Andrew Sayman wrote:
Noah Miller wrote:
Depending on your BIOS and motherboard, you may be able to use
another IRQ if you move the card to a different PCI slot.
- Noah
This is a computer meant to be rack-mounted that I'm trying to install
this on. I certainly don't
Hi,
Updating zaptel gives me this during the make. Any
ideas, the
searches and Wiki gives me no hints.
In file included from
/usr/src/linux-2.4/include/linux/fs.h:19,
from
/usr/src/linux-2.4/include/linux/capability.h:17,
from
stevanus ha scritto:
Does anyone know how to make this thing (7940) work with asterisk
(chan_sccp module) ?
I've set the configuration according to the wiki and now the phone
just keep asking for CTLSEPxxx.tlv from my tftp server.
Update the skinny firmware.
The phone has to look for
Hi all,
I have been worriyng and googling a lot but I can't find my mistake.
I am trying to regiter an X-Lite Softphone to Asterisk, but
I am getting a SIP/2.0 403 Forbidden response:
SEND TIME: 10157385
SEND 10.100.249.12:5060
REGISTER sip:10.100.249.12 SIP/2.0
Via: SIP/2.0/UDP
exten = 555,1,MusicOnHold(default)
i can hear the music, so far so good.
But when i hold an incoming call by pressing the HOLD-key on my snom
telephone - nothing happens.
No output at CLI that the MOH gets played.
When debugging SIP on asterisk, in the moment i press the HOLD-key i can
This is somewhat unique to the site installation. For example, I don't
have *69 programmed at my site because frankly there's no need for it
with the Cisco 7960's.
I do however have an automatic conference booking utility and a speaking
clock. Not often found in smaller sites.
I think you
On Wednesday 06 Jul 2005 06:07, wei li wrote:
Hi there:
I have successfully installed the Asterisk 1.0.9 on my Freebsd 5.4
box. When I tend to install the addon for mysql CDR billing, It always
return me the following errors:
SIP# gmake clean
rm -f *.so *.o .depend
gmake -C format_mp3
Hello,
is there an open-source implementation of G.729 codec for use outside
of US? I know it's a patented codec, but since there are usually no
software patents outside of the US, I don't care about the patent
license. I could use open-source implementation of the codec, if there
was some. Any
Stefan Gofferje wrote:
"If the phone just requests CTLSEPxxx.tlv and nothing else, it either have
been used on a CallManager with authentication / encryption enabled and is
now security locked because the asterisk does not provide the proper
tlv-file or the firmware is corrupted. Try to reset
Check that both sides use the same codec. I
had the same problem before L
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep
Sent: Tuesday, July 05, 2005 4:36
PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
can we use asterisk as a SIP Redirect Server?
Thanks
Erdem HAKI [EMAIL PROTECTED]
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On Wed, Jul 06, 2005 at 03:07:16PM +1000, wei li wrote:
Hi there:
I have successfully installed the Asterisk 1.0.9 on my Freebsd 5.4
box. When I tend to install the addon for mysql CDR billing, It always
return me the following errors:
My asterisk-addons deb builds independently of
--- Ursprüngliche Nachricht ---
Von: Mark Phillips [EMAIL PROTECTED]
An: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Betreff: Re: [Asterisk-Users] Users handbook
Datum: Wed, 06 Jul 2005 04:50:07 -0400
This is somewhat unique to the site
Check out http://www.readytechnology.co.uk/open/g729/
Regards,
Sahil Gupta
VoiceValley
On Wed, 6 Jul 2005, Juraj Bednar wrote:
Hello,
is there an open-source implementation of G.729 codec for use outside
of US? I know it's a patented codec, but since there are usually no
software patents
Hi all,
I try to use app_rxfax. Aplication app_rxfax start
O.K., fax trying to send, but it will stop at the beginning of page and after
few seconds it stop with error 400.
Does anybody has any suggestions?
Thanks,
Bob.
___
Before you give up, I have had good results with a Sipura 2002 ATA and
using Teliax for faxing, I tried other termination accounts with the
same setup and it didn't work.
--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759
Cell: 264-235-5670
Mark Phillips wrote:
This is somewhat unique to the site installation. For example, I don't
have *69 programmed at my site because frankly there's no need for it
with the Cisco 7960's.
