Does anyone know if you can have multiple TE110P cards in one chassis?-Thermal
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Yes - we even have a server at a clients site with 2 TE410P's in it as
an interim measure. shudder
PaulH
On Wed, 2006-10-11 at 20:01 -1000, Thermal Wetland wrote:
Does anyone know if you can have multiple TE110P cards in one chassis?
-Thermal
___
Michael,
After many months of search we decided to develop an in-house solution
for such kind of needs. For a month our solution is in production and
does everything you mentioned below. Asterisk's built-in call queue does
not provide many of the features necessary for large organizations.
Idris
Hi,
Can you paste us the error messages when you compile cdr_addon_mysql
Best regards
Le jeudi 12 octobre 2006 à 03:07 +0100, Marco Mouta a écrit :
Hi guys,
I've been installing Asterisk 1.4 with Asterisk addons, and i could
notice that in /usr/lib/asterisk/modules/ doesn't have
Hi everybody,
With SER, we can put the location table relative to the registers
messages on db with a radius server or with modparam(usrloc,
db_mode, 1) and a mysql table.
In asterisk, there is the dialplan and the sip definition on db, but the
location table is in memory. Do you know if a patch
On Wed, Oct 11, 2006 at 05:56:12PM -0400, John Kane wrote:
I am trying to write a script to attempt to make a call on a Zap channel,
and if it fails, send an alarm. I can generate the call, but because the
Zap channel accepts the call, even though the other end never answers, it
sees it as a
I am using a2billing as billing software ,and I make an 800
call service which means that the destination extension should be build
I put this code at extensions.conf
exten = 99909994,1,SetAccount(2704714849)
exten = 99909994,2,Wait,2
exten = 99909994,3,DeadAGI(a2billingp.php)
helloi want to spy on a chennel listen the voice conversation between two person.i also want talk to one of them but others will not listen my voice.as per my understanding chanspy 1.4 may solve this problem,
please answer is it possible chanspy 1.4 and how to configure it.thanks,nsthiru
Apparently (from what I gathered from #openwengo at
irc.freenode.net)Wengo's own network runs on a combination of Asterisk and
OPENSer. To get Wengophone working with your asterisk you will need to do
some code hackingz...so download the source code and change it. You will
need to change the
Thirumal Saminathan wrote:
hello
i want to spy on a chennel listen the voice conversation between two
person.
i also want talk to one of them but others will not listen my voice.
as per my understanding chanspy 1.4 may solve this problem,
please answer is it possible chanspy 1.4 and how
Wednesday, October 11, 2006, 6:26:59 PM, Samy Kamkar wrote:
Check out the pickupgroup and callgroup options in sip.conf -- these
looks good. thanks.
--
Best regards,
Csibra Gergomailto:[EMAIL PROTECTED]
___
--Bandwidth
Hi,I want to register with http://www.inphonex.com VoIP provider. I want to configure my "Trixbox" and "Asterisk" servers with "inphonex". Anybody using this service? Mainly, I want to do three tasks. They are1) Able to make calls to USA2) Able to make international dialing3) Able to receive
Hi,
I'm trying to interface an SDX/Lucent/Avaya INDeX switch with an
asterisk box using the INDeX's networking feature.
This works but all calls passed to the index are received with a 5
digits rather than the 4 they really have. Users on the INDeX dial 4
digits, the digit 6 is added as a
PH == Paul Hales [EMAIL PROTECTED] writes:
PH Yes - we even have a server at a clients site with 2 TE410P's in
PH it as an interim measure. shudder
What is particularly horrible about having 2 TE410P's?
/Benny
___
--Bandwidth and Colocation
I'm having trouble installing Beronet BN4S0 card. I have downloaded
instructions from here
http://www.beronet.com/download/card_installation_guide.pdf
And when I download install-misdn-mqueue[1].tar.gz I untar it and execute
make and make install. This is the output that I get.
