Matthew J. Roth wrote:
List users,
This post contains the benchmarks for Asterisk at high call volumes on a
4 CPU, dual-core (8 cores total) server. It's a continuation of the
posts titled Scaling Asterisk: Dual-Core CPUs not yielding gains at
high call volumes. They contain a fair amount
Philipp Kempgen wrote:
Wow. My message made it to the list after more than 3 hours.
Philipp
I noticed similar delays, no wonder we get a lot of 'me too'-s to the
list (sorry list for my bitching).
--
Remco Post
I didn't write all this code, and I can't even pretend that all of it
Hi,
do I have a chance to use iax trunking on OpenBSD where there is no zaptel
driver or ztdummy available? Do I can use sth. else as timing source?
kind regards
Sebastian
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asterisk-users
Hi,
I don't think Asterisk can do this yet but I was
wondering if one could start and stop voice recording
on demand (but via voice commands).
Our situation: some users would need to initiate a
call to a special extension/context dedicated to voice
recording. This can be done easily. However,
Noah Miller wrote:
Hi Marco -
We bought some Linsys WRTP54G-NA boxes which have WIFI, 4-port, 2 SIPs...
The two SIP ports work on A* if you call one line to talk to the other in
the same box.
When we pick up a line, dial to another phone via the A* server, this
will
ring at the other
you can use FreeBSD 6.1 its working fine for me with ztdummy and I'm able to
use IAX2 trunk.
On 6/7/07, Sebastian Reitenbach [EMAIL PROTECTED] wrote:
Hi,
do I have a chance to use iax trunking on OpenBSD where there is no zaptel
driver or ztdummy available? Do I can use sth. else as timing
i need to catch the call hold event from my asterisk-java program. Im using
net.sf.asterisk.*; for communicating with asterisk server. I need to get the
call hold status on my java program . I can able to get the music on hold
status but i cannot able to get the call hold status.
The
On 07/06/07, Stephen Bosch [EMAIL PROTECTED] wrote:
Gavin Henry wrote:
Dear all,
We seem to be getting phantom calls when a inbound caller via the
legacy pbx hangups before
the SIP handsets have answered. The extensions also seem to hear
ringing on the lines too sometimes.
We are currently connecting to TeliaSonera in Denmark, and they said it should
be supported via PRI supplementary services.
I think their platform is Ericsson.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Fredrickson
Sent: 6. juni 2007 22:02
On 6/6/07, John Novack [EMAIL PROTECTED] wrote:
Henry Cobb wrote:
Why would anybody plug a telephone line into an X100P clone?
???
What else would one plug into it?
We just use them as clock cards for MeetMe and trunking.
-HJC
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On Wed, 6 Jun 2007, Mike Lynchfield wrote:
yes on home pbx i love the s/CALLERID..
maybe you should
f($[${CALLERID(number)} = 15552221313]?15:5)
try to isolate string to strings.
this is not good i think
you need qhotes on the callerid part too if you evaluate to the 1555xxx
Am Donnerstag, den 07.06.2007, 01:15 +0200 schrieb Patrick Zwahlen:
Hi everyone,
How do you send multiline SMSs using smsq or .call files ?
smsq --motx-channel=mISDN/g:bri/ 078 line1 line2
How can I have a carriage return between line1 and line2 ? I have tried
the regular \n and
On Wed, 6 Jun 2007, Davis Sylvester III wrote:
We are evaluating starting a small VoIP consumer based platform.
What is the best codec to use with customers using primarily DSL as internet
connectivity?
I know that g729 is the king-all, but I want to know what the rest of
the professional
Hi all,
i want to save voicemails in mp3 format. Asterisk does not support mp3
format. so is there any other way to do that, or is there a cpatch for doing
that. I am using Asterisk 1.4.2
--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
___
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Hi,
I'm currently migrating to 1.4 and have problems changing deprecated
AgentCallbackLogin to AddQueueMember.
I have dynamic queues and dynamic agents (MySQL Realtime), and
pseudo-dynamic agents.conf (with huge amount of possible agent
numbers).
Agent login is done trough manager API:
*
See below:
Atis wrote:
Hi,
I'm currently migrating to 1.4 and have problems changing deprecated
AgentCallbackLogin to AddQueueMember.
I have dynamic queues and dynamic agents (MySQL Realtime), and
pseudo-dynamic agents.conf (with huge amount of possible agent
numbers).
