You are better off running a small AGI script and calling the Dialplan
functions from there.
You can always start musiconhold, process, and return to dial plan.
Alex
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Rilawich Ango
Sent:
Uhm.
Actually if I write: cat /proc/interrupts
I get:
11: 2997154835 XT-PIC libata, wctdm24xxp
Is this the problem?
How can I solve it?
The output of zztest is:
[EMAIL PROTECTED] freepbx-2.2.0]# zttest
Opened pseudo zap interface, measuring accuracy...
On Mon, 15 Jan 2007, Giuffredi wrote:
Uhm.
Actually if I write: cat /proc/interrupts
I get:
11: 2997154835 XT-PIC libata, wctdm24xxp
Is this the problem?
Potentially yes. The 2400 card is sharing interrupts with the IDE disk
system.
How can I solve it?
Try moving the
Hi,
I've troubles with setting up Asterisk Realtime and MD5 authentication.
With clear text passwords everything is working fine.
-- Registered SIP 'edwin' at 10.0.0.37 port 5060 expires 600
-- Saved useragent Cisco-CP7940G/8.0 for peer edwin
[2007-01-15 10:18:12] DEBUG[28528]:
I agree with C F - We just upgraded to our first non-internal 1.2.x
system last Friday. Mostly I am glad we waited. I imagine we may
upgrade to 1.4 in about a year :)
Really it depends on your customer. If it is a commercial operation I
would be cautious of 1.4 still, and at the very least test
Hi, i want to to this thing with php AGI:
#!/usr/local/bin/php -q
?php
set_time_limit(30);
require('phpagi.php');
error_reporting(E_ALL);
$agi = new AGI();
$agi-answer();
$cid = $agi-parse_callerid();
$agi-text2wav(Hello, {$cid['name']}.);
$agi-text2wav('Enter some numbers and then press
Its not quad band and in my opinion doesn't perform well enough to be
used for anything but basic email and phone calls. This phone, even on
the newest version of firmware (Sprint) hangs when syncing with exchange
to the point where you miss calls even though you tried to answer them.
If you turn
Which command effects whether or not the * server will lookup a
peer from the db even though the phone isn't registered locally?
I have several * servers but I want any server to be able to
lookup and send a call to phones registered on another server (SIP
cluster?).
Thanks
Tim
Hi,
I have not checked this, but I thought the intention was that 'show'
was a human readable formatted output, and 'list' was meant to be the
same data but more easily machine readable.
Of course I could be completely wrong.
Steve
On 1/13/07, Jerry Geis [EMAIL PROTECTED] wrote:
I thought I
Is there an official list anywhere specifying the Prerequisites for
installing asterisk(Specifally 1.4) on Fedora Core 4? I have been
struggling with a configure: error: termcap support not found error
when compiling 1.4 on my brand new install of FC4 fully updated, Fedora
was installed as a base
I will sum up the results of my investigations :
- When canreinvite is set to yes, I manage to make a video call between
the 2 parties, when I emit a DTMF signal, it triggers the playback of a
sound clip correctly, but I can't playback a video clip.
- When canreinvite is set to no, The DTMF I
Hi list.
some info:
zaptel 1.4.0
wanpipe 2.3.4-2
kernel 2.6.19.1
Debian
I'm trying to build wanpipe on my server, but I got a error that it
can't find config.h.. I found a post on an other unrelated mailing
list which stated that includes/linux/config.h has been removed from
2.6.19. It
Thanks for help me, well, I do all that i see on the wiki page about asterisk
and
nat troubleshooting, because did not work I connected asterisk to a public ip
for testing, but, while I get two sip phones with private ip connected to my
asterisk with public ip, I can setup calls(phones rings) but
Hello Yuan,
I have recentky spoken to a number of customers who run call-centers,
tried 1.4 test installs and concluded it's not there yet in terms of
reliability. If I were to install a production box today, I would go for
1.2.
l.
In data Mon, 15 Jan 2007 00:01:27 +0100, Yuan LIU [EMAIL
This is the error i got. I've grepped through all of my include/linux/
wanpipe_includes.h files i have on my server (there is actually a
couple of them), and replaced config.h with autoconf.h, but still i
get the same error. Looks like I'm unable to locate the include/linux/
wanpipe_includes.h
Ok, how can i do the transfer from the caller to $keys ?
