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?
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
I have a B410P card with span ports set up as
span=3,1,0,CCS,AMI
span=4,2,0,CCS,AMI
span=5,3,0,CCS,AMI
signalling = bri_cpe
switchtype = euroisdn
layer1_presence = ignore
However, I keep getting these messages over and over again:
[Dec 14 18:53:14] WARNING[22476]: sig_pri.c:1150
I used libpri 1.4.12 version with asterisk 1.8.7, after the pcap
files, i found that
in wireshark setup message, the number type always changed from
subscriber to national number.
i have set pridialplan= local and prilocaldialplan=local in
chan_dahdi.conf already. because that, the system
my scenario is below
analog phone (10 to 99)-- pbx--(77)asterisk
jitsi(2000)
i have analog telephone interface numbered 77 attached with asterisk
and
other sip user is 2000 on jitsi.
I can call from any number from 10 to 99(in intercom) on 77 and ivr
response will
On Mon, Nov 19, 2012 at 08:47:23PM +0100, Adolphus Enaboifo wrote:
.. I get errors while trying to compile Libpri 1.4.13. (check
attachment} Can you guys please help me prescribe a fix.
[snip]
gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC
-O2 -MD -MT pridump.o
this is coming rather late but I took your advice and went ahead to
install Dahdi before installing libpri-1.4.13
and the error messages are now different.(see attachment)
This is compile error is reported by newer gcc compiler versions.
It is already fixed in libpri SVN.
This is a question regarding whether there's any way within hangup
extensions to determine whether the caller or callee leg (or both) of
a
bridged call has hung up. The test case I have is running under
Asterisk 1.8.17.0, but the behaviour is observed in 1.8.18.0 (and
also
1.6.2.18).
I need some advice on how to implement something in my dialplan.
Here's the scenario. A call comes in on my [incoming] context and I
answer it. The call turns out to be for my wife and she needs to
answer it on a different
handset somewhere else in the house.
I've tried call parking but
I'm testing out a server with asterisk 1.8.15.0 on it.
I'm experiencing static occurring on almost 90% of calls on this
particular server.
All test phones are using SIP, and calls to/from PSTN servers are
delivered using IAX2.
I have other production servers running 1.4.x that do not
I want to know actual DAHDI channel number (pseudo), which received
the
call or dialed the call. Where as when Asterisk receives a call on
DAHDI
channel, it shows channel as DAHDI/i5/112-15
Is there any way / configuration to change this behavior and get
actual
channel number? Earlier we
Asterisk 1.8.10.1~dfsg-1ubuntu1
See dial plan code below. When I dial 123 from a phone in this
context,
I simply get a busy signal. Why doesn't the i extension get
triggered? Console at verbosity of 10 only shows == Using SIP RTP
CoS mark 5.
[DockPhone]
exten =288,1,NoOp(Dock Phone)
I am currently in the process of upgrading a SIP/TDM gateway from
Asterisk
1.4.23 to 1.8.17. The gateway is designed to terminate SIP calls via
TDM
through a switching equipment. Among the features for this gateway
there is
handling for redirected calls, i.e. populating the outgoing RDNIS
l can't see to get the Lua extension matching to work:
[Oct 23 19:13:12] NOTICE[4288]: chan_sip.c:23577
handle_request_invite: Call from 'user' (XXX.XXX.XXX.XXX:33962) to
extension '107' rejected because extension not found in context
'luaentry'.
extensions = {}
Considering this your real question I want to track the number of
calls up at any given time, through the AMI
Comparable Command Exists !!
You can simply run CLI commands through AMI and receive the response.
