Oh.. :/ too bad..
I'll have to look at the source..
bye,
Tomislav
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of T. Shaw
Sent: Thursday, May 11, 2006 11:20 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE:
Hello all,
Iam using this Asterisk server since three weeks and i have to clarify some thing about Asterisk
here is my problem Iam trying to use my Asterisk as a gateway to pstn
and SER as a proxy and redirection server so,here in SER i had added
three or four users by using the command serctl
Hi,
Douglas Garstang wrote:
We are using a backend MySQL database for call flow, not user agent
registration info. Just how, exactly, is a backend database going to
replicate registration data between Asterisk servers? Realtime has
been documented NOT to work with multiple Asterisk systems. If
Hi,
I wonder if anyone is using Digium's TE110P card on an E1 connection.
I have been try to, but so far it wasn't much of a success.
It only works more or less in EuroISDN as PRI CPE.
And even that config gives me some trouble with channel negotiation.
My current config:
zaptel.conf:
Hey all here's an update.
I do care to thank everyone for your information on BRI interfaces
that operate in USA/NA. I know the responses are were limited, but the
selection of hardware is also limited. (Shame because BRI would fit my
needs perfectly). To continue, it's now been over 4
Hello,
Digium does not provide snmp support to monitor their
cards !
Anybody has tried Sangoma product A104 Quad T1/E1 or
others ?
Regards
harry
___
Yahoo! Mail réinvente le mail !
did you include automon = *1 in your features.conf ?? it should be somthing like this [featuremap]automon = *1 --Giridhar Bandi
On 5/12/06, Dave Morrow [EMAIL PROTECTED] wrote:
Thanks for the response.How would I change the DTMF transfer mode?David MorrowTechnical Systems LeadAutodata Solutions
Bruce Reeves wrote:
I'm trying to add ooh323c to my asterisk 1.2.7.1 http://1.2.7.1
install and did an svn update of asterisk-addons and followed the readme
in asterisk-ooh323c and I get through the .configure with no errors. But
make causes:
rpath /usr/local/lib -L./ooh323c/src
Hi
I'm struggling with setting up SIP domains.
If I specify a domain and a context in [general], that context overrides
any set in type=peer blocks elsewhere. This results in incoming calls
from PSTN gateways I use arriving in the wrong context.
If I don't specify a context (which the docs
Help! I'm having an odd problem that I'm not seeing in any of the list
archives and thought I'd ask and see if anyone can help.
I've got Asterisk behind a NAT and an SPA-841 SIP phone behind a different
NAT. Everything works fine (incoming calls ring, outgoing calls work, audio
in both
Ok, the script below (meetme.agi) will prompt for a valid pin up to 3 times. If
the pin matches one of the defined Admin pins, it will set the dialplan
priority to 10 and exit, if User, sets to 20 and exits. Otherwise Hangs up.
In the case of admin, these MeetMe options are used:
a - Admin mode
Hi,I finally found an ATA which works really well with asterisk and its application alarmreceiver. Frankly it works just like the TDM card. It is Soundwin S800 series ATA.CheersAndrew
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For BT in the UK I use :-
zaptel.conf
loadzone = uk
defaultzone=uk
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
zapata.conf
[trunkgroups]
[channels]
language=en
context=did
priindication = outofband
usecallerid=yes
cidsignalling=v23
usecallingpres=yes
sendcalleridafter=1
switchtype =
Hello,
Try http://www.freepbx.org, its written in PHP with mysql at the end, it also uses .conf files for configurations.
regards,
Umair bari
On 5/8/06, moona ather [EMAIL PROTECTED] wrote:
Hi,I have to make a web-based management interface of configuring asteriski wanted to know if it is
Hello everyone.
I've got a HFC ISDN card that I'm using with chan_misdn and it basically
behaves like crap. Echo is waaay worst then echo I get TDM400 card,
sound is choppy (there other side is allays complaining about sound
interruptions) and to top it all it detects fake DTMF's all the
freepbx has been improved since then, and I believe if you edit/add something in original asterisk .conf files, it stays there. I've tried it long ago when it was called AMP and it worked.
regards,
Umair Bari
On 5/9/06, Emmo ather [EMAIL PROTECTED] wrote:
Hello,In older version of freebpx if you
On Fri, 2006-05-12 at 19:50 +1200, Richard Scobie wrote:
[snip]
rpath /usr/local/lib -L./ooh323c/src -version-info 1:1:0 -lpthread
make: rpath: Command not found
make: [libchan_h323.la] Error 127 (ignored)
My previous mail mentioned that this had been posted recently on the
list,
Hi!!
