cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
Announcements

379
Views
35
Helpful
15
Replies
Highlighted
Beginner

Need SIP trunk to pass Caller ID to fax server

Hi,

We have a SIP Trunk setup with a Route Pattern of 1070 to connect with our XMedius fax server.

The connection works but always sends 1070 as the DNIS/DID.  I can't figure out how to make it pass the originating caller DID to the server.  Anyone know how to make this happen?  Thanks!

1 ACCEPTED SOLUTION

Accepted Solutions
Highlighted

Re: Need SIP trunk to pass Caller ID to fax server

@LisaW_81625Looking at your document in more detail I see a few things that you should change. Yesterday I read it on my mobile.

  1. Remove the translation pattern you have for translating 8700 to 1070.
  2. As has already been noted by me and @sudaggar, you need to have either individual route patterns or summary route pattern(s) if you have a continuous range of numbers that are used for FoIP in XMedius. So create a route pattern for 8700 to send it to the RL for XMedius, or to the trunk directly if you're not using RL/RG. If you do you should reconfigure this to use RL and RG instead.
  3. Never have this set on route or translation patterns that you don't intend to play secondary dial tone.
    Snag_16570e2.png
    It will mess up your ability to get secondary dial tone for when you intend it.
  4. You should change the below setting on the SIP trunk to XMedius.
    Snag_15c01e7.png
    Screen shot from our setup for this. This will fix the problem you have will calling party being the last redirect number.
  5. Remove the called transformation you created with TAC and the use of it on the SIP trunk. Or at least the use of it on the SIP trunk, you can remove it and the partition and CSS later once you're all good with you setup.
    Snag_15dbb40.png
    The whole setup as suggested by TAC is redundant as all it does is to negate that you first do a transform of called number on the translation pattern from 8700 to 1070 and then back from 1070 to 8700 with the called party transformation pattern. It just a waste of anything to do that.
  6. Deselect PSTN Access on the SIP trunk.
    Snag_15ef18d.png
    Not related to your problem as such, but the SIP trunk to XMedius use isn't for PSTN so it's a bad habit to have it selected.
  7. Set urgent priority on the route pattern.
    Snag_16eb590.png
    Not related to your problem as such either, but will keep you from having to wait for inter-digit timeout.

With this I think you should be all good with what you ask for.

Please rate all useful posts

View solution in original post

15 REPLIES 15
Highlighted
Cisco Employee

Re: Need SIP trunk to pass Caller ID to fax server

There is an option under SIP trunk for outbound calls, "Calling Party Selection":

This can be configured as per the requirement as Originator.

 

Regards

Highlighted
Beginner

Re: Need SIP trunk to pass Caller ID to fax server

Thanks but that did not solve my problem. I have the SIP trunk "Calling Party Selection" set to Originator.
I have reset the SIP trunk and restarted the fax server prior to testing.
Any other ideas?

Thank you.


Highlighted
Cisco Employee

Re: Need SIP trunk to pass Caller ID to fax server

what happens when any internal phone (registered with CUCM) makes a call to 1070, is it still showing 1070.

Highlighted
VIP Mentor

Re: Need SIP trunk to pass Caller ID to fax server

The DNIS is correct. 1070 is the called number isn't it?

If you are saying the CLI/ANI is wrong, then that's a different issue. Are you not seeing the correct CLI? This is pretty standard and as long as the caller is sending the CLI you should see it on the fax server

Please rate all useful posts
Highlighted

Re: Need SIP trunk to pass Caller ID to fax server

Spot on @Ayodeji Okanlawon.
Didn't even pay attention that the OP wrote DNIS in the text, but Caller ID in the topic.

@LisaW_81625Would you mind to share some additional information on your setup and what you want to achieve for us to better be able to help you?

Please rate all useful posts
Highlighted
Beginner

Re: Need SIP trunk to pass Caller ID to fax server

Hello,
I currently have a TAC ticket open and hope to find resolution that way.
Thank you for responses.
Highlighted
VIP Mentor

Re: Need SIP trunk to pass Caller ID to fax server

I am sure we can help you here maybe much faster than TAC. All you need to do is describe your problem better and provide logs

You have not provided an accurate description of your problem and so we cant provide an accurate response to you

Please rate all useful posts
Highlighted
Beginner

Re: Need SIP trunk to pass Caller ID to fax server

Please see attached for details with screen shots.  Next post will have log file attached

Highlighted
Cisco Employee

Re: Need SIP trunk to pass Caller ID to fax server

There are 2 options for this:
1) create one route pattern for each directory number for Xmedius Fax server (As suggested by Roger).
2) create one generic route pattern 8xxx (if Xmedius fax server has 8XXX range of extension numbers configured) and remove called party transformation CSS from sip trunk and use called party transformation mask under route pattern configuration.
Highlighted
Beginner

Re: Need SIP trunk to pass Caller ID to fax server

Log file

Highlighted

Re: Need SIP trunk to pass Caller ID to fax server

You should create one route pattern per directory number you have for fax destination in XMedius. It won’t work to have one for all and translation patterns that translate to the route pattern number. We also use XMedius, if you’d want to discuss in a one on one call, please send me a PM and we can set it up for tomorrow.

Please rate all useful posts
Highlighted
Beginner

Re: Need SIP trunk to pass Caller ID to fax server

Thank you Roger! I will do some testing prior to sending a PM. Our team will be back in-house on Monday so we can plan a time that works with them as well.
I'm so relieved to hear you also use XMedius and are willing to help.

Thank you again!! Have a great weekend.
Highlighted

Re: Need SIP trunk to pass Caller ID to fax server

@LisaW_81625Looking at your document in more detail I see a few things that you should change. Yesterday I read it on my mobile.

  1. Remove the translation pattern you have for translating 8700 to 1070.
  2. As has already been noted by me and @sudaggar, you need to have either individual route patterns or summary route pattern(s) if you have a continuous range of numbers that are used for FoIP in XMedius. So create a route pattern for 8700 to send it to the RL for XMedius, or to the trunk directly if you're not using RL/RG. If you do you should reconfigure this to use RL and RG instead.
  3. Never have this set on route or translation patterns that you don't intend to play secondary dial tone.
    Snag_16570e2.png
    It will mess up your ability to get secondary dial tone for when you intend it.
  4. You should change the below setting on the SIP trunk to XMedius.
    Snag_15c01e7.png
    Screen shot from our setup for this. This will fix the problem you have will calling party being the last redirect number.
  5. Remove the called transformation you created with TAC and the use of it on the SIP trunk. Or at least the use of it on the SIP trunk, you can remove it and the partition and CSS later once you're all good with you setup.
    Snag_15dbb40.png
    The whole setup as suggested by TAC is redundant as all it does is to negate that you first do a transform of called number on the translation pattern from 8700 to 1070 and then back from 1070 to 8700 with the called party transformation pattern. It just a waste of anything to do that.
  6. Deselect PSTN Access on the SIP trunk.
    Snag_15ef18d.png
    Not related to your problem as such, but the SIP trunk to XMedius use isn't for PSTN so it's a bad habit to have it selected.
  7. Set urgent priority on the route pattern.
    Snag_16eb590.png
    Not related to your problem as such either, but will keep you from having to wait for inter-digit timeout.

With this I think you should be all good with what you ask for.

Please rate all useful posts

View solution in original post

Highlighted
Beginner

Re: Need SIP trunk to pass Caller ID to fax server

Thank you Roger! This really helped. Just what I needed to know.
I appreciate the time you took to help with this and the detail in your response.
Lisa