Command used for this is itself Command in AMI. If you still want
to do
Setting up a group of analog lines to use for outbound emergency
calls
(911). My current dial plan and debug output shown below. It
appears
that when the SoftHangup() is executed that the line does not really
hang up. In the case shown, I had reduced the group to a single
DAHDI
(analog)
I've tested asterisk 1.8.17.0 and I'm still getting the repeated
error message on the command line:
iax2-provision.c:266 iax_provision_version: ast_db_get failed to
retrieve iax/provisioning/cach
Are you out of disk space? I would only expect to see that message once
since it looks like
I've tested asterisk 1.8.17.0 and I'm still getting the repeated
error message on the command line:
iax2-provision.c:266 iax_provision_version: ast_db_get failed to
retrieve iax/provisioning/cach
Are you out of disk space? I would only expect to see that message
once
since it
Asterisk 1.8
(a) We will have a group of 4 analog lines into a Digium card that
will
be used for local calls. What is the best way to use those lines as
a
pool for outbound calls? Can I use ChanIsAvail(), listing those 4
channels, and then use the first one returned?
There are lots of
After upgrading to Asterisk 1.8.15.1
I'm constantly getting this error on the command line:
ERROR[2499]: iax2-provision.c:266 iax_provision_version: ast_db_get
failed to retrieve iax/provisioning/cache
Can somebody explain what it is and how to fix it?
Since you say this happens
I am trying to setup a context to take a inbound call, hold the
call,
connect to
an external number, play a sound file to the external number, then
connect
the inbound caller to the external number.
My thought was to accept the call and place them in a parking lot.
Then
call
Continuing with the saga of Digium vs MTNL Mumbai, looking for
suggestions on handling incoming Caller-ID issues. The card manages
to
grab a couple of (random) digits of the incoming CID, but they're
more
or less useless. Is there any way to fix this?
Asterisk 1.8.13, Dahdi 2.5.0.1 on
Is there a way to execute next priority in the dialplan if you have
called agi:async? I want to play warning message if adhearsion is
down. Currently I wasn't able to make it work. The dialplan
execution ends after the first priority.
[incomming]
exten = _X.,1,AGI(agi:async)
exten =
I am trying to install an AsteriskNow. When system boot up, there are
two options,
To install with Asterisk 1.8 and FreePBX type 1 ENTER
To install with Asterisk 1.8 only type 2 ENTER
If I want to install Asterisk 1.8 only for example.
After asterisk is install, I found the /etc/asterisk/
Using
exten = h,n,set(CDR(llp)=${CHANNEL(rtpqos,audio,local_lostpackets)})
gives me
[Aug 24 12:08:10] WARNING[12087]: sip/dialplan_functions.c:221
sip_acf_channel_read: Unrecognized argument
'rtpqos,audio,local_lostpackets' to CHANNEL
[Aug 24 12:08:10] WARNING[12087]:
[snip]
There also used to be a core show channels concise, this is
deprecated.
What is the correct way to do this now and get all the information?
The AMI action CoreShowChannels deprecated the CLI concise command
because the output of the AMI action is extensible without breaking
existing
The AMI action CoreShowChannels deprecated the CLI concise command
because the output of the AMI action is extensible without breaking
existing systems. The CLI command is not extensible without breaking
existing systems. Richard,
Thanks - I tried the CoreShowChannels AMI and it says:
I run a hotdesking system based on the example from Asterisk: The
Definitive Guide. Calls come into the [hotdesk] context, which
verifies the phone has a logged in user and sends the call to
users,${EXTEN},1 if there is a user logged in. The [users] context
then includes several other
I have an issue that I have been bumping up against. We have some
inbound fax services and occasionally an inbound fax that
successfully came in would fail to store it's references in the
database.
We are using a function in func_odbc to update a database table. We
call the function from
I currently run an Asterisk 10 system with hotdesking functionality
set up. Several of the users have worked with a system in the past
that supported BLF on their IP phones, and would like their current
phones to behave in a similar fashion. Right now I have a really
kludgy system that mostly
Asterisk 1.4.42
Set alwaysauthreject=yes in [general] section of sip.conf.