Agent login is done
On Wed, Jun 06, 2007 at 05:27:50PM -0400, John Novack wrote:
Henry Cobb wrote:
On 6/5/07, Jared Smith [EMAIL PROTECTED] wrote:
Most of the clone cards don't support far-end disconnect supervision,
so you'll have problems where Asterisk can't tell that the other party
has hung up the
On 6/7/07, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
Atis wrote:
I'm currently migrating to 1.4 and have problems changing deprecated
AgentCallbackLogin to AddQueueMember.
I have dynamic queues and dynamic agents (MySQL Realtime), and
pseudo-dynamic agents.conf (with huge amount of
Please post the relevant portions of your sip.conf and extensions.conf
I'll bet dollars to donuts you have the same context defined as both
your regcontext and as a context in extensions.conf (or an .ael, or
whatever).
- Brad
-Original Message-
From: [EMAIL PROTECTED]
Hi,
I cannot get attended working on my Asterisk 1.2.9.1 during an inbound
call via an ISDN card to a Snom SIP phone.
The called party is not able to transfer even if :
1 - atxfer is enabled (set to *7) in in features.conf
2 - the dial option is set to value 't'
3 - I see * and then 7 on
Hi List;
To compile the zaptel and libpri, do I have to have an
diguim card (hardware) fixed in the server?
Also, is there any problem if I compiled first
asterisk and then I tried to compile zaptel and
libpri?
Regards
Bilal
On Thu, Jun 07, 2007 at 04:29:42AM -0700, bilal ghayyad wrote:
Hi List;
To compile the zaptel and libpri, do I have to have an
diguim card (hardware) fixed in the server?
No. I build zaptel, libppri and Asterisk packages on a system with no
Digium (or similar) card...
Also note that if you
Hello,
i have a small setup which requries that agents should be added dynamically,
means their usernames and passwords using a database (MySql).
can anybody have idea please give me a hint
thanks in advance
--
Srinivas Antarvedi
___
--Bandwidth and
Hi,
I need to build Text entry application by using asterisk. I already tried
this with spandsp application along with app_dtmftotext.c file, it was not
working because of some version problem.
Is there any way of building the text entry application through touch pad.
Regards
K.Rajesh.
Thanks a million, that works like a charm.
BR, - Patrick -
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anselm
Martin Hoffmeister
Sent: jeudi, 7. juin 2007 10:25
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
Hi,
I need to build Text entry application by using asterisk. I already tried
this with spandsp application along with app_dtmftotext.c file, it was not
working because of some version problem.
Is there any way of building the text entry application through touch pad.
Regards
K.Rajesh.
I have run the ./configure (actually for zaptel it is
menuselect/configure)... There is no error message, the make just hangs
with the last message to console config.status: creating
./config.status. The last item in the config.log is configure: exit
0.
-Original Message-
From: [EMAIL
In the logs, does that phone try to re-register itself, or does it just
give up?
If its not trying to re-register, you might want to look at the
Expires, Register and Retry settings in the phone.
Rob
Laurent CARON wrote:
Hi,
One of my users is in trouble with his polycom phone hooked to an
On Thu, Jun 07, 2007 at 08:45:59AM -0500, Malcom Kemp wrote:
I have run the ./configure (actually for zaptel it is
menuselect/configure)...
No, it isn't.
it is ./configure in the toplevel directory.
Just run:
./configure
make
From the toplevel zaptel directory.
--
Tzafrir
Tobias Wolf wrote:
Hi,
we are searching for wireless IP Phones (DECT preferred) with have an
solution for an external telephone book. We don't want to enter all of
our numbers into every telephone, but have one location for all the
numbers and every phone looks them up there, e.g. an ldap
Rob Schall wrote:
In the logs, does that phone try to re-register itself, or does it just
give up?
If its not trying to re-register, you might want to look at the
Expires, Register and Retry settings in the phone.
Here is the config snippet:
server voIpProt.server.1.address=
Thank you, that does seem to do it.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Thursday, June 07, 2007 9:08 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] zaptel make problem
On Thu, Jun 07, 2007 at
Jon,
Just to be clear - CFU, CFB, CFNR, CD, and ECT (e.g., TBCT, 2BCT...) are all
ISDN PRI Supplementary Services.
John Treble
Ottawa, Canada
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jon Schøpzinsky
Sent: June 7, 2007 4:09
Matthew,
I'm not sure what you mean when you say, [u]nfortunately though, none of
the switch types support this variant of this function. Could you
elaborate please. TIA.
John Treble
Ottawa, Canada
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL
Stephen Bosch wrote:
John Novack wrote:
I don't know what configuration changes I would make to solve this. My gut
feeling is that it's an electrical problem somewhere within the system, but
perhaps I'm reaching for that too soon; in any case, I wouldn't know where to
start even if we
Rob Schall wrote:
In the logs, does that phone try to re-register itself, or does it just
give up?
The phone is giving up.
Jun 7 14:29:36 NOTICE[22015] chan_sip.c: Auto-congesting
SIP/XXYYZZAA24-08553940
Laurent
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sathish s wrote:
i need to catch the call hold event from my asterisk-java program. Im
using net.sf.asterisk.*; for communicating with asterisk server. I need
to get the call hold status on my java program . I can able to get the
music on hold status but i cannot able to get the call hold
Brad,
I can't post the entire contents of sip.conf and
extensions.conf/extensions.ael, but as you can see below, I don't have a
sip_autoreg defined anywhere in my dial plan.
[EMAIL PROTECTED] asterisk]# cat sip.conf
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=xxx.yyy.34.201
On 6/6/07, Steve Totaro [EMAIL PROTECTED] wrote:
When I fire up asterisk, I keep getting Primary D-Channel on span 1 up
repeated over and over. The B channels never come up. There are no
errors in any of the logs, zttool, or the wanpipe tools.
This is the number one problem I've had with
Any one knows how to make Meet Me video conferencing room.
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf
of Xplorium with another party by e-mail without express written confirmation
by an
I don't know if this is possible, and I can't quite get my head around
how to do it...
If I am using DUNDi for redundancy in a cluster, when Phone1 makes a
call to Phone2, both Asterisk A and B will be in the RTP stream:
+---+ +---+
| A |-| B |
/+---+ +---+\
On Thu, 2007-06-07 at 11:02 +0530, ram wrote:
is this possible ?
You can only do it with realtime static.
how can i do that, any document URL to achieve that
ram
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Static
--
Oddly enough the call was being recorded. In any case in case anyone is
having the same problem, here is what did to get rid of the errors. I am
now using Monitor instead of MixMonitor as Jaswinder suggested.
Thanks
exten =
On 6/7/07, Carlos Chavez [EMAIL PROTECTED] wrote:
On Thu, 2007-06-07 at 11:02 +0530, ram wrote:
is this possible ?
You can only do it with realtime static.
how can i do that, any document URL to achieve that
ram
Hi
I have read that, but i
Auto-congesting is a inconclusive error message from what I've found.
In this case, it probably means, I haven't heard from that phone or the
response ping failed, etc.
So in this case, I'd say you're right, the phone probably is giving up,
and isn't trying to re-enter into the sip peers.
Did
On Jun 7, 2007, at 9:43 AM, John Treble wrote:
Matthew,
I'm not sure what you mean when you say, [u]nfortunately though, none
of
the switch types support this variant of this function. Could you
elaborate please. TIA.
libpri does not support CFU, CFB, and so on. The DMS100 variant of
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
John Treble
Sent: Thursday, June 07, 2007 10:44 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Cc: [EMAIL PROTECTED]
Subject: RE: [asterisk-users] PRI Partial Re-Rounting
So it is, I was wrong. What do you get when you do a 'show dialplan
sip_autoreg'? Does it show pbx_config or anything like that, or does it
say SIP?
In theory at least (though I'd have to peek at the code again to refresh
my memory), contexts that aren't created by pbx_config should not get
Hi, John:
This feedback is brilliant. Thanks. My comments follow.
John Novack wrote:
In your case, from listening to the recording, it really seems as if it
is being generated within the card. To be sure, if you haven't already,
connect to the FXS port directly from a telephone with one of
On a zaptel TE410p, when a call is bridged PRI - PRI how much involvement does
the processor have?
We're now seeing chunks of missing audio and I can't tell whether this is due
to a kernel upgrade or to a zaptel/libpri/asterisk upgrade.
I'm not seeing missed interrupts (from a cat of the
Does anyone know how the Linksys PAP2T ATA's can be mass provisioned?
Documentation seems to be sketchy, even on the Linksys web site.
Thanks,
Doug.
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To
Stephen Bosch wrote:
Hi, John:
This feedback is brilliant. Thanks. My comments follow.
