Error loading module 'res_rtp_asterisk.so': /usr/lib64/libavformat.so.52:
undefined symbol: av_tree_node_size
This is the error I get when trying to start Asterisk 11 on centos 5.
Asterisk 11 works fine on my centos 6 box - I also verified that on centos 6
I do not have the above mentioend fi
Just for grins, do you have a softphone like xlite that you can try the
outgoing call on? I think it's an outgoing issue, not a polycom one.
I do not have a softphone. I have a yealink VP-2009 and same behavior.
Jerry
--
_
--
I have a little dialplan context now...
[check-chanisavail]
exten => s,1,ChanIsAvail(${agi_channel})
exten => s,n,System(/bin/echo ${AVAILCHAN} > /tmp/${agi_file})
exten => s,n,Hangup()
and a call file:
Channel: Local/s@check-chanisavail/n
Context: check-chanisavail
Extension: s
Priority: 1
Set
On 12/20/2012 01:00 PM, Jerry Geis wrote:
IMO the local channel call should be the lowest overhead option available.
What about:
Action: Command
Command: dahdi show channels
I can just look to see if "Extension
IMO the local channel call should be the lowest overhead option available.
What about:
Action: Command
Command: dahdi show channels
I can just look to see if "Extension" has anything for the Chan I am
interested in?
is
You should just cache the AMI DAHDIChannel event information in your
program.
If you really must you could use the CLI command "pri show channels".
However, it is not intended to be repeatedly run for performance
reasons. It blocks processing of ISDN messages while it is running.
I am not con
It is
Action: ExtensionState
Exten: 5551212
Context: fubar
This will return the status of the dialplan exten hint.
>/ and
/>/ Action: Command
/>/ Command: ChanIsAvail
/>/ Parameters: DAHDI/1
/>/
/>/ says Error
/>/ No such command "ChanIsAvail"
/
ChanIsAvail is a dialplan application not a
This was a change in v1.8 and is documented in the v1.8 UPGRADE.txt file:
* The PRI channels in chan_dahdi can no longer change the channel name if a
different B channel is selected during call negotiation. To prevent using
the channel name to infer what B channel a call is using and to av
I have a CentOS 6.3 machine I installed Asterisk 11, worked fine...
I then tried to install on Cents 5.8, seemed to go fine... Then when I
placed a call I got this:
ast_rtp_instance_new: No RTP engine was found. Do you have one loaded?
Did a search and found issues with ARM and this problem bu
In 1.4.43 I would see things from "core show channels" like
DAHDI/18/x
for line 18
in Asterisk 11 its
DAHDI/i4/
How do I get the line number back?
Jerry
--
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-- Bandwidth and Colocation Provided by http://www.api-digit
This animal might be like the OBI110 box where you set it up in users.conf
instead of sip.conf.
Something like this:
[5001]
transfer=yes
call-limit=5
registersip=no
host = 1.2.3.4
context=default
hasvoicemail=no
dtmfmode=inband
threewaycalling=no
hasdirectory=no
callwaiting=no
hasmana
The two things I would try are changing type from friend to peer and
sendrpid from no to yes. The no matching peer usually means the device
username isn't matching the sip.conf username=.
I have tried both friend and peer. I changed the sendrpid to yes
and made no difference either. Still get 40
[5001]
type=friend
username=5001
secret=XXX
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
context=incoming
host=dynamic
canreinvite=no
qualify=no
trustrpid=yes
sendrpid=no
nat=no
I did notice one more thing:
chan_sip.c:17045 handle_request_register: Registration from
I am trying to get a digital accoustics talkmaster to register to
asterisk 1.4.43
I am getting the 401 unauthorized.
I have
host=dynamic
I have verified the passwords match
What else is there?
I dont see any further clues in "sip set debug".
all it says is using request as basis request
What
Try "pedantic=no" in sip.conf.
