Why not use mysql?
Do something like this: exten = s,1,MYSQL(SELECT * FROM whatever)
If Asterisk database can handle large amount of data, I would prefer it because
of stability and speed. If Asterisk database can't handle that then I'll have
to use MYSQL (or MSSQL which I prefer because I
Hi Rizvarn,
I understand you know. Unfortunately I can't help you.
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
Ok, i'll try to make it cclear.
i think you r
On Thu, 13 Jul 2006 23:23:14 +0100, Steven wrote:
Hi Alex, that file doesn't exist on my system. Hmm that's most
likely a permissions problem then, since I'm still running httpd as
apache user rather than the suggested way of running it as the
asterisk user.
I really would rather have httpd
Hi all,
I have same problem.
1.Ring Back Tone can't hear.
2.extension telephone-SIPphone(H323) is NG but
SIPphone(H323)-extension telephone is OK.
It is one way sound.
extension telephone--PBX--GWAsterisk--SIPphone(H323)
|
Don Pobanz wrote:
I am running 1.2.9.1 and did not have any problems until setting up
queues. Within a day of doing queue logins/logouts our T1 DID trunks
(not PRI) stopped accepting calls from the local telco. Internal calls
though channel banks continued to function properly. A restart
If Asterisk database can handle large amount of data, I would prefer it
because of stability and speed. If Asterisk database can't handle that then
I'll have to use MYSQL (or MSSQL which I prefer because I already store CDR
to MSSQL).
With internal database I have code like this.
exten =
"...receiving digits from IVR
through dtmf and store it on a text file "
short idea:
1 IVR start
2 set(number=)
3 playback(press_digit_or_#_to_finish)
4 (pressed) set(number=${number}${digit_pressed})
5 playback(press_another_digit_or_#_to_finish)
6 if digit pressed goto(pressed[point
This is a FreePBX questio, not an Asterisk question. Better reprase it
as one. See below:
On Thu, Jul 13, 2006 at 09:29:23PM +0100, Steven wrote:
I'm trying to confirm the above statement, that although FreePBX is showing
that my extensions have been created they are not actually available in
On Thu, Jul 13, 2006 at 11:53:19PM -0500, Rich Adamson wrote:
shadowym wrote:
Thanks for the suggestions but I specifically asked for options OTHER than
a
second server. Your suggestions about disabling un-needed services are
good
though. I already do that. I am hoping someone has some
On Thu, Jul 13, 2006 at 05:43:09PM -0300, Gustavo Alejandro Gonzalez wrote:
Hi folks!
I'm setting up a debian box with kernel 2.6 running virtual private
servers with a TE110P card, at this moment i have two instances of
vps running and i want to run asterisk on every one,
I'm not sure
Hi, I was unable to build asterisk app_rxfax using asterisk-1.2.9.1 and spandsp-0.0.3
the makeing process kept giving me error:
app_rxfax.c: error: 't30_stats_t has no member named 'column_resolution'
where should I get the proper app_rxfax.c and app_txfax.c? thanks.
On 7/12/06, Maxim Vexler
I have an ACD asterisk system running, and if a call gets put through to an
agent and they hit the reject key (if they are busy), it puts the call to
their voicemail. I would like the call to stay in the queue and try another
agent. Is this possible?
Thanks for your help.
Dean.
On 7/12/06, Martin Joseph [EMAIL PROTECTED] wrote:
On Jul 12, 2006, at 10:18 AM, KC wrote:
I have the same problem before with 2 different providers. We resolved
this
by turning off qualify (qualify=no).
Of course this doesn't fix anything, it just stops the warnings from
showing up...
Hi all,
does Asterisk 1.2.7.1 supporting VAD? because i am
running my asterisk on VPS and i want to save
badwidth.
Khan,
__
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http://mail.yahoo.com
We have a customer routing calls trough a pri (digium board), our system
then terminates the calls in various places (let say we offer LCR).
When we route a call to an unreachable cellular phone we know it cause
we get a particular ${HANGUPCAUSE} so we don't bill that call even if
billsec is
Hello,
I have a Nokia Primicell GSM base connected to an Asterisk via FXS port
(TDM) but the unit do not detect correctly the answer of the calls
originated. Anyone has experience with this GSM unit ?
Thank you in advance !
Juan Carlos Valero.
___
Hello,
We were able to get asterisk going with
X100p cards on centos 4.2.
