[cisco-voip] Modify calling number in SIP invite on CUCM 11

Tue Jul 5 14:02:54 EDT 2016

Hi Daniel,

Please let me know if I can replace the SIPADDR in SIP INVITE with the
parameter "x-nearendaddr=1102" in the SIP INVITE.

I tried your script with the last suggestions but its manipulating the
External Phone Number Mask and its not giving me desired results.

*Thanks & Regards,*
*Deepak Maggo*

On Tue, Jul 5, 2016 at 12:47 PM, Daniel Pagan <dpagan at fidelus.com> wrote:

> That script will take care of the From: header for you. Just modify
> “Remote Party ID” to “From” and the “XXXX” to “110X”. You’ll need to modify
> the script if you have multiple ranges of course. Also note that this does
> not evaluate the calling number before applying the mask - unlike how a
> Calling Party Xform would work for outgoing calls on a trunk. This will
> simply mask the user portion of SIP URI the From header in all outgoing
> INVITE requests regardless of its current value. I would use this if you’re
> certain your From header needs to be modified, but a calling party
> transform on outgoing calls might also do the trick on the SIP trunk(I’m
> guessing since I haven’t looked at your scenario myself).
>
>
>
> Hope this helps.
>
>
>
> *From:* Deepak Maggo [mailto:dmaggo at ipcelerate.com]
> *Sent:* Tuesday, July 05, 2016 11:45 AM
> *To:* Daniel Pagan <dpagan at fidelus.com>
>
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] Modify calling number in SIP invite on CUCM 11
>
>
>
> Thanks Daniel for the reply.
>
>
>
> Let me give you brief what I am trying to achieve:
>
>
>
> I am trying to record the calls using BIB configuration on CUCM, when I
> dial the internal extensions I am getting the internal extension number as
> SIPADDR in SIP INVITE from phone. but when it makes any PSTN/outbound call,
> phone sends the "External Phone Number Mask" as SIPADDR in SIP INVITE from
> phone to the recorder. I want to display the External Phone Number Mask on
> PSTN/Outbound Calls on called numbers but on the BIB's SIP INVITE sent by
> Calling Phone to the recorder. Below is one example of SIP INIVITE which I
> am getting and what I want:
>
>
>
>
>
> *2265: Recording Profile Pattern*
>
> *888XXX4192: External Phone Number Mask on extension '1102'*
>
> *Extension # : 1102*
>
> *9173XXXXX456: Dialed PSTN Number*
>
>
>
> *Current SIP INIVITE to Recorder:*
>
>
>
> To: <sip:2265 at 10.31.X.XX>
>
> Via: SIP/2.0/UDP 10.33.X.XXX:5060;branch=z9hG4bK1f7a48bf6a
>
> CSeq: 101 INVITE
>
> Call-ID: f9461400-77b1d031-10-6e08210a at 10.33.8.110
>
> *From: "1102"
> <sip:888XXX4192 at 10.33.X.XXX;x-nearend;x-refci=32007259;x-nearendclusterid=StandAloneCluster;x-nearenddevice=SEP1CEXXXXXX4AC;x-nearendaddr=1102;x-farendrefci=32007260;x-farendclusterid=StandAloneCluster;x-farenddevice=10.31.8.1;x-farendaddr=9173XXXXX456*
> >;tag=1024~05da1bec-c875-450f-8d15-930ac584ef1e-32007264
>
> Content-Type: application/SDP
>
> Content-Length: 317
>
>
>
> v=0
>
> o=WIN-N309UNLOML3 1467716653090 1467716653090 IN IP4 10.31.9.66
>
> s=SDP
>
> c=IN IP4  10.31.9.66
>
> t=0 0
>
> a=direction:active
>
> m=audio 20488 RTP/AVP 0 8 18 101
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:18 G729/8000
>
> a=fmtp:18 annexb=no
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-11,16
>
> a=recvonly
