[cisco-voip] Securelogix-Cisco API

Does anyone know what kind of performance impact the API has on ASR or ISR 44xx routers running CUBE? I’ve used Securelogix with other SBC’s in large Enterprise and Call Center deployments. The SBCs use ENUM to query Securelogix. There is virtually no CPU impact on the SBC using ENUM even when

[cisco-voip] SIP INFO and Jabber

Is there a way to change the way Cisco Jabber clients (11.5) process SIP INFO? I see the client sending SIP INFO related to RFC 5168 (XML Schema) and would like to disable this. Thank you. ___ cisco-voip mailing list cisco-voip@puck.nether.net

Re: [cisco-voip] openldap for ucm/cuc

Yes. Using Oracle Internet Directory, which is based on OpenLDAP. > On Apr 18, 2017, at 1:28 PM, Tim Frazee wrote: > > anyone using openldap or apacheDS for directories instead of the de-facto > active directory? > ___ >

Re: [cisco-voip] Cisco CUCM SME - Call Forking

st to use a SBC or Cube , you will never get the same > level of control via CUCM > > -Ankur > > > On Tue, Dec 20, 2016 at 10:39 PM, Mark Holloway <m...@markholloway.com > <mailto:m...@markholloway.com>> wrote: > Hi all. Can SME fork SIP Invites to two di

[cisco-voip] Cisco CUCM SME - Call Forking

Hi all. Can SME fork SIP Invites to two different destinations? For example a call originates from the PSTN and SME forks the invite to CUCM and Skype. Couldn’t find anything in SRND specific to SME. Only CUCM (Mobility) and Spark. PSTN SIP TRUNK | | CUBE | | SME—— Skype | | CUCM

Re: [cisco-voip] Cisco and Lync/Skype with Media Bypass

deployment. > > >   <> > From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of > Norton, Mike > Sent: Wednesday, October 12, 2016 6:50 PM > To: Mark Holloway <m...@markholloway.com>; voip puck > <cisco-voip@puck.nether.net> >

[cisco-voip] Cisco and Lync/Skype with Media Bypass

This is more of a Microsoft question but I’ve searched everywhere and cannot find an answer. I’m currently working on a design to integrate CUCM and Skype for Biz. The Skype client will have media bypass enabled. All the Lync/Skype capacity calculators and TechNet articles talk about how many

Re: [cisco-voip] Jabber and P-RTP-Stat

; the sender and delete all copies. > > > -Original Message- > From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of > Mark Holloway > Sent: Thursday, March 03, 2016 4:12 PM > To: cisco-voip@puck.nether.net > Subject: [cisco-voip] Jabber an

[cisco-voip] Jabber and P-RTP-Stat

Is it true the latest Jabber client does not support P-RTP-Stat in a SIP BYE message? I searched (quickly, on my phone) but couldn't find the answer. Sent from my iPhone ___ cisco-voip mailing list cisco-voip@puck.nether.net

[cisco-voip] CUCM 11 and Fixed Audio Source

Does anyone know if CUCM 11 still requires a fixed audio source to be Multicast? The problem is when using an SBC other than CUBE for PSTN SIP Trunking there is no way to convert the stream to Unicast that I’m aware of. I’m open to suggestions. Thanks, Mark

[cisco-voip] CUCM Web Services API

Does anyone know if it’s possible to query CUCM over HTTP/HTTPS to see if a user is available to take a call? (ie. is the phone registered, is DND on or off, are all lines occupied and the user is busy) ___ cisco-voip mailing list

Re: [cisco-voip] CUCM Shared Line Appearance (SIP phones)

ct * from sysprocedures where procname = "sqlfunctions" > > Thanks, > Nick > > Disclaimer: IANADBA > > From: cisco-voip [mailto:cisco-voip-boun...@puck.nether.net] On Behalf Of > Daniel Pagan > Sent: Monday, September 14, 2015 10:07 AM > To: Daniel Pagan

[cisco-voip] CUCM Shared Line Appearance (SIP phones)

Hi all. When CUCM sends a SIP Invite to shared lines on SIP phones the user portion contains some sort of (what appears to b) a randomly calculated alpha-numeric string. Is there any information or documentation available on how Cisco actually calculates what this string will be? Here is an

Re: [cisco-voip] What is better than MediaSense for recording calls?