Probably by using a goto :
http://phpagi.sourceforge.net/phpagi2/docs/phpAGI/AGI.html#goto
hth
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To
These are the different meanings for the diferrent error codes:
T1 TIMEOUT = 32769
T2 TIMEOUT = 32770
T3 TIMEOUT = 32771
UNEXPECTED MF SIGNAL= 32772
UNEXPECTED CAS = 32773
INVALID STATE = 32774
SET_CAS FAILURE = 32775
SEIZE ACK
Cory Hawkless wrote:
Once all of the prereq’s were installed it compiled fine, its
frustrating I cant find a “This is what you must have installed before
beginning your Asterisk install’
It hasn't changed from Asterisk 1.2; termcap (ncurses or similar) is
pretty much the only mandatory
Hi List.
We have a small issue with making parked calls work with the new
Asterisk 1.4. I have an impression that this used to work with 1.2, so
its either I'm doing something wrong, or a regression. I hope its not
the latter and you can tell me what I'm doing wrong.
The setup is an Asterisk
Cory Hawkless wrote:
Is there an official list anywhere specifying the Prerequisites for
installing asterisk(Specifally 1.4) on Fedora Core 4? I have been
struggling with a “/configure: error: termcap support not found”
/error when compiling 1.4 on my brand new install of FC4 fully
updated,
Hi,
thanks for the help. It turns out the this device I had, an Orion GSM gateway, does not talk MFC/R2,
but some variant of R2, according to Steve U.
thanks anyways :)
Facundo Ameal wrote:
These are the different meanings for the diferrent error codes:
T1 TIMEOUT = 32769
T2
When I send a call from my TE410P using EM, the legacy
PBX answers the call but doesn't route it.
Any suggestions on what config settings to muck with?
Do you have PRI ISDN or inband signaling trunks?
Either way, it would be zapata.conf configs that would be the issue.
All,
I had used 1.4beta3 for some time. I read all the changes etc...
One of the changes was Sip show peers was changed to sip list peers.
I changed my interface to accomidate that...
Over the weekend I installed 1.4.0 release. It seems as though
the sip list peers is GONE and now it is back to
Wondering if anyone on here can help with a niggling issue: One of our
extensions is unable to make attended transfers at all.
The phone in question is an Elmeg ip290, and works fine for direct
transfers. However, on attempting to make an attended transfer, the first
leg succeeds (the inbound
Jerry Geis wrote:
Why did it revert back?
The developer community (with input from a lot of users) decided the
change was not the right thing to do, and it got changed back.
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asterisk-users
On 1/15/07, Tim Connolly [EMAIL PROTECTED] wrote:
Which command effects whether or not the * server will lookup a
peer from the db even though the phone isn't registered locally?
I have several * servers but I want any server to be able to
lookup and send a call to phones
On Fri, Jan 12, 2007 at 02:33:54PM -0500, Doug Crompton wrote:
I am using spa3000 hardware - 2.0.1(5673) firmware - 3.1.3(GWa)
I have used newer firmwares but find that 3.1.3 had less echo problems.
Thanks again Doug for that detailed explanation.
As for the DTMF playback level and DTMF
chester c young wrote:
cannot make Dial(...,,g) work correctly, although Dial(...,,gh) works
just fine. (to make matters worse, it does seem to work sometimes,
although once working or not working between changes it either works or
doesn't work all the time.)
g option to Dial only continues
Thanks Paul.
I think it was nagging because the phpagi code looks to see if there is
already a wav file before creating a new one. Since I had old ones with the
nagging, it didn't create new ones. The problem I am having now is that it
won't play it at all, just beeps.
Thanks!
On 1/12/07,
On Monday 15 January 2007 11:03 am, Eric ManxPower Wieling wrote:
g option to Dial only continues the dialplan if the destination
(called) leg of the call hangs up. It will NOT cause the dialplan to
continue if the source (calling) leg of the call hangs up.
I was going to give him the exact
Hello List,
I am stuck with this problem for several days... anybody can give me a hint
on this??
I know many of you dealt with problems similar to this, how did you address
this??
Thanks in advance!!!
-lars
-- Forwarded message --
From: Lars Knopf [EMAIL PROTECTED]
Date: Jan
On 1/15/07, Lars Knopf [EMAIL PROTECTED] wrote:
Hello List,
I am stuck with this problem for several days... anybody can give me a hint
on this??
I know many of you dealt with problems similar to this, how did you address
this??
Thanks in advance!!!