Id like to confgure my extensions.conf
file in order to handle overlap!!
Is it possible?
What should be changed?
Thanks by advance
Nicolas L.
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Asterisk-Users mailing
Hello,
IMHO, there are 2 ways to do this,
1) You can connect your VoIP modem to your asterisk box using x100p FXO card, you'll need to get one and install it properly.
2) Get SIP/IAX account from any VoIP provider and use it with asterisk.
Hope this helps.
Regards,
Umair Bari
On 5/12/06,
I don't see anything obviously wrong with your configs.
You don't want relaxdtmf. That can cause the problem, not fix it.
Hi Eric,
at the begining - Thanks for your help.
relaxdtmf is not written in my config, so it should be at the default, i
guess i remember default is yes ?
However,
I don't know which version you downloaded, but if you can get the
source from CVS on Sourceforge, and build it yourself, you may have
more luck - The CVS version has code contributed from several sources,
and is slightly better that the packaged version.
Cheers,
Steve
On 5/12/06, Tomislav
I've got a HFC ISDN card that I'm using with chan_misdn and
it basically behaves like crap. Echo is waaay worst then echo
I get TDM400 card, sound is choppy (there other side is
allays complaining about sound
interruptions) and to top it all it detects fake DTMF's all the time.
Is this a
On 5/11/06, Tim Robinson [EMAIL PROTECTED] wrote:
There is a lot of junk in your zapata.conf that you do not need, as it
relates to analogue lines. This might be causing confusion?
I have tried a similary config to yours, doesn't helps. I haven't got
this problem on an E1, just on the newer
2006/5/12, Crazy Boy [EMAIL PROTECTED]:
I am unable to understand where to give above mentioned values? What
configuration files I should use to implement this using the Vebtel SIP
provider? Do I need to provide any more values to implement this using
Asterisk from Vebtel?
In addition to your
I've got a HFC ISDN card that I'm using with chan_misdn and it basically
behaves like crap. Echo is waaay worst then echo I get TDM400 card,
sound is choppy (there other side is allays complaining about sound
interruptions) and to top it all it detects fake DTMF's all the time.
Is this a
Woodoo People .pGa! wrote:
I've got a HFC ISDN card that I'm using with chan_misdn and it basically
behaves like crap. Echo is waaay worst then echo I get TDM400 card,
sound is choppy (there other side is allays complaining about sound
interruptions) and to top it all it detects fake DTMF's
Hi aLL
I have an [EMAIL PROTECTED] box running.
When i register via SIP to the box with 2 phones both behind the same firewall
the registration goes through fine and I can see in the realtime database AND
that an IP and port has been entered for each extension, its obviously same IP
(the
Chris Bagnall wrote:
I've got a HFC ISDN card that I'm using with chan_misdn and
it basically behaves like crap. Echo is waaay worst then echo
I get TDM400 card, sound is choppy (there other side is
allays complaining about sound
interruptions) and to top it all it detects fake DTMF's all the
I also hadan issueto getmISDNworking with HFC cards without problems.
Therefor I switched to zaphfc (use bristuff), this is working perfectly with HFC cards. It does everything I need including MSN support and without problems, even with multiple HFC cards.
So my advice is to get rid of mISDN
Dear Colleagues,
I have 2 phones Cisco 12SP+ connected to my asterisk box 1.2.6 and
SCCP channel version: 20060408.
When a call is generated no audio pass through the phones, neither if
i call from a 12SP+ to another nor calling between 12SP and other
phone (ex. an x-lite). Only sometimes works
I'm not sure if you have considered this, but if you were using SIP
between the Asterisk servers you can definitely achieve this using
X-headers.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Thursday, May 11,
Eric, thank you very much. But It could help in this case me?
Regards
Josué
2006/5/12, Eric ManxPower Wieling [EMAIL PROTECTED]:
Josué Conti wrote: Dinesh, very obliged for the attention. I am using version 1.0.9 of asterisk
and it is really all good with this version, only this case of atxfer
hi
does anyone have an idea how it could be possible to do email - fax
gatewaying with asterisk + app_txfax, but still keep track of who
sent the fax? i've thought a little about smtp auth, but it doesn't
look too easy to integrate smoothly with asterisk
roy
Hi all,
i am desperating, trying to configure an OptiPoint410
with the hint priority.
