Try removing the g from the dial command:
exten = _X.,1,Dial(Zap/1/${EXTEN},60)
exten = _X.,2,Hangup ;
exten = _NXXX,1,Dial(Zap/1)
Simon Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rob Tarte
Sent: Wednesday, 2 February 2005 16:50
Try
busydetect=no
Simon Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Howard Lowndes
Sent: Monday, 31 January 2005 19:17
To: jurgen; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Zap channels in AU
Insert a Wait(2) before Answer
Simon Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Howard Lowndes
Sent: Friday, 28 January 2005 17:30
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Caller ID in AU
Is anyone in AU successfully
Cisco 837
Simon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike Dent
Sent: Friday, 3 December 2004 7:25
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Re: Asterisk crashes my router!?
Or can anybody suggest a
Kevin wrote:
There is a reboot script posted on the wiki to reboot Polycom
telephones. When I execute this script, I get the following messages.
I am concerned as this is causing issues with asterisk and the PRI.
Does anyone have any ideas why this would be happening?
asterisk
Has anyone written an equivalent script to remote reboot Cisco 79XX phones?
Simon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Baker
Sent: Monday, 29 November 2004 17:29
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Have you looked at FOP - available at www.asternic.org
It might be most or part of the way to what you want. You could work with
Nicolas on adding the features that you need and it doesn't have.
Just a thought.
Simon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
I thought I might try this, but smbclient returns an error - cannot resolve
host - a PING of the host is fine.
Any ideas?
Simon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Thomas Hutton
Sent: Thursday, 18 November 2004 15:24
To: AsteriskUserMaillist
I have been running Asterisk happily for many months and I was trying to
upgrade from CVS-HEAD-08/13/04-10
Simon Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of el Flynn
Sent: Monday, 11 October 2004 16:09
To: Asterisk Users Mailing List - Non
anyone have any ideas what is wrong?
Simon Brown
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,Hangup
exten = fax,2,Congestion
exten = fax,102,Congestion
exten = f,1,Hangup
exten = f,2,Congestion
exten = f,102,Congestion
Any ideas?
Simon Brown
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Moved it out the macro as you said and it now works.
Simon Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nathan Wharton
Sent: Thursday, 26 August 2004 12:07
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users
You need the following entries:
OS79XX.TXT
P003-07-1-00
SIPDefault.cnf
image_version: P0S3-07-1-00
Simon Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin Day
Sent: Saturday, 21 August 2004 6:01
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users
In my zapata.conf, I have
callerid=unknown
so if an incoming call doesn't set or suppresses it's callerid then my phone
will show unknown. I have found that if the callerid on the incoming call
is suppressed, then the call goes straight to Voicemail.
Has anyone seen this problem?
Simon Brown
Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Simon Brown
Sent: Friday, 20 August 2004 13:51
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] CallerId
In my zapata.conf, I have
callerid=unknown
so if an incoming call doesn't set or suppresses
Shaun,
Contact me off list - Simon.Brown at otterson.com.au
I might be able to help you.
Simon Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shaun Ewing
Sent: Friday, 20 August 2004 14:21
To: Asterisk Mailing List
Subject: [Asterisk-Users
Yes, they do perform better. Less echo, better busydetect.
Simon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christopher Lee
Sent: Friday, 20 August 2004 15:29
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Atick Certification on FXO Modules
In my .conf file I have context= below channel=. Also you have to stop and
restart asterisk to make the changes - a reload is not enough.
Simon Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronan
Sent: Thursday, 19 August 2004 14:10
I have tried with insecure=very in the [sipgate] section of my sip.conf.
This stops outgoing calls from working and still no incoming calls. Taking
it out reverts to outgoing working but incoming still reporting unable to
authenticate ...
I stopped and restarted asterisk after each change.
Simon
I am still having the problems. Putting insecure=very in the sip.conf does
not fix the problem - it actually stops the incoming call from being received
- it rings busy on the callers end.
Simon Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
,Goto(mainmenu,s,1)
exten = 4316568,4,Hangup
TIA
Simon Brown
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I've been using V7 for a couple of months now with no problems.