-lars
-- Forwarded message
I have the following code. When I call the extension, it either ignores the
first Hello there everyone, or says hello and moves on sometime stoping
before it finishes hello. The rest of the text reads fine. Anyone else
have this issue??
Thanks!
hi all,
i have the following .call file:
Channel: IAX2/[EMAIL PROTECTED]/myPOTSline
MaxRetries: 2
RetryTime: 60
WaitTime: 30
#
# Assuming that your local extensions are kept in the
# context called [extensions]
#
Context: default
Extension: 156
Priority: 1
when i drop the .call file into the
with tcpdump i could notice that invites didn't reach my * server.
After Rebooting Lan's Firewall CheckPoint problem solved.
On 1/12/07, Steven [EMAIL PROTECTED] wrote:
Is there a local dialplan on the phone?
Maybe these phones were recently upgraded or reset to factory and lost the
4XXX
Are you creating a temporary wav file? If so, look at that first. If the
wav file is truncated at least you know the problem is related to the
way swift gets invoked and passed the text argument. If the file is okay
you need to look at the way it gets handled afterwards.
blackwater dev wrote:
I
g option to Dial only continues the dialplan if the destination
(called) leg of the call hangs up. It will NOT cause the dialplan to
continue if the source (calling) leg of the call hangs up.
When the calling channel hangs up, Asterisk will send the remaining
leg of the call to exten =
This may be commonly known but I haven't come across it so here goes,
maybe it'll help someone:
I have terrible echo with asterisk 1.2, zaptel 1.2.12, and a TDM400P
with 1 FXS and two FXO modules.
The Mark2 echo canceller with Aggressive turned on was the only setting
that would make it
I have to register asterisk/sip with a sipproxy that does not support
authentication, I do not know how to tell Asterisk not to send authentication
request?
# sip.conf
[general]
insecure=very
permit=207.148.115.10/255.255.255.0
[myproxy]
type=friend
host=217.118.115.10
context=from-sip
Doug,
You are saying that RFC2833 somehow doesn't work if you have the
Asterisk AND at a distinct time (still within the same call), the callee
to see the DTMF, correct ? Would this be in any case ? (meaning, if the
voice path is going via the Asterisk or UA to UA directly ?)
I've my spa3k
Julien Chavanton wrote:
I have to register asterisk/sip with a sipproxy that does not support
authentication, I do not know how to tell Asterisk not to send authentication
request?
SIP clients never request authentication/authorization.
___
We are assigning interfaces directly to our customer service queue
through an application running on each agent's PC using the QueueAdd
Manager API command. No agents are defined in agents.conf.
Does anyone have a solution to pause or remove an interface that doesn't
answer after a defined
I use default values for both of those. The big thing is to call youself.
Use a cell, call a phone on the FXS. Hit a key on the cell and listen
on FXS for DTMF. Make changes, reboot, and repeat. Hearing is believing.
It is so much easier! I think you will find the inband will work.
Doug
On Mon,
try autopause in queues.conf
James Fromm wrote:
We are assigning interfaces directly to our customer service queue
through an application running on each agent's PC using the QueueAdd
Manager API command. No agents are defined in agents.conf.
Does anyone have a solution to pause or remove
I am not sure what you are asking? The problem is that rfc2833 does not
play well with the spa-3000 and Asterisk. I am not sure if it is limited
to just the spa3k. There is a bug causing this that has been documented.
Google spa3000 dtmf bug asterisk for more info. The bottom line is that
you need
On 1/15/07, chester c young [EMAIL PROTECTED] wrote:
g option to Dial only continues the dialplan if the destination
(called) leg of the call hangs up. It will NOT cause the dialplan to
continue if the source (calling) leg of the call hangs up.
When the calling channel hangs up,
Eureka, echo free at last! ahh
I set the rxgain by running my CO's milliwatt test to 14844 from the
original 6688.
I just looped from FXO 3 to the asterisk milliwatt() test on FXO 4, I
just found the txgain was 6686, instead of 14844.
Hello everybody
Anyone know a software for callcenter, with statistics and reports and work
with asterisk?
Regards
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NICE! That did the trick.
Thanks!