Restarted asterisk
However when I attempt to register I still get:
[2012-08-08 21:11:34] NOTICE[15689] chan_sip.c: Registration from
'sip:000333082261...@domain.com' failed for '121.98.1.1' -
Wrong
password
I know the topic comes back like boomerang , but I did not find a
nice solution.
Does someone has/knows how to achieve call back on busy otherwise
called camping?
If one is calling the extension and it is busy, then caller should
get something like Press 5 to request call back and after the
I’m trying to set my system to set a caller id using the diaplan when
calling an internal extension. In other words, when I dial Joe
Smith’s extension I want my own phone to show “Joe Smith 555”. I
have sort of managed that in the sense that my phone shows Joe
Smith’s caller id based on his
exten = 123,1,Verbose(1,Test)
exten = 123,n,Set(CONNECTEDLINE(number,i)=555-555-)
exten = 123,n,Set(rclidname=TestingB 123-444-)
This line is just setting an ordinary channel variable.
What do you think is supposed to use this value?
exten =
Has anyone seen Dahdi dropping incoming calls with Hangup cause 27?
It only drops whilst we are on the phone?
Its not every single call
Any ideas?
Libpri can generate that cause code when T309 expires. T309 starts
when the link goes down. When T309 expires, active calls are dropped
because
This is the option I will try.
I'll report my findings here.
My findings, after setting layer1_presence=ignore in
chan_dahdi.conf
are :
== Starting D-Channel on span 1
== Starting D-Channel on span 2
== Primary D-Channel on span 1 up
== Primary D-Channel on span 2
This is the option I will try.
I'll report my findings here.
My findings, after setting layer1_presence=ignore in chan_dahdi.conf
are :
== Starting D-Channel on span 1
== Starting D-Channel on span 2
== Primary D-Channel on span 1 up
== Primary D-Channel on span 2 up
foo*CLI
My previous message was incomplete.
On thing to note is I had to forbid hfcmulti in modprobe.d in the
second box to comply with a warning from dahdi. Without that, I could
see this line in the output of lsmod:
mISDN-core hfcmulti
1. What is the root cause that makes a board change
We have an Asterisk server which connects to another Asterisk server
acting as a PSTN gateway. This gateway machine has Digium TE210P card
connected to a pair of PRIs.
For the most part, all is working well, however there are some
specific
telephone numbers that my users have attempted to
Thank you for your reply Patrick!
for the first situation, I did try asterisk 1.6.2.6 and dahdi 2.3 but
with no success.
Can anyone suggest a combination that works till a patch is released?
The patch for the layer1_presence option is in Asterisk 1.8.13.0-rc1 and
10.5.0-rc1.
See
Hi Khalid,
my setup is almost identical
except for
loadzone = be
defaultzone = be
(obviously)
and in
chan_dahdi.conf:
[isdn4]
signaling = bri_cpe_ptmp
switchtype = euroisdn
group = 2
context = isdn
dahdichan = 10,11
this results into:
ERROR[1021]:
Please just reply to the mailing list.
- Original Message -
Right you are,
but when using bri_cpe I get:
[May 9 23:08:45] WARNING[2775]: sig_pri.c:6969 pri_dchannel: PRI
Error
on span 4: Received MDL/TEI managemement message, but configured for
mode other than PTMP!
This
I have read the excellent information here:
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
and believe I have an understanding of what is offered. I have a
couple of questions:
- Is it possible to update COLP/COLR when a SIP redirect occurs, or
when a
I've had an old server die on me, it was installed by someone else
then never maintained. It runs some old version of Elastix on top
of Asterisk 1.4.33 with 4x Digium T100P cards. I swapped all the
parts into a referb of the same gear and it runs great, but I want
to put it in a completely
I want to use Call Deflection with DAHDISendCallreroutingFacility
Application.