But could on centos 4.3 due to kernel
issues.
Anybody has faced this issue ?
And how do sort it out so that we
can use centos 4.3 ?
Thanks
Varun
___
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Hello,
Our asterisk server is on Centos 4.2
We want to use Astbill.
Astbill requires Drupal and mysql 5.
I could not find rpms mysql5 for centos.
We are getting mysql extensions issues
because of php-mysql.
How do we solve this ?
Any other billing software that similar
to Astbill ?
Thanks
Hi List, this is probably quite straightforward
I need to call a sip extension for 15 seconds, if unanswered
I then need to call the same sip extension and an additional sip extension for
a further 15 seconds, finally if the calls arent answered I need it to
go to a generic unavailable
When we route a call to an unreachable cellular phone we know
it cause
we get a particular ${HANGUPCAUSE} so we don't bill that call even if
billsec is 0 (the duration of the cellular is unreachable bla bla
message), but the customer says their system too records the
call as
0 and
Thanks All for your replies
A couple have mentioned backup routes, as I'm clueless on backup routes
How do I program the dialplan in extensions.conf to:
(a) try multiple provider to make an outgoing call based on current
latency between my * box and the different providers ?
(b) have if
W == Warren (mailing lists) [EMAIL PROTECTED] writes:
W So let's cut to the chase here... If you want to run a production
W server with queues, which version should you be running to get 30+
W days of uptime without needed a reset?
If you need IAX2, 1.2.9.1 is your only option in 1.2.x. We are
On Fri, 2006-07-14 at 16:50 +0800, ven wrote:
Hi, I was unable to build asterisk app_rxfax using asterisk-1.2.9.1
and spandsp-0.0.3
the makeing process kept giving me error:
app_rxfax.c: error: 't30_stats_t has no member named
'column_resolution'
where should I get the proper app_rxfax.c
Tzafrir Cohen wrote:
On Thu, Jul 13, 2006 at 11:53:19PM -0500, Rich Adamson wrote:
shadowym wrote:
Thanks for the suggestions but I specifically asked for options OTHER than
a
second server. Your suggestions about disabling un-needed services are
good
though. I already do that. I am
I'm trying to setup a call queue, but it keeps dropping calls that are
waiting for 1 min. Is there any way to make the queue unlimited amount of
time waiting? or is there a maximum?
Thanks,
Dean.
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Not True.. you can firewall the IAX port off so that not just 'anyone'
can get to it.
Also remember that the vulnerability found doesn't allow someone into
your system, it only takes the system down (which can still be bad).
However.. 1.2.9.1 crashes more by ITSELF then my unfirewalled asterisk
Yup.. well I did :) hehe.
On 7/13/06, William Piper [EMAIL PROTECTED] wrote:
I don't know about IAX, but what you are trying to do should work in the SIP
world. Obviously, it won't work for inbound... as it will send the call the
box that sent the latest registration.
There is always the ole,
Hi
all,
I have an ISDN
connection in Italy with MSN. On incoming call how can i check the dialed number
?
DNID varible could
works fine ?
Thanks in
advance
Giordano
___
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asterisk-users
Sorry - I misread it.
Have you ever had a network card fail in a way that did not lock up
every network-bound job on the system? I would think that it would be
unlikely that you could recover from that easily.
Yes, redundant drives with RAID-1 is good. If those drives are
hot-swappable and the
I replay to myself because I realize tha I also changed php vesion number: It works with php 4.4.0 and asterisk 1.2.4 and doesn't with php 5.1.4 and asterisk 1.2.9. Same code, same configuration.Any idea?
benqOn 7/13/06, Ben Q [EMAIL PROTECTED] wrote:
Hi,I was using cdr-csv through phpagi with
hi,can you describe what you want.../ArunOn 7/14/06, varun [EMAIL PROTECTED]
wrote:Hello,We were able to get asterisk going withX100p cards on centos
4.2.But could on centos 4.3 due to kernelissues.Anybody has faced this issue ?And how do sort it out so that wecan use centos 4.3
Check it ${EXTEN}
On 7/14/06, Giordano Grandis [EMAIL PROTECTED] wrote:
Hi all,
I have an ISDN connection in Italy with MSN. On incoming call how can i
check the dialed number ?
DNID varible could works fine ?