Julian Lyndon-Smith wrote:
try autopause in queues.conf
James Fromm wrote:
We are assigning interfaces directly to our customer service queue
through an application running on each agent's PC using the QueueAdd
Manager API command. No agents are defined
Hello,
There are two GPL call center suites that handle inbound and outbound
calling for Asterisk:
VICIDIAL:
http://astguiclient.sf.net/vicidial.html
GnuDialer:
http://www.gnudialer.org
MATT---
On 1/15/07, Carlos Rojas [EMAIL PROTECTED] wrote:
Hello everybody
Anyone know a software for
From:"Yuan LIU" [EMAIL PROTECTED]Another bizarry: If I run the Echo application from the console, I can hear a very long delay (upward to 1,000 ms).I can run the same application from a GrandStream phone (on the same LAN) and hear little delay.What could possibly be wrong?If it were interrupt
Yes, the wav file is fine. For some reason it's just getting cut off.
Whatever I type there seems to get cut off, strange.
On 1/15/07, Paul [EMAIL PROTECTED] wrote:
Are you creating a temporary wav file? If so, look at that first. If the
wav file is truncated at least you know the problem is
Hello all,
we're using asterisk 1.2.12.1 in an Inbound callcenter using the queue
application. If there are many calls in the queue, it sometimes takes up to 30
Seconds before a call is distributed to an agent.
For example there are 10 callers in the queue, an Agent is finishing a call and
Are these guys still around? I cannot get to www.nufone.net or
nufone.com
Thanks,
Wiley
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I have not yet seen this article posted to this list, so I thought many of
us would be interested in having a look at this project sponsored by
O'Reilly:
http://www.oreillynet.com/etel/blog/2007/01/help_create_the_asterisk_cookb.html
It seems they are looking for both problems and
We offer a commecial very detailed reporting solution that is widely
deployed and is available free of charge to small CCs / SOHOs. See
http://queuemetrics.com . Which kind of call center are you going to
implement? inbound / outbound / mixed traffic?
l.
On Mon, 15 Jan 2007 19:36:11
Hi People,
We use the Grandstream GXP-2000 phones, firmware 1.1.1.14, Asterisk
PBX, Slackware Linux 10.2, loaded on a Intel(R) Pentium(R) 4 CPU 2.66GHz
Box... The issues that we are experiencing involves our Telephone
Operator's/Receptionist whom answer multiple incoming calls... As an
I can connect to http://www.nufone.net/ just fine.
Wiley Siler wrote:
Are these guys still around? I cannot get to _www.nufone.net_
file://www.nufone.net or nufone.com
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asterisk-users
Using Asterisk 1.4, on the console 'show application queue' mentions an
option 'i' that should ignore call forward requests from queue members
and do nothing when they are requested. Does this work?
My assumption is that the member whose next according to the queue
strategy should get the call
On Mon, 2007-01-15 at 20:37 +0100, Lenz wrote:
[..]
Hello everybody
Anyone know a software for callcenter, with statistics and reports and
work
with asterisk?
Try MOR from www.kolmisoft.com
Regards,
Regards
--
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24
Looking at the phpagi 2.14 source that I have I see that right after
creating the file it does this:
$ret = $this-stream_file($fname, $escape_digits);
So if the swift-generated wav file sounds right the stream_file is where
the problem lies.
copy the wav file to a file named test.wav and create
I second that, seems to be working fine from here (Toronto/Rogers fiber
connection).
Maybe a lagging DNS or routing issue with your ISP?
Alex
On 1/15/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
I can connect to http://www.nufone.net/ just fine.
Wiley Siler wrote:
Are these guys
On 1/15/07, J. Espinal [EMAIL PROTECTED] wrote:
Hi People,
We use the Grandstream GXP-2000 phones, firmware 1.1.1.14, Asterisk
PBX, Slackware Linux 10.2, loaded on a Intel(R) Pentium(R) 4 CPU 2.66GHzBox...
The issues that we are experiencing involves our Telephone
Operator's/Receptionist
If you would bother to read my post you will see that what I am
wanting to do is not the asterisk directory cmd. I don't want them to
be able to search or anything fancy like that. I want an app that will
go through and say the recorded name for everyone that has a mailbox
one by one. I did
blackwater dev wrote:
I have the following code. When I call the extension, it either
ignores the first Hello there everyone, or says hello and moves on
sometime stoping before it finishes hello. The rest of the text reads
fine. Anyone else have this issue??
Try adding this...
From:"blackwater dev" [EMAIL PROTECTED]
Yes, the wav file is fine. For some reason it's just getting cut off. Whatever I type there seems to get cut off, strange.
