device but fails when made from Jabber. I ran all the below Utils and all came out without any significant alarms The NTP, though is at stratum 4. But again that's for both the clusters and one of them can make calls with jabber. Have ran some traces as below These are multiple failed calls. Not sure why there are so many REFER messages !! Thank s From: Ryan Huff [mailto:ryanh...@outlook.com] Sent: 24 November 2015 15:32 To: abbas wali <abba...@gmail.com <mailto:abba...@gmail.com> >; cisco-voip@puck.nether.net <mailto:cisco-voip@puck.nether.net> Subject: Re: [cisco-voip] Jabber phone mode outbound calls issue I assume this is only with the Jabber clients and not IP phones as well? The annunciation message you're getting from Call Manager is typically reserved for when the calling device does not have access to the called device (or pattern). If you're confident that you're CSS/Partitions are correct you may need to look at OS level items. I recently assisted someone who presented with similar symptoms; everything worked fine except Jabber client egress and the solution there was NTP (incorrect/unsupported NTP can cause very, very strange behavior in UCOS). I would give the cluster a quick health check (performed from the CLI of the publisher); * utils dbreplication runtimestate * Looking for everything to come back with a (2) Setup Completedi message in the Replication Setup column * utils diagnose module validate_network * Looking for it to come back with Passed (anything fails like reverse DNS ... etc and it will explain) * utils ntp status * Looking for it to show synchronized and a stratum 3 (or lower) * Windows servers (SNTP) are unsupported for NTP and may cause issues even if it shows synchronized * utils ntp server list * Looking for any ntp servers referenced by hostname/FQDN rather than IP address (you should reference ntp servers by IP address) If everything comes back healthy, I would setup a test call scenario and pull traces off of CCM and follow the call flow. If one of the health checks fail, I would resolve that and then you may have to schedule a cluster restart (if possible). = Ryan = Email: ryanthomash...@outlook.com <mailto:ryanthomash...@outlook.com> Spark: ryanthomash...@outlook.com <mailto:ryanthomash...@outlook.com> Twitter: @ryanthomashuff <http://twitter.com/ryanthomashuff> LinkedIn: r
[cisco-voip] Jabber phone mode outbound calls issue
Hi all, Jabber phone only mode (10.5.2) is unable to make any outbound calls including any internal calls even to reach the voicemail. Inbound calls are working. This is happening in CUCM 9.1 When dial anything , I get the "your call cannt be completed as dialled please consult." I have checked via the DNA and the line settings are okay and calls permitted. Hence the CSSs\DP are okay. Strangely, we have another cluster CM 9.1 with the same jabber version and setting and it has no issues making any calls. Any suggestions. Thanks ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] UCCX 9 EM agents cant login
thanks Christine/Ryan, Cluster rebooted last time due to some db sync issues but its been fine since then. talking about 3/4 weeks back that was. could have restarted the RIS Data Collector. but at this time creating and recreating new profiles will do. thanks all. On 15 September 2015 at 23:17, Christine See-Evans < christine.see.ev...@chemeketa.edu> wrote: > Delete the EM Profile then rebuild the profile, do the associations in > both CUCM and UCCX, CTI re-start. > > Check the version for your agent/supervisor CAD desktop for your version > of CUCM/UCCX (if you have them). Uninstall, re-install, restart. > > That's my last ditch effort. > > > > > *Christine See-Evans*, BCS, MBA > *Network Analyst* > Chemeketa Community College > 4000 Lancaster Drive NE, > Salem, OR 97305 > christine.see.ev...@chemeketa.edu <cseee...@chemeketa.edu> > (503)589-7776 <cseee...@chemeketa.edu> > > > > > “Make space in your life for the things that matter, for family and > friends, love and generosity, fun and joy. Without this, you will burn out > in mid-career and wonder where your life went.” > > > ― Jonathan Sacks > > On Tue, Sep 15, 2015 at 11:23 AM, abbas wali <abba...@gmail.com> wrote: > >> Cti restarts and profile+ device re association been done already. No >> luck. >> >> >> >> *From:* Ryan Huff [mailto:ryanh...@outlook.com] >> *Sent:* 15 September 2015 18:46 >> *To:* ealeather...@gmail.com; abba...@gmail.com >> *Cc:* cisco-voip@puck.nether.net >> >> *Subject:* Re: [cisco-voip] UCCX 9 EM agents cant login >> >> >> >> If the cti service in ccm has lost the state of the device, reassociating >> the device as Ed suggests, or restarting the cti manager service in ccm is >> how I have resolved these types of issues before. >> >> Thanks, >> >> Ryan >> >> >> >> Original Message >> From: Ed Leatherman <ealeather...@gmail.com> >> Sent: Tuesday, September 15, 2015 01:36 PM >> To: abbas wali <abba...@gmail.com> >> Subject: Re: [cisco-voip] UCCX 9 EM agents cant login >> CC: Ryan Huff <ryanh...@outlook.com>,Cisco VOIP < >> cisco-voip@puck.nether.net> >> >> I've had some weird, rare occasions where i've had to disassociate the >> device or profile from the rmjtapi app user and re-associate them. I'd >> suggest you try that if you haven't already just as a quick thing to do, >> although that won't tell you a root cause. >> >> >> >> On Tue, Sep 15, 2015 at 10:21 AM, abbas wali <abba...@gmail.com> wrote: >> >> Sorry that will mean !! >> >> >> >> *From:* Ryan Huff [mailto:ryanh...@outlook.com] >> *Sent:* 15 September 2015 15:18 >> >> >> *To:* abbas wali <abba...@gmail.com>; cisco-voip@puck.nether.net >> *Subject:* RE: [cisco-voip] UCCX 9 EM agents cant login >> >> >> >> Have is the subsystem in partial service? >> -- >> >> From: abba...@gmail.com >> To: ryanh...@outlook.com; cisco-voip@puck.nether.net >> Subject: RE: [cisco-voip] UCCX 9 EM agents cant login >> Date: Tue, 15 Sep 2015 15:11:33 +0100 >> >> Thanks Ryan, >> >> >> >> We have a dozen of other users who can login to them phones without any >> issues. Even I can do it on my CIPC >> >> >> >> But these new 3 agents cant. >> >> >> >> *From:* Ryan Huff [mailto:ryanh...@outlook.com <ryanh...@outlook.com>] >> *Sent:* 15 September 2015 15:05 >> *To:* abbas wali <abba...@gmail.com>; cisco-voip@puck.nether.net >> *Subject:* RE: [cisco-voip] UCCX 9 EM agents cant login >> >> >> >> Assuming this isn't a new/upgrade UCCX and it isn't a JTAPI compatibility >> issue with CCM; assuming it was working and now it is not. >> >> I would check these items; >> >> Agent/phone checks; >> >>- Make sure the RmCM user/password hasn't changed from what UCCX has >>recorded. NO >>- Does the agent have the Standard CTI Role? Yes >>- Does the agent have IPCCX defined in their profile? Yes >>- Does the agent have CTI control of the phone? Yes >>- Does the agent have control of the EM. profile? Yes >>- Is the physical phone associated to the RMCM user? Yes ( currently >>to log them in I am using my soft phone which is associated with RMCM >> user ) >> >> >> >> Server checks; yes as many other agent can login and are talking calls. >> >>- DNS ..