For call recording without analytics the Acme Packet/Oracle ISR (Interactive Session Recorder) is the way to go. It has been deployed in some of the largest CUCM/UCCE/CVP contact centers in the world. Currently it supports 24,000 concurrent calls in a single cluster in 8U of rack space. It

Re: [cisco-voip] Sip Trunk - CUCM and Third-party PBX

Typically you can disable SIP INVITE AUTHENTICATION on PBX’s. What kind of PBX is it? On May 14, 2015, at 1:44 AM, Tim Smith tim.sm...@enject.com.au wrote: Hi Claiton, I don’t think this has changed recently. You can’t do a SIP REGISTER from CUCM directly on a trunk. You need to

Re: [cisco-voip] UCCH-HCS 10.x : how different is it from ucce ?

Too bad, because Perimeta is terrible. HCS-CC requires an underlying infrastructure that is pretty extensive and requires A2Q. You have to have a Perimeta ___ cisco-voip mailing list cisco-voip@puck.nether.net

Re: [cisco-voip] CUCM 10.5 MoH

suppose there is any way to use a router to convert the stream to Unicast before it hits the SBC? On Apr 13, 2015, at 4:42 PM, Brian Meade bmead...@vt.edu wrote: Are you just talking about for fixed audio source? On Mon, Apr 13, 2015 at 4:07 PM, Mark Holloway m...@markholloway.com

[cisco-voip] CUCM 10.5 MoH

Are there any plans for Cisco to allow (bring back) Unicast MoH on 10.5? If trunking CUCM to a non-Cisco device there are cases where MoH breaks because not every endpoint will support Multicast. Thanks, Mark ___ cisco-voip mailing list

Re: [cisco-voip] Cisco Unified Border Element Session Licensing

I don’t think CUBE licensing is perpetual. You should call Cisco to get a definite answer but the folks I know had to rip and replace. On Apr 10, 2015, at 4:28 PM, Countryman, Edward edward.country...@presencehealth.org wrote: Does anyone know if I would be allowed to move SUBE session

Re: [cisco-voip] sRTP and RTP in SIP Invite

that perhaps there is a way this can be accommodated within CUCM. On May 30, 2014, at 12:20 PM, Brian Meade bmead...@vt.edu wrote: Can you send a CallManager SDI/SDL trace for one of these calls? On Fri, May 30, 2014 at 12:14 PM, Mark Holloway m...@markholloway.com wrote: Yep, it’s TLS

[cisco-voip] sRTP and RTP in SIP Invite

I’ve got a non-Cisco SIP device sending SIP Invites to CUCM (SIP Trunk). The SDP from my device includes RTP and sRTP in the SIP Invite. Reading Cisco docs it looks like the way Cisco expects sRTP to work is the SIP Invite should only include sRTP assuming if the call should be encrypted. If

Re: [cisco-voip] sRTP and RTP in SIP Invite

:41 AM, Mark Holloway m...@markholloway.com wrote: I’ve got a non-Cisco SIP device sending SIP Invites to CUCM (SIP Trunk). The SDP from my device includes RTP and sRTP in the SIP Invite. Reading Cisco docs it looks like the way Cisco expects sRTP to work is the SIP Invite should only include

Re: [cisco-voip] CUCM 7x with ACME SBC

The Acme Packet SBC has a very advanced SIP Header Manipulation regex based language. It also supports LUA. You can write “plugins” for the SBC using LUA to extend it’s functionality. Pretty cool for an edge appliance! On May 5, 2014, at 5:12 AM, Tim Smith tim.sm...@enject.com.au wrote: Hi

Re: [cisco-voip] One-way audio after BLIND transfer to SIP Trunk

I’ve seen instances where the SIP SDP contains either a=send or a=receive for Blind Transfer on CUCM and that causes one way audio. On one instance this was between CUCM and Lync. Fortunately the customer had an Acme Packet SBC and we used SIP Header Manipulation Rules to fix it. Can you get

Re: [cisco-voip] Office 365 with CUCM 9.1 Integration

This is pretty simple. Put an Acme Packet SBC on the edge of the customer premise to sit between CUCM and Lync SIP Trunks. It’s Lync certified and very robust. It can also double as an SBC for PSTN SIP Trunks. On Feb 3, 2014, at 11:19 AM, Michel L. M. B. Perez michelmbpe...@gmail.com wrote: