Ryan Booz wrote:
I have an Asterisk system for a small office with 12 extensions. For
parts of the incoming dialplan that go to “support”/”sales” we have
phones ring various people in an “additive” fashion. Example:
- snip --
exten = s,2,Dial(${E25}|18)
exten =
somebody has a hint for my problem plz?
This worked before but now it
doesn't.
Von: René Enskat [Teamware GmbH]
[mailto:[EMAIL PROTECTED] Gesendet: Dienstag, 20. Dezember 2005
15:05An: 'Asterisk Users Mailing List - Non-Commercial
Discussion'Betreff: Goto after Dial PRoblem
i want to
On Mon, 19 Dec 2005, [ISO-8859-1] Diego Andr?s Asenjo Gonz?lez wrote:
Hi everybody!
Jonathan wrote:
Hi,
I'm using a td400p card with an FXO port and asterisk 1.2.1 in South
Korea and asterisk isn't detecting when PSTN callers hangup.
I've gone through all the settings related
On Tue, 20 Dec 2005, AR Tarzi wrote:
could you please tell how it interfaces with Asterisk? Could I receive calls
into Asterisk? send calls out?
I've just downloaded it and am searching (unsuccessfully) for these on
Gizmo's site/software.
Gizmo isn't just a soft phone. Like Skype, its a
formerly, i was using a macro to jump around for zap calls. now im using the
plain old string manipulation which works. there is no difference between
the two but why did the macro suddenly create problems? the relevant part of
extensions.conf is:
(works)
exten = _4x,1,Dial(Zap/$[5 + ${EXTEN:1}])
David Allen wrote:
Hi,
I want to be able to receive incoming calls via H323 to Asterisk for SIP
Conversion and then send the Call to a seperate machine running SER to
route the call to the end user CPE. However if the call is not answered,
I want to be able to send that call back to the
Douglas Garstang wrote:
If you look at the changelist, most of the contributors are Digium employees.
The changelist reflects the committer, not the actual developer.
/O
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Trond Andersen wrote:
Does Asterisk have SIP support for PRACK/100rel?
No.
Asterisk does also seem to filter out some SIP header fields. Is there
a way I can force Asterisk to pass on ALL SIP header fields?
With the SIP_HEADER dial plan function you can reach all.
We do not parse all
Douglas Garstang wrote:
Realtime, as stated by Digium, does not work with sip users. This isn't
related to sip subscriptions.
Realtime certainly works with SIP users.
And yes, SIP subscriptions are lost in the current code. In 1.0, the
support for subscriptions was very poor so it wasn't
Douglas Garstang wrote:
You know, there isn't even a 'sip delete subscription' command available.
If there was, it might be possible to write some third party scripts
which
interact via the Manager Interface to control subscriptions.
Without even a delete function though, it's pretty much
Douglas Garstang wrote:
I don't think it's an expiry issue. When a phone reboots before the expiry,
it sends a new subscription to Asterisk. Asterisk doesn't remove the old one.
If subscriptions where keyed like registrations, the new one would just
overwrite the old one.
If the phone
Douglas Garstang wrote:
Anthony,
Your getting SIP SUBSCRIBE messages from the Polycom phones. Check the
mac-directory file and look for directory entries that have bw1/bw. For
each of these, the phone tries to send a SUBSCRIBE to the ct (contact)
field. If you don't have corresponding
1. SIP subscriptions are stored in memory and cleared when you do a 'reload'.
So, if you make any configuration changes and 'reload' you lose all your BLF
lights. People take this stuff for granted and expect it to work.
I think we cleared that up in previous postings. Saving the
In article [EMAIL PROTECTED],
Douglas Garstang [EMAIL PROTECTED] wrote:
Good grief. Your perceptions of reality are quite warped. It seems you
have a problem with me asking a lot of valid questions. Why exactly is
that?
No-one has a problem with the validity or number of your questions.
Many
Erik wrote
Create an waiting extension:
exten = _*XX*XX,1,wait(${EXTEN{1:2})
exten = _*XX*XX,1,dial($EXTEN{3:2})
Then dial using that waiting extension:
exten =
s,2,Dial(${E25}Local/*18*${E24}Local/*30*${E28}Local/*42*${E28}Local/*56*${E22})
This wil dial all the numbers at the same time,
In article [EMAIL PROTECTED],
Douglas Garstang [EMAIL PROTECTED] wrote:
Oh, and no... I can't switch to another solution. The decision was
made above my head to go with Asterisk. It's my job to make it do all
that 'enterprise-grade' stuff.