[EMAIL
On 11 Oct 2006, at 19:35, Dean Collins wrote:
Lol - use a real PC maybe :P
Nah, that would be dull.
In some ways the mac intel is nearer to a 'normal PC'
(whatever that is) than the systems I normally run asterisk on
- a NatSemi Nemiah and an arm5 :-)
Asterisk 1.2.X runs fine on the intel
As long as you have no interrupt conflicts...don't
see why not...
We have 3 TE410P cards in a Dell 2850...had to
disable hyperthreading in the bios...and then make sure we had no shared
interrupts on them...
Work fine though...See no reason why you should
have any problem with more than 1
nothing heh...
- Original Message -
From: Benny Amorsen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, October 12, 2006 5:21 AM
Subject: [asterisk-users] Re: Multiple TE110P cards in one chassis
PH == Paul Hales [EMAIL PROTECTED] writes:
PH Yes - we even
Digium sells cables to interconnect them for timing. (dunno if thats
only for the 412 cards).
zoa
Don wrote:
As long as you have no interrupt conflicts...don't see why not...
We have 3 TE410P cards in a Dell 2850...had to disable hyperthreading
in the bios...and then make sure we had no
Dear
I am using
a2billing accounting software, how can I charge on the destination target not
at the caller side
Ex: if user
A have 10$ and user B have 10$ ,and they have a trunk 127.0.0.1 with charge 1$ for the minute for on net call .
When user A call
user B for 1 minute ,user A
Hi all,
for all those using asterisk + voismart gsm cards,
we have released a new package that fixes a lot of issue
and add some new features.
take a look to voismart open source website for it:
http://open.voismart.it
Greetings,
Matteo.
--
Matteo Brancaleoni
RD Director
Tel +39.02.70633354
James Andrewartha wrote:
Jessee J Holmes wrote:
As far 1.6.7 firmware supporting multiple presences (48 i think), maybe
I was wrong on that; however, I remember reading the 2.0.1 firmware
release notes and they mentioned that feature was fixed within the 2.0
firmware. Maybe they fixed it before
Dear folks,
I have problem in fax reception. The astrisk detects the fax tone and jusmps
to the fax extension and rxfax application starts and the max machine starts
the fax but saddenly stops and seems the rxfax have died. It doesnt returns,
not files in the output dir and ..
[EMAIL PROTECTED] wrote:
I am an asterisk newbie. I have successfully installed asterisk on Freebsd.
The problem I am having is when I try to route based upon incoming DID.
CALLERID(dnid) nor CDR(dst) have a number in them. Please help.
Digium analog cards do not support DID service.
Alex Robar wrote:
Analog routes (ie. copper telco lines) do not have DID information on them.
Only digital lines (PRI, often VoIP DID) have this information sent
alongside the call.
Analog lines in the USA can support DID, but only using things like EM
Wink which the Digium cards do not
Make sure you have the correct version of libtiff installed.
On Thu, 2006-10-12 at 12:22, Mohammad Shokuie wrote:
Dear folks,
I have problem in fax reception. The astrisk detects the fax tone and jusmps
to the fax extension and rxfax application starts and the max machine starts
the fax
Thanks Eric, I didn't know that. The general answer I've always seen regarding analog lines was that they supported CID only, and never sent DID. Good to know... Thanks,Alex
On 10/12/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Alex Robar wrote: Analog routes (ie. copper telco lines) do not
On ISDN lines it's possible to prohibit the
presentation of caller id, what if I have a SIP
gateway, something like an Audiocodes Mediant
1000. How do I prohibit the caller id presentation
on that one?