Simon Brown
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg Smith
Sent: Thursday, 12 August 2004 19:54
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Convert Cisco 7960
Can anyone shed some light on this ??? Or is this not the right sort of
question to ask?
Simon Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Simon Brown
Sent: Wednesday, 4 August 2004 11:44
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users
play a background message after
answering and before ringing the extensions (between steps 2 3). But I
cannot get it to work if the extensions are rung straight away.
Any help would be greatly appreciated.
Simon Brown
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Try www.sipserve.co.nz
Simon Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Jones
Sent: Thursday, 29 July 2004 4:44
To: Asterisk User (E-mail)
Subject: [Asterisk-Users] New Zealand DIDs
Does anyone know where i can get DIDs in New Zealand
I would be very interested in the backlight add-on.
Simon Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, 29 July 2004 7:12
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Cisco 7960 backlight and list etiquette
I have my set up answer an incoming call and then ring nominated extensions.
How can I have the person ringing in be able to key in a number and be taken
to a defined menu item?
I can achieve it if I play a background message, but not if I ring an
extension straight away.
Simon Brown
-zero on 'SIP/201-c410'
It tries to Goto (macro-stdexten,s,0)
Does anyone know why?
Simon Brown
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-zero on 'SIP/201-c410'
It tries to Goto (macro-stdexten,s,0)
Does anyone know why?
Simon Brown
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There is a very good and working WinCE IP phone available from SJPhone.
Simon Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins
Sent: Wednesday, 23 June 2004 18:03
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Future WinCE IP
Then shouldn't Asterisk be changed so it jumps to unavailable in the dial
plan? Surely this would be the correct way of working.
Simon Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Wednesday, 23 June 2004 0:47
To: [EMAIL
: [Asterisk-Users] Busy message
*I* think it should go to unavailable, but it has always gone to busy.
On Tue, 2004-06-22 at 16:34, Simon Brown wrote:
Then shouldn't Asterisk be changed so it jumps to unavailable in the
dial plan? Surely this would be the correct way of working.
Simon Brown
Logged in bugtracker as Bug #1893
Simon Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Wednesday, 23 June 2004 11:37
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Busy message
The issue has been suggested several times
When I dial a SIP phone which is specified in the sip.conf, but the phone is
not connected, Asterisk gives the message The user at Extension XXX is on
the phone
Shouldn't the message be the unavailable message?
Is there something wrong with my set up or is this a bug with Asterisk?
Simon
?
Simon Brown
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as
soon as a pattern is matched (or after a defined timeout). Have you also got
the SIPmacaddress.cnf file on your tftp server?
Simon Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Randy Bush
Sent: Friday, 18 June 2004 10:43
To: splatters
Subject
And didn't the original poster of this thread state rather forcefully that
this list is for * issues, not to be hijacked - which is exactly what is
happening based on comments/demands made by the original poster that were not
on the topic of *
Simon Brown
-Original Message-
From: [EMAIL
If you take * out of the equation and put the DECT phone and a normal handset
straight on to a PSTN line, you will probably and the same thing happens -
the DECT phone keeps ringing for 3-4 rings after you answer the call on the
normal handset.
Simon Brown
-Original Message-
From
Yes they do !!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Iain Stevenson
Sent: Friday, 11 June 2004 19:53
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] XML How To for Cisco 7960
--On Friday, June 11, 2004 10:46 am +0200 Stefan de Konink
.
Was this an intentional change?
Simon Brown
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I wish to have outgoing calls try to use a SIP/IAX provider and if this
fails, then fall back to PSTN and I am not sure how the dial plan should
look.
Can someone please post a sample of how it should look.
Thanks in advance,
Simon Brown
___
Asterisk
I have successfully used
SJPhone on my iPAQ 5450 with asterisk.
Simon
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ng kar
feiSent: Monday, 31 May 2004 18:50To:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Need guides
on setting up PDA on asterisk server
Can PDAs beused
You also need a SIPDefault.cnf
Simon
A prose style is a metaphysics.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of lists
Sent: Wednesday, 26 May 2004 9:26
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] 79XX converting
Humm that SCCP to start
Can I do this with * ???