Re: [cisco-voip] UCCX 9 EM agents cant login
This doesn’t make any sense .. Created a new account with the same parameters for that agent/user and it does login. That’s absurd From: abbas wali [mailto:abba...@gmail.com] Sent: 15 September 2015 19:23 To: 'Ryan Huff' <ryanh...@outlook.com>; ealeather...@gmail.com Cc: cisco-voip@puck.nether.net Subject: RE: [cisco-voip] UCCX 9 EM agents cant login Cti restarts and profile+ device re association been done already. No luck. From: Ryan Huff [mailto:ryanh...@outlook.com] Sent: 15 September 2015 18:46 To: ealeather...@gmail.com <mailto:ealeather...@gmail.com> ; abba...@gmail.com <mailto:abba...@gmail.com> Cc: cisco-voip@puck.nether.net <mailto:cisco-voip@puck.nether.net> Subject: Re: [cisco-voip] UCCX 9 EM agents cant login If the cti service in ccm has lost the state of the device, reassociating the device as Ed suggests, or restarting the cti manager service in ccm is how I have resolved these types of issues before. Thanks, Ryan Original Message From: Ed Leatherman <ealeather...@gmail.com <mailto:ealeather...@gmail.com> > Sent: Tuesday, September 15, 2015 01:36 PM To: abbas wali <abba...@gmail.com <mailto:abba...@gmail.com> > Subject: Re: [cisco-voip] UCCX 9 EM agents cant login CC: Ryan Huff <ryanh...@outlook.com <mailto:ryanh...@outlook.com> >,Cisco VOIP <cisco-voip@puck.nether.net <mailto:cisco-voip@puck.nether.net> > I've had some weird, rare occasions where i've had to disassociate the device or profile from the rmjtapi app user and re-associate them. I'd suggest you try that if you haven't already just as a quick thing to do, although that won't tell you a root cause. On Tue, Sep 15, 2015 at 10:21 AM, abbas wali <abba...@gmail.com <mailto:abba...@gmail.com> > wrote: Sorry that will mean !! From: Ryan Huff [mailto:ryanh...@outlook.com <mailto:ryanh...@outlook.com> ] Sent: 15 September 2015 15:18 To: abbas wali <abba...@gmail.com <mailto:abba...@gmail.com> >; cisco-voip@puck.nether.net <mailto:cisco-voip@puck.nether.net> Subject: RE: [cisco-voip] UCCX 9 EM agents cant login Have is the subsystem in partial service? _ From: abba...@gmail.com <mailto:abba...@gmail.com> To: ryanh...@outlook.com <mailto:ryanh...@outlook.com> ; cisco-voip@puck.nether.net <mailto:cisco-voip@puck.nether.net> Subject: RE: [cisco-voip] UCCX 9 EM agents cant login Date: Tue, 15 Sep 2015 15:11:33 +0100 Thanks Ryan, We have a dozen of other users who can login to them phones without any issues. Even I can do it on my CIPC But these new 3 agents cant. From: Ryan Huff [mailto:ryanh...@outlook.com] Sent: 15 September 2015 15:05 To: abbas wali <abba...@gmail.com <mailto:abba...@gmail.com> >; cisco-voip@puck.nether.net <mailto:cisco-voip@puck.nether.net> Subject: RE: [cisco-voip] UCCX 9 EM agents cant login Assuming this isn't a new/upgrade UCCX and it isn't a JTAPI compatibility issue with CCM; assuming it was working and now it is not. I would check these items; Agent/phone checks; * Make sure the RmCM user/password hasn't changed from what UCCX has recorded. NO * Does the agent have the Standard CTI Role? Yes * Does the agent have IPCCX defined in their profile? Yes * Does the agent have CTI control of the phone? Yes * Does the agent have control of the EM. profile? Yes * Is the physical phone associated to the RMCM user? Yes ( currently to log them in I am using my soft phone which is associated with RMCM user ) Server checks; yes as many other agent can login and are talking calls. * DNS ... (are forward and reverse lookups working correctly) * NTP ... NTP still working? Thanks, Ryan _ From: abba...@gmail.com <mailto:abba...@gmail.com> To: ryanh...@outlook.com <mailto:ryanh...@outlook.com> ; cisco-voip@puck.nether.net <mailto:cisco-voip@puck.nether.net> Subject: RE: [cisco-voip] UCCX 9 EM agents cant login Date: Tue, 15 Sep 2015 14:53:54 +0100 That is the case, the DN is exclusive only to the profile – its not used on any phy. Phone. does anyone know, if want to get traces from RTMT which option should I use i.e. Cisco Call Manager will suffice ? thanks From: Ryan Huff [mailto:ryanh...@outlook.com] Sent: 15 September 2015 14:35 To: abbas wali <abba...@gmail.com <mailto:abba...@gmail.com> >; cisco-voip@puck.nether.net <mailto:cisco-voip@puck.nether.net> Subject: RE: [cisco-voip] UCCX 9 EM agents cant login Since you mention using extension mobility When the Agent logs in with their Ex. mobility profile, does the DN happen to be on another IP phone? The only way to "share" an ACD extension between multiple devices is to assign it to a Device Profile exclusively and then login using Extension Mobility to wha
Re: [cisco-voip] UCCX 9 EM agents cant login