Also, enable a SIP debug on the peers, check if anything out of the
ordinary appears.
seems as though pedantic=no was the issue. they are staying online.
further looking (which I seemed to miss) was in 1.4 pedantic as default no,
in 11 default is yes.
When you say "two", is it two every time? The same two? Is there something
different about the two that show this behavior? There isn't enough
information in your message.
Yes it is the same two devices every time.
I have the server running 11.0.2 , I have 8 asterisk devices (1.4.43),
I have tw
How can extensions.conf be changed to work with both
Asterisk 11 and 1.4.X such that 1.4.X calls deadagi and 11 just calls
agi as deadagi is no more.
Thanks,
jerry
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I am running 11.0.2 from 1.4.43 previous.
When I start up and do a "sip show peers" all devices are on and show an
IP Address.
After some time "sip show peers" shows two devices of my 12 as
(Unspecified).
I never had an issue with 1.4.43.
Is there some issue with 11.0.2 and registration?
Je
I am sending something like this to the AMI (along with other commands):
Variable: agi_pa_list=box1,box2,
however, in the dialplan I do:
exten => app_confbridge_call_out,n,noop(${agi_pa_list})
and it ONLY reports box1 and it should be "box1,box2"
Is something different in Asterisk 11 vs 1.4.43
Does anyone have information or successfully connected Asterisk
to TalkMaster from Digital Accoustics?
Thanks,
Jerry
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New to Asterisk? Join us for a live
By not "in sync" do you mean that there is a delay between when the
speaker speaks and when the client hears it?
There's always going to be some amount of delay. It takes time to encode
the audio, send it, mix it (in this case), receive it, decode it, and
have it pass through a jitterbuffer (w
I am running asterisk 1.4.43 on a really small network for testing, all
on same switch.
I launch a meetme between my server and 5 asterisk clients that
are all on 10 foot network cables all connected to the same switch.
The meetme is fine everything is in sync
Then I reboot one of the clients
Can you clarify what you mean by "MeetMe to be active"? What MeetMe
options are you using and what is your configuration like? With the
proper combination of options it shouldn't matter who gets into the
conference bridge first. This is what Page essentially does, with the
difference being that o
I think "I" have a race condition.
I am running something like this in my dialplan
call agi to bring my "list" of devices into my MeetMe
Playback beep
start MeetMe()
So in fact the meetme is not started before I bring the list
of devices into the meetme.
How can I do this differently so the M
hile AGI is an application, it has to be done within a call, so it is best
to do a call to the context that has the AGI command in it.
[runmyagi]
Exten => s,1,AGI...
Command: Dial
Context: runmyagi
Danny
Thanks, what might the command look like to "add" one asterisk box (sip
connected)
to
I am running 1.4.43
Trying to use AGI to do the
Application: AGI
Its telling me missing action in request.
What should the Action: be in this case
I tried
Action: Originate
but it still says the same error.
--
_
-- Ban
I am using 1.4.43 currently.
I am using the AMI to originate a call over a SIP Trunk to my cell
XXX506. works fine.
when the call is active I do a "core show channels concise" and I get:
SIP/testsystem-0ad0!smvoice-dialout!callprogress!4!Up!AGI!smvoice!0!!3!24!(None)
My AGI is called
I have an mp3 that is 128K, 44.1K stereo.
I convert that to wave 16 bit, stereo, 44.1K
The "sound" alike at this time.
I want to play them (not just over my sound port) but through asterisk
on select devices/machines that are also running asterisk over the
Console/dsp.
I converted the wave fi
What version of Asterisk are you running? There was an issue found in
February where this exact behavior could occur, two "dialplan reload"
commands would clobber each other. It was also resolved back then in all
supported branches (1.8, 10, and trunk).
http://lists.digium.com/pipermail/asterisk
Sorry to step in here but I think the 2 of you are talking at cropp
purposes
I initial query was about a dialplan reload, not an asterisk restart.
Jerry, how long does your system take to perform a dialplan reload?
surely it is under a second.