You are most welcome
Someone at Sangoma has this problem before and thinks it's the
mainboard. They want us to try putting the card into another server to see if
it will do the same thing.
You will
On Thu, 2007-06-07 at 21:41 +0530, ram wrote:
I have read that, but i dont see any examples that
give me solution for meetme.
can you just give me some examples
I think the example shown on that page, even though it is for
extensions.conf, is very clear. Just put the
I have made a little progress with this problem today, but am still looking
for suggestions as to what could be wrong:
Rebooting the phone (by the keypad, or by removing power) will
not cause it to re-register, nor will stopping asterisk and
restarting it.
If the phone that refuses to
On 5 Jun 2007, at 18:06, Arun Kumar wrote:
3. Can you post some of the CLI errors you mentioned?
iax2_trunk_queue: Maximum data space exceeded
and once this start it never gets stopped so I've to kill the
asterisk and restart the whole box. Instead of restart whole box if
I just try to
On Jun 7, 2007, at 11:30 AM, Steve Hanselman wrote:
On a zaptel TE410p, when a call is bridged PRI - PRI how much
involvement does the processor have?
We're now seeing chunks of missing audio and I can't tell whether this
is due to a kernel upgrade or to a zaptel/libpri/asterisk upgrade.
On 6/7/07, Douglas Garstang [EMAIL PROTECTED] wrote:
If I am using DUNDi for redundancy in a cluster, when Phone1 makes a call to
Phone2, both Asterisk A and B will be in the RTP stream:
Correct so far... although once the call is made, it's no longer a
DUNDi question, and is simply a
At 11:44 6/7/2007, Douglas Garstang, wrote:
Content-class: urn:content-classes:message
Content-Type: multipart/alternative;
boundary=_=_NextPart_001_01C7A923.2703ACD7
Does anyone know how the Linksys PAP2T ATA's can be mass
provisioned? Documentation seems to be sketchy, even on
How do you get PAP2T-NA's? They aren't even on Linksys's web site.
-Original Message-
From: Doug [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 07, 2007 10:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Douglas Garstang
Subject: Re: [asterisk-users] Provisioning
On 5 Jun 2007, at 22:01, Adrian Marsh wrote:
Yeah I've heard the same breaks in conversations myself. It simply
goes
silent for a few seconds - making both parties say the usual sorry..
Missed that can you say again?...
Connection quality via remote SIP (outside our network via
internet)
On Thu, 7 Jun 2007, Douglas Garstang wrote:
Does anyone know how the Linksys PAP2T ATA's can be mass provisioned?
Documentation seems to be sketchy, even on the Linksys web site.
If it's like the pap2, you can use tftp and xml. This should get you
started.
/tftpboot/spa000F66A83C90.xml:
Yes from Brazil...
On 6/6/07, Ed Nuñez [EMAIL PROTECTED] wrote:
Is anyone else having trouble going into voip-info.org today?
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To UNSUBSCRIBE or update
I'm also experiencing the same problem. Has anyone found a fix for this?
Jesus
On 2/7/07, Santiago Aguiar [EMAIL PROTECTED] wrote:
Hi everyone!
I'm using Asterisk 1.2.7.1 on a CentOS 4 server with 5 - 9 agents and I'm
having some issues with the Chanspy application. All the agents are on
Just wanted to update anyone interested in this issue.
If I monitor a g729 SIP channel using ChanSpy, I am getting the same error
as when I use MixMon.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez
Sent: Thursday, June 07, 2007 12:14 PM
To:
Steve,
Thanks, but, we can't use tftp. The ATA's (all 1500 of them) are on
remote networks. As far as I know, tftp only works across a local
subnet. I called Linksys and they told me the ATA's can be provisioned
with http/https, but only after we become a certified reseller/provider.
Gonna have
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jared Smith
Sent: Thursday, June 07, 2007 10:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and reinvites...
On 6/7/07, Douglas
On 6/7/07, Carlos Chavez [EMAIL PROTECTED] wrote:
On Thu, 2007-06-07 at 21:41 +0530, ram wrote:
I have read that, but i dont see any examples that
give me solution for meetme.
can you just give me some examples
I think the example shown on that page, even though it is for
Quoting Douglas Garstang [EMAIL PROTECTED]:
Steve,
Thanks, but, we can't use tftp. The ATA's (all 1500 of them) are on
remote networks. As far as I know, tftp only works across a local
subnet. I called Linksys and they told me the ATA's can be provisioned
with http/https, but only after we
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jared Smith
Sent: Thursday, June 07, 2007 10:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and reinvites...
