Variable Caller ID 1

[troyhieb]

Hello,

We have a dental office that has an internal call center that received calls routed in from various clinic DIDs. We'd like to be able to replace the outgoing caller id with the local clinic's DID number for the call center users if they need to call a patient back. We're planning on using a combination of shortcodes to call various ARS tables that we're hoping to send the caller id out of when they call.

For example:
Shortcut:
*100
Dial
Line Group ID: 52: SUP-FR-SIP

ARS: 52: SUP-FR-SIP
N;
NSxxxxxxxxxx"@sip.flowroute.com"
Dial
18

xxxxxxxxxx I actually put the DID number I want to pass and Line Group 18 is the SIP trunk line. If we can get it to work to pass the caller id info we're going to create pairs of shortcode/ARS to have "site" codes to dial to pass the caller id of a specific site.

Our SIP URI is set to Use User Data and we're on Server Edition 9.1.1. I also see from watching active calls that through Monitor it actually send the IP of Flowroute and in the active calls it shows caller id as ending in @fl.gg. I've tried "@fl.gg" and "@xxx.xxx.xxx.xxx" where the x's are the IP for our Flowroute connection.

Any ideas to make this direction of idea to work or any other idea to be able to "on demand" change the outgoing caller id? So far no matter what it transmits the SIP user info in the SIP tab of the user. The following is from Flowroute's support page.

"We transmit Caller-ID based on the presence of one of the following header fields in order of preference: "P-Asserted-Identity", "Remote-Party-ID" or "From:". The caller-ID number you send out on outbound calls is pulled from one of these three fields within the actual SIP packets sent to our servers, how it gets into those fields is a different story."

It appears the user SIP tab is transmitting the FROM: field so if we can get ARS to replace it or one of the higher fields in the order of preference that would be great.

Thanks for any assistance in advance!

Troy

 

[intrigrant]

I am not sure about the URI, I always use "Use Internal Data", then the modding with ARS CLI works.
Then in ARS do not use "@whatever" as it is taken from the SIP trunk>ITSP Domain Name the call goes out on, that way you can use ARS fallback.
Finally in ARS the number after S must match the providers rules. Each provider does have its own rules, some want with country code some only want the national number. And not all providers allow to send anything else as a number from the DDI range belonging to the SIP trunk.
Basically it is trial and error technology and the provider often don't know either.

 

[troyhieb]

Sorry, "Use Internal Data" is what I have. It seems Flowroute takes anything put into the SIP tab of the user. For example if you delete out the full DID in the SIP tab it automatically replaces it with the User Name.

For example: User/Extension 3001
SIP Tab:
SIP Name: 3001
SIP Display Name(Alias): Extn3001
Contact 3001

With this in the user SIP tab and I make a call and the caller id shows as +3001. So it seems like Flowroute accepts anything sent. I just need to figure out how the IP Office can replace the SIP tab info from ARS.

 

[intrigrant]

That is a good start.
Take a monitor log with SIP enabled filter (not the HEX) and enable shortcodes in the Call filter, look where in the SIP INVITE the username is send and which shortcode is used to dial out.
Normally ARS should overwrite the users SIP value unless there are user shortcodes used which don't pass ARS

 

[troyhieb]

There are no user shortcodes in the system and other then some various other shortcodes created to route into auto-attendants it appears that it's routing the call correct. The *100 shortcode I have from the original post is *100 which points to the SUP-FR-SIP ARS which I have the a N; shortcode in that does the NS6516158229 shown below. But the NS in the ARS doesn't seem to be replacing the SIP tab info from the test user phone I'm working with. If I can figure out how to correctly get the IP Office to replace the SIP tab info from the User with information from the ARS table it appears Flowroute will accept what's received.