If you look in the logs, at the end of any dialplan
Hi Jerry,
From the Asterisk CLI, enter the command "core restart when
convenient", this command will restart asterisk only when there is no
incoming call, and when it will close all outgoing calls.
With a restart of asterisk should reload all the information:
extensions, sip, agi, iax, voicemail
If I issue a "dialplan reload" and some AGI starts as its reloading
and "directs" something into the diaplan that is still reloading
what happens
I presume my context is not there?
What I see is the diaplan is messed up somehow and I goto the default
context
then after that it is messaged
I experienced a similar issue in the past, where Asterisk and DAHDI seemed
to disagree about my zone. In any case, try using Congestion() instead of
PlayTones(busy).
Chris,
This is very strange. I changed to Congestion() I still get nothing .
I even changed to playback(demo-congrats) and I st
I am using two polycom phones to call into an asterisk box and the
console/dsp.
First phone calls in and I get connected just fine.
second phone calls in and I detect the Console/dsp is busy, and i try to use
playtones(busy) and I hear nothing. (see below)
How can I hear the tones? Thanks
Jerr
If I remember correctly, dahdi dummy was removed and the functionally added by
default when you load dahdi with no TDM cards installed. I could be wrong
though.
What do you need dummy for?
I am using CentOS 5 on a machine and have no issue with alsa dropping audio.
I dual booted and have CentO
I need to use the dahdi dummy driver.
Its not being compiled at this time.
When I go into tools subdirectory under dahdi-linux-complete-2.4.1
and do make menuselect all I get is
CC="" CXX="g++" LD="" AR="" RANLIB="" CFLAGS="" make -C menuselect
CONFIGURE_SILENT="--silent" nmenuselect
make[1]: En
I have a connection between two asterisk boxes, both running 1.4.43
The connection is alive and good and working. however, I see a bunch of
401 Unauthorized messages using wireshark - then it eventually registers
again
just fine.
Why would it not successfully register the first time - every ti
If I am using asterisk (server) and then asterisk on client (sound port)
and I want to get the best MP3 sound I can get - how can I do that with
ulaw codec
and wav file conversion.
I used gst-launch to convert my MP3 to WAV (16K and mono) then playing
over ulaw
to the other client. I know mono
On 10/16/2012 10:24 AM, Jerry Geis wrote:
I am using asterisk 1.4.43.
When I call over the PRI to a single phone and play my recorded
message its heard just fine.
When I call over the PRI to a single extension (the switch then takes
3 phones offhook in intercom mode)
and play my same recorded
I was wanting to call ChanIsAvail from AMI.
I logged in and issues command,
Action: Command
Command: ChanIsAvail DAHDI/1
my response was this:
event_list=0 ret=158 Response: Follows[CR ][LF ]Privilege: Command[CR
][LF ]No such command 'ChanIsAvail DAHDI/1'
Is there any way to tell if a chann
How do I use softhangup through the AMI interface?
I am using 1.4.43. Will softhangup hangup a DAHDI channel?
I have found that "Action: Hangup" does not hangup a DAHDI channel only sip.
Thanks,
jerry
--
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-- Bandwidth and Co
I am using asterisk 1.4.43.
When I call over the PRI to a single phone and play my recorded message
its heard just fine.
When I call over the PRI to a single extension (the switch then takes 3
phones offhook in intercom mode)
and play my same recorded message the audio is dropping out and the
w
I am using 1.4.43 connected on PRI to avaya PBX.
If I call one extension through the PRI and speak a message (recorded
file) sounds fine.
If I call an extension through the PRI that brings together a group of
phones on the avaya side
and play the same recorded file the audio drops out.
What m
I am setting up with meetme a conf with X number of asterisk boxes and
"other" devices and phones. I am using the l parameter for all devices
being listen only
but I'm not sure thats happening as I am getting some feedback (some
devices are close to each other like 5 feet).