Cti restarts and profile+ device re association been done already. No luck. From: Ryan Huff [mailto:ryanh...@outlook.com] Sent: 15 September 2015 18:46 To: ealeather...@gmail.com; abba...@gmail.com Cc: cisco-voip@puck.nether.net Subject: Re: [cisco-voip] UCCX 9 EM agents cant login If the cti service in ccm has lost the state of the device, reassociating the device as Ed suggests, or restarting the cti manager service in ccm is how I have resolved these types of issues before. Thanks, Ryan Original Message From: Ed Leatherman <ealeather...@gmail.com <mailto:ealeather...@gmail.com> > Sent: Tuesday, September 15, 2015 01:36 PM To: abbas wali <abba...@gmail.com <mailto:abba...@gmail.com> > Subject: Re: [cisco-voip] UCCX 9 EM agents cant login CC: Ryan Huff <ryanh...@outlook.com <mailto:ryanh...@outlook.com> >,Cisco VOIP <cisco-voip@puck.nether.net <mailto:cisco-voip@puck.nether.net> > I've had some weird, rare occasions where i've had to disassociate the device or profile from the rmjtapi app user and re-associate them. I'd suggest you try that if you haven't already just as a quick thing to do, although that won't tell you a root cause. On Tue, Sep 15, 2015 at 10:21 AM, abbas wali <abba...@gmail.com <mailto:abba...@gmail.com> > wrote: Sorry that will mean !! From: Ryan Huff [mailto:ryanh...@outlook.com <mailto:ryanh...@outlook.com> ] Sent: 15 September 2015 15:18 To: abbas wali <abba...@gmail.com <mailto:abba...@gmail.com> >; cisco-voip@puck.nether.net <mailto:cisco-voip@puck.nether.net> Subject: RE: [cisco-voip] UCCX 9 EM agents cant login Have is the subsystem in partial service? _ From: abba...@gmail.com <mailto:abba...@gmail.com> To: ryanh...@outlook.com <mailto:ryanh...@outlook.com> ; cisco-voip@puck.nether.net <mailto:cisco-voip@puck.nether.net> Subject: RE: [cisco-voip] UCCX 9 EM agents cant login Date: Tue, 15 Sep 2015 15:11:33 +0100 Thanks Ryan, We have a dozen of other users who can login to them phones without any issues. Even I can do it on my CIPC But these new 3 agents cant. From: Ryan Huff [mailto:ryanh...@outlook.com] Sent: 15 September 2015 15:05 To: abbas wali <abba...@gmail.com <mailto:abba...@gmail.com> >; cisco-voip@puck.nether.net <mailto:cisco-voip@puck.nether.net> Subject: RE: [cisco-voip] UCCX 9 EM agents cant login Assuming this isn't a new/upgrade UCCX and it isn't a JTAPI compatibility issue with CCM; assuming it was working and now it is not. I would check these items; Agent/phone checks; * Make sure the RmCM user/password hasn't changed from what UCCX has recorded. NO * Does the agent have the Standard CTI Role? Yes * Does the agent have IPCCX defined in their profile? Yes * Does the agent have CTI control of the phone? Yes * Does the agent have control of the EM. profile? Yes * Is the physical phone associated to the RMCM user? Yes ( currently to log them in I am using my soft phone which is associated with RMCM user ) Server checks; yes as many other agent can login and are talking calls. * DNS ... (are forward and reverse lookups working correctly) * NTP ... NTP still working? Thanks, Ryan _ From: abba...@gmail.com <mailto:abba...@gmail.com> To: ryanh...@outlook.com <mailto:ryanh...@outlook.com> ; cisco-voip@puck.nether.net <mailto:cisco-voip@puck.nether.net> Subject: RE: [cisco-voip] UCCX 9 EM agents cant login Date: Tue, 15 Sep 2015 14:53:54 +0100 That is the case, the DN is exclusive only to the profile – its not used on any phy. Phone. does anyone know, if want to get traces from RTMT which option should I use i.e. Cisco Call Manager will suffice ? thanks From: Ryan Huff [mailto:ryanh...@outlook.com] Sent: 15 September 2015 14:35 To: abbas wali <abba...@gmail.com <mailto:abba...@gmail.com> >; cisco-voip@puck.nether.net <mailto:cisco-voip@puck.nether.net> Subject: RE: [cisco-voip] UCCX 9 EM agents cant login Since you mention using extension mobility When the Agent logs in with their Ex. mobility profile, does the DN happen to be on another IP phone? The only way to "share" an ACD extension between multiple devices is to assign it to a Device Profile exclusively and then login using Extension Mobility to whatever device they wish to use. Thanks, Ryan _ From: abba...@gmail.com <mailto:abba...@gmail.com> To: cisco-voip@puck.nether.net <mailto:cisco-voip@puck.nether.net> Date: Tue, 15 Sep 2015 13:54:13 +0100 Subject: [cisco-voip] UCCX 9 EM agents cant login Hi all, Urgent issue here. Ext Mob. Enabled agents cant login. Getting “Login failed due to a configuration error with your phone and JTAPI or UCM. Contact your admin..” The users profile is in the contr
Re: [cisco-voip] UCCX 9 EM agents cant login
Sorry that will mean !! From: Ryan Huff [mailto:ryanh...@outlook.com] Sent: 15 September 2015 15:18 To: abbas wali <abba...@gmail.com>; cisco-voip@puck.nether.net Subject: RE: [cisco-voip] UCCX 9 EM agents cant login Have is the subsystem in partial service? _ From: abba...@gmail.com <mailto:abba...@gmail.com> To: ryanh...@outlook.com <mailto:ryanh...@outlook.com> ; cisco-voip@puck.nether.net <mailto:cisco-voip@puck.nether.net> Subject: RE: [cisco-voip] UCCX 9 EM agents cant login Date: Tue, 15 Sep 2015 15:11:33 +0100 Thanks Ryan, We have a dozen of other users who can login to them phones without any issues. Even I can do it on my CIPC But these new 3 agents cant. From: Ryan Huff [mailto:ryanh...@outlook.com] Sent: 15 September 2015 15:05 To: abbas wali <abba...@gmail.com <mailto:abba...