Hi,
I am getting an error trying to compile the asterisk addons:
cdr_addon_mysql.c:24:22: asterisk.h: No such file or directory
make: *** [cdr_addon_mysql.o] Error 1
Can anyone suggest something I could try?
Thanks.
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From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Eric
Sent: Wednesday, March 23, 2005 1:48 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Problem compiling asterisk-addons
Hi,
I am getting an error trying to compile the asterisk addons
to memory or resource allocation.. Anyone know if there was any
work done to ast_channel_alloc() or related functions?
Thanks.
- Eric
Jan 7 07:24:50 WARNING[163850]: Unable to allocate channel structure
Jan 7 07:24:50 WARNING[163850]: Unable to start PBX on channel 0/11, span 1
Jan 7 07:24:50
being held until a moderator can view it.
Fine.
So now I get an autoresponder from the moderator telling me he's on
vacation until someone near the end of the month.
Seriously, what gives. Can we make some changes here? I'd like to
post my findings and get some help.
- Eric
Um, that's about normal here. It runs like 16 threads on a fresh startup.
Maybe you don't have threading enabled on your box?
- Eric
On Fri, 7 Jan 2005 10:04:59 -0600
Matthew Boehm [EMAIL PROTECTED] wrote:
Holy cow! Why are there so many asterisk instances running? There should
only be 1
Hi,
I recently upgraded from stable v1.0.2 to v1.0.5.
I'm seeing a bug when retransmitting my invite for proxy authentication.
Essentially, when I retransmit the request with the proxy auth, the From
number becomes the To number in the SIP message.
Here is an example (with ambiguous numbers):
functions. I can't say if that could cause performance issues
under higher load.
I'd love to hear how you make out, as well as anyone else's input.
- Eric
On Mon, 07 Mar 2005 15:05:32 -0500
[EMAIL PROTECTED] wrote:
Hello Folks,
Has anyone had production experience using * w/ MySQL Blobs
Try 0x00140014
the first 0014 applies to one line, the second to the other line.
On Thu, 2003-10-16 at 18:47, Juan J. Sierralta P. wrote:
On Thu, 2003-10-16 at 19:43, CW_ASN - Gus wrote:
Also: Which codecs are you using?
AudioModes: 0x00150014
And the codecs tested are
No! s is executed when Asterisk has no destination extension. For
example when a call comes in from the PSTN Asterisk doesn't know what
extension to send the call to, so it sends it to the s extension.
On Sun, 2003-10-19 at 11:23, rnc Info Lists wrote:
I have my sip phones going into the
Why use this rather than STREAM FILE?
On Sat, 2003-10-18 at 16:50, Paul Crick wrote:
** REPOST: A week later and no feedback - am I the only one
** who'd find this functionality useful? No other AGI stuff
** out there needing something similar?
I'd like some feedback on potentially
My asterisk process produced the following errors this morning:
Dec 8 10:44:07 WARNING[50315282]: rtp.c:829 ast_rtp_new_with_bindaddr: Unable
to allocate socket: Too many open files
Dec 8 10:44:07 WARNING[50315282]: chan_sip.c:2352 sip_alloc: Unable to create
RTP session: Too many open files
that I hit my limit of open files on this
machine. Restarting asterisk immediately solved the problem, so
I'm leaning towards a leak, however, I didn't have the opportunity,
in the moment, to check and see how many files and what type were
open.
- Eric
On Wed, 08 Dec 2004 16:48:19 +
Sean
)
++ ++ ++
| gs | - | sip0 | - | sip1 |
++ ++ ++
Nothing is allowed to reinvite in this scenerio, the call path is
exactly as you see above.
Has anyone run into a problem like this with a similar setup?
- Eric
Has there been an update on this issue yet?
Thanks
--- [EMAIL PROTECTED] wrote:
Ok Thank you very much to all people !!
I will wait for the patch, and perhaps in the
meantime I could try to
introduce the agi workaround
suggested by Jeroen, when it will be available.
Andrea
*blush* yes!
whois wes said:
your timeout is set to 1/2 a second (500 milliseconds).
change timeout to 5000
On 8/22/06, Eric [EMAIL PROTECTED] wrote:
I am prototyping with the Manager interface and pasting the
following into the telnet session.