How do I ensure tha
I place a call to a polycom phone, it answers, my AGI
calls "Exec SendDTMF 11 " but I do not hear the DTMF tones on the
phone.
Why is that?
Jerry
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com -
Nothing happens "at the same time", unless you're broadcasting information
over some transport that supports multicast sends. There's always going to
be some interspersing of transmissions, if for no other reason than each
participant's channel in the conference has to be serviced after the media
I am using Meeting on 1.4.43 with a handfull of devices, like 10 to 20
in a Meetme.
I can "tell" a difference (as two of the devices are close to each
other) that they are
not fully in "sync". One was slightly behind the other... Any way to get
them more in sync?
Is it the delay from starting
It may not be extensions.conf per se. It could be extensions-custom.conf or
any other file included in extensions.conf. Also, Asterisk generates some
of its' own "custom context" entries, so you might look into that as well.
Also check extensions.ael.
Danny
THanks, actually all of my modifcati
Unless you have configured your file systems not to, there will be a
modification
time on the extensions.conf. That might give you a clue as to *when* it got
altered.
--
AJS
The date is Aug 2 2012.
So the "file" is not changing.
Anything else?
Jerry
--
__
Has anyone ever run across where asterisk "looses" part of a diaplan???
I has this happen a couple times, so I put script in place at 2AM that dumps
the dial plan and compares it to the previous day or a know good one.
This ran fine for quite a while (multiple weeks, forget when I started
this).
I have had a case where after a hangup on the Alsa channel
asterisk still thinks the line or call is active.
I have:
rtptimeout=60
rtpholdtimeout=60
rtpkeepalive=60
in my sip.conf file to help with this but it had no effect.
How can I ensure a session HANGS up and is not stale
Is there a
Try adding qualify=yes
Eric
I added this and I still had one time last night at 3am that
said "Unspecified".
Is there something else?
I put it in the [general] section of sip.conf on both machines.
Jerry
--
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-- Bandwidth
Moving from 1.4 to 11 deadagi is deprecated.
Is there a suggest way in the dialplan to handle the "case" of either.
Was hoping to keep one extensions.conf file and just call the
appropriate agi/deadagi.
What's the best way to do that?
Jerry
--
I have a couple asterisk boxes, running sip between both boxes. 1.4.43
on both.
both are installed from source,
both have default settings,
My config for one box is:
[devgeis]
type=friend
defaultname=devgeis
username=devgeis
secret=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm
rtptimeout=60
The AMI action CoreShowChannels deprecated the CLI concise command
because the output of the AMI action is extensible without breaking
existing systems. The CLI command is not extensible without breaking
existing systems.
Richard,
Thanks - I tried the CoreShowChannels AMI and it says:
Response
With MeetMe there was a MeetMeAdmin() that could KILL a conference.
How do I do that with the new ConfBridge. I dont see a way to kill the
conference.
There also used to be a "core show channels concise", this is deprecated.
What is the correct way to do this now and get all the information?
On 08/22/2012 08:46 AM, Jerry Geis wrote:
Hi Jerry,
Firstly, in logging.conf, make sure you have a line as follows:
full => notice,warning,error,debug,verbose,dtmf,fax
If you made any changes, then in the asterisk CLI, do: reload logger
Then again in the CLI, do:
set verbose 5
set debu
Below is the backtrace of I think why I am hearing a tone in my confbridge.
I have all sounds turned off for my confbridge profile.
Looks like something with "fax" is in the mix. How do I disable that -
no faxes in my
conference and dont need any tones from it?