Thanks in advance
Giordano
___
I need to call a sip extension for 15 seconds, if
unanswered I then need to
call the same sip extension and an additional sip
extension for a further 15
seconds, finally if the calls aren't answered I need
it to go to a generic
unavailable VM.
My question is if the first sip extension is
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
It would look like this:
exten = s,1,MYSQL(Connect dbcxnid HOST USER PASS DB)
exten = s,n,MYSQL(Query resultid ${cbcxnid} SELECT\ \*\ FROM\ whatever)
exten = s,n,MYSQL(Fetch fetchid ${resultid} VAR1\ VAR2\ VAR3\)
exten =
Not sure what you want, but I have asterisk running
on Centos 4.3 and theres no problems.
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
I'm trying to setup a call queue, but it keeps
dropping calls that are
waiting for 1 min. Is there any way to make the
queue unlimited amount of
time waiting? or is there a maximum?
Hi
Make sure you are not setting the timeout parameter on
the Queue command. Failing that can you post the
Hello everybody,
I have an ISDN italian connection , with two number associated to one
line.
Is there a way to set which MSN should be used for an outcoming call? I
ask this because some companies give a different bill for each
telephone number.
I have tried with the SetCIDNum() application, but
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Even more important: base yourself on a distribution that fixes the
security problems for you. You will never have the resources to track,
test and apply all of those fixes, unless you're a full-time-job
security consultant.
What Linux
I cannot use it, I have the immediate=yes in my zapata, the extension will be
always 's'
Thanks again for all
Giordano
-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Marco Mouta
Inviato: venerdì 14 luglio 2006 15.25
A: Asterisk Users Mailing List
varun wrote:
Hello,
But could on centos 4.3 due to kernel
issues.
Anybody has faced this issue ?
And how do sort it out so that we
can use centos 4.3 ?
check out the wiki. CentOS 4.3 has an error that needs to be corrected
before you can compile zaptel.
I think it did not pick up hardware.
We googled and the hint was that we needed
to patch kernel. I hope I got it right.
BTW is there any issue with centos 4.3 and
hardware detection ?
Thanks
Varun
On Fri, 2006-07-14 at 18:46 +0530, Arun Kumar wrote:
hi,
can you describe what you want.
Anyone knows how to contact maintainers of Chan_gsm_bt?
They http://changsmbt.free.fr/ site has no contact details.
I believe I found the issue why it does not initiate SCO links
properly..
It looks to be a timing issue. It sends additional AT commands without
waiting for the responses for
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
does Asterisk 1.2.7.1 supporting VAD? because i am
running my asterisk on VPS and i want to save
badwidth.
If Asterisk supports VAD (or silence suppression) please tell me how to turn it
of! I don't care about bandwidth, I care about
I'm not personally sure, but if I recall correctly, the astDB is cleared
whenever the Asterisk server is stopped...
Anyone else?
On Friday 14 July 2006 9:30 am, Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED]
says...
It would look like this:
exten =
Is not immediate for use by FXS ports? If the line is ISDN then the
number would arrive in a setup message on the D-channel.
On Jul 14, 2006, at 8:34 AM, Giordano Grandis wrote:
I cannot use it, I have the immediate=yes in my zapata, the
extension will be always 's'
Thanks again for all
I have a question. We are going to attempt mixing some SIP
and H323 solutions here. The H323 is possibly going to be phased out sooner or
later but this is the first step. I have set up an Asterisk server that is also
running GnuGK so we have one machine doing both SIP and acting as a
Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
does Asterisk 1.2.7.1 supporting VAD? because i am
running my asterisk on VPS and i want to save
badwidth.
If Asterisk supports VAD (or silence suppression) please tell me how to turn it
of! I don't care about
I believe that with immediate=yes Asterisk does not know what number is
dialed and so that information is not available. Stop using immediate=yes.
Giordano Grandis wrote:
I cannot use it, I have the immediate=yes in my zapata, the extension will be always 's'
Thanks again for all
Giordano
On Fri, Jul 14, 2006 at 03:34:04PM +0200, Giordano Grandis wrote:
I cannot use it, I have the immediate=yes in my zapata, the extension
will be always 's'
Why would you set immediate=yes ?
--
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755 iax:[EMAIL PROTECTED]
Hi Andrea,
maybe you set fromuser= field inside your sip.conf??
Giorgio Incantalupo
Andrea Spadaccini wrote:
Hello everybody,
I have an ISDN italian connection , with two number associated to one
line.