Action: Originate
Channel: SIP
How would I set up a call between two extensions which are both pstn
phones (and not peer devices)?
Thanks
Eric
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I want to be able to playback a certain soundfile for
all parties in a call to hear.
How would I do that?
Eric
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I would like to initiate a call in asterisk (say with cron)
so that this call rings on the destination number _and_
on an asterisk extension.
How would I achieve this?
thanks
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this could be happening?
Thanks a lot.
Also, if IAX2/4506:[EMAIL PROTECTED] is your
real username and password, change them asap, you just made it available
to 1+ people and the archives ;)
Don't worry it was not real - thanks for the warning.
--
Eric Smith
This problem was solved by changing the preferred codec from
G729A to ulaw.
Eric Smith said:
We are using the following to record conversations.
exten = _1XXX.,1,SetVar(CALLFILENAME=call_to_${EXTEN:1}_dated_${TIMESTAMP})
exten = _1XXX.,2,Monitor(wav,${CALLFILENAME},m)
exten = _1XXX.,3,Dial
Hi
Is it possible to start recording a call during the conversation?
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calls. So I would like to terminate an asterisk call with
say a * and then be returned to dialtone.
How would I define that rule?
Thanks.
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Asterisk
,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
==
What gives?
Thanks
Eric
Adam Dobrin said:
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
Eric wrote:
I would like to initiate a call in asterisk
Thanks for the help.
Ok, it is now authenticating.
But the command:
Channel: SIP/snom
Context: default
Exten: 2412
causes no action and nothing in the logs.
Any idea?
Thanks a lot
Eric
Tony Mountifield said:
In article [EMAIL PROTECTED], Eric [EMAIL PROTECTED] wrote:
Hi
I have
Is this possible to do with the latest asterisk?
thanks
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What is a good way to set up in the dialplan for the case where a
call fails (say due to congestion or whatever) and then asterisk
immediately dials again, with a different trunk or perhaps another
destination number?
Thanks
--
Eric Smith
and the
phone rings once and then shows a missed call.
Below is the response I get from the Manager:
Any ideas what is causing this hangup?
Thanks.
Eric
= Manager response ==
Response: Success
Message: Originate successfully queued
Event: Newchannel
Channel: SIP/dualphone-9524
On March 13, 2005 09:57 am, Nigel Burgess wrote:
[door]
exten = s,1,Dial (SIP31,15)
exten = s,2,Playtones(dtmf)
However the call hangsup before trying to play the DTMF tone.
When a Dial happens, the dialplan stops until the call is
disconnected. See show application dial to see how you can send
.
Every provider that I know of in the USA has an e-mail - text message
feature.
--Eric
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Robert Hajime Lanning wrote:
quote who=C F
Well, as far as I know there is no such service in the USA. Take in
mind that SMS is not so popular in the states, email is, and every
cell phone in the US that I have seen that supports SMS, supports SMS
to email from the phone as well.
um, backwards.
Robert Hajime Lanning wrote:
quote who=Eric Wieling
Robert Hajime Lanning wrote:
um, backwards. E-Mail to SMS. I have not seen the other way
around.
Both Cingular and Verizon supports both.
I have not tried this, nor have I seen any documentation mentioning
it. Do you or anyone else have
Raoul Bönisch wrote:
Hello!
I'd like to Flash() a modem line (BRI) with Asterisk. It is a
passive ISDN-card connected to a hardware PBX. I use ISDN4Linux.
I recognised that unfortunately the Flash() application flashes
Zap devices only. Now I am wondering how I could flash Modem/ttyI0.
The source
.
I don't know of this has been fixed in CVS-HEAD or not.
--Eric
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is pretty useless. nat=yes
combined with qualify=yes should cause enough traffic on the right
ports to keep the NAT translations open on your NAT router.
Now, if ASTERISK is behind NAT it's a whole other set of issues and
fixes, but you don't mention that so I won't cover it.