@gmail.com> >; cisco-voip@puck.nether.net <mailto:cisco-voip@puck.nether.net> Subject: RE: [cisco-voip] UCCX 9 EM agents cant login Assuming this isn't a new/upgrade UCCX and it isn't a JTAPI compatibility issue with CCM; assuming it was working and now it is not. I would check these items; Agent/phone checks; * Make sure the RmCM user/password hasn't changed from what UCCX has recorded. NO * Does the agent have the Standard CTI Role? Yes * Does the agent have IPCCX defined in their profile? Yes * Does the agent have CTI control of the phone? Yes * Does the agent have control of the EM. profile? Yes * Is the physical phone associated to the RMCM user? Yes ( currently to log them in I am using my soft phone which is associated with RMCM user ) Server checks; yes as many other agent can login and are talking calls. * DNS ... (are forward and reverse lookups working correctly) * NTP ... NTP still working? Thanks, Ryan _ From: abba...@gmail.com <mailto:abba...@gmail.com> To: ryanh...@outlook.com <mailto:ryanh...@outlook.com> ; cisco-voip@puck.nether.net <mailto:cisco-voip@puck.nether.net> Subject: RE: [cisco-voip] UCCX 9 EM agents cant login Date: Tue, 15 Sep 2015 14:53:54 +0100 That is the case, the DN is exclusive only to the profile - its not used on any phy. Phone. does anyone know, if want to get traces from RTMT which option should I use i.e. Cisco Call Manager will suffice ? thanks From: Ryan Huff [mailto:ryanh...@outlook.com] Sent: 15 September 2015 14:35 To: abbas wali <abba...@gmail.com <mailto:abba...@gmail.com> >; cisco-voip@puck.nether.net <mailto:cisco-voip@puck.nether.net> Subject: RE: [cisco-voip] UCCX 9 EM agents cant login Since you mention using extension mobility When the Agent logs in with their Ex. mobility profile, does the DN happen to be on another IP phone? The only way to "share" an ACD extension between multiple devices is to assign it to a Device Profile exclusively and then login using Extension Mobility to whatever device they wish to use. Thanks, Ryan _ From: abba...@gmail.com <mailto:abba...@gmail.com> To: cisco-voip@puck.nether.net <mailto:cisco-voip@puck.nether.net> Date: Tue, 15 Sep 2015 13:54:13 +0100 Subject: [cisco-voip] UCCX 9 EM agents cant login Hi all, Urgent issue here. Ext Mob. Enabled agents cant login. Getting "Login failed due to a configuration error with your phone and JTAPI or UCM. Contact your admin.." The users profile is in the controlled list for RM application user. The phone they are loggin in - is used by other agents with their profiles and they can get through. Not sure what else I can check. Please help !! Thanks ___ cisco-voip mailing list cisco-voip@puck.nether.net <mailto: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] UCCX 9 EM agents cant login
Hi all, Urgent issue here. Ext Mob. Enabled agents cant login. Getting "Login failed due to a configuration error with your phone and JTAPI or UCM. Contact your admin.." The users profile is in the controlled list for RM application user. The phone they are loggin in - is used by other agents with their profiles and they can get through. Not sure what else I can check. Please help !! Thanks ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] UCx 9 - email notifications
thanks Dan, found another way do it via the CH and in the message settings, set the Message Recipient to either another user with Mailbox or a DL. thanks On 9 September 2015 at 17:44, Daniel Pagan <dpa...@fidelus.com> wrote: > Accomplish this with the Accept and Relay option for user message actions > for VM1 and specify VM2’s SMTP address. This should keep the message > available for VM1 while forwarding a copy to VM2. You’ll need to setup CUC > with a SMTP smart host in order to relay messages, and will likely need to > make changes to ensure SMTP connections are accepted from your Unity > Connection server but I can’t provide much assistance on Office 365. Keep > in mind this solution **won’t** provide any MWI or TUI feedback for VM1 > when VM2 reads/deletes the message or marks it unread. > > > > As for the DL question, two things I can think of… > > > > 1. Configure VM1 to Accept and Relay to a SMTP address that > resolves to a distribution list in Office 365 instead of going directly to > VM2. > > > > Or > > > > 2. Design the call flow so that voicemail calls to VM1 are **not** > routed to a VM1 user and greeting, but rather to a Call Handler with > message delivery to a Distribution List. This Call Handler’s greeting > sounds like VM1 (have the user record the greeting), but it sends all > messages to a DL where the VM1 user account is also a member. There’s a few > ways to set this up, and it’s far from perfect, but it’s one way to > accomplish this. > > > > Depending on the privacy requirements of VM1, you a 3rd option might be > to configure a 2nd extension on VM2’s phone in CUCM and then add this new > extension as an alternate extension to the CUC user account for VM1. This > allows VM2 to dial into CUC and access VM1’s mailbox. For MWI, just add a 2 > nd MWI extension for VM1 referencing the new line you added for VM2’s IP > phone. I know this doesn’t give a voicemail attachment to VM2 user, but I > figured the outcome this