Thanks,
Jerry
---
Hi Jerry,
Firstly, in logging.conf, make sure you have a line as follows:
full => notice,warning,error,debug,verbose,dtmf,fax
If you made any changes, then in the asterisk CLI, do: reload logger
Then again in the CLI, do:
set verbose 5
set debug 5
Then try your scenario and look afterwards
On 08/17/2012 03:21 PM, Jerry Geis wrote:
On 08/17/2012 06:36 AM, Jerry Geis wrote:
On 08/13/2012 04:58 PM, Jerry Geis wrote:
On 08/13/2012 01:13 PM, Jerry Geis wrote:
I am getting a "beep beep beep" (like a busy or hangup sound) when
I am using my
AGI to start up a conf. (did
On 08/17/2012 06:36 AM, Jerry Geis wrote:
On 08/13/2012 04:58 PM, Jerry Geis wrote:
On 08/13/2012 01:13 PM, Jerry Geis wrote:
I am getting a "beep beep beep" (like a busy or hangup sound) when I
am using my
AGI to start up a conf. (did not happen with Meetme).
The confbridge work
On 08/13/2012 04:58 PM, Jerry Geis wrote:
On 08/13/2012 01:13 PM, Jerry Geis wrote:
I am getting a "beep beep beep" (like a busy or hangup sound) when I
am using my
AGI to start up a conf. (did not happen with Meetme).
The confbridge works, but the beep beep beep is mixed in with
Is it possible to miss a UDP SIP packet to hangup a call?
Using 1.4.43 I had a call from on asterisk box (server) to a
low end client (chan_alsa) not hangup.
Could this be due to missed UDP SIP packet to hangup?
Is there anyway for a client asterisk (chan_alsa again) to
monitor the connection an
All,
I ran into a case today where using the "Console/DSP" that the SIP
channel had been
hung up a long time ago. I tried to call back into the Console/Dsp and I
got busy.
There was no active channel any longer. Some how it did not get the hangup.
I am running 1.4.43
How can I make sure my c
On 08/13/2012 01:13 PM, Jerry Geis wrote:
I am getting a "beep beep beep" (like a busy or hangup sound) when I
am using my
AGI to start up a conf. (did not happen with Meetme).
The confbridge works, but the beep beep beep is mixed in with the audio.
I have turned off every so
I am getting a "beep beep beep" (like a busy or hangup sound) when I am
using my
AGI to start up a conf. (did not happen with Meetme).
The confbridge works, but the beep beep beep is mixed in with the audio.
I have turned off every sound in the confbridge.conf file.
How can I find out where th
I am starting to use ConfBridge and not MeetMe in asterisk 10.
I have everything converted over EXCEPT.
I am using an AGI and AMI to bring phones into a conf automatically.
When I do that the conf is going just fine - however - I head beep,
beep, beep.
I have every sound listed in confbridge.
On 08/10/2012 11:23 AM, Jerry Geis wrote:
I have a profile in confbridge
[MessageNetConfBridge]
and more...
Asterisk is reading it at startup.
[1;30m == ^[[0mParsing '/etc/asterisk/confbridge.conf': ^[[1;30m ==
^[[0mFound
^[[1;30m ^[[0mapp_confbridge.so => (^[[0;33mConf
I have a profile in confbridge
[MessageNetConfBridge]
and more...
Asterisk is reading it at startup.
[1;30m == ^[[0mParsing '/etc/asterisk/confbridge.conf': ^[[1;30m ==
^[[0mFound
^[[1;30m ^[[0mapp_confbridge.so => (^[[0;33mConference Bridge
Application^[[0m)
When I try to use it I get a wa
On 08/10/2012 09:00 AM, Jerry Geis wrote:
I just downloaded and compiled from source asterisk 10.7.0
after installing and running I tried to do a meetme, did not work.
I looked in the apps/app_meetme* and there is only the C file, there
is no .o
seems like it did not compile.
Is that a new
I just downloaded and compiled from source asterisk 10.7.0
after installing and running I tried to do a meetme, did not work.
I looked in the apps/app_meetme* and there is only the C file, there is
no .o
seems like it did not compile.
Is that a new default behavior?
Looking for the trick to ge
When I use a call file to start a call I set the
CallerID: field and the polycom phone shows the correct information.
When I use a call file to start a conf call I set the
CallerID: field and my polycom phones show "asterisk" not the callerID I
have set.