S,1,answer call
S,2,play thanks for calling, we'll be with you soon
S,3,play music while caller waits and ring nominated extensions at same time
S,101,if not answered go to voicemail
I can't find a way to play music and ring extensions at the same time.
Any help would
When I start * I get 6 mpg123 processes start as well. Is this normal?
Often after a couple of days these mpg123 processes start to consume cpu and
I have to kill them off.
I do not have a sound card in the server and I have noload = chan_oss.so
Simon
I, too, am in Australia. I have used the X100P card and now have recently
swapped to use the TDM400P card with one FXS and one FXO. Others in
Australia are also using Asterisk with the Zaptel cards.
Regards,
Simon Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Yager
Sent: Monday, 24 May 2004 9:57
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] NetJet and RAS
Thanks! That's good to know. Please excuse my ignorance - if we have two
telstra ISDN2 lines, which card should I get?
Thanks,
Andrew
On 24/05/2004, at 9:53 AM, Simon Brown wrote:
I, too, am
Ever since I updated to CVS-head from 10 May, something weird has been
happening...
Every night at 1:10 AM (Eastern Australian time) my phone rings, there is no
callerid, and it results in a message in Voicemail which is just the
disconnect beeps (due to the inability of being able to detect
I have set up an extension so I can dial it and listen to my MusicOnHold from
any handset. This is what is in the extensions.conf:
exten = 997,1,MusicOnHold()
exten = 997,2,Hangup
After 180 seconds of playing, the call terminates. Why does this happen?
Simon
] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Simon Brown
Sent: Monday, May 17, 2004 4:38 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Music on hold
I have set up an extension so I can dial it and listen to my
MusicOnHold from any handset. This is what is in the extensions.conf
Try eBay - there are lots on there
Regards,
Simon Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Gardiner
Sent: Wednesday, 12 May 2004 22:05
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Where to get 48 volt Power Supplies for Cisco IP
Cisco 827/837 works fine for me .
Simon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean Magnuson
Sent: Thursday, 13 May 2004 14:48
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Sonicwall with Firmware 6.6.02 - SIP?
Sonicwall now has SIP
I just came across the same problem. I fixed it by changing from sipgate.net
to sipgate.de in the sip.conf file.
Simon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of nicolas
Sent: Tuesday, 11 May 2004 21:10
To: [EMAIL PROTECTED]
Subject:
.
Has anyone got any ideas?
TIA
Simon Brown
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Are you using the 7905 with a SIP image?
Simon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alric
Sent: Friday, 23 April 2004 4:37
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco phones
Nick Knight wrote:
I haven't used cisco phones
I am looking for a SIP/IAX termination provider in New Zealand. Does anyone
know of one?
TIA
Simon
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You could try
these:
voiptalk - www.voiptalk.org
sipgate - www.sipgate.de
Simon
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ron
McMillinSent: Thursday, 15 April 2004 15:29To:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Most
Reliable Proxy Server?
Hi all,
Do you know
# Proxy Registration (0-disable (default), 1-enable)
proxy_register: 1
Simon Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Roger
Sent: Wednesday, 7 April 2004 6:20
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] registration failure
I feel I'm
I put an externip=xxx.xxx.xxx.xxx in my sip.conf so I can register with FWD
from behind a NAT
With this entry my PSTN calls have a problem in that the other party cannot
hear me - I can hear them.
It does not matter whether I make the call or the other party does.
Any ideas ?
TIA
Simon
I have quite successfully set up the Services button to work on the 7940
running SIP.
I have a metric-imperial converter, a foreign exchange rate calculator, a
calendar etc available to users.
The XML is really fussy though.
Simon
-Original Message-
From: [EMAIL PROTECTED]
Here is a copy of one that works perfectly.