provides might meet the requirements of the user. > > > > Dan > > > > > > *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] *On Behalf > Of *abbas Wali > *Sent:* Wednesday, September 09, 2015 10:57 AM > *To:* cisco-voip@puck.nether.net > *Subject:* [cisco-voip] UCx 9 - email notifications > > > > Hi all, > > basic question > > how to send email notifications (O365 is setup already) to another > subscriber who already has got their own VM box. > > so e.g. > > VM1: 12345 > > VM2: 98765 > > both are setup with mailbox. VM1 when receives a VM, needs to send > notification via email to VM2 email box with the attachment. > > > > also, VM1 receives VM and can send it to a group of users's email > > I have tried to create a system Distribution list and add the memebers > into it but then how to link VM1 to that DL? > > thanks > > > -- > > > -- *Abbas Wali* ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] Cisco Unity 9
how to NOT play the Sorry "user" is not avalible, record your message at the tone... just want to play a personal recording OR a recorded name and take a message this is for a system call handler and have tried the untick [ Play the "Record Your Message at the Tone" Prompt] but it doesn't work. any ideas, thanks -- ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] UCx 9 - email notifications
Hi all, basic question how to send email notifications (O365 is setup already) to another subscriber who already has got their own VM box. so e.g. VM1: 12345 VM2: 98765 both are setup with mailbox. VM1 when receives a VM, needs to send notification via email to VM2 email box with the attachment. also, VM1 receives VM and can send it to a group of users's email I have tried to create a system Distribution list and add the memebers into it but then how to link VM1 to that DL? thanks -- ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] ILS multiple Hubs with SME
hi all, bit unclear about having a primary and secondary hubs for a single spoke in ILS. with us - there are ( or will be ) SMEs acting as Hubs for their region. have seen a leaf can point to redundant SME in case of a failure but if you run ILS on top of that - then similarly we can have a redundant HUB in the far side SME. thanks PS. when go to the configuration - can see step 1. configure hub step 2. configure additional hubs does that mean hubs from different cluster or same -- ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] isdn channels full message
Hi folks, CUCM 9 and H323 gateways with ISDN channels to PSTN. We want to redirect to VM when all the channels are full. Can CUCM do it or Dpeer? Thanks ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] 1/4 calls to DDI while rest to ext.
[image: Inline images 1] have done something like that for testing and fiddled around with the iMax integer i.e from 2 to 5 but the results are random and there is no controlled distribution. Scott, I am not sure about the circular HG as it distributes evenly. On 22 May 2015 at 17:33, Scott Voll svoll.v...@gmail.com wrote: I think you could also use a global variable and look up which call your on and route 1-3 to one and number 4 somewhere else. But my UCCx is a little rusty as my partner has managed UCCx since I came here. or a circular hunt group in CM might work to. Haven't' tried this. YMMV Scot On Fri, May 22, 2015 at 8:36 AM, Brian Meade bmead...@vt.edu wrote: You can definitely do this in UCCX. Use Java to create a random number 0 to 3 for each call. Random rand = new Random(); int value = rand.nextInt(4); Then create an if statement to match if the random value is 0-2 or 3 and use a redirect step to send the call to 2 different places based on which random number comes up. Brian On Fri, May 22, 2015 at 11:26 AM, abbas wali abba...@gmail.com wrote: Thanks Phil, Its just calls coming into a particular number – calling party doesn’t matter and no IVR for this and no TOD 75% to a diff number/ext/ddi or the rest 25% to another though out. We got UCCX, UCx, ARC and CM v 9. Can use any for it!! *From:* Walenta, Philip [mailto:philip.wale...@polycom.com] *Sent:* 22 May 2015 16:14 *To:* abbas wali; cisco-voip@puck.nether.net *Subject:* RE: [cisco-voip] 1/4 calls to DDI while rest to ext. It would help if we had a little more understanding of what you are trying to accomplish. Are you trying to distribute calls ¼ and ¾ over an hour, a day, a month? Will there be any other quantification on the call itself (calling number, called number, any IVR entry?)? *From:* cisco-voip [mailto:cisco-voip-boun...@puck.nether.net cisco-voip-boun...@puck.nether.net] *On Behalf Of *abbas wali *Sent:* Friday, May 22, 2015 10:01 AM *To:* cisco-voip@puck.nether.net *Subject:* [cisco-voip] 1/4 calls to DDI while rest to ext. Hi all, Is there a way to distribute ¼ of calls to one number/ddi and rest to a different set of numbers. CUCM 9 HP cant do that for me. Anything in UCCX 9!! TIA. ___ 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 -- *Abbas Wali* ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
[cisco-voip] 1/4 calls to DDI while rest to ext.