I do however have an automatic conference booking utility and a
speaking clock. Not often found in smaller
I'm planning on implementing an Asterisk system at a couple of small
offices, and a couple of homes, in the near future... and I don't have
any documentation yet. What you're suggesting sounds wonderful, to me. I
would contribute, if I had anything... but making it an inclusive manual
would be
good day all,
my asterisk is working greatly..
and i want to put a billing..
but i have this error when i try 'make'
[EMAIL PROTECTED] asterisk-addons-1.0.7]# make
cc -fPIC -I../asterisk -D_GNU_SOURCE
-I/usr/include/mysql -I/root/asterisk/include/ -c
-o app_addon_sql_mysql.o
On Wed, Jul 06, 2005 at 06:19:53AM -0400, Chris Mason (Lists) wrote:
If one is implementing an Asterisk solution in an office scenario, it
has to have fairly similar features to another Asterisk installation.
It's easy enough to edit and remove the parts that are different. What I
am
I just wonder what can i do with asterisk and its limits. For example i
really don't know yet is asterisk used as redirect server?
Thanks for your reply,
Erdem HAKI - [EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew M
Stemen
Hi all,
I have a bit of a problem with chan_capi. Details: chan_capi-cm-0.5.3
from sourceforge, zaptel, libpri and asterisk cvs HEAD from July 4,
2005. Everything works fine except...
[app_capiHOLD.so]Jul 4 22:56:58 WARNING[1013]: loader.c:313
__load_resource:
Hello!
Following the instructions on voip-ip.org I have implemented Realtime
with MySQL for my Asterisk server. The individual extension
configuration is managed in a table called extensions.
Still I have to keep some data in the extensions.conf, namely the switch
and the include statements. Is
Patrick ha scritto:
Hi all,
I have a bit of a problem with chan_capi. Details: chan_capi-cm-0.5.3
from sourceforge, zaptel, libpri and asterisk cvs HEAD from July 4,
2005. Everything works fine except...
stop asterisk,
rm /usr/lib/asterisk/modules/*
rm /usr/include/asterisk/*
cd asterisk
John Daragon wrote:
Hi;
I'm looking for a Polycom distributor in the UK who can supply a small
number (around 20) IP301 / IP501 handsets. Can anyone recommend someone ?
jd
I have been buying from Zycko - very efficient and on the ball.
--
Chris Mason
NetConcepts
(264) 497-5670 Fax:
I stand corrected... It was late :)
PAP2-NA = Useful
PAP2 = Useless/Vonage
-Original Message-
From: Brian Capouch [mailto:[EMAIL PROTECTED]
Sent: Wednesday, July 06, 2005 1:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk on
Chris Mason (Lists) wrote:
John Daragon wrote:
Hi;
I'm looking for a Polycom distributor in the UK who can supply a small
number (around 20) IP301 / IP501 handsets. Can anyone recommend someone ?
jd
I have been buying from Zycko - very efficient and on the ball.
Ta.
jd
--
John
Hi,
I am using a sip provider that offers voicemail. They send me a sip
notify that there are voicemails, and I would like this notify to be
sent to one of the extensions on asterisk (a sipura 2100 or cisco
7960), to light a lamp/give stutter dial tone.
The provider is running * too and is
Hi
Is there a phone comparison matrix I could consult
I have a series of features that I would like to evaluate on the most
common phones on the market
example:
dual-ethernet
POE / direct power / both
number of lines
speed dials programmable buttons
BLF LEDS
Headset plug
conference call
Good day all
Im looking for someone in the UK that knows asterisk and thats willing
to do a quick job for us,its in at tele city
--
Thanks
Altus Snyman
Stormcorp Network Solutions
+27 11 8071141 exten 301
___
Asterisk-Users mailing list
Hi,
i was successful in compiling app_conference and setting up an
conference was quite easy. :-)
Does anyone knows if it is possible to have an IVR accessable from
inside the conference. So, if i dialed into an conference i want to be
able to press '*' and then the actual discussion is
I m using oh323 and i m receiving incoming calls at windows NetMeeing and at SJPhone from SIP IAX softphones but what should i do to be able to call from NetMeeting or any H323 softphone .when i dial any extension... it starts OH323/R4096 and then fails and plays demo-congrates from
Hi all,
Just want to share with all of you a new hot DECT VoIP gateway
available from www.broad-tel.com/index_en.php.
The DECT VoIP gateway is capable of handling both SIP and the H.323
calls. Up to 4 registrations to the SIP proxy or H.323 Gatekeeper.