Regards,
Kristian
--
Kristian Larsson KLL-RIPE
Hello everybody,
I have a problem and already browsed the mailing list archives but
didn't find any help. So I ask here
My new * Box ist up runnig. Got access to the SIP server of my
Internet provider (Userid, password, phone number, ...). And yesterday I
tried my first calls to the outside
On Wednesday 11 October 2006 15:16, Douglas Garstang wrote:
Are you serious? Would you really just wait until a system looked like it
was on shaky ground before deciding to build a new one? What about if some
other component failed? What about the myriad of other failures you didn't
think of
On Wednesday 11 October 2006 17:56, Douglas Garstang wrote:
I have no data to prove it, but isn't the time between failures on this
type of TDM PBX equipment far better than a commodity server? Do they have
any moving parts? A server has moving parts, and moving parts fail.
Norstar's voicemail
You may have better luck asking the a2billing list.
Try here: http://forum.asterisk2billing.org/
bp
On 10/12/06, Khaled Chehab [EMAIL PROTECTED] wrote:
Dear
I am using a2billing accounting software, how can I charge on the destination target not at the caller side
Ex: if user A have 10$ and
On Wednesday 11 October 2006 20:30, Jay R. Ashworth wrote:
On which topic: do *you* know who to call and what to tell them to get
your lead DID forwarded to your cell phone when your span (or switch)
goes down?
I've already got that covered; it was in the PRI install. If the D channel
goes
I have had this problem before and it always turns out to be the fire wall.
You SIP registration and signaling (port 5060) is going thru okay but the
audio signals use a range of different ports which (if blocked) will cause
the problems you experience. Try putting * in DMZ to test this theory
helloi want to spy on a chennel listen the voice conversation between two person.i also want talk to one of them but others will not listen my voice.please answer is it possible chanspy
1.4 and how to configure it.thanks,nsthiru
___
--Bandwidth and
Frankly waiting for the box to break will loose you the client.
I would change the box but use the original Hard Drive, it only takes a
couple of minutes on a small system.
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada
On Wednesday 11 October 2006 15:16, Douglas
hello Jason.
It is using DMA. When the stutter happens it happens across all inbound
or sip lines at the exact same time. I thought it might be the IDE drive
but I have noticed it happening with no disk activity at all.
...Boyd
-Original Message-
From: [EMAIL PROTECTED]
Hi,
I'm getting a lot of messages like this:
Forcing Marker bit, because SSRC has changed
I searched on internet but found nothing useful.
I have an A102 beronet card on an Asterisk 1.2.9.1 box.
What does that message mean? Is it connected to a problem with the PRI
line (fastweb italia) or
On Wed, Oct 11, 2006 at 12:20:10PM +0200, Csibra Gergo wrote:
ps.: sorry for starting new thread with reply, but I can not send
mails to this list otherwise.
If your mail program will let you, try to delete the In-Reply-To
header; that will unthread your message.
Cheers,
-- jra
--
Jay R.
Sounds more likely to be a problem on the SIP/RTP side of your setup to
the PRI side (at a guess).
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Giorgio
Incantalupo
Sent: 12 October 2006 14:21
To: asterisk-users@lists.digium.com
Subject:
On Wed, Oct 11, 2006 at 09:00:20AM +0200, Olle E Johansson wrote:
The new channel will have configurations for trunks, services and
phones. It will
Does that mean that it will make a distinction concerning the
difference in administrative span of control between trunks, which go
to the
On Thursday 12 October 2006 09:02, Henry.L.Coleman wrote:
Frankly waiting for the box to break will loose you the client.
I would change the box but use the original Hard Drive, it only takes a
couple of minutes on a small system.
Exactly. Keeping some extra TDM hardware around for several
Hi Frank, I sent a patch updated here:
http://bugs.digium.com/view.php?id=6082
But that was some months ago, I havent seen a bugmarshall for a while
there, so I keep patching my own Asterisk for several stuff. New
features are never added to release branches, so you need to patch
1.2.12.1
On Wed, Oct 11, 2006 at 02:17:11PM -0400, Andrew Joakimsen wrote:
Asterisk can only be the proxy/server for MGCP, you connect other
devices to it. Asterisk can not be a user agent connecting to other
MGCP server.
