asterisk users - Apr 2008 - Sending caller name out PRI?

I have a PRI connected to a traditional PBX using NI-2 and a typical
config (further below).  When I call from a SIP/IAX phone to an extension
on the PBX, only the number makes it through.  If I plug that same port on
the PBX to a carrier the PBX presents both name and number.

Hints or pokes to relevant chapters in documentation?  My config is
essentially like one found here:
  http://www.voip-info.org/files/nortel-asterisk-0.2.pdf
The author makes no reference to CNID, so I'm assuming that he wasn't
bothered by it not working.

Ideas?

The system is a trixbox 2.6.0.7 which includes zaptel-1.4.9.2-8.

...
    -- Executing [s at macro-outbound-callerid:22]
NoOp("SIP/3991-b7900488",
"CallerID set to "Peter" <NPANXX3991>") in ne
 Executed application: Noop
 Executed application: Macro
    -- Executing [s at macro-dialout-trunk:12]
AGI("SIP/3991-b7900488",
"fixlocalprefix") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
  == Begin MixMonitor Recording SIP/3991-b7900488
    -- AGI Script fixlocalprefix completed, returning 0
 Executed application: AGI
    -- Executing [s at macro-dialout-trunk:13]
Set("SIP/3991-b7900488",
"OUTNUM=4342") in new stack
 Executed application: Set
    -- Executing [s at macro-dialout-trunk:14]
Set("SIP/3991-b7900488",
"custom=ZAP/g14") in new stack
 Executed application: Set
    -- Executing [s at macro-dialout-trunk:15]
GotoIf("SIP/3991-b7900488",
"1?gocall") in new stack
    -- Goto (macro-dialout-trunk,s,17)
 Executed application: GotoIf
    -- Executing [s at macro-dialout-trunk:17]
Macro("SIP/3991-b7900488",
"dialout-trunk-predial-hook|") in new stack
   Context 'macro-dialout-trunk-predial-hook' for macro
'dialout-trunk-predial-hook' lacks 's' extension, priority 1
 Executed application: Macro
    -- Executing [s at macro-dialout-trunk:18]
GotoIf("SIP/3991-b7900488",
"0?bypass|1") in new stack
 Executed application: GotoIf
    -- Executing [s at macro-dialout-trunk:19]
GotoIf("SIP/3991-b7900488",
"0?customtrunk") in new stack
 Executed application: GotoIf
    -- Executing [s at macro-dialout-trunk:20]
Dial("SIP/3991-b7900488",
"ZAP/g14/4342|300|") in new stack
    -- Requested transfer capability: 0x00 - SPEECH
    -- Called g14/4342
Queuing frame from PRI_EVENT_PROCEEDING on channel 0/1 span 4
    -- Zap/73-1 is proceeding passing it to SIP/3991-b7900488
    -- Zap/73-1 is ringing
Echo cancellation already on
    -- Zap/73-1 answered SIP/3991-b7900488
...


zapata.conf (includes inline and comments purged)  Spans 3 and 4 connect
to the PBX.  (Yes the restating of the defaults are redundant, but I'm
willing to try any goofiness to make it work.)
-------
[trunkgroups]

[channels]

language=en
context=from-zaptel
signalling=fxs_ks
rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no
faxdetect=incoming

group=0,11
context=from-carrier-custom
callerid=asreceived
usecallerid=yes
hidecallerid=no
switchtype = national
signalling = pri_cpe
pridialplan=national
prilocaldialplan=national
channel => 1-23
groupcontext=default

; Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2"
group=0,12
context=from-carrier-custom
callerid=asreceived
usecallerid=yes
hidecallerid=noswitchtype = national
signalling = pri_cpe
pridialplan=national
prilocaldialplan=national
channel => 25-47
groupcontext=default

; Span 3: TE4/0/3 "T4XXP (PCI) Card 0 Span 3"
group=0,13
context=from-nortel-custom
callerid=asreceived
usecallerid=yes
hidecallerid=no
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
faxdetect=incoming
switchtype = national
signalling = pri_net
pridialplan=national
prilocaldialplan=national
channel => 49-71
groupcontext=default

; Span 4: TE4/0/4 "T4XXP (PCI) Card 0 Span 4"
group=0,14
context=from-nortel-custom
callerid=asreceived
usecallerid=yes
hidecallerid=no
switchtype = national
signalling = pri_net
pridialplan=national
prilocaldialplan=national
channel => 73-95
groupcontext=default

group=1

Hi all,

got some issues with meetme, i created meetme 8000, i have users 200 and 201
dial to 8000, meetme works fine.

but i need an outside user to join, so that user will dial-in via the sip trunk.
sip trunk has did e.g. 55554444, outside user dials-in to that DID, asterisk
rcvs it forward it to 8000.  meetme prompt will play asking for pin etc etc.
once connected to the meetme, after 1-2 secs the outside user will be
disconnected from the conference. but local extension 200 and 201 is still ok.

during disconnect i saw this on the logs,  Hungup
'Zap/pseudo-1147226235'

how can i fix these prob? TIA.


regards,
nhadie

       
---------------------------------
Be a better friend, newshound, and know-it-all with Yahoo! Mobile.  Try it now.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20080430/1c192067/attachment.htm

On Tue, 2008-04-29 at 23:49 -0500, Peter A Eisch wrote:
> I have a PRI connected to a traditional PBX using NI-2 and a typical
> config (further below).  When I call from a SIP/IAX phone to an extension
> on the PBX, only the number makes it through.  If I plug that same port on
> the PBX to a carrier the PBX presents both name and number.
> 
> Hints or pokes to relevant chapters in documentation?  My config is
> essentially like one found here:
>   http://www.voip-info.org/files/nortel-asterisk-0.2.pdf
> The author makes no reference to CNID, so I'm assuming that he
wasn't
> bothered by it not working.
You could set "pri debug span 1" in the Asterisk CLI (assuming that
this
is the first PRI span) and see if the name is actually being transmitted
to the PBX.

-- 
Jared Smith
Training Manager
Digium, Inc.