Re: [asterisk-users] QuadBRI and Zyxel Wifi phone stop working togetherafter 3 calls
Just tried it. When I run sip show channels it doesnt show any open channels. Thanks, Frederik On 14 Sep 2006, at 03:27, Bill Gibbs wrote: Make those calls then check the CLI sip show channels and see if the channels are stay up -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Frederik Fix Sent: Wednesday, September 13, 2006 8:35 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] QuadBRI and Zyxel Wifi phone stop working togetherafter 3 calls Hi, I have a strange problem that I have no idea how to debug: I have a Zyxel Prestige 2000W Wifi telephone that is connected to my Asterisk server which has a Junghanns.net QuadBRI card. I can make exactly 3 calls to the outside over the QuadBRI. Any calls after that fail with the log saying that all lines are busy. Turning the phone off and on solves the problem and I can make 3 calls again before it repeats. This problem does not occur when I make calls from my Cisco 7960G phones using SCCP or using eyebeam and SIP. Also making calls from the Zyxel through a cheap Cologne chipset ISDN card using zaphfc does not show this problem. I am using the following versions: Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1r Zyxel Prestige 2000W (version 1) Zyxel-Firmware: Wj.00.11 Any help is very much appreciated, Frederik ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] QuadBRI and Zyxel Wifi phone stop working together after 3 calls
Hi, I have a strange problem that I have no idea how to debug: I have a Zyxel Prestige 2000W Wifi telephone that is connected to my Asterisk server which has a Junghanns.net QuadBRI card. I can make exactly 3 calls to the outside over the QuadBRI. Any calls after that fail with the log saying that all lines are busy. Turning the phone off and on solves the problem and I can make 3 calls again before it repeats. This problem does not occur when I make calls from my Cisco 7960G phones using SCCP or using eyebeam and SIP. Also making calls from the Zyxel through a cheap Cologne chipset ISDN card using zaphfc does not show this problem. I am using the following versions: Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1r Zyxel Prestige 2000W (version 1) Zyxel-Firmware: Wj.00.11 Any help is very much appreciated, Frederik ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Receiving faxes and then sending them on
Hi,
I'm trying to setup a system where incoming faxes are received using
SpanDSP and then send on to another (remote) fax machine. The SpanDSP
part is working excellently, however I dont seem to be able to get
the forwarding part to work. Heres what I put into my extensions.conf:
exten = s,4,Answer()
exten = s,5,Set(FAXFILE=/tmp/fax-${UNIQUEID}.tif)
exten = s,6,Set([EMAIL PROTECTED])
exten = s,7,Set(EMAILADDR=${ARG1})
exten = s,8,rxfax(${FAXFILE}|debug)
exten = s,9,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR} $
{CALLERIDNUM})
exten = s,10,Dial(${ARG2})
exten = s,11,txfax(${FAXFILE}|caller)
exten = s,12,Hangup
Asterisk does start dialing at priority 10 however as soon as the
remote fax hangs up that call gets destroyed as well.
Is there anyway to do something like this?
Kind regards,
Frederik Fix
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[Asterisk-Users] Problem with integrating ISDN PBX using NT mode
Hi,
I'm just in the process of replacing a crappy Siemens PBX with a new and
shiny Asterisk system. To connect Legacy equipment I hooked up a small
ISDN PBX (DeTeWe OpenCom 36) to one port on a Junghanns.net quadBRI
card. That port is configured for NT Point to Multipoint
(Mehrgeraeteanschluss) mode. Now I can place calls from the ISDN PBX to
the other Asterisk extensions but the other way around does not work.
Whenever I call from the Asterisk server to one of the extensions
connected through the ISDN PBX that extension rings for a split second
and then the call is dropped. Here is what I get on the console:
-- Executing Macro(SCCP/13-002f, standard|Zap/g2/40) in new
stack
-- Executing Dial(SCCP/13-002f, Zap/g2/40|20) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g2/40
== Primary D-Channel on span 4 up for TEI 64
== Primary D-Channel on span 4 up for TEI 66
-- Zap/10-1 is proceeding passing it to SCCP/13-002f
-- Zap/10-1 is ringing
-- Channel 0/1, span 4 got hangup request
-- Hungup 'Zap/10-1'
== No one is available to answer at this time (1:0/0/0)
-- Executing Goto(SCCP/13-002f, s-NOANSWER|1) in new stack
-- Goto (macro-standard,s-NOANSWER,1)
-- Executing VoiceMail(SCCP/13-002f, u40) in new stack
-- Executing Goto(SCCP/13-002f, default|s|1) in new stack
-- Goto (default,s,1)
== Channel 'SCCP/13-002f' jumping out of macro 'standard'
== Primary D-Channel on span 4 down for TEI 65
== Primary D-Channel on span 4 down for TEI 64
== Primary D-Channel on span 4 down for TEI 66
I think I properly configured the ISDN PBX (theres not much to configure
there). Can someone help me here? What's causing the hangup request? How
could I find out?
Below is the relevant configuration.
Thanks in advance,
Frederik Fix
zapata.conf:
[channels]
switchtype = euroisdn
pridialplan = local
prilocaldialplan = local
nationalprefix = 0
internationalprefix = 00
;usecallingpres=yes
echocancel = yes
echocancelwhenbridged = yes
echotraining = 100
debug = 2
; Festnetzanschluss
signalling = bri_cpe
context=extern
group = 1
; S/T port 1
channel = 1-2
; S/T port 2
channel = 4-5
; S/T port 3
channel = 7-8
; Interner S0-Bus
signalling = bri_net_ptmp
context = intern-isdn
group = 2
; S/T port 4
channel = 10-11
extensions.conf:
[macro-standard]
exten = s,1,Dial(${ARG1},20)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(b${MACRO_EXTEN})
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${MACRO_EXTEN})
[nebenstellen-intern]
; Konferenzzimmer
exten = 13,1,Macro(standard,SCCP/13)
; Ingrid
exten = 17,1,Macro(standard,${INGRID})
exten = 57,1,Macro(standard_ohne_ab,Zap/g2/57)
; Gavigo
exten = 60,1,Macro(standard,SCCP/60)
; Woelm
exten = 66,1,Macro(standard,SCCP/66)
exten = 68,1,Macro(standard_ohne_ab,Zap/g2/68)
; van de Beeck
exten = 40,1,Macro(standard,Zap/g2/40)
exten = 44,1,Macro(standard_ohne_ab,Zap/g2/44)
; Rohan
exten = 50,1,Macro(standard,Zap/g2/50)
exten = 59,1,Macro(standard_ohne_ab,Zap/g2/59)
; Hinterhaus
exten = 58,1,Macro(standard,Zap/g2/58)
exten = 22,1,Macro(standard,Zap/g2/22)
; fuer Testzwecke
exten = 61,1,Macro(standard,SIP/eyebeamtest)
; virtuelle Nebenstellen
exten = 30,1,Macro(virtuell)
exten = 35,1,Macro(virtuell)
exten = 48,1,Macro(virtuell)
exten = 25,1,Dial(Zap/g2/58)
[intern-isdn]
exten = 25,1,Dial(SCCP/13)
exten = s,1,DISA(no-password|intern)
[dialout]
exten = _0.,1,Dial(Zap/g1/${EXTEN:1})
[intern]
include = dialout
include = nebenstellen-intern
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