With plain dialplan (no AGI), I notice that the first few syllables from Playback() or Background()could be eaten up, sometimes
I just put in a Audiocodes Mediant 1000, which seems to be working well except
for one annoyance. I am using Polycom 501's and 601',s and if I do a
supervised transfer of a PSTN call where I complete the transfer before the 3rd
party has answered, the PSTN party hears dead air until the call
Doug Crompton wrote:
I am not sure what you are asking? The problem is that rfc2833 does not
play well with the spa-3000 and Asterisk. I am not sure if it is limited
to just the spa3k. There is a bug causing this that has been documented.
Google spa3000 dtmf bug asterisk for more info. The
Wiley Siler wrote:
Are these guys still around? I cannot get to _www.nufone.net_
file://www.nufone.net or nufone.com
Not only can I get to their website, but yesterday I called their
customer service and for the first time ever it was actually answered by
a live person.
Steve
[EMAIL PROTECTED] wrote:
Hello all,
we're using asterisk 1.2.12.1 in an Inbound callcenter using the queue application. If there are many calls in the queue, it sometimes takes up to 30 Seconds before a call is distributed to an agent.
For example there are 10 callers in the queue, an Agent
OK... I understand. As I remember I did try other methods like INFO. It
has been awhile. I think INBAND is the only one that worked for me.
Doug
On Mon, 15 Jan 2007, Julio Arruda wrote:
Doug Crompton wrote:
I am not sure what you are asking? The problem is that rfc2833 does not
play well
On 1/15/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
I just put in a Audiocodes Mediant 1000, which seems to be working well
except for one annoyance.
I don't have any experience with an Audiocodes Meidant 1000, but I'll try to
help you
I am using Polycom 501's and 601',s
We have a
Strange. I can get there too now... Must have been DNS problem
Now to figure out where my DID has gone
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Prior
Sent: Monday, January 15, 2007 2:08 PM
To: Asterisk Users Mailing List -
I am trying to connect 2 asterisk servers through OpenVPN, the VPN should
carry 16 channel, however when active channels reached 4 concurrent
channels, the connection became unstable, with a very high latency (around
900ms), the internet bandwidth is 1Mbps on each server, I have upgraded the
Hi Guys:
I'm using Asterisk with Addpac 2620 as gateway, internally I'm using
Grandstream BT200, unfortunately when I called to external phones (PSTN),
and I have to choose some extensions, the Phone don't dial the extensions, I
believe that DTMF relay in ADDPAC is not working well. I'm using
On Monday 15 January 2007 19:22, [EMAIL PROTECTED] wrote:
Hello all,
For example there are 10 callers in the queue, an Agent is finishing a call
and it takes up to 30 seconds before his phone rings again. We're already
set the wrapuptime parameter in queues.conf to 0, for my point of view
an
This is being forwarded to my People who should be banned from using
technology folder.
- Original Message -
From: Colin Anderson
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Sent: Thursday, January 11, 2007 12:10 AM
Subject: RE: [asterisk-users] Directory
Get me a F*ckin human being seems to work well for me with Verizon.
- Original Message -
From: Andrew M Stemen [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, January 12, 2007 4:54 AM
Subject: Re:
I have a problem where my recorded queue calls stop recording once the
call is transferred to a different extension. Is there some additional
parameter I need to set so recording continues? Is it even possible to
do this?
Thanks,
Jay
___
Am Montag, den 15.01.2007, 13:38 -0700 schrieb Andrew Niemantsverdriet:
If you would bother to read my post you will see that what I am
wanting to do is not the asterisk directory cmd. I don't want them to
be able to search or anything fancy like that. I want an app that will
go through and
Thank you, I wasn't aware of the prune command!
-Lars
On 1/15/07, David Thomas [EMAIL PROTECTED] wrote:
On 1/15/07, Lars Knopf [EMAIL PROTECTED] wrote:
Hello List,
I am stuck with this problem for several days... anybody can give me a
hint
on this??
I know many of you dealt with
O.Kamal wrote:
I am trying to connect 2 asterisk servers through OpenVPN, the VPN
should carry 16 channel, however when active channels reached 4
concurrent channels, the connection became unstable, with a very high
latency (around 900ms), the internet bandwidth is 1Mbps on each
server, I
1.2 series ?
I think that 1.4 has that fixed. At least, that's what my team leaders
are telling me ;)
Julian.
Jay Moore wrote:
I have a problem where my recorded queue calls stop recording once the
call is transferred to a different extension. Is there some additional
parameter I need to
On 1/15/07, James Fromm [EMAIL PROTECTED] wrote:
Using Asterisk 1.4, on the console 'show application queue' mentions an
option 'i' that should ignore call forward requests from queue members
and do nothing when they are requested. Does this work?