To bring the users most flexibility, the
I am trying to use language=it in asterisk
I downloaded the sound package and installed it
I added
country=it in indications.conf
language=it in sip.conf
language=it in iax2.conf
everything ok in call from sip and from iax
The problem arises in outside call, coming trom CAPI Trunk
I try
you can find it under VoIP Products/Wireless IP phone.
On 7/6/05, IM.Nobody [EMAIL PROTECTED] wrote:
Hi all,
Just want to share with all of you a new hot DECT VoIP gateway
available from www.broad-tel.com/index_en.php.
The DECT VoIP gateway is capable of handling both SIP and the H.323
Would it be a good replacement of expensive WiFi phones? How much is it??
On 7/6/05, IM.Nobody [EMAIL PROTECTED] wrote:
Hi all,
Just want to share with all of you a new hot DECT VoIP gateway
available from www.broad-tel.com/index_en.php.
The DECT VoIP gateway is capable of handling both
Hello to all,
Does Asterisk support QSIG? and the configuration is in capi.conf?
and if it supports it, do you have samples of the configuration?
I had my Asterisk connecting to a Siemens PBX with ETSI and it was
working fine, but peolpe said to me that QSIG could implement more
features and
Get the handbooks pdf's files, open and modify it regarding your
configuration, and put that on a web page accessible for everyone on the
office.
You can look for the files on Voip-info.org
Carlos
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew M
On Wed, 2005-07-06 at 15:51 +0200, [EMAIL PROTECTED] wrote:
I found a german forum, and it seems to be a common problem ( I don't
undertand a lot of german, anyway)
( cfr: http://www.ip-phone-forum.de/forum/viewtopic.php?t=17639)
I also try to put
exten = .,1,SetLanguage(it)
in some
But doesnt Asterisk supports QSIG already?
I just whant to know how to configure it.
João
George Lin wrote:
Joao,
We have developed some QSIG stack over asterisk. It will be a paid system.
would you be interested in ?
Regards
George
-Original Message-
From: [EMAIL PROTECTED]
Ok , I solved by myself
just put
[from-pstn-custom]
exten = s,1,SetLanguage(it)
in extensions_custom.conf
Andrea
--
I am trying to use language=it in asterisk
I downloaded the sound package and installed it
I added
country=it in indications.conf
language=it in sip.conf
On Wed, 2005-07-06 at 15:05 +0100, Joao Pereira wrote:
Hello to all,
Does Asterisk support QSIG? and the configuration is in capi.conf?
and if it supports it, do you have samples of the configuration?
QSIG is not an option in capi.conf. It is an option in the configuration
of my Eicon Diva
Cher Areski,
Je m'excuse si je me suis trompé sur vos intentions. Cependant j'aimerais souligner que je ne suis pas le seul a avoir trouvé l'installation de votre application un peu trop compliquéeet les instructions du "Idiots Guide" imprécises.Il faut dire que Linux n'est dejà pas facile pour
thank you !
I solved by myself in a different (slightly different) way
Now I understand why my first solution didn't work: I didn't put the _ in
front of the .
thank you again,
Andrea
Patrick
Thanks for the help.
I also have a Eicon Diva Server BRI and I know it can be used with
chan_capi and asterisk, but the QSIG configuration is not direct.
Of course I googled before asking to the list and I didnt found any
direct explanation if QSIG is supported.
Voip-info.org sais that
I have never have the /ext work for me.
register=1234:[EMAIL PROTECTED]/ext
It has never worked for me.
David
On Wed, 2005-07-06 at 10:27 +0200, Christian Peter wrote:
I dislike the statement in the bug reports you can easily add /ext to
your register statement as a workaround because it
Yep, along with 6 other distros.
W
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of JD Austin
Sent: Tuesday, July 05, 2005 5:53
PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Epia
C3 Linux
Tried knoppix?
Wiley
On 7/6/05, Gundemarie Scholz [EMAIL PROTECTED] wrote:
Hello!
Following the instructions on voip-ip.org I have implemented Realtime
with MySQL for my Asterisk server. The individual extension
configuration is managed in a table called extensions.
Still I have to keep some data in the
Hi guys,
I'm new to Asterisk, so I'm hoping someone can guide me :-)
Currently, I am having the configuration as follows :
PSTN - Cisco router - Sip Express Router - Asterisk Voicemail
I'm able to get the part from PSTN to Sip Express Router working, but
I can't integrate Asterisk with Sip
Rob,
How in the world did you know that
I just ran the memtest86 and it is nothing but error after error.