>
>
>
>
>
>
>
> *Desired SIP INVITE to Recorder:*
>
>
>
> To: <sip:2265 at 10.31.X.XX>
>
> Via: SIP/2.0/UDP 10.33.X.XXX:5060;branch=z9hG4bK1f7a48bf6a
>
> CSeq: 101 INVITE
>
> Call-ID: f9461400-77b1d031-10-6e08210a at 10.33.8.110
>
> *From: "1102"
> <sip:1102 at 10.33.X.XXX;x-nearend;x-refci=32007259;x-nearendclusterid=StandAloneCluster;x-nearenddevice=SEP1CEXXXXXX4AC;x-nearendaddr=1102;x-farendrefci=32007260;x-farendclusterid=StandAloneCluster;x-farenddevice=10.31.8.1;x-farendaddr=9173XXXXX456*
> >;tag=1024~05da1bec-c875-450f-8d15-930ac584ef1e-32007264
>
> Content-Type: application/SDP
>
> Content-Length: 317
>
>
>
> v=0
>
> o=WIN-N309UNLOML3 1467716653090 1467716653090 IN IP4 10.31.9.66
>
> s=SDP
>
> c=IN IP4  10.31.9.66
>
> t=0 0
>
> a=direction:active
>
> m=audio 20488 RTP/AVP 0 8 18 101
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:18 G729/8000
>
> a=fmtp:18 annexb=no
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-11,16
>
> a=recvonly
>
>
> *Thanks & Regards,*
>
> *Deepak Maggo*
>
>
>
> On Tue, Jul 5, 2016 at 11:19 AM, Daniel Pagan <dpagan at fidelus.com> wrote:
>
> It depends on which headers this external platform is parsing for
> caller-ID. If it’s using Remote Party ID then this will help you:
>
>
>
> M = {}
>
> function M.outbound_INVITE(msg)
>
> msg:applyNumberMask("Remote-Party-ID", "XXXX")
>
> end
>
> return M
>
>
>
> I’ve tested this script before, but *cannot say it will fit your needs
> and resolve this issue*. What this will do is apply a four-digit mask to
> your external party number on the Remote-Party-ID header.
>
>
>
> You can replace this will “From” if needed.
>
>
>
> It actually parses the actual SIP URI, not the display field which, in
> your case, already contains a four digit extension.
>
>
>
> Applying this requires a reset of your SIP trunk - just something to keep
> in mind.
>
>
>
> - Dan
>
>
>
>
>
>
>
> *From:* cisco-voip [mailto:cisco-voip-bounces at puck.nether.net] *On Behalf
> Of *Deepak Maggo
> *Sent:* Friday, July 01, 2016 3:24 PM
> *To:* Brian Meade <bmeade90 at vt.edu>
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] Modify calling number in SIP invite on CUCM 11
>
>
>
> In earlier versions of CUCM we used to get extension number as SIP address
> for external calls, but after upgrading the CUCM to v11.0 we are getting
> external phone number mask.
>
> If anyone can help me with SIP normalisation script where I can replace
> the external phone number mask with extension number in SIP INVITE, which I
> can apply on SIP trunk to my call recorder coz I want external phone number
> mask to be displayed on PSTN calls.
>
> Appreciate all help in this.
>
> Thanks & Regards,
> Deepak Maggo
>
> On 1 Jul 2016 14:47, "Brian Meade" <bmeade90 at vt.edu> wrote:
>
> It's got the extension in there too in a few fields.  I think it does the
> Invite this way so the recorder can match up the call from the CTI messages.
>
>
>
> On Fri, Jul 1, 2016 at 2:30 PM, Deepak Maggo <dmaggo at ipcelerate.com>
> wrote:
>
> Here is the SIP Invite from extension number "1102" with External Phone
> Number Mask "8889184192", when I dial the internal extension I do receive
> the extension number in SIP invite:
>
>
>
> Jul 01 07:30:24.631 [INFO] JSIPListerner: processRequest() requestType
> (INVITE) = INVITE