Aha, so *this* is where the big chip on the shoulder
On 21/12/05, Tony Mountifield [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],
Douglas Garstang [EMAIL PROTECTED] wrote:
Oh, and no... I can't switch to another solution. The decision was
made above my head to go with Asterisk. It's my job to make it do all
that 'enterprise-grade'
Does anybody know how to configure an ISDN card (Eicon Diva 2.0 S/T PCI)?
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On Wed, 21 Dec 2005, Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Douglas Garstang [EMAIL PROTECTED] wrote:
Oh, and no... I can't switch to another solution. The decision was
made above my head to go with Asterisk. It's my job to make it do all
that 'enterprise-grade' stuff.
Hi all,
I am testing my hands on asterisk , but got stuck. Let me tell you i am
only using its VOIP functionlities
I have registered the asterisk server at a remote proxy server. My clients
registered at asterisk server can make outgoing calls , but the calls made
from outside is not
#!/bin/shecho ---echo installing Eicon Diva CAPI ISDN supportecho this has not been tested!echo ---yum -y update
yum -y install isdn4k-utils-devel kernel-unsupported kernel-smp-unsupportedecho
Please stop replaying to mesage. If you plan to open thread do so by
writing mail to this address
asterisk-users@lists.digium.com
--
Tomislav Parcina
[EMAIL PROTECTED]
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In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
zttest result worst 99.86, average 99.95.
How do you run zttest?
--
Tomislav Parcina
[EMAIL PROTECTED]
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Ok, I have the following setup:-
Realtime CDR and Realtime Extensions.
I have two commands for each extension, the first is Set for the callerID, and the secondis Dialto tell asterisk where to divert the calls to.
Problem is, only the Set command is being added to the CDR and the Dial command
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
exten =
s,2,Dial(${E25}Local/*18*${E24}Local/*30*${E28}Local/*42*${E28}Local/*56*${E22})
This wil dial all the numbers at the same time, however eacht local number
waits a bit longer before executing the dial, hence it hunts :)
So
I'm having no luck playing MP3s both through MP3player and Musiconhold. When playing through MP3player I get -- Executing MP3Player("IAX2/192.168.1.40:4569-1", "/var/lib/asterisk/moh-native/test.mp3") in new stack Dec 21 03:36:48 NOTICE[15621]: chan_iax2.c:3105 iax2_read: I should never be
Install Quotefix. Google is your friend.
On Tue, Dec 20, 2005 at 10:56:13PM -0500, C F said:
In M$outlook click on Tools Options select Preference then
Email-Option then play around with on Replies and Forwards.
Again you forgot to RTFM.
On 12/20/05, Douglas Garstang [EMAIL PROTECTED]
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
As mentioned before, the new Sangoma card coming out over New Year is a
reliable and very good quality card.
It isn't out yet but you allready now that is reliable?
--
Tomislav Parcina
[EMAIL PROTECTED]
I have just setup asterisk on a debian sarge box. I am running
Asterisk 1.21 with AMP and chan_capi_cm 0.6.1 using a BT Speedway (AVM Fritz) ISDN card, connected to
a BT ISDN2e line. Currently we have 6 extensions configured all using CounterPath(Xten)
eyebeam softphone.
After many
I believe Kevin stated cvs would be around for possibly up to six monthss
and its updated once daily from svn. Should be more then adequate time
to get svn installed and tested.
Apparently there is a daily mirror copying svn - cvs, so it will not be
truly bleeding edge, but really close.
Tomislav Parcina wrote:
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
zttest result worst 99.86, average 99.95.
How do you run zttest?
cd /path/to/asterisk/zaptel
./zttest
Doug
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[EMAIL PROTECTED] wrote:
hi Steve
- 1ms ensured good EC convergence, using software EC. Adding delay
really degrades the performance of an EC adaption loop. It may be a
block size of 2 or
Yes, but...
As mentioned some pentiums are not that happy about a 1ms block and
some oy the
Doug Lytle wrote:
cd /path/to/asterisk/zaptel
../zttest
That should have read ./zttest with one period, not two.