I use Asterisk:1.8.11 libpri:1.4.12 facilityenable=yes transfer=yes
my dialplan is like this:
You should always specify the switchtype and signaling parameters for
ISDN issues as well. In this case it is not
On 04/06/12 18:05, Joseph wrote:
When I start asterisk 1.8.10 I see:
== Parsing '/etc/asterisk/users.conf': == Found
-- Registered extension context
'app_dial_gosub_virtual_context'; registrar: app_dial
-- merging incls/swits/igpats from
On Fri, Mar 16, 2012 at 11:43 AM, Richard Mudgett
rmudg...@digium.com wrote:
snip
pri intense debug:
TEI: 0 State 7(Multi-frame established)
V(A)=31, V(S)=31, V(R)=42
K=7, RC=0, l3_initiated=0, reject_except=0, ack_pend=0
T200_id=0, N200=3, T203_id=8192
[ 00 01 54 3e
snip
pri intense debug:
TEI: 0 State 7(Multi-frame established)
V(A)=31, V(S)=31, V(R)=42
K=7, RC=0, l3_initiated=0, reject_except=0, ack_pend=0
T200_id=0, N200=3, T203_id=8192
[ 00 01 54 3e 08 02 01 b3 62 1c 66 9f aa 06 80 01 00 82 01 00 a1
31
02 02 01 3a 02 01 0c 30 28 0a
I currently have an Asterisk 1.6.2.18 server running a patched (see
below) libpri 1.4.10.2 connected to a Toshiba Strata CTX670. All
external calls come in via the Strata and then are routed to the
Asterisk server over a single PRI link using Q931. This setup is
working and has been working
I'm having issues with asterisk 1.8.4 dropping calls during transfer,
and
transfer to park extension. We're using polycom soundpoint IP 650.
when the
park button is hit the response is i'm sorry not an extension at
the same
time number 7 appers on the lcd.
Please use a newer version of
snip
/etc/asterisk/chan_dahdi.conf
==
;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
;autogenrated on 2012-02-29
;Dahdi Channels Configurations
;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak
?
Yes. It does look like the carrier is not sending the Reverse Charging
Indication ie.
Richard
Att,
Rafael Saraiva
2012/2/17 Richard Mudgett rmudg...@digium.com
I had not noticed that you switched to direct email earlier.
- Original Message -
Switchtype
I'm trying to emulate the functionality of our existing phone system,
which is somewhat different than what Asterisk provides with a
trivial parking configuration. I'd like each user to have three park
buttons, park 1, park 2, park 3. The snom 870s I'm using have a
Park+Orbit button, which
exten = _*70[123],1,NoOp(parking in ${EXTEN:1})
same = n,Set(PARKINGEXTEN=${EXTEN:1})
same =
n,GotoIf(${DEVICE_STATE(park:${PARKINGEXTEN}@parkedcalls)}=INUSE?busy)
same = n,Park()
same = n(busy),Busy()
What I'm hoping to
accomplish is have Asterisk respond to the
I have been trying to get the dynamic parking working.
For some reason when I park a call using this method the console says
it is using the default parking context not the one I am trying to
specidfy. It also is playing the parked extension to the caller. I
am transfering the call to an
The value is always -1. I must enable something in chan_dahdi to pass
the correct value?
++
[PABX]
exten=_X.,1,Gotoif([${CHANNEL(reversecharge)} = -1]
?entrada,${EXTEN},1:hangup,${EXTEN},1)
+++
rssr305*CLI -- Accepting call from '5132083300' to '1584' on
Reversecharge not appear in debug .
I'm in Brazil , the signaling is different here ?
Please capture the incoming SETUP from libpri for the collect call.
pri set debug on span x
Richard
--
_
-- Bandwidth and Colocation
How to block collect calls on ISDN trunk?
You need Asterisk v1.8 or later and check the value of CHANNEL(reversecharge)
in your dialplan.
https://wiki.asterisk.org/wiki/display/AST/Function_CHANNEL
Richard
--
_
-- Bandwidth
How to block collect calls on ISDN trunk?
