On 2012-03-13 18:38, Chris Bagnall wrote:
Greetings list,
I'm trying to source a very basic ISDN BRI - SIP gateway. Unfortunately,
everything I've seen seems to want to do lots of other things - registering
handsets, IVRs, voicemail, etc. I only want it to
present an ISDN BRI as a SIP account
Hi,
I'm using a packet interception module for modifying udp packets coming
to asterisk sip port. now my packet modification application
successfully forwards the packet but somehow there is no response from
asterisk. it may be that the modifications destroyed sanity of the sip
packet so asterisk
Original-Nachricht
Datum: Tue, 13 Mar 2012 22:40:27 +0200
Von: Tzafrir Cohen tzafrir.co...@xorcom.com
An: asterisk-users@lists.digium.com
Betreff: Re: [asterisk-users] DAHDI_SPANCONFIG failed on span 1: No such
device or address (6)
On Tue, Mar 13, 2012 at 05:10:14PM
Am 14.03.2012 00:34, schrieb James Sharp:
ping + arp isn't going to work if they're on a different VLAN.
I believe this will work:
Too complicated. Just have a look on the switch(es) the phones are
connected to.
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Hello list,
I have a client who's taking intermittent errors on their PRI. The server is
configured with one PRI from the TELCO, and two PRI connecting to their Iwatsu
ADIX legacy system. The odd thing is, the system can run for days, weeks or
months without a reported error and then just bomb.
I have a customer that has a CX3000 IP that was designed for MS Lync.
Anyone know if these can run as standard SIP so we can use it with
Asterisk?
Thanks
Bryant
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According to the specifications, it should connect with little difficulty.
http://www.voipsupply.com/polycom-cx3000
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant
Zimmerman
Sent: Wednesday, March 14, 2012 12:32 PM
To:
Hi ,
When I make a call to an extension, which is on another call, the called
party (who is on call waiting) will get a BEEP sound. But the caller gets
the normal ringing tone. Is there any way to have a different dialer tone
for the Caller, which lets him know that the other person is on a
snip
/etc/asterisk/chan_dahdi.conf
==
;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
;autogenrated on 2012-02-29
;Dahdi Channels Configurations
;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak
Try adding
resetinterval = never
here. The default is to reset any idle channels every 3600 seconds.
Unfortunately, if a channel is being reset just as an incoming call
arrives, there is a chance that the channel will get stuck in a
resetting state and block any further use of that channel.
On 03/13/2012 11:06 PM, Dave Platt wrote:
Ouch. That isn't going to be so easy to spot, then! You would have to guess
a bunch of likely passwords, fake up a challenge with some known nonce, and
compare the response against those you would expect with each of the various
possible passwords.
Hi,
I'm getting the messages listed below after login to asterisk cli;
-- Remote UNIX connection
-- Remote UNIX connection disconnected
Usually verbose level is set to 4, after setting to 2, I'm not getting
these messages.
Is there other way to stop these messages? because
Those messages someone or something is running asterisk -r or similar.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir
Sent: Wednesday, March 14, 2012 3:13 PM
To: asterisk-users@lists.digium.com
Depending on your Asterisk version, add hideconnect = yes to asterisk.conf
and restart.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir
Sent: Wednesday, March 14, 2012 2:13 PM
To: asterisk-users@lists.digium.com
Subject:
Any one having this problem.
The Grandstream Firmware revision is 1.2.5.3.
We have the registration time set to 5 minutes, and every time after a
reboot, the BLF's will initially indicate the correct state, then stop
working a few minutes later.
The workaround has previously been to reboot
Markus unive...@truemetal.org writes:
Does such a thing exist?
How does a2billing do it? It should be pretty easy in an AGI. If you can
afford a linear lookup per call, just grep through the array of prefixes
to find the ones matching a particular call, then pick the cheapest from
the results.
Hello,
Is anyone aware of an AMR-NB codec for Asterisk 10? The old patch floating
around for older versions of Asterisk doesn't seem to work anymore.
Best regards,
Jan Blom
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Am 15.03.2012 00:35, schrieb Benny Amorsen:
Markusunive...@truemetal.org writes:
Does such a thing exist?
How does a2billing do it? It should be pretty easy in an AGI. If you can
afford a linear lookup per call, just grep through the array of prefixes
to find the ones matching a particular
On Wednesday 14 Mar 2012, NaJIm wrote:
When I make a call to an extension, which is on another call, the
called party (who is on call waiting) will get a BEEP sound. But
the caller gets the normal ringing tone. Is there any way to have a
different dialer tone for the Caller, which lets him
A2Billing doesn't do that. A2Billing in fact has a lot of shortcomings one
of which is this exact issue.
I would suggest running rate sheets against each other for finding true LCR
and then only uploading the rates that are cheaper into the system. In most
cases there are not such high
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