Has anyone successfully had a SIP phone fail over from Asterisk
Server A to Server B using DNS SRV?
Definitely we have been doing this for quite a while.
If so, which phone worked for you? I'm assuming you set up your
DNS SRV records so that the IP
addresses of A and B
I too have heard of people persuading a vonage tech to provide the
password to log into and factory reset their device, but I get the
impression that it is an uncommon occurrence.. you'd be lucky, basically.
I have an ATA-186 that Vonage unlocked for me. They used to just charge
$20 or so (on
I'd say that would depend on the configuration you are considering. We
have a number of fax machines running off of sipura spa-2000's that
connect to a remote asterisk server and terminate to the pstn via voip
as well.
I'd say it's about 90% reliable at this point. However, we've noticed
is not possible, then send fax to PSTN destination using voip;
Did you use this http://tafm.sourceforge.net/ to make ATA sepura work ?
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Steven
Kokinos
Envoyé : mercredi 23 février 2005 15:46
À : Asterisk
Does anyone know where to obtain the 2.5 / 1.3.1 bootrom/app? My
account on polycom's site keeps pointing me at documentation only.
Regards,
-Steve
On Sep 1, 2004, at 10:00 AM, Matthew Marlowe wrote:
I'm using the same SIP version, everything is running great except as
I've said before that
Does anyone know of
any provider(s) that can provide DID's for the Czech
Republic?
Regards,
-Steve
I too have the same problem on a few units, but not on others. I also
have been having difficulty hooking up multiple lines from one Sipura
to the same multi-line phone system (seems to create a line cross) but
have no problems with either cisco or dlink boxes. In general they are
nice units,
i have been trying to get a newly liberated (from vonage) cisco ata-186
(sip ios v3.1) working properly with asterisk. my client is behind a
linksys wrt-54g, which up to this point hasn't proven to be a problem
(i have several sipura spa-2000's and polycom phones working just fine
behind
Beyond this, you can still just use the NAT keepalive in the Sipura.
While It only provides for either a NOTIFY or REGISTER (which both
generate errors in asterisk) if you change it to something else (I just
have it send blank, but a few ... or anything will do) asterisk won't
complain and
hello all,
I'm having problems getting my polycom soundpoint ip 500 working, and
was wondering if anyone would be willing to share their config files
with me (the polycom configs). I have managed to get my boot server up
and running, and the phone successfully updated its ROM, and downloaded
I currently have
several asterisk servers geographically distributed (for automatic fail-over in
the event of either a network or server problem). My carrier delivers to each
server based on the same priorities that I have set inthe DNS SRV records
which the clients point to.
Users always
does anyone out
there using zaprtc know how to go about initializing it at boot time? i have it
compiled and working properly, but there is very limited documentation.
-Steve
-04-20 at 17:23, Steven Kokinos wrote:
does anyone out there using zaprtc know how to go about
initializing
it at boot time? i have it compiled and working properly,
but there is
very limited documentation.
Yup, works great for me :)
Add this to rc.local to get it initialised at boot
i have a quick question from the latest build in the stable branch. in all
of the previous builds of asterisk i have used, calling either asterisk
itself or safe_asterisk spawns one asterisk process, like this:
root 11218 0.0 0.1 5244 936 pts/0S20:55 0:00 /bin/sh
, 2004, at 6:52 PM, Iain Stevenson wrote:
--On Thursday, April 15, 2004 6:43 pm -0400 Steven Kokinos
[EMAIL PROTECTED] wrote:
Actually, after rebooting my machine music on hold started working
properly. Not sure what the issue was. As for ztdummy, I am having a
more
substantive issue
I have had problems with the NOTIFY packet being sent (not problems,
just annoying warnings) when using keepalive with the Sipura SPA-2000.
Asterisk will complain both using the Register and Notify methods
(which are the two that the Sipura uses).
The NOTIFY warning isn't actually causing any
thanks to those who replied. I have managed to get the functionality I
was looking for working with a series of Macros. However, it doesn't
work as simply as I would like. There are two issues I've run into:
(1)Goto provides no way to pass variables between one context and
another.