As it happens, I have a 4 port VoiPACK gateway that speaks MGCP (legacy
of an
I worked for some one that installed servers. He has several fully built
machines with clean installs ready to go if a client needs a loaner.
- Original Message -
From: Andrew Kohlsmith [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, October 12, 2006 3:56 PM
On Wed, Oct 11, 2006 at 09:11:43AM -0500, Aaron Daniel wrote:
That's a bug with the 7.5 firmware. I would suggest upgrading to the
8.4 version, we've been running it for a few weeks in a test environment
and everyone's been pretty satisfied with the new firmware (read:
nobody's complained).
On Wed, Oct 11, 2006 at 12:07:50PM -0400, Bob Chiodini wrote:
We had a power failure that took down the internet connection and local
DNS server. My local Cisco phones could not register (IP addresses are
hard-coded) and, because of the DNS failure I could not register with my
SIP provider.
On Thu, Oct 12, 2006 at 02:26:16PM +0200, Norbert Zawodsky wrote:
As soon as the connection is up and the receiver is lifted on both
sides, the leds of the DSL Modem between Asterisk and my ISP, and the
leds of the switch between Asterisk and the SNOM phone start rapidly
flashing. So I assume
Hello,
I have recently a practical with students about Asterisk and MGCP, I m
used to SIP and asterisk concepts, but I d like to know if someone has
some experience or success story about Centrex features with MGCP (as
far I know it s the most used protocol for centrex).
I' d like to show the
On Thu, 2006-10-12 at 10:20 -0400, Jay R. Ashworth wrote:
As a general rule, if you aren't already, you should have your Linux
box running a local DNS server, to which everything in your net should
be pointed, and that server *should have an authoritative zone for your
local RFC 1918 network
Heh, well, I actually just started a blog to keep track of various
goings on, but I just started it so it's kinda scarce.
I intend to update it in and out with various information I email to
people so everyone can benefit from the questions and answers people
use. I'd like to see other people
On Thu, 2006-10-12 at 10:20 -0400, Jay R. Ashworth wrote:
On Wed, Oct 11, 2006 at 12:07:50PM -0400, Bob Chiodini wrote:
We had a power failure that took down the internet connection and local
DNS server. My local Cisco phones could not register (IP addresses are
hard-coded) and, because of
As far as I know any signaling protocol you can get on a Channelized
Voice T-1 you can also get on analog lines. In fact these signaling
protocols originated on analog lines and are simply emulated on voice
T-1s. Notice I said Channelized Voice T-1, not ISDN PRI.
Granted, analog lines with
Hi,
I have some (5-10 per day on an average 250 calls/day) incoming calls
dropped after 25 to 60 seconds.
Asterisk is 1.2.10 + BriStuff 0.3.0-PRE1s on one hand (with 4 ISDN lines...)
Snom 320 SIP IP Phone (release 6.2.3) on the other.
With SIP Debug on, it *_looks_* like a normal call
Avi Miller wrote:
On 04/10/2006, at 1:55 AM, Matt wrote:
How can I make * aware of the other ext on the remote box so the DID
caller can access them like he can with the local box?
On each box, define the other range:
Box A:
exten = _9XX,1,Dial(IAX2/BoxB/${EXTEN})
Box B:
exten =
On Thu, Oct 12, 2006 at 04:53:51PM +0200, Dave Cotton wrote:
It's more likely directly linked with how asterisk deals with
registrations to external SIP/IAX servers it appears to sit there for
ever trying to do the registration, then when an internal phone tries to
re-register it can't, in the
-Original Message-
From: Dovid B [mailto:[EMAIL PROTECTED]
Sent: Thursday, October 12, 2006 8:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How big is *your* dialplan??