Hi all, Is there a way to distribute ¼ of calls to one number/ddi and rest to a different set of numbers. CUCM 9 HP cant do that for me. Anything in UCCX 9!! TIA. ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] 1/4 calls to DDI while rest to ext.
Thanks Phil, Its just calls coming into a particular number calling party doesnt matter and no IVR for this and no TOD 75% to a diff number/ext/ddi or the rest 25% to another though out. We got UCCX, UCx, ARC and CM v 9. Can use any for it!! From: Walenta, Philip [mailto:philip.wale...@polycom.com] Sent: 22 May 2015 16:14 To: abbas wali; cisco-voip@puck.nether.net Subject: RE: [cisco-voip] 1/4 calls to DDI while rest to ext. It would help if we had a little more understanding of what you are trying to accomplish. Are you trying to distribute calls ¼ and ¾ over an hour, a day, a month? Will there be any other quantification on the call itself (calling number, called number, any IVR entry?)? From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of abbas wali Sent: Friday, May 22, 2015 10:01 AM To: cisco-voip@puck.nether.net mailto:cisco-voip@puck.nether.net Subject: [cisco-voip] 1/4 calls to DDI while rest to ext. Hi all, Is there a way to distribute ¼ of calls to one number/ddi and rest to a different set of numbers. CUCM 9 HP cant do that for me. Anything in UCCX 9!! TIA. ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] 1/4 calls to DDI while rest to ext.
Its about a phased migration from one service desk to another. One main DDI will need to be migrated to a new SD. To start with 25% of calls to new and rest to the same SD. There are no IVRs or agents just plain call distribution. Thanks From: Scott Voll [mailto:svoll.v...@gmail.com] Sent: 22 May 2015 16:34 To: abbas wali Cc: Walenta, Philip; cisco-voip@puck.nether.net Subject: Re: [cisco-voip] 1/4 calls to DDI while rest to ext. use case might help us to understand... But things that come to might are using UCCx for sending calls where you want them to go. you could also look at using Hunt groups native to CM but knowing what your trying to accomplish will help us to give a better answer. YMMV Scott On Fri, May 22, 2015 at 8:26 AM, abbas wali abba...@gmail.com mailto:abba...@gmail.com wrote: Thanks Phil, Its just calls coming into a particular number – calling party doesn’t matter and no IVR for this and no TOD 75% to a diff number/ext/ddi or the rest 25% to another though out. We got UCCX, UCx, ARC and CM v 9. Can use any for it!! From: Walenta, Philip [mailto:philip.wale...@polycom.com mailto:philip.wale...@polycom.com ] Sent: 22 May 2015 16:14 To: abbas wali; cisco-voip@puck.nether.net mailto:cisco-voip@puck.nether.net Subject: RE: [cisco-voip] 1/4 calls to DDI while rest to ext. It would help if we had a little more understanding of what you are trying to accomplish. Are you trying to distribute calls ¼ and ¾ over an hour, a day, a month? Will there be any other quantification on the call itself (calling number, called number, any IVR entry?)? From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of abbas wali Sent: Friday, May 22, 2015 10:01 AM To: cisco-voip@puck.nether.net mailto:cisco-voip@puck.nether.net Subject: [cisco-voip] 1/4 calls to DDI while rest to ext. Hi all, Is there a way to distribute ¼ of calls to one number/ddi and rest to a different set of numbers. CUCM 9 HP cant do that for me. Anything in UCCX 9!! TIA. ___ cisco-voip mailing list cisco-voip@puck.nether.net mailto: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
Thanks mate. From: Anthony Holloway [mailto:avholloway+cisco-v...@gmail.com] Sent: 15 April 2015 21:46 To: abbas Wali Cc: cisco-voip@puck.nether.net Subject: 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 mailto: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 mailto: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 mailto: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 mailto: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 mailto: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 mailto: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
Re: [cisco-voip] PER CALL BANDWIDTH CONSUMPTION OVER ETHERNET+802.1Q
Thanks Dennis, that’s interesting figures. Surprised that Cisco in their SRNDs and even the end to end Qos book, have used a 128k everywhere, without any explanation that this depends on the codecs/no. of calls. From: Heim, Dennis [mailto:dennis.h...@wwt.com] Sent: 16 April 2015 12:42 To: Abbas Wali; 'Anthony Holloway' Cc: cisco-voip@puck.nether.net Subject: RE: [cisco-voip] PER CALL BANDWIDTH CONSUMPTION OVER ETHERNET+802.1Q With BiB it is 3x your codec. G.711 example: 1. 80k down far-end audio (remote party-current user) 2. 80k up current user audio (current user-remote party) 3. 80k up bib far-end audio (current user-recording server) 4. 80k up bib current user audio. (current user-recording server). On a G.711 call you would have need 80k down and 240k up. Dennis Heim | Emerging Technology Architect (Collaboration) World Wide Technology, Inc. | +1 314-212-1814 https://twitter.com/CollabSensei xmpp:dennis.h...@wwt.com tel:+13142121814 sip:dennis.h...@wwt.com Innovation happens on project squared -- http://www.projectsquared.com/ http://www.projectsquared.com https://wwt.webex.com/meet/dennis.heim Click here to join me in my Collaboration Meeting Room From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Abbas Wali Sent: Thursday, April 16, 2015 5:16 AM To: 'Anthony Holloway' Cc: cisco-voip@puck.nether.net mailto:cisco-voip@puck.nether.net Subject: Re: [cisco-voip] PER CALL BANDWIDTH CONSUMPTION OVER ETHERNET+802.1Q Thanks mate. From: Anthony Holloway [mailto:avholloway+cisco-v...@gmail.com] Sent: 15 April 2015 21:46 To: abbas Wali Cc: cisco-voip@puck.nether.net mailto:cisco-voip@puck.nether.net Subject: 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 mailto: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 mailto: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 mailto: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 mailto: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