On 6/7/07, Douglas
On 6/7/07, Douglas Garstang [EMAIL PROTECTED] wrote:
I called Linksys and they told me the ATA's can be provisioned
with http/https, but only after we become a certified reseller/provider.
Gonna have to work on that I guess.
Well, I'm pretty sure that http provisioning works *without* becoming
Hi All,
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+cdr
Under example:
exten = s,2,Set(CDR(MyFavoriteBand)=Foo Fighters)
exten = s,3,Set(CDR(MyFavoriteSong)=Hero)
and under description:
-userfield: The channel's user specified field.
-any custom value that you wish to store.
My
On Thu, 7 Jun 2007, Douglas Garstang wrote:
Steve,
Thanks, but, we can't use tftp. The ATA's (all 1500 of them) are on
remote networks. As far as I know, tftp only works across a local
subnet. I called Linksys and they told me the ATA's can be provisioned
with http/https, but only after we
On Thu, 7 Jun 2007, Jon Pounder wrote:
Quoting Douglas Garstang [EMAIL PROTECTED]:
Steve,
Thanks, but, we can't use tftp. The ATA's (all 1500 of them) are on
remote networks. As far as I know, tftp only works across a local
subnet. I called Linksys and they told me the ATA's can be
Hi all --
I'm having awesome fun with Asterisk voicepulse connect together.
So cool.
I'm trying to have the caller id read back to me.Do I need to do
something to have this sent across in the sip.conf? Or is there
something I need to do somewhere to enable the reading of this
Remco Post wrote:
I guess that if I read these stats correctly, the bottleneck for * is
not so much cpu power, it's the cpu cache. As I see it, the cpu cache
becomes far less efficient for larger call volumes, eg. the cache is
unable to keep the most frequently used code and data in cache, due
On Thu, Jun 07, 2007 at 12:51:28PM -0400, John Novack wrote:
Stephen Bosch wrote:
Hi, John:
This feedback is brilliant. Thanks. My comments follow.
You are most welcome
Someone at Sangoma has this problem before and thinks it's the
mainboard. They want us to try putting the
We use G.729. Consumes only 35kbps of bandwidth and has a level 4 (from
0 to 5) of voice quality. We still have very poor public data networks
here in Brazil that makes G.711 a very high bandwith consunption codec
for us.
Another point that is good for G.729 is that we can bridge calls from
There is talk about combining Vmukti and web-meetme.
This should make a very good audio-video solution.
--
--
Steven
http://www.glimasoutheast.org
Khaled Chehab [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Any one knows how to make Meet Me video conferencing room.
Regards
On 6/7/07, Steve Edwards [EMAIL PROTECTED] wrote:
I tried it with my xml file and it complains about the file being corrupt.
I had this problem too with some of the Linksys phones, and it did
turn out to be a problem with the XML file... it's seems they're
fairly picky about things like single
On 6/7/07, Matthew Pease [EMAIL PROTECTED] wrote:
I'm having awesome fun with Asterisk voicepulse connect together.
So cool.
I'm glad you're having fun!
I'm trying to have the caller id read back to me.Do I need to do
something to have this sent across in the sip.conf? Or is
Lucky the answer to your problem is simple -- you are using an old
format for the caller id -- they are now functions like
${CALLERID(num)} etc. -- see the documentation for more information.
on Thursday 06/07/2007 Matthew Pease([EMAIL PROTECTED]) wrote
Hi all --
I'm having awesome fun
Hi List,
I am having issues sending calls to my carrier who is using a Nextone switch to
handle the session and a Cisco box for the RTP stream. He said that he keeps
seeing a Q931 cause 44 error which he said he never received before. All of
his other clients are able to get through so its not
Douglas Garstang wrote:
Let's just say we only configured the originating phone with
canreinvite=yes, which hopefully means the originating phone would
reinvite with the second Asterisk server. That's all fine and good until
it becomes the receiving phone, and the other phone (as an
Tzafrir Cohen wrote:
BTW - my testing was on Asterisk 1.2.17 with special pulse dial drivers
provided by Sangoma.
Why do you need special drivers for that? Doesn't Zaptel handle that?
Though it will happily identify pulse digits larger than 10...