I worked for some one that installed servers. He has several
thanks Andrew,
I realize the #1 change, my problem is that for some reason, when
under a minimal amount of load, say 4 to 5 simultaneous calls, the
transfer capability starts working somewhat different, either won't get
the # or won't get the redirection digits. Since I'm dialing from
This are the things that make me believe in technology. I wonder if Ubuntu Linux advocates will help with the development of the controlling modules.
*
Reuters 16:55 PM Oct, 11, 2006
AMSTERDAM -- Palm and pumpkin seed oil could soon be generating electricity to help power cell phone
Matt wrote:
Avi Miller wrote:
On 04/10/2006, at 1:55 AM, Matt wrote:
How can I make * aware of the other ext on the remote box so the DID
caller can access them like he can with the local box?
On each box, define the other range:
Box A:
exten = _9XX,1,Dial(IAX2/BoxB/${EXTEN})
Box B:
On Thu, 2006-10-12 at 12:00 -0400, Jay R. Ashworth wrote:
On Thu, Oct 12, 2006 at 04:53:51PM +0200, Dave Cotton wrote:
It's more likely directly linked with how asterisk deals with
registrations to external SIP/IAX servers it appears to sit there for
ever trying to do the registration, then
Dave Cotton wrote:
On Thu, 2006-10-12 at 12:00 -0400, Jay R. Ashworth wrote:
On Thu, Oct 12, 2006 at 04:53:51PM +0200, Dave Cotton wrote:
It's more likely directly linked with how asterisk deals with
registrations to external SIP/IAX servers it appears to sit there for
ever trying to do the
I've read alot of comments on the SPA-3000, many if not all saying they had echo
issues, but I've not seen anyone comment on the SPA-3102. Does anyone have any
comments or issues with these?
Tim
___
--Bandwidth and Colocation provided by Easynews.com
Well, it looks like AstriCon 2006 is going to be big. We've sold out
the entire Westin Park Central -- every last room. So, here are some
nearby hotels to check if you're planning on coming down to Dallas for
the big Asterisk-fest.
Wyndham Garden Hotel-Park Central
8051 Lyndon B Johnson Fwy
on an analog Zap PSTN channel, you have no real way of determining if
the remote side answered, because, as you discerned, it IS considered
answered as soon as asterisk opens the channel.
How about you contact another asterisk server through the PSTN, and dial
through to an extension on that
Hi,
We just upgraded from 1.2.7 to 1.2.12.1. Everything is fine, except
that asterisk seems to just crash at random. Often I can make it
crash by using the ChanSpy function (which we use to monitor agents).
Sometimes it will just crash on its own.
The reason we were initially running 1.2.7
We have seen more random crashing on 1.2.12.1 as well as compared to 1.2.7.1
It's not that bad, we only have one crash per week out of 6 servers
that are on 1.2.12.1, but it is much more than when we were running on
1.2.7.1.
It's never the same reason when we do a gdb backtrace on it though so
Steven,
How many people does this make? Eg how many rooms in the Westin?
Regards,
Dean Collins
Cognation Pty Ltd
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Steven Sokol
Sent: Thursday, 12 October 2006 1:13 PM
To: Asterisk
Same thing here.. I can't pin it down.. other then I can make it
happen by using ChanSpy. This is an e-mail I got from our CSR Manager
today..
Technician is talking to the customer.
The call drops.
The technician still receives calls as if they were logged-in to the
queue, but the callcenter
Digium doesn't recommend timing for instance...all timing off say off 1...
But we do it and it works flawlessly...
- Original Message -
From: Zoa [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, October 12,
Tzafrir Cohen a écrit :
[...]
Did you know a good GPLed softphones which works on Windows ?
IAXcomm should. So should wengophone and mozphone.
And Kiax and Ekiga
--
Daniel
___
--Bandwidth and Colocation provided by Easynews.com --
On Thu, 2006-10-12 at 11:56 -0500, Eric ManxPower Wieling wrote:
This has been talked about quite a bit on this mailing list. Search the
archives.