My assumption is that the member whose next
Silly question: how are the calls going out? If they're going out
through an analog line without the ability to detect hang-ups, then,
that's the problem.
calls are coming in and out thru an Asterisk server using iax2. have
tried two different DID providers and have same problem.
Just as a followup, I'll intimate to the list what went down: (paraphrased)
I'm in Edmonton, the users are in Calgary, so I conferenced in to a Calgary
conference with all of the suits in the big boardroom. I basically let them
argue themselves in circles about how the Directory app should work
On Mon, 15 Jan 2007, O.Kamal wrote:
I am trying to connect 2 asterisk servers through OpenVPN, the VPN should
carry 16 channel, however when active channels reached 4 concurrent
channels, the connection became unstable, with a very high latency (around
900ms), the internet bandwidth is 1Mbps on
Yeah. 1.2.14.
I heard bad things about 1.4 not being all that stable. I'm hesitant to
move to it.
Jay
Julian Lyndon-Smith wrote:
1.2 series ?
I think that 1.4 has that fixed. At least, that's what my team leaders
are telling me ;)
Julian.
Jay Moore wrote:
I have a problem where my
Hi,
I have a TDM21B (A new TDM20B that just arrived plus a X400M that I
already had). This card sometimes works, and sometimes only the FXO
module is seen.
By works, I mean all 3 modules have the green lights on the ports on
and when zaptel is loaded the log shows:
Freshmaker
-Original Message-
From: BJ Weschke [mailto:[EMAIL PROTECTED]
Sent: Monday, January 15, 2007 3:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queue cmd option 'i'
On 1/15/07, James Fromm [EMAIL PROTECTED] wrote:
Using Asterisk
Am Montag, den 15.01.2007, 14:22 -0800 schrieb chester c young:
Silly question: how are the calls going out? If they're going out
through an analog line without the ability to detect hang-ups, then,
that's the problem.
calls are coming in and out thru an Asterisk server using iax2.
Anselm Martin Hoffmeister wrote:
Am Montag, den 15.01.2007, 14:22 -0800 schrieb chester c young:
Silly question: how are the calls going out? If they're going out
through an analog line without the ability to detect hang-ups, then,
that's the problem.
calls are coming in and
I know when you read that subject everyone thinks NAT, but that isn't the
case here. Incoming calls get 2 way audio, but outbound calls do not have
incoming audio. below is the flow
callee -- asterisk -- firewall/router -- provider
Callee is firewalled, but not NAT. callee is on the
I have a TDM21B (A new TDM20B that just arrived plus a X100M that I
already had). This card sometimes works, and sometimes only the FXO
module is seen.
(correction to previous message that incorrectly listed X400M/S400M when
I'm using only the single port versions of these modules
--- Paul [EMAIL PROTECTED] wrote:
Anselm Martin Hoffmeister wrote:
Curious - is this still a $50 thread?
yes.
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Colleagues,
We're in the process of standardizing on Sprint PCS and Cingular phones on a
national basis (~ 50 properties, 1000's of lines). I manage an Asterisk
install at one location.
I've been looking at the Multitech CellFinder CDMA for Sprint as a dial
backup solution. Basically, it's a
Correction, that's Multitech CALLFinder CDMA, not CellFinder. Sorry for the
misquote.
EKG
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Germann
Sent: Monday, January 15, 2007 8:22 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users]
Thanks all, I'll give this a shot.
On 1/15/07, Yuan LIU [EMAIL PROTECTED] wrote:
From: *blackwater dev [EMAIL PROTECTED]
Yes, the wav file is fine. For some reason it's just getting cut off.
Whatever I type there seems to get cut off, strange.
With plain dialplan (no AGI), I notice that
Why not use DBGet / DBPut? I use it for Caller ID and I have over 50K
entries in the DB, and there is no appreciable load hitting the DB in the
dialplan. And my one install (admittedly modest) hits the DB a few thousand
times a day, with up to 46 concurrent calls.
-Original Message-
Eric Germann wrote:
Colleagues,
We're in the process of standardizing on Sprint PCS and Cingular phones on a
national basis (~ 50 properties, 1000's of lines). I manage an Asterisk
install at one location.
I've been looking at the Multitech CellFinder CDMA for Sprint as a dial backup
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