Here what i have...
OptiPoint410std- exten 2001
X-Lite - exten 2002
But unfortunately no LED ON on my OptiPoint410
sip.conf
[2001]
type=friend
context=local
host=dynamic
dtmfmode=rfc2833
incominglimit=1
[EMAIL PROTECTED] wrote:
Hello,
Digium does not provide snmp support to monitor their
cards !
That's like saying Toyota doesn't provide gas with their cars. You can
setup snmp with in linux and have it execute commands that you want to
determine whether or not the hardware is functioning
Hi all,
I've been having problems with my A20002D lately - callers from the PSTN
don't hear me when I answer, but I hear them. Disabling echo
cancellation in zapata.conf brings the audio (and echo) back. This used
to work fine, until two days ago.
The only weird thing in the logs is this:
Hello,
In a fit of optimism I recently purchased a X100-FX v2
(http://x100p.com/products_2.htm) despite the lack of reviews I was able
to find on the device. The feature set made it hard to resist. I have
since been experiencing audio quality issues with it.
Do any other mailing list
Hi all,
I've been testing various codecs to eliminate choppiness that I
sometimes get on my Asterisk IAX2 DSL provider (Exgn) connections,
and Speex seems to work the best, so far - but Speex seems oddly unpopular.
Can anyone share their experiences with Speex (good and bad)? Is anyone
Hello, there is a way to send notification(not
email) when it's received an voice mail? Maybe a SIP message to
inform?
Best REgards
Ever Zalazar
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Asterisk-Users mailing list
To
Yes. I did.
David Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodatasolutions.com
Tel: (519) 963-3020
Fax: (519) 451-6615
Lead, follow or get out of the
way!
This message has originated from Autodata Solutions. The attached material is
the
On 5/12/06, Ben Holt [EMAIL PROTECTED] wrote:
Hello,In a fit of optimism I recently purchased a X100-FX v2(http://x100p.com/products_2.htm) despite the lack of reviews I was ableto find on the device.The feature set made it hard to resist.I have
since been experiencing audio quality issues with
All I see when I press *1 is
-- Attempting native bridge of
SIP/8001-252e and SIP/3020-5171
I still cannot make this work.
David Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodatasolutions.com
Tel: (519) 963-3020
Fax: (519) 451-6615
Lead,
Josh,
Thank you!
I think the AGI could be bypassed by doing a realtime() to get the PIN from
mySQL, also returning the variable that defines admin or user and jumping in
the dialplan accordingly. Otherwise I would just end up having the AGI do the
query because there is a need to store the
Hello,
How can I park a call or put on hold a caller from an
analogue to sip agents ?
PSTN===FXO/asterisk=sip agents
When I press hold key or #800 the channel is hangup ??
Harry
Regards
Hi,
Last time I had this problem was following a unclean powerdown and the
solution was:
- Kill Asterisk
- Stop wanpipe
- cd /etc/wanpipe/wan_ec
- In there there should be 2 files:
wan_ec_pid
wan_ec_socket=
- Delete those files
- Perform a reboot of
I just bought a couple of these units. It seems to work fine but I could
not really test it as the phones were too close together so could not
get a clear idea of the call quality.
Phoning comedian mail seemed fine and certenly acceptible considering
the gsm codec was being used.
One minor
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Dr. Michael J. Chudobiak
Sent: Friday, May 12, 2006 8:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Speex fans?
Hi all,
I've been
Hello,
How can I park a call or put on hold a caller from
an
analogue to sip agents ?
PSTN===FXO/asterisk=sip agents
When I press hold key or #800 the channel is hangup
??
Harry
Regards
Can someone give me some direction on automon filenames? I
would like them to be the dialed number if possible. I saw a patch available
for changing this but havent quite figured out how to use it.
Can someone point me in the right direction?
Thanks,
Dave
Same problem with audio quality. Got rid of them. Also the context
line only allowed 12 characters and we need more than that for some
installations, I didn't want to have to rename 100 contexts to less
than 12 characters.
On 5/12/06, Gareth Blades [EMAIL PROTECTED] wrote:
I just bought a
Damon Estep wrote:
Can anyone point me to a sample or information on using MeetMe like this?
Conference room is set up with 2 PINs, one for the moderator and one
for the participants.
Participants get music until the moderator joins (to avoid wild,
un-moderated tangents).