--Eric
--
Always
Roman Zhovtulya wrote:
Hello,
I wonder if I would have to sacrifice anything if I set NAT=yes for
all sip clients I have, regardless of whether they are behind the NAT or
not.
The idea is to have the setting that works regardless of whether the
user is behind the NAT or not, since I'm not sure
Raoul Bönisch wrote:
* Eric Wieling [EMAIL PROTECTED] [2005-03-14 16:56]:
Raoul Bönisch wrote:
Flash is an analog thing. It does not even apply to ISDN.
So how does the R key on my ISDN-telephone work then?
I suspect it sends an ISDN specific put call on hold or take call
off hold message
Ron Joffe wrote:
Hey folks
I have a new setup with a TDM400P for a pair of analog extensions and a few
SIP phones. We seem to be experiencing a bunch of Crackeling when talking
between the analog and SIP extensions.
Any ideas?
Yes. Check the suggestions given to the other guy that posted this
people on this list would be
familiar with that as would Skype.
--Eric
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[EMAIL PROTECTED] wrote:
Hello,
I upgraded my office from Asterisk 1.0.0 to Asterisk
CVS-HEAD-03/13/05-13:14:04 this weekend, and are now
experiencing some problems accessing voicemail. The web based interface
works fine, in addition to dialing 8500,
which is mapped to:
exten =
/asterisk/features.conf.
--Eric
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unable to navigate through my menus !!!
dtmfmode=inband ONLY works with ulaw and alaw codecs. You want
dtmfmode=rfc2833 if you want DTMF over other codecs.
--Eric
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Mark Twain
] lines at the top of the zaptel.conf.sample? You
need it.
--Eric
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Once you run Dial from an AGI script, you lose control of the call via
the AGI script.
Jean-Michel Hiver wrote:
(obviously if you do other magic in your dialplan this needs to be
adjusted. The important part is the 'g' flag to Dial (go on after
hangup), and the NoOp which echos the
Richard J. Sears wrote:
Hey Everyone,
I am using NuFone for 866 inbound service and I am trying to figure out
the callerid part of it. Any call into my * system just shows Toll Free
Call and will not give me the calling party's caller ID info.
Is this just something I have to live with using
/US/tech/tk652/tk653/technologies_tech_note09186a00800e2560.shtml
Eric
On Thu, 17 Mar 2005 13:17:37 -0600, Scott Nelson [EMAIL PROTECTED] wrote:
I am an Asterisk newby, and I cannot seem to get Caller ID information
from our T1 line. When calls appear at the phones, they say the call
came
, blah) or
play a busy tone or something.
Hope that helps, but unfortunately, I don't have enough experience
with * to troubleshoot this much more.
Eric
On Thu, 17 Mar 2005 11:26:12 -0800, Trevor Peirce [EMAIL PROTECTED] wrote:
Trevor Peirce wrote:
Anyhow, they are seeing the RELEASE
; similarly, it appears that * may not
be providing the in-band message to playback to the calling party when
an extension is out of service or something.
Eric
On Fri, 18 Mar 2005 00:06:24 +0100 (CET), Peter Svensson
[EMAIL PROTECTED] wrote:
On Thu, 17 Mar 2005, Eric Knudson wrote:
Trevor
-cvs mailing list, you would have seen a
fix being added yesterday. See: http://www.lists.digium.com/
--Eric
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John Goerzen wrote:
Over at http://www.voip-info.org/wiki-Asterisk+tips+911, I see these
extensions.conf lines:
exten = s,1,SetVar(SET_EMERG_FLAG=0)
exten = s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
exten = s,n,SetGlobalVar(EMERGENCY=1)
exten = s,n,SetVar(SET_EMERG_FLAG=1)
exten =
Thomas Andrews wrote:
If I comment out the following line in zapata.conf I would expect
asterisk to forget the cli information for that channel when I reload:
callerid=Uniden Dead (256) 428-6125
... but it doesn't; I have to restart asterisk for it to take effect.
The funny thing is that the
Anton Krall wrote:
What do you think?
CPU0
0: 16148159 XT-PIC timer
1: 4 XT-PIC keyboard
2: 0 XT-PIC cascade
5: 0 XT-PIC usb-uhci
8: 1 XT-PIC rtc
10: 161351663 XT-PIC usb-uhci,
Asterisk is not a SIP proxy.