1837074308mS PRN: IPOKeepaliveTask::Main sending keepalives at 5000 ms
1837076580mS CMARS: Case 1 - dial_delay_time: 4000 has expired - Resolve what we might have - Called Number: 12182696157 Addr: dd93b408
1837076580mS CMARS: FORM: SUP-FR-SIP - Received Number: 12182696157
1837076580mS CMARS: FOUND A SHORT CODE - short_code: N; - Tel: NS+6513158229 - Called_Party: 12182696157 - Line Group Id: 18
1837076580mS CMARS: SetCurrentTarget: Short_Code: N; - Line_Group_ID: 18
1837076580mS CMARS: FindActiveARSByGroupID GroupID=18 - Not Found
1837076580mS Sip: SIP Line (10): License, Valid 1, Available 5, Consumed 2
1837076580mS Sip: SIP Line (10): sip_trunk_config_items 00020001, voip.flags 00000948
1837076580mS Sip: SIPDialog dd947c70 created, dialogs 3
1837076580mS CMARS: ModifyCMARSTarget: Short_Code: N; - Line_Group_ID: 18 set line status to CMARS_TRYING
1837076580mS CMARS: FOUND LINE - Line Id: 10 - using line group id: 18 (code N;, line 18) - Called Number: 12182696157 - Calling Number: +6513158229
1837076580mS CMARS: SEND Setup TO LINE
1837076580mS Sip: 0a89010a00004f54 10.20308.0 3851 SIPTrunk Endpoint(f4bedb98) received CMSetup
1837076581mS Sip: 0a89010a00004f54 10.20308.0 3851 SIPTrunk Endpoint(dd947c70) SetLocalRTPAddress to 216.251.161.65:41878
1837076581mS SIP Call Tx: 10
INVITE sip:12182696157@216.115.69.144 SIP/2.0
Via: SIP/2.0/UDP 216.251.161.65:5060;rport;branch=z9hG4bK9b36da0baa679dec4374eda1d868d3a7
From: "Extn3001" <sip:3001@216.115.69.144>;tag=486c6d4251f4cd10
To: <sip:12182696157@216.115.69.144>
Call-ID: 2f66848568287049de8e73295aac8072
CSeq: 115783600 INVITE
Contact: "Extn3001" <sip:3001@216.251.161.65:5060;transport=udp>
Authorization: Digest username="xxxxxx",realm="sip.flowroute.com",nonce="VdHk7lXR48K9ls3wjyNThpdX4RMKI34P",response="5eddc53ca618c081844d9b34c3d14b3e",uri="sip:12182696157@216.115.69.144",algorithm=MD5,qop=auth,nc=00000005,cnonce="cd
3ed4e3f88f04a33c72"
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
User-Agent: IP Office 9.1.1.0 build 10
Content-Type: application/sdp
Content-Length: 279

v=0
o=UserA 2078181536 2103081283 IN IP4 216.251.161.65
s=Session SDP
c=IN IP4 216.251.161.65
t=0 0
m=audio 41878 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 0-15

 

[Pepp77]

You cannot use S to send CLI for SIP.

You either have to create a dedicated URI with the number you want to display and point the shortcode to an ARS that uses that line group ID, or you need to amend the user SIP programming to the number you want to display and then point the ARS to the SIP URI set to internal data.

My SIP trunks always have 3 entries

URI 1 - *** - Used for incoming calls. I normally set this to 100 and 100
URI 2 - Use Internal Data - Used for outgoing calls using the users SIP information for CLI. This would be set to 100 and 101
URI 3 - Dedicated number - ie 02074014780. This would be set to 100 and 102.

An ARS pointing to 100 as its line group would display whatever the SIP provider wants to display. Pointed to 101 would display the CLI set against the user. Pointing the ARS to 102 would display 02074014780 as the outgoing CLI.

| ACSS SME |

 

[derfloh]

You can send a CLI with a *** URI and NSxxxx or NSixxxx short codes expending on the providers needs. But as always with SIP... It always depends on what the provider expects.

 

[Pepp77]

derfloh. Interesting - never had any success with that on any of the SIP providers I have used (Voiceflex, Gamma, Intelepeer and Colt).

| ACSS SME |

 

[intrigrant]

I see "1837076580mS CMARS: FOUND A SHORT CODE - short_code: N; - Tel: NS+6513158229 - Called_Party: 12182696157 - Line Group Id: 18"
I don't see the CLI number +6513158229 in the SIP request so there must be a setting in the SIP trunk wrong.
And yes, NS+6513158229 in ARS should work on SIP.

 

[troyhieb]

Our SIP trunks we always configure similar to how Pepp77 mentioned. Although looking at it and with this discussion we might be configuring out SIP trunks incorrectly.

URI 1 - Use Internal Data -- Incoming/Outgoing Group ID = 18
URI 2 - *** -- Incoming/Outgoing Group ID = 18

We have around 10 systems that we have SIP trunks. We've always configured the trunks to have the 2 URI and had to make sure the *** was the second but also used the same incoming/outgoing groups for both. If the *** was first then we had issues passing the User SIP tab info.

 

[intrigrant]

I always use 1 URI and still the ARS number sending works.
I do not have a exampla at hand, it is on my work laptop which I left in the office.
Right now it is extremely busy but if I can spare a moment tomorrow I will post the SIP line settings here.

 

[janni78]

If you have "Use Internal Data" on your URI it will use the User SIP tab for CLI.
If you put another outgoing ID on your second URI you should be able to use that ID to change the CLI with a shortcode.

"Trying is the first step to failure..." - Homer

 

[troyhieb]

Janni78,

That was my problem. We didn't realize that if we've set the default URI path to "Use Internal Data" that no matter what it'll use the User SIP tab info. I created a 3rd SIP URI using *** set to a new group ID and updated the ARS I was using to test with. It worked correctly now.

Also created a second shortcode/ARS pair to test passing two different Caller ID's. Worked perfectly.

So FYI, FlowRoute which is a national provider does seem very easy to pass what you want for Caller ID.

Thanks all for the help. This was huge for the client we're working with to be able to do on their system.

Thanks,
Troy

 

[Pepp77]

Just retested voiceflex using the Ns to send the CLI and it now works with them.

I like it.

| ACSS SME |