Of Benjamin
Lawetz
Sent: August 29, 2006 1:08 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] DTMF between cisco and sipura going
throughasterisk
Hello all,
we're having an issue with DTMFs being sent to Sipura's. Calls are
originating from
]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg Boehnlein
Sent: August 31, 2006 12:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DTMF between cisco and sipura going
throughasterisk
On Tue, 29 Aug 2006, Benjamin Lawetz wrote:
Hello all,
we're
Hello all,
we're having an issue with DTMFs being sent to Sipura's. Calls are
originating from a Cisco AS5300 being sent to asterisk which in turn sends
it to the Sipura. Connected to the Sipura is a legacy PBX (or actually shows
the same problem with a cheap answering machine). The DTMFs
Still no answers huh?
I've asked a couple of time how to do this, and by the lack of answers, I'm
guessing there is no way.
The workaround unfortunately is to create an entry for each IP address in
the range (I hope you don't have to open up a whole C class)
-Original Message-
From:
: [asterisk-users] Trunk with multiple IPs?
Benjamin Lawetz wrote:
Still no answers huh?
I've asked a couple of time how to do this, and by the lack of
answers, I'm guessing there is no way.
The workaround unfortunately is to create an entry for each IP address
in the range (I hope you
I'm in the process of upgrading an asterisk to 1.2.10 and started by
upgrading libpri-1.2.3 (make make install) and zaptel (make make
install).
Was about to install asterisk, but doing a ls I get the following error:
ls: relocation error: /lib/libpthread.so.0: symbol _h_errno, version
GLIBC_2.0
: [Asterisk-Users] Early media after a dial command
Hi Benjamin,
How do you setup early media in asterisk ?
Harry
--- Benjamin Lawetz [EMAIL PROTECTED] a écrit :
Hello all,
I've been playing around with early audio, and I'm able to get some
things working
We have PSTN calls coming
Hello all,
I've been playing around with early audio, and I'm able to get some things
working
We have PSTN calls coming in to asterisk in SIP from a Cisco AS5300. If I do
the following:
Exten = i,1,Playback(ss-noservice,noanswer)
Exten = i,2,Congestion(15)
Exten = i,3,Hangup()
The PSTN caller
I will tell you straight up that NFS mounted volumes will cause asterisk to
croak if it needs access to something that's not mounted. The first time
the NFS share disappears for a moment, you're going to be restarting
services and losing time on the asterisk machines that need the mounts. It
I had made a patch a while back so it retrieved the emailbody and
emailsubject from the users table in mysql.
Let me see if I can dig it up
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: April 18, 2006 9:05 AM
To:
Had this working also at some point, but had one killer problem... NAT
issues! Most of our clients are natted, and depending on the router, they
only allow traffic to return from the server that the traffic was sent to.
So the invites coming from other servers were being dropped.
But besides that
As BJ mentionned, it could be your MWI of depending on your profiling, it
might be scheduled to download it's profile every hour, and therefore might
reboot and ring after each download
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Koski
Sent:
I'd change your definition to something
like
[providerX]
context=providerX-inbound
host=11.222.222.23
in your providerX-inbound context you can match the
different extensions
[providerX-inbound]
exten =
514907,1,NoOp(514907)
exten =
55,1,NoOp(55)
Now a question
I don't think they have a specifig Provisioning Guide for each device. They
have a general provisionning guide and you can generate an example from the
Sipura Profile Compiler for the available options though
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Actually got my hands on one, it's not that bad size wise. About the size of
a big smartphone
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel
Hiver
Sent: January 10, 2006 11:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Or you can treat everything as a 10 digit number retaining in a variable
whether the user dialed one or not
exten = _1NXXNXX,1,SetVar(ONPRESSED=TRUE) *** skip this step if you
don't care whether the one was pressed in any of your dialplans
exten =
Hello,
In the process of upgrading a couple of voicemail servers from CVS (end of
august 2005) to 1.2.1
This is a purely voicemail system using mysql configurations.
All my mailboxes are in the default context and it worked fine under the
CVS version. But with 1.2.1 the voicemailmain fails to
Title: Linksys SPA-841 Disconnects from Asterisk
Check in you console or your logs when this happens. I'm
guessing it's a Stale Nonce
If this is the case, Sipura supposedly fixed the bug on
it's most recent firmware (At least for the SPA-1001 and SPA-2100, but I'm
guessing the SPA-841 also)
Hello,
I'm trying to configure one of our providers for incoming calls only.
He's sending me SIP calls from a certain range of IP addresses (let's say
192.168.5.0/255.255.255.0 for example purposes).
And I'm trying to configure sip.conf to send his calls in a specific
context, but he still keeps
I have a dialplan entry that currently when you call, sets some variables
and then calls DISA.
I'm having a bit of a unnatural timeout from there.
I would want from the DISA to match certain patterns to know when the number
is complete
A 311 or 911 call would be dial as soon as the third digit
I had written a patch to add an else argument to gotoiftime (like the
goto command)
http://bugs.digium.com/view.php?id=5223nbn=11
But the feedback was split on the need for this/the way to do it.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
, 2005 9:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Redundency
Benjamin Lawetz wrote:
Since I can't do that, what I've settled on is heartbeat + mon.
