[asterisk-users] Cisco Gateway sending call to * without CID Name

[JR Richardson]

> You can not send caller ID name through the outgoing calls. You need
> to engage the telco and tell them to set up the number-to-name
> associations. Then you just send the right number and basically they
> set the caller ID name for you (yes I know technically the name isn't
> transmitted end-to-end as part of the call flow, your carrier sets it
> in their LIDB ... so on and so forth)
>
> For the inbound calls (from PSTN to your Cisco) have you made sure in
> your config you have:
>
> isdn supp-service name calling
>
> Ref: http://www.cisco.com/en/US/docs/ios/12_3/sip/configuration/guide/chapter9.html#wp1063971
>

Just so we are all on the same page, here is my setup:

legacyPBX> PRI <Cisco2600GW> SIP <Asterisk> SIP <PSTN

I was able to debug isdn on the GW, there is no CID name coming from
the PBX to the GW on the PRI.  I do see the CID Name from asterisk to
the GW to the PRI.

I tried the commands:
!
voice service voip
signaling forward unconditional
!

in conjunction with:
!
interface Serial1/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-ni
isdn protocol-emulate network
isdn incoming-voice voice
isdn supp-service name calling *************
isdn outgoing ie facility
no isdn outgoing display-ie
isdn outgoing ie connected-number
no cdp enable
!

But that broke outbound calls, I got this on the asterisk CLI:

WARNING[13430]: chan_sip.c:3517 process_sdp: Insufficient information
for SDP (m = '', c = '')

So I disabled 'signaling forward unconditional'

I read the Cisco reference and it looks like that what is needed to
fix this, but it doesn't.

I think I'll pick up testing again next week, do some more reading
cisco IOS over the weekend, like I have nothing better to do.

Also I will get with the PBX vendor and see if they can send me any name info.

Thanks all for your valuable input, I really appriciate the effort.

WIAT! HOLD THE PHONE!

Just got it working, while I was composing this, I was also in the
gateway poking around and found under the sip-ua where to set the
outbound CID name:

!
sip-ua
nat symmetric role active
nat symmetric check-media-src
calling-info pstn-to-sip from name set Name ************************
!

Now the CID Name is coming into asterisk in the first SIP invite.  I
realize this is only good in this setup because it's only 1 customer
so I can set the name statically and not be concerned about what, if
anything, they send to me on the PRI.  If this were a PSTN GW, I would
still be screwed I think.

I'm putting this one to bed.  Have a good weekend.

-- 
Thanks.
JR
---------------------
JR Richardson
Engineering for the Masses