Call is
Asterisk 1.2.7.1 and Zaptel 1.2.5
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile
Sent: Thursday, May 11, 2006 6:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Paging and Auto Answer on Grandstream
Hi!
Anyone meet with the following problem?
May 12 15:51:44 WARNING[14399] channel.c: Unable to find a codec translation
path from ulaw to unknown
May 12 15:51:44 WARNING[14399] app_txfax.c: Unable to restore read format on
'SIP/neopost1-8083'
May 12 15:51:44 WARNING[14399] channel.c: Unable
I believe the hint priority must be in the same context as the phones
extension number, in this [local]
On May 12, 2006, at 6:58 AM, richard Coco wrote:
Hi all,
i am desperating, trying to configure an OptiPoint410
with the hint priority.
Here what i have...
OptiPoint410std- exten 2001
Last time I had this problem was following a unclean powerdown and the
solution was:
- Kill Asterisk
- Stop wanpipe
- cd /etc/wanpipe/wan_ec
- In there there should be 2 files:
wan_ec_pid
wan_ec_socket=
- Delete those files
- Perform a reboot of your
does anyone have an idea how it could be possible to do email - fax
gatewaying with asterisk + app_txfax, but still keep track of who
sent the fax? i've thought a little about smtp auth, but it doesn't
look too easy to integrate smoothly with asterisk
i don't know what your problem
Hello,
How can I park a call or put on hold a caller from
an
analogue to sip agents ?
PSTN===FXO/asterisk=sip agents
When I press hold key or #800 the channel is hangup
??
Harry
Regards
Tom Vile wrote:
Same problem with audio quality. Got rid of them. Also the context
line only allowed 12 characters and we need more than that for some
installations, I didn't want to have to rename 100 contexts to less
than 12 characters.
Which audio codecs were you using? I'm using g729 to
There is some additional functionality coming in future firmware
versions. See
http://www.voip-info.org/wiki/index.php?page=Asterisk+Paging+and+Intercom
On Fri, 2006-05-12 at 14:54, Forrest Beck wrote:
Asterisk 1.2.7.1 and Zaptel 1.2.5
-Original Message-
From: [EMAIL PROTECTED]
Hello,
How can I park a call or put on hold a caller from
an
analogue to sip agents ?
PSTN===FXO/asterisk=sip agents
When I press hold key or #800 the channel is
hangup
??
Harry
Regards
This is what I use:
[ext-paging]
exten = PAGE203,1,Set(__SIPADDHEADER=Call-Info: answer-after=0)
exten = PAGE203,n,Set(__ALERT_INFO=Ring Answer)
exten = PAGE203,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = PAGE203,n,Dial(SIP/203,5)
exten = Debug,1,Noop(dialstr is LOCAL/[EMAIL PROTECTED])
exten
We did communicate this to the manufacturer and they fixed 1 issue
with bad power supplies. We tried multiple codecs but it was still
unreliable, so we went back to the IAXy and no issues. All calls came
in over a PRI.
Did not want to waste to much time on these, maybe we will look again
in
Ok, this is my voicemail.conf:
[general]
attach=yes
charset=ISO-8859-1
emailbody=${VM_NAME}:\n\nUd ha recibido un nuevo mensaje de voz, de
${VM_DUR} segundos.\n\nTeléfono: ${VM_CALLERID}.\nFecha:
${VM_DATE}.\n\nEl mensaje ha sido adjunto a este correo.\n\nPor favor,
no responda este mensaje,
Harry,Please note that you have sent this message to the group several times today. If anyone has an answer for you, they will reply. There is no need to continually send it to the group.Alex
On 5/12/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hello, How can I park a call or put on hold a
On 5/12/06, stoffell [EMAIL PROTECTED] wrote:
On 5/11/06, Tim Robinson [EMAIL PROTECTED] wrote:
There is a lot of junk in your zapata.conf that you do not need, as it
relates to analogue lines. This might be causing confusion?
I have tried a similary config to yours, doesn't helps. I haven't
On 5/12/06, Jerry Jones [EMAIL PROTECTED] wrote:
I believe the hint priority must be in the same context as the phones
extension number, in this [local]
Additionally, it may not be the first 'exten =' line, at least in
some versions, so best to put them at the end of the context.
PLUS: Avoid
It's quite strange. When I press *1 I do not hear a tone
indicated that it's even trying to record.
David Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodatasolutions.com
Tel: (519) 963-3020
Fax: (519) 451-6615
Lead, follow or get out of the
way!
Our asterisk server has been up and running for over a year and it works
great. I have emails going to my account as an attachment and I can
listen to them on the desktop and it works fine. I just got a T-Mobile
MDA that runs Windows Pocket (or whatever they call it) and it can check
email.
I am using the m option on the dial command to play a message instead of
ringing. The message is something like please wait while I try to locate your
party so I need it to start at the beginning for each call. I think there
might be a way in 1.2.x be we are not ready to upgrade yet so a
Our system is running all of the latest code and freepbx and would send the
attachment to my MDA just fine and I was able to play it without any
problem. My problem was that the MDA is a worthless turd and a complete joke
as a phone. I took it back and switched to the backberry 8700g which has its
Hello all,
I have asterisk working well with, Sipura, but I do not manage to form
several phones avaya 4606, someone could have formed one avaya with
asterisk?
is it possible?
update the firmware of the phone, but I do not achieve that it registers,
I hope that someone could help me
Your welcome. It certainly could be done entirely in the dialplan using similar
logic, but this required a bit less mental horsepower. If your desire to avoid
AGI, is based on performance concerns, note that I have systems (Dell 2850
2xXEON 3.0) that terminate 8 PRIs and have had ALL channels
Has anyone had problems with a Cisco 7970 running sip image
SIP70.8.0-2SR1S hanging up zap channels?
Calls to SIP and IAX
are fine. Just when the call goes out via the zap channels
I have some Cisco
7960 running SIP and they work fine.
Any
ideas?
Thanks-Eric Hall
hi Dave i get the following log on *CLI -- Attempting native bridge of SIP/200-39f4 and SIP/204-2ce4 -- Playing 'beep' (language 'en') -- User hit '*1' to record call. filename: wav|auto-1147452537-200-204|m
-- Playing 'beep' (language 'en') -- User hit '*1' to stop recording call. --
I have one Sipura SPA-841 which is configured to use
dtmfmode=info and one Cisco 7905 which is using the default signalling (I
believe this is rfc2833)
I have also set relaxdtmf=yes in
sip.conf
I've tried pressing *1 on both phones (they are both on my
desk) and both behave the same.
;;
Today i move our central server to 1.2.7.1 , and im having some issues
with SPA Phones and RinbackTone. Without r option, it also happens. Is
having anyone this issue? I think it has not been changed anything
sustancially to happen this to me.
It is happening between extensiones
hi david,
can you explain me this please?
If Sangoma hardware support snmp ithink it would be a
better choice than digium .
How can we know the state of the sangoma cards with an
snmp agent ?
Harry
--- David Yat Sin [EMAIL PROTECTED] a écrit :
Hi Harry,
The Sangoma Card when used for TDM
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kerry Garrison
Sent: Friday, 12 May 2006 12:07 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Voicemail WAV to PDA Problems
Our
Thanks. I like that method.
Do you think if I add all my extensions (say 40 of them) to new Dial
commands after exten = PAGE203,n,Dial(SIP/203,5)
Like this:
exten = PAGE203,n,Dial(SIP/203,5)
exten = PAGE203,n,Dial(SIP/204,5)
exten = PAGE203,n,Dial(SIP/205,5)
exten = PAGE203,n,Dial(SIP/206,5)
Hi all. I was
reading a sample config someone had posted relating to call forwarding, and in
it, they use a Dial command with components that I cannot find any reference
to.
Can someone point me
to a reference which could explain the difference between
Dial(SIP/100|20|Ttr,,wW) and
I found the issue.
It was my Dial command!
In my dialplan I had Dial(SIP/100|20|Ttr,,wW) as this was
something I gleaned from a sample config for call forwarding. I removed
the |20|Ttr andnow the call recording works! Anyone know what the
|20|Ttr did anyhow?
David Morrow
Technical Systems
make the dial command like so:
exten = PAGE203,n,Dial(SIP/203SIP/204SIP/205SIP/206,5)
On 5/12/06, Forrest Beck [EMAIL PROTECTED] wrote:
Thanks. I like that method.
Do you think if I add all my extensions (say 40 of them) to new Dial
commands after exten = PAGE203,n,Dial(SIP/203,5)
Like
Hi, in the menu of voicemailmain, appear a
lot of options, there is a way to leave only some of them?
Also I want to know if there is a option that erase
all message in a user box.
