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peter at boku.net
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 Posted: Tue Apr 29, 2008 11:49 pm Post subject: [asterisk-users] Sending caller name out PRI? |
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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
group=
context=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
group=
context=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
group=
context=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
group=
context=default
group=1 |
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jsmith at digium.com
Guest
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| Posted: Thu May 01, 2008 11:26 am Post subject: [asterisk-users] Sending caller name out PRI? |
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On Tue, 2008-04-29 at 23:49 -0500, Peter A Eisch wrote:
| Quote: | 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. |
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peter at boku.net
Guest
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| Posted: Thu May 01, 2008 4:17 pm Post subject: [asterisk-users] Sending caller name out PRI? |
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On Thu, 1 May 2008, Jared Smith wrote:
| Quote: | 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.
|
Thanks for your response.
Yes, with 'pri intense debug span 1' I do see the name in the setup
message. With it is the ANI as well as the DID in that specific.
Is there a way to delay (or resend) the name much like the carrier does?
This would then be closer to what the carrier does (as in how I need to
have a Wait(1) before using ${CALLERID(name)}). This assumes that it's a
timing issue I guess.
peter |
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peter at boku.net
Guest
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| Posted: Sat May 03, 2008 10:43 pm Post subject: [asterisk-users] Sending caller name out PRI? |
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On 5/1/08 4:17 PM, "Peter A Eisch" <peter at boku.net> wrote:
| Quote: | Is there a way to delay (or resend) the name much like the carrier does?
This would then be closer to what the carrier does (as in how I need to
have a Wait(1) before using ${CALLERID(name)}). This assumes that it's a
timing issue I guess.
|
[following up to myself]
I've debugged on the PRI from a telco carrier and they never send the name
message in the setup with the ANI and DID -- it always comes after.
Looking more at the message as zaptel writes it to the wire, if the logging
is right, the first datum in the setup message is the name. Is there a way
I can get the name to come after the DID and ANI even in the same message?
My current guess is that the PBX discards the name because there's no call
already existing.
Is there a syntax that I can use to copy the carrier behavior?
Thanks! |
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peter at boku.net
Guest
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| Posted: Tue May 06, 2008 12:04 am Post subject: [asterisk-users] Sending caller name out PRI? |
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On Thu, 1 May 2008, Peter A Eisch wrote:
| Quote: | | Quote: | 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.
|
...
Is there a way to delay (or resend) the name much like the carrier does?
This would then be closer to what the carrier does (as in how I need to
have a Wait(1) before using ${CALLERID(name)}). This assumes that it's a
timing issue I guess.
|
For fun, I switched the switchtype on both systems to DMS100 and the name
comes across fine then. I have no idea where to take this thread from
here, so let's call it resolved. <grin>
peter |
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