Simon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Gardiner
Sent: Friday, 2 April 2004 15:35
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] CISCO 7940 and directory/services problem
Thanks for
The password has to match what you have in sip.conf
Simon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Roger
Sent: Thursday, 1 April 2004 4:03
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] setting up 7940
Simon Brown wrote:
Make sure that you
Make sure that you have the following (or equivalent in the
SIPmac_addr.conf
# Line 1 appearance
line1_name: 202
# Line 1 Registration Authentication
line1_authname: 202
# Line 1 Registration Password
line1_password: 202
and this in the SIPDefault.conf
# Proxy Registration (0-disable (default),
I cannot successfully register with, or even make calls to, a SIP provider
(such as FWD) with my * server sitting behind a NAT. The firewall is a Cisco
827 router running 12.3 IOS.
Has anyone successfully got their server behind NAT to register or make a
call to a SIP provider?
TIA
Simon
When I compile screen on my * server I get the following errors.
Any pointers would be greatly appreciated.
[EMAIL PROTECTED] screen-3.9.15]# make
gcc -c -I. -I.-g -O2 screen.c
In file included from screen.h:45,
from screen.c:85:
term.h:40:1: warning: d_CUP redefined
When I try to compile the cdr_mysql addon, I get the following error:
[EMAIL PROTECTED] asterisk-addons]# make
cc -shared -Xlinker -x -o cdr_addon_mysql.so cdr_addon_mysql.o -lmysqlclient
-lz -L/usr/local/mysql/lib
/usr/bin/ld: cannot find -lmysqlclient
collect2: ld returned 1 exit status
make:
I don't know what I have done, but when I try to start Asterisk I get
Ouch Error writing audio data: Broken pipe
This scrolls endlessly and I cannot stop the screen except by killing the
terminal session.
TIA
Simon
-
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When I start or reload * I always get this error (once).
Can someone point me in the right direction to fix this.
WARNING[1150495040]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on
call [EMAIL PROTECTED] for seqno 102 (request)
Simon
-
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When you use firefly in SIP mode it does not un-register with * on exiting
the software
Simon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Hart
Sent: Friday, 26 March 2004 11:48
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] New minor release
rver. When on an outside call and another outside call comes in, I can
hear the call waiting indicator, but cannot find any way of swapping calls
(sending a hook flash signal through the X100P).
Any help would be greatly
appreciated.
Simon Brown
-This mail
was content checked for mali
] On Behalf Of Simon
Brown
Sent: Tuesday, March 23, 2004 5:20 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Call waiting
I have threewaycalling=yes in my zapata.conf file.
When I hear the call waiting indicator I dial *0# on the phone, but it
is ignored.
How can I get it to work
: [Asterisk-Users] Call waiting
On Tue, 2004-03-23 at 16:36, Simon Brown wrote:
How do you do a hook-flash? If I quickly press the hangup button
(like you
Most SIP PHONES do not support sending a flash by briefly hanging up the
call. Devices like the Cisco ATA-186 do, since you plug analog phones
documentation but cannot find the answer.
Any help will be greatly appreciated.
Simon Brown
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http
I have found the problem and it is now working. I did not have my access
list specified correctly.
Thanks for the help from those who responded.
Simon Brown
-Original Message-
From: Simon Brown
Sent: Tuesday, 23 March 2004 13:42
To: '[EMAIL PROTECTED]'
Subject: Asterisk behind
help would be greatly
appreciated.
Simon Brown
-This mail was content checked for malicious code and virusesby GFI MailSecurity.
of swapping calls
(sending a hook flash signal through the X100P).
Any help would be greatly
appreciated.
Simon Brown
-This mail
was content checked for malicious code and virusesby GFI
MailSecurity.
I had exactly the same problem. I tried removing and reinstalling several
times but it always crashed. I sent an email to verbiage asking for help and
all I got in response was Have you got it working yet? from them. I have
been unable to get a reply since.
Simon Brown
-Original Message
My system uses an X100P card connected to a PSTN line. I have call waiting
on the line. I am using Cisco 7940 (SIP) phones with this system.
Can someone tell me how I get the Cisco phone to send the correct Hook/Flash
signal to the X100P to pick up the call waiting?
TIA
Simon Brown
My system uses an X100P card connected to a PSTN line. I have call waiting
on the line. I am using Cisco 7940 phones with this system.
How can I get the Cisco to send the correct Hook/Flash signal to the X100P to
pick up the call waiting?
TIA
Simon Brown
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