Is there a way to set which MSN should be used for an outcoming call? I
ask this because
In the UK I have 'Call sign' from BT
on my house POTS line. This gives two different nos. to call the same physical
line. One no. gives the usual UK ring-ring, ring-ring, etc. pattern. The
other no. gives a ring,ring,ring north American style of ring. I would
like to be able to detect the ring
If the SIP or IAX peer are registered as extension 37, the generated
channels would be
SIP/37- or IAX2/37-
The last 4 digits are for making a difference in case that the same
peer is active in more than 1 call.
Regards
On 7/13/06, Reynaldo Baquerizo [EMAIL PROTECTED] wrote:
Hi
I've
Ciao Giorgio,
Hi Andrea,
maybe you set fromuser= field inside your sip.conf??
Right now I'm not at work, so I don't have an Asterisk box at hand, but
voip-info.org seems to suggest that fromuser is a sip-only option:
quote from=Asterisk config sip.conf
fromuser = from_ID : Specify user to put
Moises Silva wrote:
If the SIP or IAX peer are registered as extension 37, the generated
channels would be
SIP/37- or IAX2/37-
The last 4 digits are for making a difference in case that the same
peer is active in more than 1 call.
Regards
On 7/13/06, Reynaldo Baquerizo [EMAIL
Chris:
One issue you might find, depending on the SIP phone at 4902, is that
it will show a missed call for the first 15 second attempt. If 4902
answers in the second 15 second attempt, it will still show a missed
call, when the incoming call was actually answered. If extension 4903
answers the
Rushowr wrote:
I'm not personally sure, but if I recall correctly, the astDB is cleared
whenever the Asterisk server is stopped...
This is not correct.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty
Hi,
Has anybody configured caller id on a Sangoma analog FXO card?
Does it support both DTMF and FSK based caller id?
Thanks
Mun
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On Fri, 2006-07-14 at 23:44 +1000, Boris Bakchiev wrote:
Anyone knows how to contact maintainers of Chan_gsm_bt?
They http://changsmbt.free.fr/ site has no contact details.
http://changsmbt.free.fr/index.php?l=4
Here you can add that your LG phone is failing and in the notes section
add an
I did some poking around on the Googleweb and was unable to find a concise
answer to my situation. I have some guesses and some theories about what
will work and what might not work, but I'm sure that others have followed
this path before.
Currently we have a large number of customers that we
I seem to remember reading somewhere about a setting on Cisco gateway's
(with PRI) where you can have it send inbound (from PSTN) callerID name
via SIP to *. Does anybody know what that setting is? I searched the
archives and can't quite find the right set of keywords to locate that
info.
Anyone have any thoughts on this?
On 7/13/06, voiplist [EMAIL PROTECTED] wrote:
We have a situation where the wrong account code is being passed from
Asterisk to our AGI and then on into the accountcode field in the CDR.
Here is the situation, best I can explain it..
We have 3 user records in
Transferring a call from 80014154 to 2944051.
Asterisk is sending an ACCEPT message to the party transfer the call,
immediately followed by a DECLINED message. There appears to be NOTHING logged
in between. Anyone got any ideas?
Jul 14 08:06:23 VERBOSE[16688] logger.c: Transfer to 2944051 in
Thanks, it was the parameter on the queue command, it was set for mins, not
seconds.
Regards,
Dean.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jon Farmer
Sent: 14 July 2006 14:35
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Sounds to me that the incoming call is providing the wrong userid/password.
voiplist wrote:
Anyone have any thoughts on this?
On 7/13/06, voiplist [EMAIL PROTECTED] wrote:
We have a situation where the wrong account code is being passed from
Asterisk to our AGI and then on into the
Ciao Andrea,
when I read MSN I thought of Microsoft messenger and I thought I used
SIP protocolI never used it, sorry for my stupid advice ::)
Giorgio Incantalupo
Andrea Spadaccini wrote:
Ciao Giorgio,
Hi Andrea,
maybe you set fromuser= field inside your sip.conf??
Right
FYI: If you get no help here and want to search, in the US this is
called Distinctive Ring.
W
Paul Lakra wrote:
In the UK I have 'Call sign' from BT on my house POTS line. This gives
two different nos. to call the same physical line. One no. gives the
usual UK ring-ring, ring-ring, etc.