You need Asterisk v1.8 or later and check the value of
CHANNEL(reversecharge) in your dialplan.
https://wiki.asterisk.org/wiki/display/AST/Function_CHANNEL
Can you give me an example of how to use this function?
exten = 100,1,Proceeding()
same
No, unfortunately that's not quite correct. The UPGRADE files list
*important* changes that users need to know about because they are
changes in behavior of existing functionality. New features, even
really
useful and widely anticipated ones, that don't cause backwards
compatibility issues
Dear All,
i have created one SS7 link on E1 bye using
DADHi/libss7/Asterisk1.8/DigiumTE205.
Link is up and voice is working, but regarding SMS, Provider ask to
send SMS on DChannel-16(Signaling Channel) by use below mentioned
command but its giving error.
./smsq
Is there a document that sums up the major changes made to the four
main releases available (1.4, 1.6, 1.8, and 10), to check if it's
worth upgrading?
www.asterisk.org/downloads
The UPGRADE.txt and CHANGES files do just that. They have been a part
of the Asterisk source files for a long
I currently have an Asterisk 1.8.8.1 system set up with SIP accounts
as well as a Wildcard TDM400P REV I card with both FXS and FXO
ports - FXO is connected to outside lines, FXS connected to inside
analog phones. Everything about the setup works fine except one thing
-
after making calls to
Where to find meaning of /n in Local/6613@from-queue/n ?
See https://wiki.asterisk.org/wiki/display/AST/Local+Channel+Modifiers
Richard
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
I am trying to collect information regarding a bug report for
Elastix (http://bugs.elastix.org/view.php?id=1146). In this bug,
an user has asterisk-1.8.7 and dahdi-2.4.1.2. He is trying to make
an
outbound call through an ISDN trunk, by placing
Dial(DAHDI/g0/12345w) in
On a brand new system, I met an issue I've never met before.
My setup is :
debian 6.0.3
asterisk 1.8.8.1
dahdi 2.6.0
libpri 1.4.12
freepbx 2.9.0.4
TE420FB (with hardware EC)
This is the very first time I'm using Freepbx and the whole
configuration was first generated by a make
I´m trying attended transfer with asterisk 1.6 and see the message
WARNING[] features.c: Failed to play transfer sound! once in a
while when the transfer failes.
Any idea what can be happening?
Asterisk tried to play the features.conf xfersound configured sound file.
By default this is the
On a recently updated system , I'm now reading lines as this one
(never noticed them before):
[Dec 19 19:01:52] WARNING[10828]: chan_dahdi.c:11302 pri_dchannel:
Ring requested on unconfigured channel 0/0 span 4
My setup is:
Asterisk 1.6.1.18
Libpri 1.4.12
Dahadi 2.5.0.2
My card is a
OK, read all about the patch, thanks for the fix Richard.
I would like to apply this patch to my current 1.8.7.1 but I am
afraid I don't have a clue how.
https://issues.asterisk.org/jira/browse/ASTERISK-17557
Get the patch by following the reviewboard link in the issue and
download it
Hi there.
I started the console today to reload the extensions.conf file ; only
to be greeted with extremely verbose console.
Seems related to the zaptel card:
Example:
Supervisory frame:
SAPI: 00 C/R: 0 EA: 0
TEI: 000EA: 1
Zero: 0 S: 0 01: 1 [ RR (receive ready) ]
*
Summary:
I need to be able to ring multiple numbers in followme.conf at the
same time, even if one of the SIP extensions is unreachable.
This works in 1.4.8 but not in 1.8, just barfs and sends to voice
mail instead of ringing the other 2 extensions on the same line in
since version 1.4.12 the libpri package supports ETSI Explicit Call
Transfer feature:
http://downloads.asterisk.org/pub/telephony/libpri/ChangeLog-1.4.12
Does anyone know, how to use this feature in the dialplan? I can not
find any hints in the asterisk doc.
Which Asterisk version are
snip
if the chan_dahdi.conf transfer=yes option is set and a call is
natively
bridged on the same span.