Switched out the ram and I am getting no
errors on memtest86 now.
I am back in the saddle. Fedora Core 3 is
installing as we speak Thank you!
Wiley
On the snom phones you could use the Action URL's to start some process when
the phone receives a call.
Nils
On Tuesday 05 July 2005 23:52, [EMAIL PROTECTED] wrote:
Does anybody know if there is an app that will cause similar to occur on
users PC?
I have a scenario where users will have
It's a common (and commonly overlooked) problem and whenever there appears to be no logic behind irrational behavior, the RAM is the first place I look. Because the RAM is effectively changing the running program's code at the bit level, any and all actions are unpredictable, along with their
I'm having a little problem. I have a dial-plan with a lot of SetVar's and
loops, and under certain circumstances (reproducible) it makes asterisk
crash. Wanting to debug this, I compiled using make valgrind. But doing
so, I eliminated the crashes and the dial-plan works perfectly.
Now from what
Does anyone know if it is possible to send variables over an IAX trunk?
Is there a setting in iax.conf that allows this or is there another hack
to allow this?
Thanks in advance.
Jeremiah
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Asterisk-Users mailing list
How adventurous would a person have to be to try to use the 1.1 from cvs? I
want to implement our phone system with the database connections built in,
which as I understand is being made very easy in the 1.1 code that is under
development.
thanks,
hello all,
another desperate request for help debugging my dialplan...
from a certain extension i do the following:
DBput(CFIM/${CALLERIDNUM}=${CALLERIDNUM})
a NoOp to the console says
DBput: family=CFIM, key=2122022001, value=2122022001
and database show says
/CFIM/2122022001
On Wed, 2005-07-06 at 16:02 +0100, Joao Pereira wrote:
Voip-info.org sais that zapata.conf is for configuration of Digium cards
Yup you need a T1/E1 card for qsig stuff in zapata.conf. Don't know how
it is done with an Eicon Diva Server card, if possible at all.
I also searched the list
according to Polycom the IP301,IP501 are not going to be released in the UK
(EMEA) until Q4 this
year...
try calling hardware.com if they have them available.
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
Bruce,
I too am interested in the telephone number for SimpleTelecom, as my company
had put quite a large prepayment to them. You said you posted the number on
this list; I searched for all post by you and did not find the posting which
contained a phone number. Would you be so kind as to
Ian White wrote:
The use of the nonce looks right to me. Can somebody point out what is
going wrong here?
Yes, I agree, it looks correct. However, what version of Asterisk are
you testing against? Current CVS HEAD adds 'stale=true' to the 401
response, and I don't see that in your trace.
In article [EMAIL PROTECTED],
Chris Gamble [EMAIL PROTECTED] wrote:
How adventurous would a person have to be to try to use the 1.1 from
cvs? I want to implement our phone system with the database
connections built in, which as I understand is being made very easy in
the 1.1 code that is under
Hi
We are about to buy several Snom phones.
Does anyone have warnings or advices against these phones ?
Our finalists were Cisco, Polycom and Snom.
We will be using only the SIP protocol.
Thanks
Patrick
___
Asterisk-Users mailing list
Greetings,
We are just finishing a roll-out of 25 of the SNOM 190s with a SNOM 220
w/sidecar.
The only gotcha that I found is that the SNOM 190s use rfc2833 for a
default dtfm mode and not inband which is the default for the asterisk
server.
I haven't ironed out the Mass deployment
Tzafrir Cohen wrote:
Hey, compile is what computers do, not humans. :-)
Is there any existing program that could either from /etc/asterisk or
using the manager interface figure out enough about the asterisk
configuration to generate such a manual (using some templating engine)?
I've often
Lars Boegild Thomsen wrote:
On Wednesday 06 July 2005 15:09, Erdem HAKİ wrote:
I need to set up Asterisk to serve and register for 1000 users(not
simultaneus). What kind of specifications do my server need.
Well - the interesting number is the number of simultaneous users really and
to
rm -rf /usr/include/asterisk
do a fresh checkout and try again.
/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”
On Jul 5, 2005, at 12:02 PM, Chris Coulthurst wrote:
Lately when I issue a 'reload' from the CLI, I find that it will
[EMAIL PROTECTED] wrote:
Does anybody know if there is an app that will cause similar to occur on users
PC?
Maybe have a look at the Flash Operator Panel.
It has the capability for web pops, and can even be shrunk so you can't
see it if neccesary.