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
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] 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
Re: [cisco-voip] 7800s ip phones 8.5
i think i have added the DP only to the pub as when i check in the os admin files it only show the new files for 7800 in the pub and none of the rest. On 7 October 2014 14:49, Ryan Ratliff (rratliff) rratl...@cisco.com wrote: The COP file is going to be phone firmware and only requires a TFTP restart as Bala stated. The devpack will add the new device to the database and does require a server reboot, but if you are seeing the phone in CCMAdmin to be added then you are past that step. You're going to have to look at ccm traces to confirm why the phone isn't registering. -Ryan On Oct 6, 2014, at 9:05 AM, Bala Singaram mmailb...@gmail.com wrote: Yes you are right. On Mon, Oct 6, 2014 at 3:57 AM, abbas Wali abba...@gmail.com wrote: thanks Bala, so you have to upload/install that to all the nodes/subs and no reboot required. I have also just seen a cop file for it. in which case, I can upload the cop file to all the nodes again and restart the TFTP services?? On 6 October 2014 11:49, Bala Singaram mmailb...@gmail.com wrote: Hi Abbas, Install the pack in PUB first then other nodes [ SUB ], no need to reboot the server, since the device pack will be active version itself. Regards, Bala On Mon, Oct 6, 2014 at 3:31 AM, abbas Wali abba...@gmail.com wrote: hi all, added new Device Pack for 7800 SIP phones on CUCM 8.5. can see the new phones appearing in the drop down list and in the device defaults. but still the phones are not registering. after the Dev Pack i restarted the TFTP services. do i need to reboot the whole cluster!! also I uploaded the Dev pack only on the TFTP node. does that needed to be on all the nodes (which i dont think so) thanks all. -- ___ 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 -- *Abbas Wali* ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] 7800s ip phones 8.5
thanks Bala, so you have to upload/install that to all the nodes/subs and no reboot required. I have also just seen a cop file for it. in which case, I can upload the cop file to all the nodes again and restart the TFTP services?? On 6 October 2014 11:49, Bala Singaram mmailb...@gmail.com wrote: Hi Abbas, Install the pack in PUB first then other nodes [ SUB ], no need to reboot the server, since the device pack will be active version itself. Regards, Bala On Mon, Oct 6, 2014 at 3:31 AM, abbas Wali abba...@gmail.com wrote: hi all, added new Device Pack for 7800 SIP phones on CUCM 8.5. can see the new phones appearing in the drop down list and in the device defaults. but still the phones are not registering. after the Dev Pack i restarted the TFTP services. do i need to reboot the whole cluster!! also I uploaded the Dev pack only on the TFTP node. does that needed to be on all the nodes (which i dont think so) thanks all. -- ___ 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] 7800s ip phones 8.5
Gary, and upload/install the DPack on all of them right. how about just going for the COP file option? On 6 October 2014 12:31, Gary Parker g.j.par...@lboro.ac.uk wrote: On 6 Oct 2014, at 11:57, abbas Wali abba...@gmail.com wrote: thanks Bala, so you have to upload/install that to all the nodes/subs and no reboot required. I have also just seen a cop file for it. in which case, I can upload the cop file to all the nodes again and restart the TFTP services?? I believe that if you’re installing a new device pack: - updates to existing devices do not require a reboot - adding *new* devices *does* require a reboot So, if the 7800s devices were not previously available, but you need to be able to register these devices, you will need to reboot the pub and subs. --- /-Gary Parker--f--\ | Unified Communications Service Manager | n Loughborough University IT Services | | Tel: +441509635635 Mob: +447989172258 o | http://delphium.lboro.ac.uk/pubkey.txt | \r--d-/ -- *Abbas Wali* ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] dial into ddi and get a secondary dial tone to call a new number
thanks all, I have implemented using MVA hairpinning. works well On 12 February 2014 04:38, Buchanan, James jbucha...@presidio.com wrote: I agree with Matthew--I had completely forgotten about MVA. It works reasonably well. James Buchanan | Sr. Network Engineer Presidio | www.presidio.com 12 Cadillac Drive Suite 130, Brentwood, TN 37027 D: 615.866.5729 | C: 931.797.2326 | F: 615.866.5781 | jbucha...@presidio.com PRESIDIO Practical thinking for a connected world. Follow Us: www.twitter.com/presidio From: abbas Wali [abba...@gmail.com] Sent: Tuesday, February 11, 2014 5:20 AM To: Matthew Collins Cc: Buchanan, James; cisco-voip@puck.nether.net Subject: Re: [cisco-voip] dial into ddi and get a secondary dial tone to call a new number nice one thanks Matt. is another solution be to use Unity 8 call handlers !! heard somewhere On 11 February 2014 11:06, Matthew Collins mcoll...@block.co.ukmailto: mcoll...@block.co.uk wrote: Hi Abbas, Bit long winded but yes it can be done, Create another loopback address on the gateway, Set up a h323 router in CUCM using the new loopback address and send that specific dial in number down to the H323 router. The TLC scrit can then pick up the call and route accordingly. You will also need to create a dial peer to send the public call back to the CUCM to then be routed out MGCP. Regards Matthew ǀ BLOCK ǀ Cisco Global EMEAR Partner of the Year 2013 From: abbas Wali [mailto:abba...@gmail.commailto:abba...@gmail.com] Sent: 11 February 2014 11:01 To: Matthew Collins Cc: Buchanan, James; cisco-voip@puck.nether.netmailto: cisco-voip@puck.nether.net Subject: Re: [cisco-voip] dial into ddi and get a secondary dial tone to call a new number TCL scripts on MGCP gateway ?? On 11 February 2014 10:40, Matthew Collins mcoll...