Hi All
I am looking at a replacement for a hotel PBX which requires at least 60
analogue extensions.
I tend to use Sangoma equipment but haven't tried this many analogue
extensions before. I am interested in anyone's experience of which
server platform literally fits and copes well with
2009/3/14 Vieri rentor...@yahoo.com
--- On Sat, 3/14/09, Olivier oza-4...@myamail.com wrote:
If I understand correctly, you're suggesting to
implement the h priority
instructions (or a hangup macro) to:
1) run a deadagi or a system() script to see if
someone has left a request
Duncan Turnbull wrote:
Hi All
I am looking at a replacement for a hotel PBX which requires at least 60
analogue extensions.
I tend to use Sangoma equipment but haven't tried this many analogue
extensions before. I am interested in anyone's experience of which
server platform literally
Hi,
Xorcom make what you are looking for.
Steve
On 3/15/09, Duncan Turnbull dun...@e-simple.co.nz wrote:
Hi All
I am looking at a replacement for a hotel PBX which requires at least 60
analogue extensions.
I tend to use Sangoma equipment but haven't tried this many analogue
extensions
Hi,
Is there any way to tell Asterisk not to generate additional headers like:
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
I can't find any relevant option in sip.conf file :-(
Thanks for help.
Chris
___
-- Bandwidth and
2009/3/15 Chris Maciejewski ch...@wima.co.uk
Hi,
Is there any way to tell Asterisk not to generate additional headers like:
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
I can't find any relevant option in sip.conf file :-(
For curiosity's sake, what are the
I am trying to change the SIP response to an incoming call and according
to the docs and a quick scan of the chan_sip.c code one is supposed to
be able to set the PRI_CAUSE variable and invoke Hangup but regardless
of the value in PRI_CAUSE the SIP rejection is always 603.
What is wrong here?
2009/3/11 Tzafrir Cohen tzafrir.co...@xorcom.com
On Wed, Mar 11, 2009 at 02:56:58PM -0500, Kevin P. Fleming wrote:
Vieri wrote:
- use the latest release of misdn v1
- upgrade to the latest stable kernel and use the built-in misdn v2
There is no support for mISDN v2 in Asterisk to
Got this in the log, with no calls active. Is it a problem with my isdn
line, or * ?
[Mar 15 11:36:18] ERROR[29161]: chan_dahdi.c:8735 dahdi_pri_error: ACK
received for '0' outside of window of '39' to '40', restarting
[Mar 15 11:36:18] == Primary D-Channel on span 1 down
[Mar 15 11:36:18]
On 15 Mar 2009, at 11:30, George Pajari wrote:
I am trying to change the SIP response to an incoming call and
according
to the docs and a quick scan of the chan_sip.c code one is supposed to
be able to set the PRI_CAUSE variable and invoke Hangup but regardless
of the value in PRI_CAUSE the
Hi,
We've implemented a 'page-all' function for some of our customers, and
we've noticed that
on occasion the page-all will cause asterisk to crash (safe_asterisk
then restarts it again).
The particular customer has about 20 phones, and also has 5 Linksys
932 to monitor the state of these
On Sun, 15 Mar 2009, Duncan Turnbull wrote:
Hi All
I am looking at a replacement for a hotel PBX which requires at least 60
analogue extensions.
I tend to use Sangoma equipment but haven't tried this many analogue
extensions before. I am interested in anyone's experience of which
server
For curiosity's sake, what are the troubling consequences of having those
headers included ?
My PSTN termination provider sometimes replies with SIP/2.0 513
Message too big to my BYEs with additional headers included. Just
wanted to check if this is the reason, or maybe it is related to
Hello all,
Ok it is Sunday afternoon and I am going crazy. I have been running in
circles so long that I can't think straight. As an example, I sent this
message to the wrong address the first try, AAAGGH. I have
Asterisk 1.6.0.6 and DAHDI Tools Version - 2.1.0.2,
DAHDI Version:
Hi I looked at few emails related to this subject. And still not sure
how to solve the loop detect problem for my case
iqb...@improvise:/etc/asterisk$ cat sip.conf
[general]
context=line1
[phone]
type=friend
context=phone1
secret=g00dpazzwerd
bindport=5060
host=192.168.1.106
dtmfmode=rfc2833
John Millican wrote:
# /etc/init.d/dahdi start
Loading DAHDI hardware modules:
wctdm: modprobe wctdm
What is the output of the 'dmesg' command at this point?