--
Cheers,
Matt Riddell
Hi all,
i have a problem with 2 peers conecting to an
asterisk machine, both are conected behind nat without any port mapping in the
router, and the * is conected behind other nat with the port 4569 mapped to it
address, the problem is:
when a peer register to the asterisk the other cant
http://software.newsforge.com/software/05/07/06/1235211.shtml?tid=150tid=147tid=136
http://www.theregister.co.uk/2005/07/06/eu_bins_swpat/
begin:vcard
fn:Vahan Yerkanian
n:Yerkanian;Vahan
org:ARMINCO Global Telecommunications;Head, Research Development dept.
adr:;;28, Isahakian ave., PO BOX
On Wed, 2005-07-06 at 13:34 +0200, Sergio Chersovani wrote:
[snip]
stop asterisk,
rm /usr/lib/asterisk/modules/*
rm /usr/include/asterisk/*
cd asterisk
make clean
make upgrage
cd chan_capi-cm-0.5.3
make clean
make install
now run asterisk again
Thanks Sergio. I removed everything,
Tony Mountifield wrote:
Anyone here in the know about when HEAD will be branched to 1.2?
Very soon. We are actively trying to clean up the open bugs and issues
so we can prepare a release candidate.
___
Asterisk-Users mailing list
The README in the source code states:
app_conference doesn't have DTMF-activated features or anything like
that.
I'm curious how you got audio working on your compliation. I am running
CVS HEAD + app_conference in a Xen virtual machine. I can connect to the
channel but there is no audio. Here are
Juan,
That is not going to work. Asterisk shouldnt
be behind a NAT to get registration of boxes behind NAT.
Put the asterisk on DMZ zone of their
router to make that happen.
Carlos Alperin
[EMAIL PROTECTED]
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
On Tue, Jul 05, 2005 at 08:39:15PM -0400, Michael Stahl wrote:
Take a look at the via arena web site. Your processor may look like a
586 to the installer but may not support all of the instructions
(causing a crash). The via arena site gives instructions on how to
compile and get it
Our setup:
We have a DS3 from Global Crossing terminating into a Adtran MX2800 M13
mux. From there, groups of 4 T1's run into T410P digium cards to 7
individual servers. Each trunk is configured as ISDN PRI, B8ZS/ESF,
D-channel being chan 96 with B-channels of 1-95 (we're using NFAS). The
This did wind up being a matter of memory...
Thanks,
Wiley
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Wednesday, July 06, 2005 10:14 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Epia C3 Linux
On Tue,
On Wed, Jul 06, 2005 at 11:48:27AM -0500, Brian West wrote:
/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”
On Jul 5, 2005, at 12:02 PM, Chris Coulthurst wrote:
Lately when I issue a 'reload' from the CLI, I find that it will
Why not do your research instead of asking the list to do it for
you lazy ass!
/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”
On Jul 6, 2005, at 2:09 AM, Erdem HAKİ wrote:
Hello;
I need to set up Asterisk to serve and
I have a P4 HT system running * on a 2.6.9-1.667smp kernel (Fedora Core
3). Everythng was testing out and the configuration was working. After
running YUM update, kernel 2.6.11-1.35_FC3smp was installed. Now Zaptel
cannot find /dev/zap.
Waiting for zap to come online...Error: missing /dev/zap!
On Wed, 2005-07-06 at 13:00 -0400, Lee Azzarello wrote:
[snip]
I can connect to the
channel but there is no audio. Here are my configs and Asterisk's
output:
http://lee.97montrose.org/hacking/app_conference.txt
Maybe it is a codec problem:
translate.c:134 ast_translator_build_path: No
Hello,
Quick Diagram:
Telco-PRI - Asterisk - Norstar PRI - Norstar PBX
(DMS100) (TE405P) (DMS100)
|
|
V
Cisco 7960G
(SIP)
I'm trying to change the Origination Number on my outgoing PRI, and running
into a weird
Vahan Yerkanian wrote:
http://software.newsforge.com/software/05/07/06/1235211.shtml?tid=150tid=147tid=136
http://www.theregister.co.uk/2005/07/06/eu_bins_swpat/
Hear, hear!
I'm not in general much of a praying kind of person, but do pray that
someday the US will wake up to the damage
HI ALL;
I have problem converting a windows .wav file to
.gsm format by Sox.
Could anyone help.
Cheers,
Mohammad
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To
you re using an Eicon Diva Server BRI, what are you using to connect?