Doug
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Buenas
Estoy por instalar el Sistema operativo Ubuntu
Alguien me puede informar donde consigo un manual completo
Gracias
Atte
Will
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Hi,
How can I turn off message Silence suppression ... on Asterisk console?
--
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Eugene Prokopiev
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Hello!
Is it possible to connect Siemens HiPath 4000 to Asterisk? What
equipment required on Siemens side? I mean IP not E1.
Sorry for asking here. Siemens-related websites use salesperson
language. There is no technical information.
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hi everyone,
i am trying to configure an * server to route calls from the PSTN to our internal PBX
This is the IDEA
Currently we have a PANASONIC KT1232 PBX that provides intercom calls and facility for call out lines on it (8 call out lines are pressent on it)
we also currently have 4 remote
I have seen in several places where the analog adapters you can use for faxes
and
modems have intermittent problems. Is this an issue with asterisk and are there
work-arounds?
We currently have side-stepped the issue by sharing the analog line between
the tdm
card and the fax machine
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
i have _partially_ implemented AOC into the libpri and chan_zap part of
asterisk (the IEs for AOC units are decoded and encoded and you will see
the AOC info on the console if you have increased verbosity to 5).
unfortunately it was
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
cd /path/to/asterisk/zaptel
./zttest
Thank you!
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Tomislav Parcina
[EMAIL PROTECTED]
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Hi,
I am new to Asterisk problems. Could anyone tell me how to install asterisk
with postgres cdr feature. Because I install asterisk 1.2 from newest Bristuff
and I do not have it
Thanks in advance
Cheers
Andrutto
That's nonsense.
The Eicon Diva 2.0 is a passive card and Hisax/mISDN must be used.
The 'eicon' drivers are for the active cards 'DIVA Server' only.
(I'm not sure if the source level RPM from eicon.com also provides the
driver for the passive cards...)
Also, I really would suggest not to use
On Wed, 21 Dec 2005, Andrew Gough wrote:
I have just setup asterisk on a debian sarge box. I am running Asterisk
1.21 with AMP and chan_capi_cm 0.6.1 using a BT Speedway (AVM Fritz)
ISDN card, connected to a BT ISDN2e line. Currently we have 6 extensions
configured all using CounterPath(Xten)
Hello,
Is Asterisk able to provide virtuals IPBX ?
I mean one hardware server which handle one IPBX per
enterprise .
Regards
Harry
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On Wednesday 21 December 2005 14:47, Dmitry Ivanov wrote:
Hello!
Is it possible to connect Siemens HiPath 4000 to Asterisk? What
equipment required on Siemens side? I mean IP not E1.
I am also to deal with HiPath, but 3750.
I came to the conclusion that buying anything on the HiPath side
is
My /var/log/messages log is very full of a lot of line regarding php agi
scripts, i.e
Dec 21 10:36:00 asteriskge03 php: agi Object
Dec 21 10:36:00 asteriskge03 php: (
Dec 21 10:36:00 asteriskge03 php: [request] = Array
Dec 21 10:36:00 asteriskge03 php: (
Dec 21 10:36:00 asteriskge03
Hi all,
I am testing my hands on asterisk , but got stuck. Let me tell you i am
only using its VOIP functionlities
I have registered the asterisk server at a remote proxy server. My clients
registered at asterisk server can make outgoing calls , but the calls made
from outside is not
Actually had to upgrade to latest SVN branch across the board (not just
sound files) and now all works fine. But I still agree with Matt
- if temp greeting is in place, maybe state Remove or Change Temp
Greeting instead of Record Temp Greeting.
Thanks for your help!
PedroOn 12/21/05, Kevin P.
Hi,
I would like to forward a calling from a specific
number to an extension.
The dialplan syntax should be:
exten=_*21*X.,1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:4})
exten=_*21*X.,2,Hangup
In my case,
the phone number to forward is 3473774567, and the extension is 105, hence the
syntax
On Wednesday 21 December 2005 15:11, [EMAIL PROTECTED] wrote:
Is Asterisk able to provide virtuals IPBX ?
I mean one hardware server which handle one IPBX per
enterprise .
Yes, just create separate context for each enterprise.
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That can only happen if EC is done in hardware. For most cards it is
being done in software. You don't seem to be grasping the hard real
time nature of how EC works.