>
> Jul 01 07:30:24.631 [INFO] JSIPListerner: processSIPInvite() callIDHeader
> = f89b3e00-77619055-42-6e08210a at 10.33.8.110
>
> Jul 01 07:30:24.631 [INFO] JSIPListerner: processSIPInvite() sipFromHeader
> = "1102" <sip:8889184192 at 10.33.8.110
> ;x-nearend;x-refci=29335531;x-nearendclusterid=StandAloneCluster;x-nearenddevice=SEP1CE85DC824AC;x-nearendaddr=1102;x-farendrefci=29335532;x-farendclusterid=StandAloneCluster;x-farenddevice=10.31.8.1;x-farendaddr=917325797456>
>
> Jul 01 07:30:24.631 [INFO] JSIPListerner: checkHalfCall() sipFromHeader =
> "1102" <sip:8889184192 at 10.33.8.110
> ;x-nearend;x-refci=29335531;x-nearendclusterid=StandAloneCluster;x-nearenddevice=SEP1CE85DC824AC;x-nearendaddr=1102;x-farendrefci=29335532;x-farendclusterid=StandAloneCluster;x-farenddevice=10.31.8.1;x-farendaddr=917325797456>
>
> Jul 01 07:30:24.631 [INFO] JSIPListerner: processSIPInvite() First
> sipFromHeader = "1102" <sip:8889184192 at 10.33.8.110
> ;x-nearend;x-refci=29335531;x-nearendclusterid=StandAloneCluster;x-nearenddevice=SEP1CE85DC824AC;x-nearendaddr=1102;x-farendrefci=29335532;x-farendclusterid=StandAloneCluster;x-farenddevice=10.31.8.1;x-farendaddr=917325797456>
>
>
>
>
> *Thanks & Regards,*
>
> *Deepak Maggo*
>
>
>
> On Fri, Jul 1, 2016 at 2:16 PM, Brian Meade <bmeade90 at vt.edu> wrote:
>
> What does the Invite actually look like?  I wouldn't expect this setup to
> show the external number mask to the recorder.
>
>
>
> On Fri, Jul 1, 2016 at 1:33 PM, Deepak Maggo <dmaggo at ipcelerate.com>
> wrote:
>
> No, I am not using any CSS Transformations.
>
> Thanks & Regards,
> Deepak Maggo
>
> On 1 Jul 2016 13:29, "Brian Meade" <bmeade90 at vt.edu> wrote:
>
> Do you have a calling party transform CSS on the SIP trunk?
>
>
>
> On Fri, Jul 1, 2016 at 11:09 AM, Deepak Maggo <dmaggo at ipcelerate.com>
> wrote:
>
> Hi Florian,
>
>
>
> I verified my trunk as well and it is also selected as "OnNet":
>
>
>
> [image: Inline image 1]
>
>
> *Thanks & Regards,*
>
> *Deepak Maggo*
>
>
>
> On Fri, Jul 1, 2016 at 10:56 AM, Deepak Maggo <dmaggo at ipcelerate.com>
> wrote:
>
> Florian,
>
>
>
> Thanks for the response. My route pattern is set to OnNet only:
>
>
>
> [image: Inline image 1]
>
>
> *Thanks & Regards,*
>
> *Deepak Maggo*
>
>
>
> On Fri, Jul 1, 2016 at 10:50 AM, Florian Kroessbacher <
> florian.kroessbacher at gmail.com> wrote:
>
> Maybe your trunk to the Recorder is set to Offnet, try to set it Onnet
>
>
>
>
> Am 1. Juli 2016, 15:47 +0200 schrieb Deepak Maggo <dmaggo at ipcelerate.com>:
>
> Hi,
>
>
>
> The problem I have is that we have active recording system which
> communicates with the CUCM cluster using a SIP trunk and uses
> Built-in-bridge configuration.
>
> The invites that CUCM sends via the SIP trunk show the "External Phone
> Number Mask" rather than the four digit extension when make PSTN calls
> which will not work according to  the recording system. We need the
> External Phone Number Mask on phones to display the correct Caller ID.
>
> Can anyone suggest a way to fix this?
>
> Ideally I need a way to modify the number in the SIP invite but I cannot
> find any example of how to do this.
>
> Any suggestions are welcome.
>
> Thanks
>
> *Thanks & Regards,*
>
> *Deepak Maggo*
>
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