ETSI, QSIG or someting else?
I use my Eicon Diva Server card to hook up my ISDN/BRI line through ETSI
and capi.conf to asterisk.
I had that configuration too, but isnt QSIG better? because QSIG can
send the
Valued Colleagues,
Can anyone tell me whether the Maximum Number of
Mailboxes in Asterisk
is hardcoded or configurable?!
I suppose the maximum number of allowed voice messages per
mailbox is hardcoded as
#define MAXMSG 100
in ~asterisk/apps/app_voicemail.c
Thanks
ramin
Well you could get a backtrace of the core to give us a little bit of
clue why its crashing!
/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”
On Jul 6, 2005, at 10:44 AM, Benjamin Lawetz wrote:
I'm having a little problem. I have a
http://chan-sccp.berlios.de/
20050705 ftp://ftp.berlios.de/pub/chan-sccp/chan_sccp-20050705.tar.gz
- Added support for distinctive rings
on stable: SetVar(ALERT_INFO=inside) or outside or feature
on head: SetVar(_ALERT_INFO=inside) or outside or feature
- Added support for native transfer
Valued Colleagues,
Can anyone tell me how the asterisk keeps track of the
number of existing old (read)
and new (unread) messages in a mailbox? Is there a database
table or somewhere
else from which this data can be retrieved by an application?
Thanks
ramin
Can you be a bit more specific as to what the problems is?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mohammad
Sent: July 6, 2005 2:10 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] converting windows .wav to .gsm
HI ALL;
Valued Colleagues,
Can anyone tell me whether the Maximum Number of
Mailboxes in Asterisk
is hardcoded or configurable?!
I suppose the maximum number of allowed voice messages per
mailbox is hardcoded as
#define
MAXMSG 100
in ~asterisk/apps/app_voicemail.c
Thanks
ramin
Chris Gamble wrote:
How adventurous would a person have to be to try to use the 1.1 from cvs? I
want to implement our phone system with the database connections built in,
which as I understand is being made very easy in the 1.1 code that is under
development.
thanks,
Not adventurous at
I have an incoming 800-number over IAX from Teliax and I'm experiencing
the large packet loss on connection.
When a call comes in there is no ring tone and the first few words of
the welcome message are cut off, regardless of the delay I set.
Standard call (not 800-number) coming over IAX with the
Juhu, Jippi Jippi Yeah! I am going to dance all night.
Cheers
S.
At 13:04 06.07.2005 -0500, Brian Capouch wrote:
Vahan Yerkanian wrote:
http://software.newsforge.com/software/05/07/06/1235211.shtml?tid=150tid=147tid=136
http://www.theregister.co.uk/2005/07/06/eu_bins_swpat/
Hear, hear!
Paul Belanger wrote:
[6c 0c 21 ff 36 31 33 32 37 31 38 38 35 33]
Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony
Numbering Plan
(E.164/E.163) (1)
Presentation: Unknown (127) '613271' ]
It _is_ being changed, but the
Hello,
Is anybody there using quadBRI form Junghanns.net with Asterisk ?
I would like to order that card but first would like to hear some
opinions.
Thank you in advance
Bartosz
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I use wavepad all the time on my windows box. I've never had a problem
using it to convert and edit the files.
Darren Wiebe
[EMAIL PROTECTED]
mohammad wrote:
HI ALL;
I have problem converting a windows .wav file to .gsm format by Sox.
Could anyone help.
Cheers,
Mohammad
On Wed, 2005-07-06 at 10:56 -0700, Howard Ratzlaff wrote:
I have a P4 HT system running * on a 2.6.9-1.667smp kernel (Fedora Core
3). Everythng was testing out and the configuration was working. After
running YUM update, kernel 2.6.11-1.35_FC3smp was installed. Now Zaptel
cannot find
Elwin Andriol wrote:
Don't know if this will help you any further, but. After some trouble
with IRQ sharing mayhem we solved our little problem by tinkering the
linux kernel. I forgot the names of the actual modules, but after
disabling modules for APIC support and something about IRQ sharing
Did you try tailing the /var/log/dmesg to see what happened when you
loaded zaptel and wctdm with modprobe?
Check that /etc/modprobe.conf still contains the correct module entries.
Does /lib/modules/2.6.11-1.35_FC3smp/misc still contain and correct
wctdm.ko files?
-Original Message-
My apologies for any bounced mail from me today. My mail server was
having a bit of a fit.
MARK.
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