Never looked to close into these standards, but my understanding of echo
cancel is that you need to buffer voice for 2
On Wed, 2005-12-21 at 14:15 +0100, [EMAIL PROTECTED] wrote:
My /var/log/messages log is very full of a lot of line regarding php agi
scripts, i.e
Dec 21 10:36:00 asteriskge03 php: agi Object
Dec 21 10:36:00 asteriskge03 php: (
Dec 21 10:36:00 asteriskge03 php: [request] = Array
Dec 21
Headers like Supported or Require are not passed on with SIP methods
like INVITE of 183 Session Progress. In my dialplan I am able to read
and add headers, but I think it only works to add one header in each
message?
Any tips on how I should move forward to make sure all headers are
transmittet
Hello
I wonder if it is possible to generate MWI (Message Waiting Indication) when using mgcp.
Lets say a user has got a voicemail then I want current user to get notifiedis that possible?
If this is possible what should I write in *.conf-files?
Thanks for your help!
Regards /MiMaCa
Hi, Thanks for the reply...
The clicks are on every call and every few minutes or so, I guess you
could call it regular intermittance? :)
The only option for DTMF on my IAX phones are inband and
outband-neither of which will respond to a menu prompt. I discovered I
had to use outband just
What is the meaning of a SL greater than
100%?
lv09*CLI show queue
cobrancacobranca has 0 calls (max unlimited) in
'leastrecent' strategy (6s holdtime), W:0, C:69, A:2, SL:102.9% within
45s
Dov
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You need to manage this variable on Asterisk DB in order to make call
forwarding.
It must be done in extensions.conf . In voip-info you could find how to
do that.
Androtech wrote:
Hi,
I would like to forward a calling
from a specific number to an extension.
The dialplan
On Thursday 22 December 2005 04:45, abhishek wrote:
Hi all,
I am testing my hands on asterisk , but got stuck. Let me tell you i am
only using its VOIP functionlities
I have registered the asterisk server at a remote proxy server. My
clients registered at asterisk server can make
Hi,
we have interconnected Asterisk with a HiPath4000 V1.0
using a H.323 Trunk. You have to install the oh323
channel from [1]. On your HiPath4000 V1.0 or V2.0 you
need a HG3550 board for IP-Trunking.
If you have the version 3.0 then the HiPath supports
SIP-Trunking but i have not tested it yet.
In sip.conf file in general context is it right to put port or
bindport? Is this version specific or is the same for all Asterisk
versions?
--
Tomislav Parcina
[EMAIL PROTECTED]
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I already tried to switch proxies, but it did not help.
But I never tried to change signaling.
I will check this out later today.
thx
Bart,
We have has similar issues with BroadVoice in the past. From what I
understand they had problems with DTMF depending on which proxy you
register
to.
In my case, the phone number to forward is 3473774567, and the extension is
105, hence the syntax should be:
exten =
3473774567,1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:105})
Let me explain you what this syntax is saying :
presuming this number is called from extension 7001
- Put in the DB, under
Hi,
why does asterisk always give the state Unabailable?
asterisk-er*CLI
-= Registered Asterisk Dial Plan Hints =-
31 : SIP/31exten = 31 State:Unavailable Watchers 0
23 : SIP/23exten = 23 State:Unavailable Watchers 0
22
New Asterisk 1.2.1 with AMP
Voice ok by MOH plays for a second, then pauses for a second, eventually gives
up all together. Any Ideas??
Regards
Andrew Gough
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Es un timeout...necesitas incrementarlo...en la libreria de unicall existe un archivo qe se llama mfcr2.c...
#define BLOCKING_RELEASE_TIME 450
#define
ANSWER_GUARD_TIME
100
#define
DEFAULT_T1
5000 -Dale una valor mas alto...2 por ejemplo
#define
DEFAULT_T1A
150
#define
DEFAULT_T1B
Hi Tim,
probably my information are not quite clear; 3473774567 is a mobile phone
and 105 is an extension. I would like to forward any outside calling from
this mobile (3473774567) to the extension 105.
When you talk about DB, what do you mean exactly?
Could you be so kind to post some
Is it possible from within the dialplan to determine if an Agent channel is
already a member of
a queue? Would like to use this as part of a check that will play a message
if the agent is the
last person to log off the queue.