@block.co.ukmailto: mcoll...@block.co.uk wrote: This can also be set up as part of Single number reach using Mobile Voice Access, You need to install a TCL script on the router. Users dial in, Prompted for extension number and pin then get secondary dial tone to dial out. Regards Matthew -Original Message- From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.netmailto: cisco-voip-boun...@puck.nether.net] On Behalf Of Buchanan, James Sent: 11 February 2014 10:21 To: abbas Wali; cisco-voip@puck.nether.netmailto: cisco-voip@puck.nether.net Subject: Re: [cisco-voip] dial into ddi and get a secondary dial tone to call a new number Not really. Some have done this using call handlers with pre-programmed speed dials. Some have also tried changing the restriction tables in Unity to allow for this, but that has a tone of security issues. UCCX would be your best place to do this, but you would want to make sure you secured it with a PIN. James Buchanan | Sr. Network Engineer Presidio | www.presidio.comhttp://www.presidio.com 12 Cadillac Drive Suite 130, Brentwood, TN 37027 D: 615.866.5729 | C: 931.797.2326 | F: 615.866.5781 | jbucha...@presidio.commailto:jbucha...@presidio.com PRESIDIO Practical thinking for a connected world. Follow Us: www.twitter.com/presidiohttp://www.twitter.com/presidio From: cisco-voip [cisco-voip-boun...@puck.nether.netmailto: cisco-voip-boun...@puck.nether.net] on behalf of abbas Wali [ abba...@gmail.commailto:abba...@gmail.com] Sent: Tuesday, February 11, 2014 4:11 AM To: cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net Subject: [cisco-voip] dial into ddi and get a secondary dial tone to call a new number hi all, is there a feature in CUCM 8.5 where you call into a tollfree number into cucm cluster and then from there you call out to different destinations i.e. getting a secondary dialtone after the first call connects Thanks -- This message w/attachments (message) is intended solely for the use of the intended recipient(s) and may contain information that is privileged, confidential or proprietary. If you are not an intended recipient, please notify the sender, and then please delete and destroy all copies and attachments. Please be advised that any review or dissemination of, or the taking of any action in reliance on, the information contained in or attached to this message is prohibited. ___ cisco-voip mailing list cisco-voip@puck.nether.netmailto:cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip -- Abbas Wali -- Abbas Wali -- *Abbas Wali* ___ cisco-voip mailing list cisco-voip@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip
Re: [cisco-voip] dial into ddi and get a secondary dial tone to call a new number
nice one thanks Matt. is another solution be to use Unity 8 call handlers !! heard somewhere On 11 February 2014 11:06, Matthew Collins mcoll...@block.co.uk wrote: Hi Abbas, Bit long winded but yes it can be done, Create another loopback address on the gateway, Set up a h323 router in CUCM using the new loopback address and send that specific dial in number down to the H323 router. The TLC scrit can then pick up the call and route accordingly. You will also need to create a dial peer to send the public call back to the CUCM to then be routed out MGCP. Regards Matthew ǀ BLOCK ǀ Cisco Global EMEAR Partner of the Year 2013 *From:* abbas Wali [mailto:abba...@gmail.com] *Sent:* 11 February 2014 11:01 *To:* Matthew Collins *Cc:* Buchanan, James; cisco-voip@puck.nether.net *Subject:* Re: [cisco-voip] dial into ddi and get a secondary dial tone to call a new number TCL scripts on MGCP gateway ?? On 11 February 2014 10:40, Matthew Collins mcoll...@block.co.uk wrote: This can also be set up as part of Single number reach using Mobile Voice Access, You need to install a TCL script on the router. Users dial in, Prompted for extension number and pin then get secondary dial tone to dial out. Regards Matthew -Original Message- From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of Buchanan, James Sent: 11 February 2014 10:21 To: abbas Wali; cisco-voip@puck.nether.net Subject: Re: [cisco-voip] dial into ddi and get a secondary dial tone to call a new number Not really. Some have done this using call handlers with pre-programmed speed dials. Some have also tried changing the restriction tables in Unity to allow for this, but that has a tone of security issues. UCCX would be your best place to do this, but you would want to make sure you secured it with a PIN. James Buchanan | Sr. Network Engineer Presidio | www.presidio.com 12 Cadillac Drive Suite 130, Brentwood, TN 37027 D: 615.866.5729 | C: 931.797.2326 | F: 615.866.5781 | jbucha...@presidio.com PRESIDIO Practical thinking for a connected world. Follow Us: www.twitter.com/presidio From: cisco-voip [cisco-voip-boun...@puck.nether.net] on behalf of abbas Wali [abba...@gmail.com] Sent: Tuesday, February 11, 2014 4:11 AM To: cisco-voip@puck.nether.net Subject: [cisco-voip] dial into ddi and get a secondary dial tone to call a new number hi all, is there a feature in CUCM 8.5 where you call into a tollfree number into cucm cluster and then from there you call out to different destinations i.e. getting a secondary dialtone after the first call connects Thanks -- This message w/attachments (message) is intended solely for the use of the intended recipient(s) and may contain information that is privileged, confidential or proprietary. If you are not an intended recipient, please notify the sender, and then please delete and destroy all copies and attachments. Please be advised that any review or dissemination of, or the taking of any action in reliance on, the information contained in or attached to this message is prohibited. ___ 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