Is there something additional needed to
Is there a limit to the number of connections
that manager can handle at one time?
In my logs I see "connect error" but then try again
in a few seconds and it works.
I could have quite a number of connections at one time.
How can I up the limit.
Jerry
--
__
Is there a limit to the number of connections
that manager can handle at one time?
In my logs I see "connect error" but then try again
in a few seconds and it works.
I could have quite a number of connections at one time.
How can I up the limit.
Jerry
--
__
I have a meetme running that is taking audio from a PC running asterisk
(console) as input
to my server that is then feeding it using meetme to two other asterisk
PC's going out the console.
All running 1.4.43
I have noticed that when the meetme first starts if I change the input
audio (new so
Is there a way to have ALSA accept more than one
incoming call?
I have asterisk running a box with an audio source input.
So the incoming call just connects the audio feed.
Issue is I want "at times" to source that feed to more than
one call.
Can I do that? How is it accomplished?
Thanks,
Jer
I have a process that runs on a server and does a simple 'asterisk -rx
"sup show peers' > /tmp/peers"
and then looks for any "(Unspecified)" items and reports them as having
lost connection.
My server is running 1.4.43 and the two boxes I am monitoring are also
running 1.4.43.
Once in a great wh
If I have a connection from Asterisk to Cisco Call Manager using SIP
can I send a text message using "SendText" from asterisk across the SIP
trunk to CCM and it convert that to text message to the skinny cisco phones?
Does that work?
Thanks,
Jerry
--
_
I used my cell to call in and create a CDR record here from asterisk 1.4.43:
"","317XXX","s","default","""GEIS JERRY ""
<317XXX>","DAHDI/23-1","","BackGround","SM_ATTENDANT","2012-01-05
18:12:09","2012-01-05 18:12:10","2012-01-05
18:12:18",9,8,"ANSWERED","DOCUMENTATION","1325787129
On 01/03/2012 08:48 AM, Jerry Geis wrote:
I have a USB to serial converter attached to my box. pl2303: Prolific
PL2303 USB to serial adaptor driver
if I login to the box and send/receive serial commands over this unit
it works without error EVERY time.
however, if I run the same command set
I have a USB to serial converter attached to my box. pl2303: Prolific
PL2303 USB to serial adaptor driver
if I login to the box and send/receive serial commands over this unit it
works without error EVERY time.
however, if I run the same command set from with-in the extensions.conf
with Syste
On 12/28/2011 03:57 PM, Jerry Geis wrote:
I ran into a "rare" situation today.
A really short message is being played over the ALSA or console
channel from one asterisk box to another. Both running 1.4.30.
the incoming context on the ALSA or Console port box first runs an A
I ran into a "rare" situation today.
A really short message is being played over the ALSA or console channel
from one asterisk box to another. Both running 1.4.30.
the incoming context on the ALSA or Console port box first runs an AGI
before connecting the audio path.
The AGI got hung up for a
On 12/07/2011 10:16 AM, Jerry Geis wrote:
I am using AMI to call a phone and play a wave file. That works fine
to SIP/401.
Now I am trying to "redirect" that call that is ringing to another
phone (SIP/404).
When I do it the other phone rings but the first phone continues to
I am using AMI to call a phone and play a wave file. That works fine
to SIP/401.
Now I am trying to "redirect" that call that is ringing to another phone
(SIP/404).
When I do it the other phone rings but the first phone continues to ring
also.
Then when I answer on SIP/404, I get a ring not
I am using my first PAP2 device from linksys. Used many polycom phones...
I configured the PAP2 device with asterisk. I have the registration,
thought I was good to go.
Plugged in my Valcom 2924 public address analog connection, called the
extension
and I got busy... very strange I thought.
Is there a way with the command (1.4.42) for sip show peers to
see the FULL "Name/Username" field???
I have long names and mine are being truncated.