I can sorta do it by using AddQueueMember and checking
complete restart of asterisk
Kib Eki wrote:
Hi,
why does asterisk always give the state Unabailable?
asterisk-er*CLI
-= Registered Asterisk Dial Plan Hints =-
31 : SIP/31exten = 31 State:Unavailable
Watchers 0
23 : SIP/23exten = 23
Erik,
This looks like a great
option! Thanks. Im wondering about two things (a bit of a
newbie I guess) and am hoping for a bit more clarity.
1.
In your example is Local
the context? Ive seen Local/ referenced in the
documentation online, but dont understand what it is. Im
sure this
Doug,
If you stop complaining and listen to what people are saying, you would be
able to accomplish your goals. Some of your points have merit, but you are
asking for help in all the wrong ways. Please remember, this list is for
users, not developers. The user community is quite extensive in
I see what you mean and already have the option turned on. However the entries
in astdb
are a bit odd:
//Agents/40042: [EMAIL PROTECTED];4004
//Agents/4005 : [EMAIL PROTECTED];4005
//Agents/4011
Hi Ollie. No, Realtime does not support the use of multiple Asterisk systems
all accessing SIP users. Digium, including Kevin Fleming, have confirmed this.
-Original Message-
From: Olle E. Johansson [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 21, 2005 1:30 AM
To: Asterisk Users
Ollie.
It does now. There's a new chan_sip.c available that I tested last night. Phone
reboots, and subsequent new subscriptions do not cause accumulations of sip
subscrptions on Asterisk.
-Original Message-
From: Olle E. Johansson [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December
Ollie.
I said it in a previous post. Just to make it clear... :) Realtime does not
support having multiple Asterisk systems all accessing a central database for
SIP users/registration information. Digium have admitted it doesn't work and
have said that it will take the better part of a year to
Can someone help me understand how to identify frame slips from pri debug,
pri intense debug, or any other method?
I have 2 Dell 18xx servers (at different locations) connected to the PSTN
via PRI's. One is connected with a T100P and the other via a TE110P. Both
are in production and seem to
I should really make those alterable when the channel is set up. They
just switch with the national variant right now, and that doesn't give
enough flexibility for all the weird stuff people face. A lot of places
require a long timeout to give a call time to be routed and handled.
However,
OK I know this is going to be a silly one, but I'll ask anyway. Asterisk 1.2.1,
AMP with SIP Eyebeam 6 line Softphones and CAPI ISDN. When I am on a call, and
another call comes in it goes straight to voicemail, without alerting me. I
would like it to ring, and let me handle both calls if I
Eugene Prokopiev wrote:
Hi,
How can I turn off message Silence suppression ... on Asterisk console?
By turning it off in the phone's setup.
/O
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Will. Here you have to post questions, news or stuff directly related
to Asterisk PBX. For information about ubuntu linux distro please look
in google or ubuntu.com
regards,
Will, aqui debes escribir preguntas, noticias o cosas directamente
relacionadas con Asterisk. Para informaci'on
[EMAIL PROTECTED] wrote:
Hello,
Is Asterisk able to provide virtuals IPBX ?
I mean one hardware server which handle one IPBX per
enterprise .
A lot of service providers do that. One caveat is the parking function,
that only supports one parking lot for all virtual PBXs.
/O
Trond Andersen wrote:
Headers like Supported or Require are not passed on with SIP methods
like INVITE of 183 Session Progress. In my dialplan I am able to read
and add headers, but I think it only works to add one header in each
message?
You can add several headers, but I advise you not to
Hi,
can anybody help me with the allcall.agi script from
http://www.voip-info.org/wiki/view/Polycom+auto-answer+config
When I run the script offline from asterisk it seems to work but inside my
dialplan it does nothing - it does not write any calling file.
-- Executing AGI(SIP/31-ee46,
The workaround for the parking limitation is app_valetparking.so from
http://www.pbxfreeware.org/app_valetparking.c
instructions on how to install is on the wiki.
On 12/21/05, Olle E Johansson [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote:
Hello,
Is Asterisk able to provide virtuals
Douglas Garstang wrote:
Hi Ollie. No, Realtime does not support the use of multiple Asterisk systems
all accessing SIP users. Digium, including Kevin Fleming, have confirmed this.