Wei Su wrote:
We encouter a situation where we need to use SIP info to convey infomation
for one end point to another endpoint. I use asterisk to do the test and
find asterisk does not forward the SIP info to another endpoint, but act as
UAS and returns a 4xx error
disconnect cause code and see if you can find something that works.
Don't know how relevant this is, but are you configured for user side
or network side signaling?
Eric
On Fri, 18 Mar 2005 08:28:27 -0800, Trevor Peirce [EMAIL PROTECTED] wrote:
Peter Svensson wrote:
The two issues are only somewhat
C F wrote:
Now consider this (this works with the cisco 7960, even if you put a
7914 with it, it will still use all 20+ plus buttons this way, if CW
is disabled on the phone):
exten = 123,1,Dial(SIP/${EXTEN},30,tr)
exten = 123,2,Voicemail(u${EXTEN})
exten = 123,3,Playback(goodbye)
exten =
likes one of
the newbie problems of using allow=all or bandwidth=low. DON'T DO THAT!
Use disallow=all and then allow= lines for the one or more codecs that
you actually want to use.
Asterisk does not fully support G723.1. fully means transcode.
--Eric
--
Always do right. This will gratify
ChanSpy was NEVER in the official Asterisk CVS nor in
any officially released version of Asterisk.
--Eric
--
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files to get new versions to be noticed
(since there are no timestamps available via TFTP). FTP is much cleaner,
you can just edit files and the phones will notice the changes.
This was fixed in 1.4.1. TFTP and FTP now work the same for deciding
to download the firmware or not.
--Eric
Brian McCrary wrote:
Hello,
I'm wondering if anyone has benn able to successfully get g726-16
passthrouhg to work? I am wanting to use this codec instead of g729 as
I'm running out of DSPs using a high complexity codec on the Ciscos. I
would think it would work just as g729 does, which has been
It means the caller hung up in the middle of the voicemail app.
Anton Krall wrote:
So far, nobody has been able to tell us what this error means.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of MF Hulber
Sent: Lunes, 21 de Marzo de 2005 02:54 a.m.
To:
Alessandra Grasso wrote:
My objective is to estimate the performances of *
How much the trancoded can influence the performances?
Thanks,
show translation recalc 30
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Mark Twain
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Sys Admin wrote:
I am setting up a new asterisk based call center. I just read:
http://www.voip-info.org/wiki-IAX+versus+SIP
After reading this and other google results for IAX vs SIP is there
any reason why i should use SIP anywhere !!
Because most equipment doesn't support IAX
--
Always do
. Locking interrupts
for a long time will mess up Asterisk no matter WHAT you do.
--Eric
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Probably poor headsets or integrated speaker/microphone - do you have
any hard phones?
On Fri, 25 Mar 2005 01:16:50 +0500, Rizwan Chaudhry [EMAIL PROTECTED] wrote:
I have configured Xlite phones with my Asterisk server.The problem is
that i am gettting a terible echo when i call from one
television to mpeg-4 on
my mythtv box at home. Twice, leaving room for scheduled jobs. Has anyone some
references to documentation to put these figures into perspective?
Thanks in advance,
Eric.
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with iax, with
codec translation and *without* zap can rule out zap as part of the cause.
Saddly enough, i still didnt find the time to do any load measurements
on pri cards. Although i have a test setup ready to go.
I can continue testing using your hints, thanks.
Eric.
Zoa
I have a simple 4 person ACD queue using the AgentCallback function. No
matter what strategy I use, anytime someone calls into the queue
asterisk dials the agents in the order that they are listed in the
agents.conf file. This doesn't seem right to me, or am I wrong.
come into play.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Rees
Sent: Tuesday, March 29, 2005 9:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] ACD queue question
I have a simple 4 person ACD queue
/sbin/ztcfg -v
/usr/sbin/asterisk
Eric.