Why? I don't have a problem I've solved it in my case. But my solution
will be of no use whatsoever for most others. I don't need to register
I shall assume, then, from the lack of response any of the four things:
A) I'm doing this correctly
B) No one knows for certain
C) No one does E164 caller IDs
D) No one read this.
;)
I'm hoping this is right. I looked about and nothing I found seemed to either
confirm or deny that this was
If you downgrade, let us know if it fixes things for you.
It's strange that there were so many changes in the 1.2 SVN branch
after 1.2.7.1 that seem to be complete changes in how some things
operate(like the transcoding optimization mess for Asterisk 1.2.11 and
1.2.12 that was fixed in
Hello !
I 've some questions how bridging of ISDN calls is done.
Assume an asterisk system with a TE405 card equipped.
(PRI1 - PRI4)
An incoming ISDN call on PRI1 is transfered back to
PRI3. Unless there is DTMF detection or other things
involved, the bridging is done without Asterisk. Does
this
On 10 Oct 2006, at 22:33, Barry D. Hassler wrote:
I was playing around with that idea myself, but I can't find a way
to place the call which will actually play the recording. What I'd
like to accomplish is that somewhere in a conference call, I'd like
to be able to say let me play this
On 11 Oct 2006, at 07:44, Martin Joseph wrote:
On 2006-10-10 18:12:23 -0700, hugolivude [EMAIL PROTECTED] said:
An Internet browser uses port 80. I might have two or more behind a
NAT both using port 80. Isn't that the same thing?
Remember that the browser INITIATES all activity on the
Dave Cotton wrote:
On Thu, 2006-10-12 at 11:56 -0500, Eric ManxPower Wieling wrote:
This has been talked about quite a bit on this mailing list. Search the
archives.
Why? I don't have a problem I've solved it in my case. But my solution
will be of no use whatsoever for most others. I don't
On Thu, 2006-10-12 at 13:33 -0400, Matt Florell wrote:
We have seen more random crashing on 1.2.12.1 as well as compared to 1.2.7.1
It's not that bad, we only have one crash per week out of 6 servers
that are on 1.2.12.1, but it is much more than when we were running on
1.2.7.1.
Even once
I set up facilityenable=yes in zapata.conf, but it still didn't work for caller name.
Searched google and found out that when Asterisk is configured as switchtype National with signalling pri_net, it does not send the Display information in the Facility message. Asterisk insead puts the Caller
sip wrote:
I shall assume, then, from the lack of response any of the four things:
A) I'm doing this correctly
B) No one knows for certain
C) No one does E164 caller IDs
D) No one read this.
I would fall into the categories of (B) (C) and (E)
E} I don't know what E164 Caller IDs are.
12 okt 2006 kl. 03.36 skrev Andrew Joakimsen:
What are your T.38 plans with this?
That's top secret... :-)
The T38 will be handled the same way as today - in passthrough mode -
until we have more T38 implementation code within the core. That's a bit
outside of the SIP scope.
/O :-)
12 okt 2006 kl. 15.51 skrev Jay R. Ashworth:
On Wed, Oct 11, 2006 at 09:00:20AM +0200, Olle E Johansson wrote:
The new channel will have configurations for trunks, services and
phones. It will
Does that mean that it will make a distinction concerning the
difference in administrative span of
I thought I would list my issues so all of you that know more than me
might be able to help.
1. I have 6 Swissphone ip10 they disconnect calls at either 70 seconds,
120 seconds or 180 seconds I have polycom Phones that go forever
2. When I try and transfer calls I have a LONG delay before the
Thursday, October 12, 2006, 6:58:57 PM, Tim wrote:
I've read alot of comments on the SPA-3000, many if not all saying they had
echo
issues, but I've not seen anyone comment on the SPA-3102. Does anyone have any
comments or issues with these?