Erik Jacobs wrote:
Options (in no particular order):
1) Connect Asterisk to existing 6 PSTN lines using FXO. Connect existing
modems to Asterisk using FXS. Data speeds will probably be sub 14.4k, which
is not acceptable.
Yes, this would connect modems at slow speeds (or not at all) due to
Rich Adamson wrote:
[-snip-]
Then, back up your config files on something else and wait for your
server to be compromised. ;)
For cases where you expect something to be compromised, and potentially
overwritten, perhaps by an automated script, a trick that I have found
worthy of using is to move
All,
Anyone have any experience with the Digium TDM400P?
We have a Digium TDM400P up and working with asterisk. We've fxotune'd the
interface and pretty much eliminated all of the echo on the channel.
Our latest issue is that all calls that run over the zap channels sound
muffled and distant.
Ciao Giorgio,
when I read MSN I thought of Microsoft messenger and I thought I used
SIP protocolI never used it, sorry for my stupid advice ::)
Don't worry, the subject is not punctual. :)
--
Andrea Spadaccini
Multimedia Technologies Institute s.r.l.
The Centos-plus repository has mysql5 rpms.
W
varun wrote:
Hello,
Our asterisk server is on Centos 4.2
We want to use Astbill.
Astbill requires Drupal and mysql 5.
I could not find rpms mysql5 for centos.
We are getting mysql extensions issues
because of php-mysql.
How do we
I wish it were that simple..
We see the username coming in, it's in the channel etc..
We see the call come into one account and we see * set an account code
for another account.. Really..
It seems that it has something to do with the fact that accounts
registering from the same IP get mixed
Thanks Warren - I think I found the
answer here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+ZAP+channels
It is part of Zaptel configuration.
Warren (mailing lists)
[EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
14/07/2006 16:38
Please respond to
Asterisk Users Mailing List -
Paul Lakra wrote:
Thanks Warren - I think I found the answer here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+ZAP+channels
It is part of Zaptel configuration.
Glad to help. Why don't you update the wiki now and add dsomething
about the different name and where it is called
At 08:07 AM 7/14/2006, you wrote:
3) Connect Asterisk *AND* modems to PSTN using splitters. Does anyone know
what happens if someone is using a PSTN with the modem and Asterisk tries to
use an FXO? Is Asterisk smart enough to detect that the PSTN is currently
in use? Or is it like your little
I have deployed 5 Polycom 301 phones manually and I
would now like to provision them via my ftp server. My question is: How do I
get the current config files the phone is using off the phone? If I do an ftp provisioning
all the phones info will be lost true? So basically I need to get the
Hi,
My LinkSys SPA941 is working fine, but i always have a kind of Static
Metalic Noise, even if i'm not talking...
It becomes better when i reduce handset volume in the Phone. Also
There are lots of audio parameters in the webpage of the phone. like
audio volume, handset gain ...
Could you
Hi,
Assuming that you are using chan_capi-cm, the following from the README
file works for me with MSN.
snip
The Dial string
===
Example:
Dial(CAPI/ggroup/[callerid:]destination[/params])
Or:
Dial(CAPI/contrcontroller/[callerid:]destination[/params])
Or:
Ciao Peter,
Assuming that you are using chan_capi-cm, the following from the
README file works for me with MSN.
Thanks for your help, but I forgot to mention that I'm using mISDN, not
CAPI, so I can't use your Dial() options.
But I looked at the mISDN Dial() options, and I found this:
snip
How do I login to a polycom phone to retrieve the
config files?
Stephen Murphy
VP Operations
Cell: 604
790 3070
wVoIP: 604
638 8181
web:
expansivenetworks.com
501 905 West Pender St
Vancouver, BC V6C 1L6
In the interface Serial section add:
isdn outgoing display-ie
This will put the Display IE in codeset 0... if you need it in CodeSet 6 add:
isdn outgoing ie display codeset_0 shiftcodeset codeset_6
Mark
I seem to remember reading somewhere about a setting on Cisco gateway's
(with PRI) where
Aha - get rid of the leading comma for each entry..
= ,Front Desk
= ..
A.
On Jul 13, 2006, at 1:00 PM, Kevin Savoy wrote:
I've X'd out the extensions and passwords but this is all I have in
there.
Thanks
[default]
=,,Front Desk,,
has any one tried installing asterisk on a VPS
mechine ?
what is the minimum RAM and hard disk space needed
to install asterisk if i am going to install it on a VPS mechine ?