-Original Message-
From: David Masure [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 30, 2005 3:18 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Bristuff and startup scripts
Hi,
I'm not the kind of Linux guru and I was wondering
19:33:42 -0600, Eric Rees [EMAIL PROTECTED]
wrote:
I tried leastrecent. I did change the strategy, but didn't make a
difference.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe
Dennick
Sent: Wednesday, March 30, 2005 6:49 AM
To: 'Asterisk
True. I think Digium's USA bias is clearly demonstrated by their lack
of a BRI ISDN product. Most of the rest of the world use it in
abudnace yet Digium do not see fit to service this market because it
is not big in the US. very poor...
On Thu, 31 Mar 2005 18:32:40 +0900 (JST), Isamar Maia
?
Thanks
Eric
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Default is U-law, but I also switched it to A-law with the exact same
results.
Sean Kennedy wrote:
Eric Mason wrote:
I'm having a problem with my Polycom phones and hoping someone else
has experienced the same thing: Outbound calls are fine, and inbound
calls originating from another SIP phone
speex towards
optimized speex or gsm my spike period goes up from 1 to 10 minutes. If this
increase is related to (decreasing) translator costs, I guess a few hour period
for G711 is quite possible. I guess I should ask the dev-list...
Eric.
-Original Message-
From: Steve Kann
, noise cancellation or voice
detection in your voip client.
Eric.
-Original Message-
From: 1 2 [mailto:[EMAIL PROTECTED]
Sent: Friday, April 01, 2005 3:46 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Maybe an echo cancellation problem?
Hi
Was hoping
allowed to play with Asterisk
- so performance issues are just personal :)
A question about your snake load tests: have you seen any unexplainable
spikes in processor load, or machine hangups every few hours?
Eric.
-Original Message-
From: David [mailto:[EMAIL PROTECTED]
Sent: Friday
. It just sounds very distorted, like a cross between a robot
and Donald Duck.
It really seems to be a problem with the way Asterisk is bridging the
call from IAX to the phone. It does SIP - SIP bridges (not
reinviting) just fine.
Noah Miller wrote:
Hi Eric -
I'm having a problem with my
There's no transcoding going on. It's ulaw on IAX with Sixtel and ulaw
on SIP to the phone. I considered that as a possibility originally, and
even tried using GSM with Sixtel to force it to do transcoding, but had
the exact same problem.
The Asterisk box is a 2.4ghz P4 with 512MB RAM,
You need to upgrade these phones to the latest firmware for it to work
well with asterisk.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Thore
Sent: Sunday, April 03, 2005 3:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
this will shed some light on the issue.
Eric
Noah Miller wrote:
There's no transcoding going on. It's ulaw on IAX with Sixtel and ulaw
on SIP to the phone. I considered that as a possibility originally, and
even tried using GSM with Sixtel to force it to do transcoding, but had
the exact same
Hakem Taourchi wrote:
Hello,
Do you confirm there is a way to send information and update it while
the call is ongoing using the caller Id information ?
I strongly doubt this will work on anything except an analog phone. I
also strongly doubt that Asterisk supports this at all.
--
Always do
Daryll Strauss wrote:
Yep, I've seen it and from reading http://www.voxilla.com it's a
pretty common problem.
If you turn on debugging what you'll see is that the Sipura has
mistakenly detected a DTMF code in the audio stream and is relaying it
by repeating the signal (very loudly I might add)
So
to register to receive calls from your DID (if you have one)
--Eric
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Matt Loretitsch wrote:
Looking for some help any way I can. I've been closely following
digium's troubleshooting steps and seem to be okay there. I am
connecting, via PRI, to a Definity system. When I release the board on
the Definity side I get this in Asterisk:
*CLI Apr 7 10:17:23
Matt wrote:
I have a STUN server running on my Asterisk box which seems to work
for most of my SIP clients.. but some of them seem to require NAT=yes
turned on. If I go further and turn QUALIFY=yes to on, is there a
reason I need to keep running a STUN server? If so, what's the
difference?
I
I have asterisk installed on a Dell 2850 dual-Xeon 3.0Ghz box with 2GB
of memory. This is serving about 75 sip clients, Polycom500's and
600's. We are running into problems with the memory. Asterisk, right
now, is using about 1.8GB of system memory. I am using Asterisk 1.0.7,
Zaptel 1.0.7 with
Andy Hamilton wrote:
I imagine that you are using SIP, which has numerous issures with NAT.