It has happened that Kevin has been wrong about the SIP channel, you
know. Or you might have asked the wrong
Douglas Garstang wrote:
Ollie.
It's Olle :-)
I said it in a previous post. Just to make it clear... :) Realtime does not support
having multiple Asterisk systems all accessing a central database for
SIP users/registration
information. Digium have admitted it doesn't work and have said
On Wed, Dec 21, 2005 at 10:45:29AM -0500, C F wrote:
The workaround for the parking limitation is app_valetparking.so from
http://www.pbxfreeware.org/app_valetparking.c
instructions on how to install is on the wiki.
On 12/21/05, Olle E Johansson [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED]
s/should/shouldn't/
--
Chris
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Douglas Garstang wrote:
Maybe some from Digium will read this email and it will make a 0.001%
contribution towards some of these things being fixed. Oh, and no... I can't
switch to another solution. The decision was made above my head to go with
Asterisk. It's my job to make it do all that
Hi,
I'm experiencing a strange issue with static realtime. Seems that some
values belonging to 'general' category (like, for example, rtptimeout,
rtpholdtimeout, realm) are ignored. Running asterisk 1.2.1, I've tried
both res_odbc and res_mysql (that one from asterisk-addons tarball)
without luck.
anybody know a discussion group in spanish of asterisk?
to suscribe me?
- Original Message -
From: Jason Becker [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, December 21, 2005 11:22 AM
Subject: Re:
I've got realtime sipfriends running
pretty well. One this I noticed is that if I make a change to the DB, the
server's 'sip show peer 1234' never shows the update until after I do a 'sip
reload'. My info, cvs-head from 12/17/05on a Dell 1750. the mysql db is
on a seperate server, as so is
Checked the code and the queuename is not included regardless. I looked
at the public SVN and it appears to be the same there as well. So
I will have to come up with an alternative solution in the mean time.
--johann
Johann wrote:
I see what you mean and already have the option turned on.
I see what you mean and already have the option turned on. However the
entries in astdb
are a bit odd:
//Agents/40042: [EMAIL PROTECTED];4004
//Agents/4005 : [EMAIL PROTECTED];4005
//Agents/4011
Hi to everybody, I have set up Asterisk that when I am in conversation pressing * 1 activates in an automatic way the recording of the call. (as expressed in feature.conf) the file which is registered is a wav file with a name produced in an automatic way by Asterisk. Is there the possibility to
Hi to all,
the following is the last thing we see from Asterisk befor it crashes:
$$$ find_chan_holded: No channel found for oad:017670014533 dad:7051538
-- ch-state CONNECTED, bc-holded 0
$$$ Bchan deActivated addr 51400101
-- cause 16
I SEND:RELEASE port:1 pid:88 mode:TE addr:51400101
Hi to all,
the following is the last thing we see from Asterisk befor it crashes:
$$$ find_chan_holded: No channel found for oad:017670014533 dad:7051538
-- ch-state CONNECTED, bc-holded 0
$$$ Bchan deActivated addr 51400101
-- cause 16
I SEND:RELEASE port:1 pid:88 mode:TE addr:51400101
On Wed, 21 Dec 2005, Andrew Gough wrote:
I have just setup asterisk on a debian sarge box. I am running
Asterisk
1.21 with AMP and chan_capi_cm 0.6.1 using a BT Speedway (AVM
Fritz)
ISDN card, connected to a BT ISDN2e line. Currently we have 6
extensions
configured all using
Hi to all,
I am having problems with the caller id using IAX. The
caller id feature does not function for an incoming
IAX2 call when the incoming caller hides the caller
id.
The caller id is presented as blank on my phone
instead of the number i set it to be. It works fine
otherwise and also
How do you configure aastra.cfg to download directory list entries to
each phone? The Aastra documentation is very sketchy. Anyone have an
example???
You can use the Aastra Web UI (Operation-Directory) or the
configuration files
(aastra.cfg and mac.cfg) to download the Directory List.
You
Not sure if this was just an error in your email, but your priorities go
1, 1, 2. Make 1, 2, 3?
Also, What version asterisk? I think it's become
Set(CALLERID(name)=blah) and
Set(CALLERID(num[ber?])=12345) lately. Maybe see if these work
differenly or better ;)
Moj
ahmed kassim wrote:
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