The Sangoma off the shelf patches
You didn't say what VERISON OF ASTERISK you are using, so I will assume
1.2. When you watch the console, what is the output on the CLI of the
SayDigits lines? The only reason I can see for CALLERIDNUM to be empty
is if the information was not received with the call.
I'll bet you got this
On Thu, 7 Jun 2007, Jared Smith wrote:
On 6/7/07, Steve Edwards [EMAIL PROTECTED] wrote:
I tried it with my xml file and it complains about the file being corrupt.
I had this problem too with some of the Linksys phones, and it did
turn out to be a problem with the XML file... it's seems
If he's using 1.4, I hope he read UPGRADE.txt, which covers this and all
the other changes between 1.2 and 1.4
Jared Smith wrote:
If you're using Asterisk 1.4, the syntax has changed:
exten = _XX.,1,Answer()
exten = _XX.,n,Playback(hello-world)
exten = _XX.,n,SayDigits(${CALLERID(num)})
Hi all -
Searching for java agi in the mailing list archives turns up ancient posts.
Anyone else using java for their AGI? How well is it working
what are you using?
My script is pretty simple, and I could write it with perl easy
enough, but I just would feel better if I can keep most
1) How can I get a list of currently set channel variables for a specific
channel in Asterisk, including custom variables set by the dialplan? I
don't want a static list of variables from a web site, I need the current
dynamic list that shows custom variables that are specific only to this
Hi Jared-
Awesome! Thanks so much for saving me hours of scratching my head.
I have the asterisk book, but evidently its 1.2 based. boo.
matt
On 6/7/07, Jared Smith [EMAIL PROTECTED] wrote:
On 6/7/07, Matthew Pease [EMAIL PROTECTED] wrote:
I'm having awesome fun with Asterisk
Matthew Pease wrote:
Hi all -
Searching for java agi in the mailing list archives turns up ancient
posts.
Have a look at http://asterisk-java.org and the tutorial at
http://asterisk-java.org/development/tutorial.html - it include a hello
world AGI script in Java.
=Stefan
signature.asc
Tzafrir Cohen wrote:
To generate a FXS dialtone without Asterisk, use fxstest (make fxstest)
from the zaptel source directory.
Can I break this dial tone with DTMF?
-Stephen-
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-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling
Sent: Thursday, June 07, 2007 2:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and reinvites...
Douglas
hello to all,
i am geting this NOTICE while i am running asterisk.
agents are able to here the customer voice but the customer is unable
to here agent voice
plz somebody help me
#rtp.c:331 process_rfc3389: Comfort noise support
incomplete in Asterisk (RFC 3389). Please turn off on client if
Hi all
I have a network with nodes with different network-interfaces (e.g. node17
with interfaces A and B and node18). Asterisk listens to 17.A, 18's DUNDi
knows 17 by knowing ip B.
When I start a DUNDi request from 18 to 17 I get a response from A via B. So
B knows that the number can be
On 6/7/07, Douglas Garstang [EMAIL PROTECTED] wrote:
That's all fine and good until
it becomes the receiving phone, and the other phone (as an originator)
also has canreinvite set to yes. Then, your back to both Asterisk
servers being completely taken out of the loop again!
While I haven't
On 6/7/07, Nick Seraphin [EMAIL PROTECTED] wrote:
1) How can I get a list of currently set channel variables for a specific
channel in Asterisk, including custom variables set by the dialplan?
Use the DumpChan() dialplan application.
2) Where can I find a comprehensive list of problems and
On 6/7/07, Matthew Pease [EMAIL PROTECTED] wrote:
Awesome! Thanks so much for saving me hours of scratching my head.
I have the asterisk book, but evidently its 1.2 based. boo.
plug type=shameless
Yeah, but don't worry too much... a second edition of the book will be
out shortly, which
Hi all,
I have a IAX trunk between two asterisk servers, both with dynamic IP
and both have a DNS name associated with it.
In the iax.conf file I configure the host parameter with the DNS name
of the servers. Everything works fine until one of these servers get a
new IP, so the other can't
On Wed, 2007-06-06 at 12:43 +0100, Steve Davies wrote:
On 5/31/07, Carlos Chavez [EMAIL PROTECTED] wrote:
Sometimes I get the following error on the console:
[May 31 11:14:01] ERROR[23502]: cdr_addon_mysql.c:230 mysql_log:
mysql_cdr: Failed to insert into database: (1062)
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