Well, I have had echo issues. Then I find out
On 12 Oct 2006, at 17:16, Douglas Garstang wrote:
-Original Message-
From: Dovid B [mailto:[EMAIL PROTECTED]
Sent: Thursday, October 12, 2006 8:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How big is *your* dialplan??
I worked for
Assuming that you are in the US (don't know about elsewhere), the Calling Party Name is obtained by a database lookup performed by the PSTN switch terminating the call. Calling Party Name is the name assigned to your line or location by the phone company. You can't "send" it. You can only receive
On Oct 12, 2006, at 2:30 AM, Jay R. Ashworth wrote:
On Wed, Oct 11, 2006 at 05:08:32PM -0500, Lacy Moore - Aspendora
wrote:
As a carrier, I would expect you to have an abundance of
redundancy, but not an SMB. SMB's don't have the money to cover
everything. That's what cellphones are
Hi list,i noticed from the cli my asterisk box is accepting unauthenticated calls how can i prevent this?CLI:-- Accepting UNAUTHENTICATED call from 192.168.0.2:
requested format = gsm, requested prefs = (), actual format = ulaw, host prefs = (g729|ulaw|alaw), priority = mine
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Tim Panton
Sent: Thursday, 12 October 2006 3:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How big is *your* dialplan??
How long do
On Thu, 2006-10-12 at 21:30 +0200, Csibra Gergo wrote:
Thursday, October 12, 2006, 6:58:57 PM, Tim wrote:
I've read alot of comments on the SPA-3000, many if not all saying they had
echo
issues, but I've not seen anyone comment on the SPA-3102. Does anyone have
any
comments or
Isnt that an internal ip address?
Cheers,
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Quitoriano
Sent: Thursday, 12 October 2006
4:04 PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users]
The way that I've done it is to set the context= line under [general] in sip.conf to a context that just gives the congestion command and hangs up the call, something like this:exten = s,1,Answerexten = s,n,Wait(2)
exten = s,n,Congestionexten = s,n,HangupI suppose you could really just use Hangup
The problem with the extra ptime descriptions in the SDP has been fixed in
Asterisk (see
http://lists.digium.com/pipermail/svn-commits/2006-October/017694.html). I've
got the latest version of the 1.4 branch from SVN and have verified that the
codec negotiation is working again.
If you don't
How about some hardware/sofware info guys? What hardware and OS versions
are you guys having these problems on? Are there any combinations out there
that are stable?
-Original Message-
From: Joseph [mailto:[EMAIL PROTECTED]
Sent: Thursday, October 12, 2006 11:35 AM
To: Asterisk Users
Can anyone share sysctl tuning params for asterisk and unix ?trying to see if we have differences in them -- MikeSales Managerhttp://www.theclubvoip.com
Making it happen1.877.807.VOIP (8647)
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yes.. actualy use 1 did for each proxy to check..then inbound for each use the method he described..On 10/12/06, Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:
on an analog Zap PSTN channel, you have no real way of determining ifthe remote side answered, because, as you discerned, it IS
Matt Florell wrote:
If you downgrade, let us know if it fixes things for you.
It's strange that there were so many changes in the 1.2 SVN branch
after 1.2.7.1 that seem to be complete changes in how some things
operate(like the transcoding optimization mess for Asterisk 1.2.11 and
1.2.12 that
Zeeshan Zakaria wrote:
I set up facilityenable=yes in zapata.conf, but it still didn't work for
caller name.
Searched google and found out that when Asterisk is configured as
switchtype
National with signalling pri_net, it does not send the Display information
in the Facility message.
Dave,
Are you in the US?
On 10/12/06, Dave Cotton [EMAIL PROTECTED] wrote:
On Thu, 2006-10-12 at 21:30 +0200, Csibra Gergo wrote: Thursday, October 12, 2006, 6:58:57 PM, Tim wrote:
I've read alot of comments on the SPA-3000, many if not all saying they had echo issues, but I've not seen anyone
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