Thanks
Jerry
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-- Bandwidth and Colocation Provided by http://www.api-dig
I have had a couple thunderstorms take out a card, again last night.
The card with "dahdi show status" still report OK both times.
When calling into the card I get all circuits are busy.
However, simply replacing the card did the trick. Before that
I stopped asterisk, restarted DAHDI, rebooted
On 10/17/2011 10:33 AM, Jerry Geis wrote:
show channels is giving me a line like:
Local/call_out@playfile-981a,2!local-playfile!call_out!4!Up!MeetMe!PA0008|1qt!3175661010!!3!2!(None)
I extract from that the following ID:
Local/call_out@smvoice-local-public-address-playfile-981a
I am
show channels is giving me a line like:
Local/call_out@playfile-981a,2!local-playfile!call_out!4!Up!MeetMe!PA0008|1qt!3175661010!!3!2!(None)
I extract from that the following ID:
Local/call_out@smvoice-local-public-address-playfile-981a
I am trying to do the AMI "Action: Hangup" command (which
Can you post the call file with the pertinent info blacked out? I'm on
1.4.41 so I might be able to assist.
I am attempting to use a local call, start a meetme, bring others in the
conf, and run a background
agi to play the wav file in the conference. All automated.
Channel: Local/call_out@
On 10/13/2011 03:25 PM, Jerry Geis wrote:
How do I specify command line arguments to the MEETME_AGI_BACKGROUND?
I'm using the 1.4.42
I am setting the value in a call file:
SetVar: MEETME_AGI_BACKGROUND=myagi,-myarg
also tried
SetVar: MEETME_AGI_BACKGROUND=myagi -myarg
In both cases th
How do I specify command line arguments to the MEETME_AGI_BACKGROUND?
I'm using the 1.4.42
I am setting the value in a call file:
SetVar: MEETME_AGI_BACKGROUND=myagi,-myarg
also tried
SetVar: MEETME_AGI_BACKGROUND=myagi -myarg
In both cases the CLI said Failed to execute
Thanks,
Jerry
--
_
If the asterisk box starts up a MeetMe conference
with the 't' flag for talk only mode does asterisk send some
kind of SIP command to the devices joining the conference
to say "dont send me audio back as I'll ignore it anyway" as
I am the only one doing the talking.
Does that happen?
I'd like t
I am starting up 4 calls over the AMI.
It "appears" as though the first 3 start up and go out right away.
The 4th call is delayed like 15 seconds.
Any thoughts on why this fourth call might be getting delayed...
Everything is working its just delayed.
Jerry
--
Depending on the cell phone you are calling, the DTMF length may need to be
set to LONG (I know this applies to Verizon phones).
Danny
I am not familiar with this setting - where is it exactly.
I looked in my sip.conf and did not see anything.
Thanks-
Jerry
--
_
I am running asterisk 1.4.41.2 and dahdi 2.4.1 (64 bit centos)
I only have one small issue.
I initiate a call over AMI, call is answered and I run my AGI.
"sometimes" when I make calls out to cell phones I ask to press 1 to confirm
the user hears the message and press 1 but I never get the 1 bac
The cause code says Unallocated (unassigned) number. You are dialing an
invalid number. Is the 9 supposed to be in your called number?
Richard
Richard
I am supposed to be dialing 9 for an outside line.
Jerry
--
_
-- Ban
I am not getting calls going out my PRI.
I am getting an error condition.
There are no errors in /var/log/asterisk/messages.
more /etc/dahdi/system.conf
loadzone=us
defaultzone=us
span=1,1,0,esf,b8zs
bchan=1-2
dchan=24
echocanceller=mg2,1-2
more /etc/asterisk/chan_dahdi.conf
[channels]
pridi
Is there a method to "lock" asterisk into memory
such that once its loaded it does not get paged out?
I have ran into a couple times where it seems like asterisk needs to be
paged back into memory to start answering a call.
This is on a machine that is using the ALSA port to send audio over.
Som
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