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SIP Trunk OUtbound CLI presentation- ad hoc 1

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IanRichIcom

IanRichIcom

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Oct 9, 2012
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Morning all, I have a client who is based in the UK and has SIP trunks from Voiceflex. They also have a US SIP number that they wish to present when they Dial the US but when they dial the UK they present the Individual DDI. They also want to withhold the numbers on an ad hoc basis.

System is IP500V2 8.1 (73) With 1 SIP Trunk and 40 DDI. The SIP uri form is */*/* with Incoming Grp 2 and Outgoing Grp 0, as well as UseInternalData/UseInternalData/UseInternalData (for DDI)and line Grp 2 for both incoming and Outgoing. I have also added the US number as a seperate URI with the number in all fields and Incoming Grp 2 and Outgoing Grp 4. I have a short code which says 00N;/DIAL/N/LINE GRP 4 but when they call out the DDI for the users is sent to Line rather than the US Number. I have tried this both on the system shortcodes as well as user.

So i have a number of questions:
1) Can the IPO send a US Number that does exist on the SIP Trunk using shortcodes?
2) Can they dial a shortcode to withold (anonymous) the number being sent on an ad hoc basis- and what programming is required please?

Hope you clever people can assist

Regards Ian

Ian@Icom
 
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Have the same situation, customer has 12 sip trunks and I need to have at least two groups of phones/extension that when they dial out they need to present a specific telephone number for outbound calls. I have experimented with "si" and "S" via the ARS putting it front and back on our system but so far nothing has worked.

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Have the same situation, customer has 12 sip trunks and I need to have at least two groups of phones/extension that when they dial out they need to present a specific telephone number for outbound calls. I have experimented with "si" and "S" via the ARS putting it front and back on our system but so far nothing has worked.

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I have been doing some reading and I found the following piece of information in the knowledge-base "S - Calling Number; Place any following digits into the outgoing call's calling number field. Using S does not alter any allow or withhold CLI setting associated with the call, the short code characters A or W should be used respectively. Note that for SIP trunks, the SIP URI configuration options override this setting."

I just called Catalyst to confirm and it seems the SIP trunks do now work like the PRI, what I wast told is that I need to contact the carrier to make this happen, my problem is that the numbers I want to use cannot be ported to the SIP because they are part of a PRI/DID set.

If this is correct this kind of sucks and it will be very disappointing...!

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Well back to square one...just talked with the carrier and they say that it can be done with the IP Office...but they do not know how to do it...great..!!

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The following short code should work to set the CLID on a SIP trunk. Using the example in the original post:

Code: 00N;
Feature: Dial
Telephone Number: N"@sipdomain.com"Z"Name"S2135551234
Line Group ID: 0

In this example Line Group ID 0 = the URI form */*/* with Incoming Grp 2 and Outgoing Grp 0

Z - the SIP Display Name
S - URI user

Change sipdomain.com to the domain the provider expects to see in the TO header.

If the user dialed 0012125554321, the SIP URI's would look something like this:

From: "Name" <sip:2135551234@ITSPdomain.com>;tag=b3cc9ff2a75eb5cc
To: <sip:12125554321@sipdomain.com>
Contact: "Name" <sip:2135551234@<IP_Address>:5060;transport=udp>
 
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Redphone I setup the Z"Name"S5551234567 in ARS and it did not work...what did work for me was to setup the outbound CLI under SIP URI's PAI, I found out after a dozen reboots that what you put under SIP Line , Send Caller ID is irrelevant, as a matter of fact if you set it up as P asserted ID as it should be and if you add a second URI and a second PAI for second outboud CLI it doesn't allow you to do it, it kept giving me an error "Send Caller id must not be set to P Asserted ID, if PAI is not None".

What worked for me: I set Send Caller ID under SIP Line to none, added any number of SIP URIs to match your outbound CLI, making sure you use different outgoing Group IDs, I used the same Incoming group ID as the rest, created my different ARS entries with its corresponding shortecode to dial, and it worked and I am happy!!
The only strange thing is that once I added the second URI system status started to show 6 channels instead of 3 but I was able to make a call and get the a different outbound CLI without any problems, not sure how the extra 3 channels are going to reacts if the somebody tries to make a that 4th call out.
I did not experiment further but I believe that if you setup PAI to "Use Internal Data" you should be able to setup the outbound CLI under the "user's SIP tab" but I looked everywhere and I could not find that elusive SIP tab otherwise I would have tried it.
By the way. carrier said that charges may apply if the caller ID is not part of the DID

IanRichIcom hope this information helps...!!

Thanks,


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Surprise, surprise...Release 5 doesn't have all the goodies I was playing with on Release 8.1, they start on release 6. Redphone, I tried the same at the ARS at our customer's system and it is not working. I found the SIP tab under users...just had to go 3 releases down to find it.

Thanks,

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@Trinetintl- You are a genius- it works. Programmed my International Number under the URI,programmed the shortcode 00N;/DIAL/./line GRPID 4 (as set by the International URI) and it works. Thanks a bunch.

By the way ignore my post regarding the */*/* Uri adding the SIP Tab to the users- it is actually the UseInternalData/UseInternalData/UseInternaldata URI that addds this

Regards

Ian@Icom

Ian@Icom
 
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@Trinetintl- You are a genius- it works. Programmed my International Number under the URI,programmed the shortcode 00N;/DIAL/./line GRPID 4 (as set by the International URI) and it works. Thanks a bunch.

By the way ignore my post regarding the */*/* Uri adding the SIP Tab to the users- it is actually the UseInternalData/UseInternalData/UseInternaldata URI that addds this

Regards

Ian@Icom

Ian@Icom
 
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Using the all internal URI for your outgoing URI, just put the CLI number required into the users SIP Names field. I've done this with Voiceflex and Gamma in the UK. UK number for CLI need to leading zro striping in here. If you want to present numbers that are not on the SIP, both these providers will do this with CLI flexibility. I think with VF this is just a tick box on the VF portal. We don't use them anymore so can;t check.

Jamie Green

[bold]A[/bold]vaya [bold]R[/bold]egistered [bold]S[/bold]pecialist [bold]E[/bold]ngineer
 
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@jamie77, actually I tried that on the release 5 site I worked on last night and that did not work...after a ton of reboots and in one last act of desperation I added the outbound CLI number on all the fields; Local URI, Contact and Display Name and that did the trick.
I do not know why I was always under the impression that the SIP URI was critical to the trunk being up and register, but now I know it is totally independent and you can mess with it as much as you like without dropping the connection.

@IanRichIcom, thanks for the complement but it was just trial and error and a ton of reboots on both our lab system and the customers IPO.

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With a default SIP trunk ( version 7+ )set the outgoing SIP URI's with a * and in ARS use the sCLI parameter to send a specific number.
I do that all the time and it never fails.
 
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I ran into a little problem...customer finally got around forwarding one of the main numbers to the SIP trunk and he just called me to tell me that it sounded busy and SIP was reporting a 404 on the Invite. I went into the system and changed the Local URI from the CLI to the SIP trunk number or "Use Authentication Name" and no system seems to be accepting calls now but the outgoing CLI is not the correct one even though Contact and Display Name stayed the same.


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