No hardware timing source found in /proc/dahdi, loading dahdi_dummy
Running dahdi_cfg: /usr/sbin/dahdi_cfg
If the dmesg
Shaun Ruffell wrote:
John Millican wrote:
# /etc/init.d/dahdi start
Loading DAHDI hardware modules:
wctdm: modprobe wctdm
What is the output of the 'dmesg' command at this point?
No hardware timing source found in /proc/dahdi, loading dahdi_dummy
Running dahdi_cfg:
John Millican wrote:
Shaun Ruffell wrote:
John Millican wrote:
# /etc/init.d/dahdi start
Loading DAHDI hardware modules:
wctdm: modprobe wctdm
What is the output of the 'dmesg' command at this point?
All I see in dmesg is:
dahdi: Telephony Interface Registered on major 196
On Sun, Mar 15, 2009 at 6:28 PM, Asif Iqbal vad...@gmail.com wrote:
Hi I looked at few emails related to this subject. And still not sure
how to solve the loop detect problem for my case
iqb...@improvise:/etc/asterisk$ cat sip.conf
[general]
context=line1
[phone]
type=friend
Thanks very much Rob Stephen
The channel banks look good. I am not sure if they are easily availble
in NZ but we can get some in I am sure.
Xorom make very positive comments about their astribanks and that you
can have multiple channel banks on a server so they look pretty good (if
they are
On Sun, Mar 15, 2009 at 8:28 PM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
On Sun, Mar 15, 2009 at 6:28 PM, Asif Iqbal vad...@gmail.com wrote:
Hi I looked at few emails related to this subject. And still not sure
how to solve the loop detect problem for my case
I am probably missing something, being a newbie. I have a 4 port
fxs/fxo (2/2) card.
My land line is going to one of the FXO port and my home phone is connected
to one of the FXS port.
I want to be able to call my phone number from external phone (cell phone)
and have my home phone ring.
Hi,
problem is that you are saying that phone in sip.conf is at the same
ip address of your asterisk box so you are dialing into a loop to your
self asterisk box
[phone]
type=friend
context=phone1
secret=g00dpazzwerd
bindport=5060
host=192.168.1.106
dtmfmode=rfc2833
what you need is:
[phone]
Hello Asif,
I have experienced 'loop detected' when the peer where I want to send
the calls to, and the asterisk Box have both the same IP address (That
would make a loop).
Could you please verify?
Regards,
Asif Iqbal wrote:
Hi I looked at few emails related to this subject. And still
hi,all
i have just set up asterisk 1.6.0.7 rc1 with spandsp 0.0.5 pre4 to ReceiveFAX,
link to a E1 (DE410P) using dahdi
this can receive the fax from E1 successfully, but i see many error message in
the log like this:
[Mar 16 09:24:38] ERROR[23540] channel.c: ast_read() called with no recorded
asterisk 1.4.23.2 and spandsp 0.0.4 get the same error nowbut less times
than other version ...
[Mar 16 10:12:50] DEBUG[23749]: chan_dahdi.c:7115 do_monitor: Monitor doohicky
got event Alarm on channel 1
[Mar 16 10:12:50] DEBUG[23752]: chan_dahdi.c:4731 __dahdi_exception: Exception
on 11,
On Sun, Mar 15, 2009 at 9:57 PM, MaxGao ss...@126.com wrote:
and many times when reciving tax , the E1 card will down , all the channel
get red alarm...
[Mar 16 09:49:19] DEBUG[20928] chan_dahdi.c: Monitor doohicky got event
Alarm on channel 2
[Mar 16 09:49:19] WARNING[20928] chan_dahdi.c:
Shaun Ruffell wrote:
John Millican wrote:
Shaun Ruffell wrote:
John Millican wrote:
# /etc/init.d/dahdi start
Loading DAHDI hardware modules:
wctdm: modprobe wctdm
What is the output of the 'dmesg' command at this point?
All I see in dmesg is:
dahdi: Telephony Interface
Again, if I am interpreting this correctly, he is not using SIP. A
four port card 2fxo/2fxs means to me that he is not using SIP at all.
If by card, you mean some kind of SIP gateway, then I misunderstood
and the problem, but seeing DAHDI channels leads me to believe that
SIP is not required and
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