[Asterisk-Users] Problems with zaptel and TE210P

Alexander Lopez Alex.Lopez at OpSys.com
Mon May 1 12:17:51 MST 2006


His PRI span is showing down, If you forget to add the ${EXTEN} as you
said it would show as connecting and he _should_ get an intercept from
the telco.

 

 

________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Colin
Anderson
Sent: Monday, May 01, 2006 2:53 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Problems with zaptel and TE210P

 

Just a stab:

 

exten => _9NXXXXXX,1,Dial(ZAP/g1/${EXTEN})

 

Note uppercase ZAP and explicitly specifying the dialled number. 

 

hth

	-----Original Message-----
	From: Dan Brummer [mailto:dan.brummer at vegas.com]
	Sent: Monday, May 01, 2006 12:48 PM
	To: Asterisk Users Mailing List - Non-Commercial Discussion
	Subject: RE: [Asterisk-Users] Problems with zaptel and TE210P

	Some more info to my problem:

	 

	ipt-dev01*CLI> zap show status

	Description                              Alarms     IRQ
bpviol     CRC4

	T2XXP (PCI) Card 0 Span 1                OK         0          0
0

	 

	ipt-dev01*CLI> pri show span 1

	Primary D-channel: 24

	Status: Provisioned, Down, Active

	Switchtype: National ISDN

	Type: CPE

	Window Length: 0/7

	Sentrej: 0

	SolicitFbit: 0

	Retrans: 0

	Busy: 0

	Overlap Dial: 0

	T200 Timer: 1000

	T203 Timer: 10000

	T305 Timer: 30000

	T308 Timer: 4000

	T313 Timer: 4000

	N200 Counter: 3

	 

	Any ideas? ZTCFG looks good.

	 

	
________________________________


	From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dan
Brummer
	Sent: Monday, May 01, 2006 10:41 AM
	To: asterisk-users at lists.digium.com
	Subject: [Asterisk-Users] Problems with zaptel and TE210P

	Hello,

	I'm just starting out with asterisk and I'm playing around with
the system.  Currently I have a Digium TE210P connected to a PRI on the
Asterisk server.  I have a SIP soft phone on my laptop for testing that
is working fine.  When I try to place a call from my soft phone I get
this from Asterisk:

	 

	May  1 09:11:41 NOTICE[20098]: app_dial.c:1029 dial_exec_full:
Unable to create channel of type 'Zap' (cause 34 - Circuit/channel
congestion)

	  == Everyone is busy/congested at this time (1:0/1/0)

	  == Auto fallthrough, channel 'SIP/test-3a26' status is
'CONGESTION'

	 

	 

	#/etc/zaptel.conf:

	span=1,0,0,esf,b8zs

	bchan=1-23

	dchan=24

	 

	#/etc/asterisk/zapata.conf:

	[channels]

	switchtype=national

	context=default

	signalling=pri_cpe

	group=1

	channel => 1-23

	 

	#/etc/asterisk/extensions.conf:

	[general]

	static=yes

	writeprotect=no

	autofallthrough=yes

	 

	[default]

	exten => 123,1,Answer()

	exten => 123,2,Playback(hello-world)

	exten => 123,3,Hangup()

	 

	exten => _9NXXXXXX,1,Dial(Zap/g1)

	 

	 

	Any ideas?  Thank you in advance, your help is greatly
appreciated.

	 

	-Dan

	 

	 

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