(2)I can't
i've been searching the archives but can't find anything substantive on
this. most of the music on hold documentation discusses integrating
with zap hardware, but i am trying to send it across a sip channel.
I have the following in extensions.conf:
exten = 2100,1,Answer
exten =
, 2004, at 5:27 PM, Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Steven Kokinos [EMAIL PROTECTED] wrote:
i've been searching the archives but can't find anything substantive
on
this. most of the music on hold documentation discusses integrating
with zap hardware, but i am trying to send
I'm working on setting up a macro that will allow users to call their
own DID number, and when they hear their voicemail greeting hit the *
key and be prompted for their password to check vmail.
For some reason though the background command isn't working as I'd
expect it to:
[macro-vmessage]
it local to the macro.
-Steve
On Apr 14, 2004, at 6:11 PM, Chris A. Icide wrote:
On 02:44 PM 4/14/2004, Steven Kokinos wrote:
I'm working on setting up a macro that will allow users to call their
snip
exten = 1,*,Macro(checkmessage,${ARG1})
exten = 2,*,Hangup
try:
exten = *,1,.
exten = *,2
I've been intermittently seeing the following warning:
Apr 14 18:35:04 WARNING[1192437440]: rtp.c:386 ast_rtp_read: RTP Read
error: Resource temporarily unavailable
which doesn't appear to have any effect on the current call, but isn't
anything I've seen before either. Any thoughts on whether
Hello-
I have several Sipura SPA-2000's at different locations (all behind Linksys WRT54G
boxes). When setting:
nat=yes
qualify=yes
Things work properly about 90% of the time, however, if a remote end loses the
connection briefly, then asterisk can't see the adapter until the next
Hello,
I have been running into a problem on my server (which I believe was
the cause of an O/S crash earlier today). I am consistently seeing the
following messages in /var/log/messages:
Apr 8 05:24:04 east insmod: /lib/modules/2.4.20-8/misc/torisa.o:
insmod char-major-196 failed
Apr 8
Hello-
As part of the
asterisk build/installation instructions it mentions that the zaptel drivers
should be built and configured first. My question is whether they are required
at all, in the case of a system with no hardware cards at all (as is the
situation in my case).
With them
in my setup i have
several users with DID lines coming in from various sip/iax providers. within
our old phone system, a user could call their own DID line, then hit the * key
when they hear their voicemail greeting and be prompted for their password.
is there any way
this could be
Hello-
I am in the process of adding a new provider to my asterisk box (both
for outbound termination as well as inbound DID). They are going to be
delivering and receiving traffic via SIP only.
Now, in IAX via Voicepulse or others I know that I can simply have one
registration statement
Hello-
I'm obviously doing something wrong here in trying to get an inbound
DID to work with voicepulse.
I have an outbound context set-up for those calls in iax.conf, and the
appropriate register in- statement.
within extensions.conf I am doing something like this:
exten =
Scott-
Thanks for the tip. It was in fact that I didn't have the two contexts
matching. Once I resolved that everything is working great.
-Steve
On Apr 2, 2004, at 6:21 PM, Scott Laird wrote:
On Apr 2, 2004, at 3:12 PM, Steven Kokinos wrote:
Hello-
I'm obviously doing something wrong here
Hello,
I've just compiled and configured my asterisk box, and so far things
have gone pretty smoothly. Now I am working on being able to make a
call via voicepulse, and have run into an issue.
When I connect via my sip phone (using xlite) and dial a pstn number I
can see that it is being
actually figured this one out on my own, had incorrectly labeled
secret as password in iax.conf.
On Apr 1, 2004, at 6:14 PM, Steven Kokinos wrote:
Hello,
I've just compiled and configured my asterisk box, and so far things
have gone pretty smoothly. Now I am working on being able to make
There are a lot of wholesale VOIP providers now, they are probably just
buying capacity from one of them and either re-branding, or are using a
custom gateway and/or commercial solution.
-Steve
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zac
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