This is interesting as I didn't know that.
What if a call comes in a BRI span in which one B-channel is already
used ? Is dahdi still capable to ask Explicit Call Transfer using
Hello all,
I'm having trouble setting the callerid name and number independently
with the following configuration:
Asterisk 10.0.0-beta2
DAHDI Version: 2.5.0 Echo Canceller: HWEC, MG2
libpri version: 1.4.12
Allstream PRI 23+D / dms100
Test cases:
(1) Using the form:
same =
When you perform an attended transfer, the extension of the person
transferring is displayed to the co-worker.
Can I override the caller ID to display the caller’s callerID during a
blind transfer?
Upgrade to 10.0 – this isn’t available in any of the 1.X flavors
because they had to
When you perform an attended transfer, the extension of the person
transferring is displayed to the co-worker.
Can I override the caller ID to display the caller’s callerID
during a
blind transfer?
Upgrade to 10.0 – this isn’t available in any of the 1.X flavors
because they had
As promised, here is a follow up on my quest to get CallerID
correctly
presented when forwarding calls to cellphones.
Here is a reminder of the issue at hand:
Alice (GSM handset) calls Bob (ISDN-connected Asterisk extension)
which forwards to Cory (GSM handset)
What I
2. As I feel specically new to this RDNIS concept, how should I set
CALLERID(RDNIS), before or after Answer() statement ?
It does not matter in this case. Asterisk v1.6.1 will keep both legs
of the call anyway.
If you ultimately want to get the call entirely off of your Asterisk
We have realtime queue architecture on asterisk 1.8.7.0
I noticed that when we change strategy from any other to 'linear' it
requires Asterisk restart take the change in effect.
I have one realtime queue '1' with strategy set to 'ringall' and I
change its strategy to 'linear'. Now when check
As promised, here is a follow up on my quest to get CallerID correctly
presented when forwarding calls to cellphones.
Here is a reminder of the issue at hand:
Alice (GSM handset) calls Bob (ISDN-connected Asterisk extension)
which forwards to Cory (GSM handset)
What I would like to get
As promised, here is a follow up on my quest to get CallerID
correctly
presented when forwarding calls to cellphones.
Here is a reminder of the issue at hand:
Alice (GSM handset) calls Bob (ISDN-connected Asterisk extension)
which forwards to Cory (GSM handset)
What I would like
Hi,
in asterisk 1.8.7.0 option 'd' works as expected: Pressing a key while
in ringing state puts the call to an one digit extension.
In asterisk 1.8.8-rc2 this is not working anymore. After doing a diff
on
the code it seems to me, that in version 1.8.7 there is an autoanswer
in
Hi,
in asterisk 1.8.7.0 option 'd' works as expected: Pressing a key
while
in ringing state puts the call to an one digit extension.
In asterisk 1.8.8-rc2 this is not working anymore. After doing a
diff
on the code it seems to me, that in version 1.8.7 there is an
autoanswer in
Hi Team,
i have been facing issues with sangoma card with 16 E1.
used LibSS7
asterisk 1.6
with 8 E1 the links are stable , but moment i add another card of 8 E1
for to support 16 E1. link keeps fluctuating
any idea why ?
Your 16th channel may be mismatched with the network.
Hi Team,
i have been facing issues with sangoma card with 16 E1.
used LibSS7
asterisk 1.6
with 8 E1 the links are stable , but moment i add another card of 8
E1
for to support 16 E1. link keeps fluctuating
any idea why ?
Your 16th channel may be mismatched
[trunkgroups]
[channels]
[my-phones](!)
usecallerid = yes
hidecallerid = no
callwaiting = yes
usecallingpres = yes
callwaitingcallerid = yes
threewaycalling = yes
transfer = yes
canpark = yes
cancallforward = yes
callreturn = yes
echocancel = yes
[trunkgroups]
[channels]
[my-phones](!)
usecallerid = yes
hidecallerid = no
callwaiting = yes
usecallingpres = yes
callwaitingcallerid = yes
threewaycalling = yes
transfer = yes
canpark = yes
cancallforward = yes
callreturn = yes
echocancel = yes
echocancelwhenbridged = yes
Upgrade to 1.8.7.1 There was a bug fixed recently (I think in 1.8.6,
but might have been 1.8.7) which caused Asterisk to sometimes not
transcode when it should.