Best REgards
Ever Zalazar
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Can someone tell me if passwords are sent in plain text when using IAX?
I have been told already that SIP automatically encrypts the password?
Anyone know of some good Asterisk security links, docs, articles?
Thanks!
___
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On 5/12/06, Florian Overkamp [EMAIL PROTECTED] wrote:
Douglas Garstang wrote:
We are using a backend MySQL database for call flow, not user agent
registration info. Just how, exactly, is a backend database going to
replicate registration data between Asterisk servers? Realtime has
been
On 5/12/06, Florian Overkamp [EMAIL PROTECTED] wrote:
Douglas Garstang wrote:
We are using a backend MySQL database for call flow, not user agent
registration info. Just how, exactly, is a backend database going to
replicate registration data between Asterisk servers? Realtime has
been
I must be missing something. Seems to me that only one phone would connect. This is just a plain dial command that rings all those extensions and when one answers, the rest stop ringing.
Right?
On 5/12/06, Tom Vile [EMAIL PROTECTED] wrote:
make the dial command like so:exten =
You are correct, That is why the PAGE()
Application was made. It creates a MeetMe room, calls the Technologies on the
list and transfers the calls to the temp MeetMe.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora
Sent: Friday, May
-Original Message-
From: Leif Madsen [mailto:[EMAIL PROTECTED]
Sent: Friday, May 12, 2006 12:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Dialling a DUNDi Route
On 5/12/06, Florian Overkamp [EMAIL PROTECTED] wrote:
Douglas
Ugh. We thought we'd fixed some problems by using regexten and DUNDi. Guess not.
We have a configuration with three Asterisk boxes. Phones register with a
single, primary asterisk box under normal conditions. For voicemail deposit,
retrieval, we trunk the calls over to our asterisk voicemail
I'm having a big problem where digits dialed from certain cell phones
are too short to be recognized by my asterisk server. I'm running AAH
2.8. Some cell phones don't allow the caller to hold down the digits
and have the tones play as long as they hold them down for. They just
play a short
Thanks to everyone who responded. I was able to modify the freepbx
paging code to use something like the suggested macro and it worked
well. For those who may be interested, the following Page macro works
for Linksys SPA942 phones:
[macro-page];
;
; Paging macro:
;
; Check to see if SIP device
You were doing so good too.
The voicemail application has a function to run an external app to notify
about voicemail. We have scripts on the main servers that recieve
notification from a voicemail server script that particular phones have a
certain number of messages. That script then runs
Just installed fc5, installed correct kernel source, and trying to
compile zaptel-1.2. Changed the link in /lib/modules/2.6.15-1.2054_FC5
to point to /usr/src/redhat/SOURCES. Like:
lrwxrwxrwx 1 root root 23 May 12 15:21 build - /usr/src/redhat/SOURCES
A 'make install' still complains with:
I just installed rpm binaries in a new mandriva and I
see a frew error messages with asterisk -vvvcfg,
btw I would also like a little guidance to just set up
a couple sip phones to start playing with soft phone
communication with 3 pcs on the network
thanks :)
ng
Here is a CLI of the problem:
Here is a CLI of the problem:
== Timeout for Zap/47-1 parked on 5401. Returning to park-dial,Zap/47,1
-- Executing Dial(Zap/47-1, Zap/47||t) in new stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Hungup 'Zap/47-1'
Zap/47-1 could be any of
Thanks Aaron. That'd probably work. However, we also have an asterisk box
dedicated to ACD, and we face the same problem with that. Phones don't register
with it directly, but it still needs to know their location. Ideally we need
one solution to address both the voicemail and acd servers.
I did a test with ParkAndAnnounce and the call back is not going to fly here.
Does anyone have a version that talks back during the transfer like Park() does?
I piggybacked off of another feature request in the bug system that is very
similar.
http://bugs.digium.com/view.php?id=6953
I hope
In article [EMAIL PROTECTED],
Mike Clark [EMAIL PROTECTED] wrote:
Damon Estep wrote:
Can anyone point me to a sample or information on using MeetMe like this?
Conference room is set up with 2 PINs, one for the moderator and one
for the participants.
Participants get music until the
Rich,
Check what is the content of /lib/modules/2.6.15-1.2054-FC5/build?
If it is empty, then you need to do yum install kernel-devel again.
Also you can check running uname -a to see if you have the same release that
the one that you're checking.
Regards,
Carlos Alperin
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