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Charles K Green wrote:
All,
Anyone have any experience with the Digium TDM400P?
We have a Digium TDM400P up and working with asterisk. We've fxotune'd the
interface and pretty much eliminated all of the echo on the channel.
Our latest issue is that all calls that run over the zap channels
,
recordingcheck|20060714-135108|115290.9581) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20060714-135108|115290.9581: Outbound recording
not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp(SIP/518-1acd
Someone using these phone Snom 300 with his own headset ?We got horrible static noise on them?P.D.Got silence as answer from Snom by now... maybe on holidays or with in theEuropean Football championship.
-- Adrià Vidaladriavidal at gmail.com
___
Were using Polycom 601s and I was wondering if
there was a way to do transfers by simply pressing the Transfer
button followed by the extension. Currently you need to hit Transfer,
extension, and then a transfer soft key. That extra soft key is
really confusing the users.
Two follow on questions:
1. Wouldn't that be for calls from * to the gateway out to the PSTN? I
want incoming calls from the PSTN to the gateway to deliver CNAM via SIP
to my * box.
2. How would I know if I want/need codeset 6?
[EMAIL PROTECTED] wrote:
In the interface Serial section
Hello all,Recently hear that specialised vendor ATCOM has introduced their new ATA with Infineon chipset.http://www.atcom.cn/En_products_AG188.html
Has anybody had any experience with it. Would like to know about this product and its quality.Infineon is better than PA1688 chip?Thanks in
Hi,
someone out there has a patch for chan_sccp to work with trunk? Sergio seems
to have abandoned the project and chan_skinny is still a long way from
beeing really usable :-S
Regards,
Andreas.
_
Need more speed? Get Xtra
[EMAIL PROTECTED] wrote:
Hello all,
Recently hear that specialised vendor ATCOM has introduced their new
ATA with Infineon chipset.
It's hardly surprising, afaict, the PA1688 is being discontinued, and
the AR1688 will not have an ATA firmware.
http://www.atcom.cn/En_products_AG188.html
Has
Hello all,
Wanted to toss out a question that I've been looking into for some time now
with no real results. When a variable is given a value in the dialplan, that
obviously will take up a little memory. If you're running a rather
large/complex dialplan, you may end up with variables you don't
Stephen,
As far
as I know, there's no way to pull the current config directly from Polycom IP
phones. You can connect to the internal web server on the phone, and get/set
some parameters through that interface, but not _all_ the settings. You really
need to start with a standard set of xml
Does anyone know if it's possible to transfer a caller, who's call has been
answered by an agent, to another phone?
We're seeing some very evil things when we try this. Spurious SIP messaging,
Asterisk locking up completely, etc.
Or... is the only way to do it, to press the '#' key?
Thanks,
I've setup an internal access through DISA as below but I get a
misleading announcement (no messages) in my mailbox.
[disa]
exten = 123,1,Answer
exten = 123,2,DigitTimeout,8
exten = 123,3,ResponseTimeout,30
exten = 123,4,Authenticate()
exten = 123,5,DISA(xx|disa-access)
[disa-access]
The phones will upload their XML files to an FTP server that you specify automatically upon boot. If you don't include other files there, there shouldn't be any configuration wipe, and you should be left with a set of XML files for usage.
AlexOn 7/14/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Alex,
The
phones don't upload their XML config files upon boot. They download them from
the FTP server.
Douglas.
-Original Message-From: Alex Robar
[mailto:[EMAIL PROTECTED]Sent: Friday, July 14, 2006 3:33
PMTo: Asterisk Users Mailing List - Non-Commercial
They will upload any changes you made at the phone. If a file does not exist on the server before rebooting while dumping the logs.On 7/14/06, Douglas Garstang
[EMAIL PROTECTED] wrote:
Alex,
The
phones don't upload their XML config files upon boot. They download them from
the FTP
They
will upload any changes made via the web interface, or via the menu's on the
phone itself.
They
upload an 'overrides' file. They do not upload a new copy of the original XML
files phone1.cfg and sip.cfg.
Therefore, if you had no XML (ie sip.cfg, phone1.cfg) files in the first
place,
The Asterisk development team is pleased to announce new releases of
Asterisk and Zaptel: Asterisk 1.2.10 and Zaptel 1.2.7.
These releases incorporate a number of bug fixes, and the Asterisk
release contains a new option to help avoid a potential denial of
service vulnerability in the IAX2
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