SIP Trunk IPO to Asterisk CallerID issue 2

[critchey]

I followed the FAQ on this site to get a SIP trunk connected between an IP 500 V2 and an Asterisk system. I was able to get the SIP trunk up and running and I was able to dial an extension on the Asterisk side. I was even able to transfer the call from the IPO to the Asterisk extension. The issue is when I do this the CallerID is not retained (it shows on the Asterisk side as blank). The customer is insisting that we get this fixed and the Asterisk side is insisting they have other IPO people doing the same thing and can send the correct CallerID. I asked if the Asterisk side could ask one customer how they got it to work but of course they refuse to do it.

IP 500 V2
9.1.9
SIP trunk license 5 channels
Only sending from IPO to Asterisk (never other way)
Shortcode 1000/dial 3k1/1000/17 to get across SIP to Asterisk works fine even for transfer

I can not seem to wrap my brain around how to maintain the CallerID in this situation. Any and all help would be greatly appreciated.

The truth is just an excuse for lack of imagination.

 

[RodneyMcSnow]

What version of Firmware is running on the IP Office?

Do you have access to, can you reach across their network to another site OR can someone send you a copy of one of the other IP Office systems so you can review the SIP LINE?

In the IP Line TAB: What is selected for:
Send Caller ID? DEFAULT=NONE
Caller ID from From Header? DEFAULT=NOT SELECTED

In the SIP URI TAB: What is selected for:
Local URI? DEFAULT=USE INTERNAL DATA
Contact? DEFAULT=USE INTERNAL DATA
Display Name? DEFAULT=USE INTERNAL DATA
PAI? DEFAULT=USE INTERNAL DATA

 

[RodneyMcSnow]

By the way, in the USER, SIP TAB: Have you filled out each persons caller id info here?

 

[Westi]

SIP Advanced
Routing method: to Header should send the correct info that is selected in the URI

Joe W.

FHandw, ACSS (SME)


"This is the end of the world, make sure to buy your T-shirt before it is too late"
Original expression of my daughter

 

[Westi]

sorry forget about it.
It is late, that setting is for incoming

Joe W.

FHandw, ACSS (SME)


"This is the end of the world, make sure to buy your T-shirt before it is too late"
Original expression of my daughter

 

[critchey]

@RodneyMcSnow
Already stated 9.1.9
Already stated that no I can not access other working configs
Send CallerID was set to none; I asked customer to change to Diversion header last night no response yet
Local URI, Contact, Display name are all set to use internal data

I also think that I forgot the ss in the shortcode so I also asked customer to add that.

I think using diversion header and ss in shortcode may resolve the issue I am still waiting to hear from my customer to find out.

Thanks for all the help.

The truth is just an excuse for lack of imagination.

 

[critchey]

Just as an update I got my test IPO connected to our Asterisk system and was able to get this figured out. I changed the Local URI, Contact, Display Name to * and it then passed along the CallerID even without the SS in the shortcode. I did also have the CallerID set to Diversion Header but it did not work till I changed the SIP URIs. Hopefully this helps someone down the road.

The truth is just an excuse for lack of imagination.

 

[Westi]

Nice one critchey

get some pink lovin' for thinking of others

Joe W.

FHandw, ACSS (SME)


"This is the end of the world, make sure to buy your T-shirt before it is too late"
Original expression of my daughter

 

[critchey]

Absolutely I get a lot of help around here I like to return the favor whenever possible.

The truth is just an excuse for lack of imagination.