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Multiple Outbound Caller ID on a SIP Trunk

Unanswered Question
Oct 31st, 2013
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Hello,


We've got CUCM 8.6.2 and have a SIP trunk from our provider going through a CUBE.  Today everything is working well.  Because of a few ordering/design issues, we're getting ready to port one of our remote site's numbers to our SIP Trunk and then route them over the WAN.  Inbound calls won't be an issue but we'd like all the remote sites outbound calls to show their name and caller ID information. 


Right now I have the name and caller ID specified on the trunk but I don't see anyway to have it change based on the origin of the call.  I've seen where you can go to the line level to define the caller ID but I rather do it somewhere in-between if it is possible.  I tried adding another trunk but it didn't like the same destination IP.   I don't have much expereince with SIP as this is our first trunk. 


Thoughts?


Thanks!

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Carlo Poggiarelli Thu, 10/31/2013 - 13:56
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Hi.
Do you mean caller id to send to provider or caller id of internal calls between sites?
Let me know.
regards

Carlo

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ecornwell Fri, 11/01/2013 - 06:18
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Hi Carlo,


It is caller id and name sent to the provider.    Internal calls won't change and work as expected.


Thanks!

Jaime Valencia Thu, 10/31/2013 - 17:15
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Calling name is defined at DN level, no way around that.

Calling number can be defined at DN, route list, route pattern or GW/trunk level.

Easiest way would be to just define the external phone number mask at each DN and check that option at the RP.


Otherwise you'd need to create separate RPs to modify calling info based on what each phone can reach via CSS/partitions.



HTH

java

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Carlo Poggiarelli Fri, 11/01/2013 - 06:33
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So... please share your cube config and calling info you want to send to the provider


Thanks

Regards

Carlo

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ecornwell Fri, 11/01/2013 - 06:50
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Just as an FYI, we're running IOS 15.1(2)T5.


What I'd like to see is something like this:


For all calls execpt extns 1100-1199 - Name: Company A, Number: 555-555-1234

For calls from extns 1100-1199 - Name: Company B, Number 111-111-1234


All the 1100-1199 extensions are in their own Device pool but we use local routing on the CUCM side.


I can't post the entire config but here are the revalant sections:


voice service voip

address-hiding

mode border-element

allow-connections h323 to h323

allow-connections sip to sip

no supplementary-service sip moved-temporarily

redirect ip2ip

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none

sip

  bind control source-interface Loopback10

  bind media source-interface Loopback10

  error-passthru

  asserted-id pai

  early-offer forced

  midcall-signaling passthru

  privacy-policy passthru

  g729 annexb-all

!

voice class codec 4

codec preference 1 g711ulaw

codec preference 2 g729r8 bytes 30

codec preference 3 g726r32

!

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g729r8 bytes 30

!

voice class sip-profiles 1

request INVITE sdp-header Audio-Attribute modify "a=ptime:20" "a=ptime:30"

response ANY sdp-header Audio-Attribute modify "a=ptime:20" "a=ptime:30"

request REINVITE sdp-header Attribute modify "a=T38FaxFillBitRemoval:0" ""

!

voice class sip-profiles 2

response ANY sip-header Allow-Header modify "UPDATE," ""

!



dial-peer voice 1999 voip

description Outgoing voice / fax to SIP Provider

destination-pattern .T

session protocol sipv2

session target ipv4: (Provider IP)

voice-class codec 1

voice-class sip asserted-id pai

voice-class sip privacy-policy passthru

voice-class sip early-offer forced

voice-class sip profiles 1

dtmf-relay rtp-nte

fax-relay sg3-to-g3

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none

no vad

!


dial-peer voice 732 voip

description incoming voice call from SIP

destination-pattern (Incoming Phone number pattern)

session protocol sipv2

session target ipv4: (CUCM-IP)

incoming called-number (Incoming Phone number pattern)

voice-class codec 1

voice-class sip asserted-id pai

voice-class sip privacy-policy passthru

dtmf-relay rtp-nte

fax rate 14400 bytes 48

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none

no vad


sip-ua

no remote-party-id

disable-early-media 180

retry invite 2

!

Carlo Poggiarelli Sun, 11/03/2013 - 13:58
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Hi.

Sorry for my late.


Now.. a solution could be:


- For calls from all extension except 1100-1199 create a partition OUT_COMP_A

- Create a CSS CSS_OUT_COMP_A

Create a route pattern 9.! and add a calling party transform mask 5555551234 and a prefix 1070

-Associate this CSS to all phones except 1100-1199

(If you are using local route group, the above steps are not necessary because you can add both calling party transform mask and prefix digit in the route list --->route group configuration page)


Repeat the same steps for phones from 1100 to 1199 changing calling party transform mask with 1111111234 and prefix digit into 1071


(we'll use prefix digit to differentiate outgoing dilapeer on cube config)



Now on cube config add 2 sip-profiles where we modify the calling name


voice translation-rule 10  (with this rule we remove the prefix added on route pattern on CUCM)

rule 1 /^1070/ //

rule 2 /^1071/ //


voice translation-profile strip-prefix

translate called 10


voice class sip-profile 100 (this will be for company A)

request INVITE sip-header Remote-Party-ID modify "\"(.*)\" " "\"Company A\" "

request INVITE sip-header From modify "\"(.*)\" " "\"Company A\" "


voice class sip-profile 101(this will be for company B)

request INVITE sip-header Remote-Party-ID modify "\"(.*)\" " "\"Company B\" "

request INVITE sip-header From modify "\"(.*)\" " "\"Company B\" "



Now create 2 dial peers matching outgoing prefix



dial-peer voice 1070 voip

translation-profile outgoing strip-prefix

session protocol sipv2

session target ipv4: (Provider IP)

voice-class codec 1

voice-class sip profiles 100

dtmf-relay rtp-nte

no vad


dial-peer voice 1071 voip

translation-profile outgoing strip-prefix

session protocol sipv2

session target ipv4: (Provider IP)

voice-class codec 1

voice-class sip profiles 101

dtmf-relay rtp-nte

no vad




HTH



Regards


Carlo















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ecornwell Mon, 11/04/2013 - 10:59
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Hi Carlo,


That was extremly helpful, thank you for the response!!!


I had one question, on the dial-peers, there's something missing to match properly isnt there?  We'd need something like:


dial-peer voice 1071 voip

translation-profile outgoing strip-prefix

session protocol sipv2

session target ipv4: (Provider IP)

incoming called-number 1071.T

voice-class codec 1

voice-class sip profiles 101

dtmf-relay rtp-nte

no vad


Is that correct?

Carlo Poggiarelli Mon, 11/04/2013 - 11:11
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Hi.
Sorry i missed that :)
Yes on DP 1070 configure destination-pattern 1070T and on DP 1071 configure destination-pattern 1071T

Sorry again :)

HTH

Regards

Carlo

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ecornwell Mon, 11/04/2013 - 11:13
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Great, thanks!  I've got a couple weeks before this needs to go into production so I'll let you know how it works out after that.


Your help is much appreciated!!!

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