Heartbeat will monitor for a system level failure and switch to the
backup
an automatic test call or to
simulate a call somehow would be appreciated.
Benjamin Lawetz wrote:
Ok, I tried something slightly different.
I modified the existing the udp.monitor (or was it the tcp.monitor) of mon
and basically sending a sniffed SIP Registration packet which I send to
the asterisk
On Tue, 25 Oct 2005, Benjamin Lawetz wrote:
Well what I was thinking of doing in the future was to have a cron job
drop a call file that would call another asterisk server that would
auto-answer and either generate some kind of network answer to MON or
connect another call to the first
Since I can't do that, what I've settled on is heartbeat + mon.
Heartbeat will monitor for a system level failure and switch to the backup
machine if neccesary; and mon will watch the asterisk (or any
other) service and restart it and/or alert me if it fails.
What kind of monitor are you
I'm looking for a way to transmit a user specific variable to my dialplan
If we use the example of the hair color, I was thinking of having something
like:
[bob]
context=users
host=dynamic
secret=password
type=friend
username=bob
hair=brown
[lary]
context=users
host=dynamic
secret=password
I'm running multiple asterisk servers and need to use the CheckGroup
function (and other group functions) across multiple servers
Ex:
- there are 5 channels in group test on server 1
- there are 8 channels in group test on server 2
I would need a checkgroup to return me 13.
Any way to
I seem to recall this problem on the mailing list a couple of months ago,
I'd point you towards, but can't seem to find it.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Max Braz
Sent: September 21, 2005 9:29 PM
To: asterisk-users@lists.digium.com
I've got a friend who's spending 6 months on the other side of the world. So
before he left I configured him a softphone on his laptop to connect to my
asterisk so he can call home free of charge.
Unfortunately, he just found out he has horrible internet connection.
Bandwith and latency is ok,
We're currently working on a SIP load-balancing system using ipvsadm and so
far it seems to work pretty well.
We load balance the port 5060 registration (tracking the connection for a
little more time than the registration retry so that it goes back to the
same server) using tunneling.
So a
The catch all extension I use is _. (match everything).
That's a nono, but that is not the problem :-)
and also tried _X. (match any numeric) don't match special extensions.
Much better!
From voip-info.org on the cmd VoiceMail page:
If, during the recording the caller presses:
'#' - or
Just put 203 in his own context which reacts to those numbers, and then
include your normal context
[restrict]
Exten = _00.,1,goto(unauthorised,1)
Exten = _087.,1,goto(unauthorised,1)
Exten = _086.,1,goto(unauthorised,1)
Exten = unauthorised,1,Playback(invalid)
Exten = unauthorised,2,wait(2)
Was wondering if anyone else was getting these kinds of messages with
Sipuras (SPA-1001 and SPA-2100):
WARNING[26867]: Forbidden - wrong password on authentication for INVITE to
'sip:[EMAIL PROTECTED];tag=as25fb4c1a'
It's a warning, so I'm guessing it's not more critical than that. But I'm
Hi,
I've been having a little problem with my asterisk servers, I have 4
identical asterisk servers setup (same hardware, same OS, same config). Once
in a while (once or twice a day) one of the server crashes on the cron job
reload. But I realized this only happens on 3 of the 4 servers. Tried to
Is asterisk able to forward it's ALERT_INFO data to another asterisk server
?
My situation should look like the following:
Call comes into asterisk1 in SIP. Asterisk1 sets the ALERT_INFO=Bellcore-r2,
Asterisk1 dials Asterisk2 (SIP), Asterisk2 dials our SIP device which should
ring with the
I'm having a little problem. I have a dial-plan with a lot of SetVar's and
loops, and under certain circumstances (reproducible) it makes asterisk
crash. Wanting to debug this, I compiled using make valgrind. But doing
so, I eliminated the crashes and the dial-plan works perfectly.
Now from what
, 2005, at 10:44 AM, Benjamin Lawetz wrote:
I'm having a little problem. I have a dial-plan with a lot of SetVar's
and loops, and under certain circumstances (reproducible) it makes
asterisk crash. Wanting to debug this, I compiled using make
valgrind. But doing so, I eliminated the crashes
Was scratching my head trying to figure out why what I was doing wasn't
working. Ran a couple of tests, and found out that the string length
function has a limit of 79!!
If the string you are checking is longer than 79 characters, it will
return 79.
Is this a bug (using Asterisk 1.0.5) or
Hi everyone,
I'm having a little problem and was wondering whether anyone would have
any ideas or pointers for me.
I've been working on load-balancing asterisk and have had a pretty
successful setup using LVS and IP tunneling (plus a bit of iptables
nating).
I am only load balancing the SIP
Hi,
I'm setting up multiple asterisk servers and trying to do the classic
DIAL(IAX2/asterisk1/${EXTEN}IAX2/asterisk2/${EXTEN}IAX2/asterisk3/${EXTEN},15)
After googling a bit, I fell on a discussion about putting this in a
variable so that adding additionnal servers would be easy. I can't seem
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