Consider using IAX2; one of it's benefits is working with NAT, which I
gather is your problem.
Or he could just read the Wiki and the mailing list archives to see
the simple fixes for a lot of NAT related
snacktime wrote:
So far it seems that the major thing affecting voice quality on my *
box is codec translation. How much cpu is required to translate even
a single channel without getting static like sounds or other obvious
translation issues? I know this probably depends on the codecs
Ronald Wiplinger wrote:
Any idea?
-- SIP Seeding peers from Astdb: '3366' at
[EMAIL PROTECTED]:64440 for 3600
-- Saved useragent Sipcom/ATA2000-1.6.11 for peer 3366
-- SIP Seeding peers from Astdb: '886229421761' at
[EMAIL PROTECTED]:5060 for 3600
-- Saved useragent
Requirements
On Fri, Apr 08, 2005 at 07:01:08PM -0500, Eric Rees wrote:
I have asterisk installed on a Dell 2850 dual-Xeon 3.0Ghz box with 2GB
of memory. This is serving about 75 sip clients, Polycom500's and
600's. We are running into problems with the memory. Asterisk, right
now, is using
Ugur GUNCER wrote:
How can play music when is clients phone ringing in dial progress.
Usually you read the documentation.
At the Asterisk CLI do a show applications to show you what Asterisk
apps are available. Also see musiconhold.conf.sample in the Asterisk
source directory (in the configs
Drew Einhorn wrote:
The ATA generates it's own dialtone, and waits for
the user to dial a number, before sending anything
to the * box. So one of the first examples in the
in the Brief Introduction to Dialplans from
Vol. 1 of the Asterisk Documentation Project.
[incoming]
exten =
izo wrote:
I just checked digium's site. Looks like next big thing is coming to town
DS3 on single card. Would be nice to know how many channels it can handle.
Anybody had his hands on this card or knows some details ?
Please God, if you can hear me, don't let them use a TigerJet chipet.
--
Jim Sturtevant wrote:
I was hoping someone might help me diagnose a NAT issue with an SPA-2000 and
my * server.
My SPA is behind a NAT accessing a server which is also behind a NAT but SIP
and RTP ports are forwarded to it.
My SPA can successfully register. It can call another extension which
Jim Sturtevant wrote:
Thank you for your reply. There is a wealth of information on the wiki,
etc. I turned on RTP debug and the SPA is not sending it's public IP it is
sending it's NAT IP (192.168.1.100) so *'s RTP packets are going nowhere...
nat=yes makes Asterisk use the public IP that is
of the USA and want to relocate to Europe.
Eric Wieling
[EMAIL PROTECTED]
--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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Brian McSpadden wrote:
On Apr 9, 2005 5:03 PM, Bellows, Jared [EMAIL PROTECTED] wrote:
I'm looking into the TelIAX pay-as-you-go plan. I'm assuming that they charge incoming calls minutes as well? Is there the $0.02 connection fee for the incoming call as well?
That's the only thing they do
Andrew Kohlsmith wrote:
On April 9, 2005 02:13 pm, Eric Wieling wrote:
izo wrote:
I just checked digium's site. Looks like next big thing is coming to town
DS3 on single card. Would be nice to know how many channels it can
handle. Anybody had his hands on this card or knows some details ?
Please
Andrew Kohlsmith wrote:
On April 9, 2005 08:25 pm, Eric Wieling wrote:
Which specific Digium card does not use the TigerJet chip (as shown in
lspci)?
TE405P:
05:03.0 Communication controller: Xilinx Corporation: Unknown device 0314 (rev
01)
I imagine the TE410 and TE110 are both also similarly
snacktime wrote:
On Apr 10, 2005 5:28 PM, Paul [EMAIL PROTECTED] wrote:
I have a Sipura SPA-841. I have configured my SIP.conf and extensions.conf
to allow the phone to dial an outside number via my Zap/2 PSTN. When I pick
up the handset I get a dialtone, however, when I press 9, the dialtone
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