A regression introduced in v1.8.7 broke the ability of the ./configure
script to generate the HAVE_PRI_xxx defines for ISDN. Fix
On 11-10-16 01:51 AM, Michael C. Robinson wrote:
[Oct 15 22:44:31] ERROR[29013] res_config_pgsql.c: PostgreSQL
RealTime:
Failed to connect database asterisk on 127.0.0.1:
[Oct 15 22:44:31] WARNING[29013] res_config_pgsql.c: PostgreSQL
RealTime: Couldn't establish connection. Check
I've got Asterisk 1.6.2.9 with RT SIP running, and for the most part,
it's working great.
However, once in a while, I'll get some strange output from sip show
peer ..
For example:
===
*CLI sip show peer 687F74D9848C-1
* Name : 687F74D9848C-1
Sorry for the resend, but i don't have got any response, so i try to
re-open the same problem.
Hello all,
Form 2-3 weeks i have some problems with incoming ISDN calls, it
interrupts after 1-2 minutes of call. I have tried to debug this with
pri set debug on span 1, i have noticied much
Actually it doesn't say AGI(async:script) it says AGI(async:agi)
and than continues further to setting up an AMI user so the script is
executed through the manager interface?? Than it says
AGI(agi:async).?? Well most importantly it says Cons of async AGI:
It is the most complex method of
In our office, we were running an Asterisk 1.6.2.14 machine with DAHDI
2.3.0.1, FreePBX 2.8.1 and an analog DAHDI card with 8 FXO ports. This
machine had several DAHDI trunks defined in the FreePBX interface,
each one containing a single DAHDI channel. It
also had a few outgoing routes
- Original Message -
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
I am using goautodial, I am using 20channels telcom PRI line and in my
server DIgium TE120 PRI card which is for 31 channel. with this
configuration
I am able to call from server . but problem whenever i restarted the
server that time is Asterisk is stop then I am not able to call
outside.
Can you please point me to the patch that you just made?
The patch is committed to v1.6.2 SVN branch.
Patch for v1.6.2 only.
r330490 | jrose | 2011-08-01 16:08:10 -0500 (Mon, 01 Aug 2011) | 12 lines
Asterisk 18103 - Fix reload crash caused by destroying default parking lot
Default parking
There is no event for Asterisk to recognize. The PROGRESS message just
says that there is an audio message available for the caller to listen
to. Asterisk just passes the indication to the peer channel and opens
the audio path. It is the caller who must recognize any audio message
that their
We enable pri intense debug with the standard asterisk PRI dialplan,
collected the logs and you can find the logs attached to the mail.
After the call was made, the called party cut the call, and asterisk
doesn't seem to recognise the event.
I can't make much sense of the logs given my
What I am looking for is:
If my IP Phone is related to a pickup group #1 and a call is ringing
at pickup group #2, so I can pickup the call that is ringing at group
#2 and I do not know its extensions?
In other words, how can I pickup any phone that is ringing without
knowing its
I would suggest Two B-Channel Transfer (TBCT), transferring to a
unique
number (received as DNIS by the other server) that would identify the
call
as transferred from the first server and, perhaps, the reason for the
transfer.
It looks like TBCT may not have been implemented in Asterisk
In article 296076780.5348.1310743930593.JavaMail.root@zimbra,
Richard Mudgett rmudg...@digium.com wrote:
I would suggest Two B-Channel Transfer (TBCT), transferring to a
unique
number (received as DNIS by the other server) that would identify
the
call
as transferred from
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