Is it possible to set the outgoing caller ID for a Mobility conference call?

[ChuckHaynes]

Greetings,

When we use Mobility to transfer a call from a softphone (IP Communicator) to a cell phone, it works fine. However, if we have multiple people on a call (like a conference) and then try to use Mobility to transfer to a cell phone, the call is flagged by AT&T’s Call Protect application (on the cell phone). It looks like this is happening because the outbound caller ID from a Mobility conference call to a cell phone shows up as  +1 (000) 062-4220. It is being classified as Spam. I assume this is because of the (000) area code. I contacted AT&T and asked them about removing that number from the Spam list, but they said we would have to prove that we own the number (which we don’t). It seems to be some sort of Cisco internal CUCM generated thing. Do you know if it’s possible to change or set the outgoing caller ID for this number?

Thanks

[Roger Kallberg]

Not knowing how you connect with PSTN or what protocol you use for your voice gateway(s) its little bit of a guessing game.

If you have a SIP trunk connection to your PSTN you could modify this with a sip profile attached to the outbound dial peer. If you use ISDN and SIP or H.323 as the control protocol you should be able to use voice translation profile to modify the calling number. Please provide a little bit more details about your setup for us to better help you.

[ChuckHaynes]

CallManager - CUCM 10.5.2

Voice Gateway - Cisco 2921

Connection to PSTN - SIP (AT&T IP flex)

[Roger Kallberg]

Do a debug ccsip message and analyse what you send as calling number to your Telco for these calls. From that decide what is needed to change the sent calling number to fit your needs.

[ChuckHaynes]

Sorry for the delay. Can you provide more information regarding this? I did perform a debug ccsip and here is the output.

[Roger Kallberg]

Please provide information about the call, such as calling and called number and what is displayed on receiving end to be able to digest the provided debug.

[Roger Kallberg]

Hi Chuck,
I masked out any sensitive information in the reply. Your not sending any number information for the conference call to AT&T, that's why they block it as a spam call.

You would need to modify what is sent to the provider by using a SIP profile. See this superb document for details on how to do this. https://www.cisco.com/c/en/us/support/docs/voice/ip-telephony-voice-over-ip-voip/211306-In-Depth-Explanation-of-Cisco-IOS-and-IO.html

This is an example of a SIP profile that we use to clean off any name related information prior to sending it along to the Telco. Please note that this example will not fit you're need straight off, you'd need to modify it.

voice class sip-profiles 10
 request ANY sip-header From modify "From:(.*)(<sip:.*@.*>)" "From: \2"
 response ANY sip-header From modify "From:(.*)(<sip:.*@.*>)" "From: \2"
 request ANY sip-header Remote-Party-ID modify "Remote-Party-ID:(.*)(<sip:.*@.*>)" "Remote-Party-ID: \2"
 response ANY sip-header Remote-Party-ID modify "Remote-Party-ID:(.*)(<sip:.*@.*>)" "Remote-Party-ID: \2"
 request ANY sip-header P-Asserted-Identity modify "P-Asserted-Identity:(.*)(<sip:.*@.*>)" "P-Asserted-Identity: \2"
 response ANY sip-header P-Asserted-Identity modify "P-Asserted-Identity:(.*)(<sip:.*@.*>)" "P-Asserted-Identity: \2"

Once your happy with how your SIP profile looks attach it to your outbound dial peer with this command.

dial-peer voice 110 voip <-Change the dial peer number to fit your setup.
 voice-class sip profiles 10

[ChuckHaynes]

Please excuse my ignorance. I think I can hold my own in Switching, Routing, Security, and Wireless... but Voice is currently a weak area for me.

I looked at the link you provided, as well as some other stuff I found.

It's my understanding that the value in the first set of quotes will be replaced with the value in the second set of quotes. For example:

request ANY sip-header From modify "Conference" "Test"

From: "Conference" would be replaced with From: "Test". Is this correct?

[Roger Kallberg]

Your on the right track. Apart from changing the alphabetical part of the string sent you’d also should add a numeric value. Have a look at the SIP trace you captured and check how it’s formatted for other From or any other field that doesn’t contains the word Conference to get an idea on how it should look. It’s not the word Conference as such you need to change, it’s the lack of a numeric value that is your cause of the problem.

[ChuckHaynes]

Ok, I orignally thought I needed more, but then I found this post and it made me second guess myself.

https://bmcallister.com/tag/sip

voice class sip-profiles 100

  request ANY sip-header From modify "172.16.32.2" "198.51.100.2"

  request ANY sip-header Via modify "172.16.32.2" "198.51.100.2"

  request ANY sip-header Remote-Party-ID modify "172.16.32.2" "198.51.100.2"

  request ANY sip-header Contact modify "172.16.32.2" "198.51.100.2"

  response ANY sip-header Contact modify "172.16.32.2" "198.51.100.2"

  response ANY sip-header Remote-Party-ID modify "172.16.32.2" "198.51.100.2"

  request ANY sdp-header Audio-Connection-Info modify "172.16.32.2" "198.51.100.2"

  request ANY sdp-header Connection-Info modify "172.16.32.2" "198.51.100.2"

  request ANY sdp-header Session-Owner modify "172.16.32.2" "198.51.100.2"

  response ANY sdp-header Session-Owner modify "172.16.32.2" "198.51.100.2"

  response ANY sdp-header Connection-Info modify "172.16.32.2" "198.51.100.2"

  response ANY sdp-header Audio-Connection-Info modify "172.16.32.2" "198.51.100.2"

  request ANY sip-header Call-Info modify "172.16.32.2" "198.51.100.2"

  request ANY sip-header P-Asserted-Identity modify "172.16.32.2"

Is this any better?

request ANY sip-header From modify "From:(Conference)(<sip:xx.xx.37.53>)" "From:(Conference)(<sip:0005551234@10.10.150.6>)"

[Roger Kallberg]

Try with this.

voice class sip-profiles 10
request ANY sip-header From modify "From: ("Conference" <sip:)(.*)" "From: \11023456789@\2"
response ANY sip-header From modify "From: ("Conference" <sip:)(.*)" "From: \11023456789@\2"
request ANY sip-header P-Asserted-Identity modify "P-Asserted-Identity: ("Conference" <sip:)(.*)" "P-Asserted-Identity: \11023456789@\2"
response ANY sip-header P-Asserted-Identity modify "P-Asserted-Identity: ("Conference" <sip:)(.*)" "P-Asserted-Identity: \11023456789@\2"

Where 1023456789 is the number that you want to present to the caller for these conference calls. I checked your debugs and the fields where you sent the word "Conference" to ATT is From and P-Asserted-Identity, so these should be the once that you'd need to modify.

[ChuckHaynes]

Ok, thanks for clarifying.

What about Remote-Party-ID? You highlighted it above for including the word "Conference" as well.

[Roger Kallberg]

I verified that this does not pertain to this by looking through your debug once more. But if you need to modify it just add two rows to the SIP profile and change out the header type.

[ChuckHaynes]

% Invalid input detected at '^' marker.

I assume maybe I don't need the quotes in red.

request ANY sip-header From modify "From: ("Conference" <sip:)(.*)" "From: \11023456789@\2"

[Roger Kallberg · Accepted solution]

Okay then we need to escape the " as the regex doesn't like it when it's a part of the match string. Try with this.

request ANY sip-header From modify "From: (\"Conference\" <sip:)(.*)" "From: \11023456789@\2"

I actually didn’t test this or the prior post in a router, just wrote it out as is, so it could need to be adjusted for syntax errors.