Hi Kapil,
AFAIK, there are no such PDF's that exist unless someone has really spent
time compiling such information, which will be great to see.
However, if you check out www.voip-info.org, its a complete mine of useful
information regarding doing what you wish to.
Regards,
Sahil Gupta
Voi
On Wed, 21 Sep 2005, Shaun Ewing wrote:
> On 9/21/05, Armin Schindler <[EMAIL PROTECTED]> wrote:
> > Hi all,
> >
> > it took a while, but on sourceforge.net I added the new release 0.6 of
> > chan_capi-cm driver.
>
> Doesn't seem to work with 1.0.8:
>
> Sep 21 10:25:13 WARNING[16435]:
> /usr/lib
Hi List
I am very new to Asterisk but have been alloted a job to replace my
traditional PBX with it. Kindly provide me some useful info (PDF's etc) to
setup Asterisk with FXO and FXS both.
I have to cater some 60 users with 10 simultaneous calls.
Regards
Hi
I was doing some bandwidth testing, and my incomming usage is
36% more than my outgoing bandwidth.
The setup is IAX2 trunking using GSM codec.
Is there any obvious reason I am overlooking to figure out why
there is such a big difference between the two.?
I am using CVS-head September 3rd,
I dont believe its limiting but if you want to block users in real time when their credit runs out, you need to use the realtime config. Thats what i assume anyway.
Dan
On 9/21/05, Steven <[EMAIL PROTECTED]> wrote:
I found configuration via MySQL too limiting.I went back to text files.I do not kno
Hi,
Thanks all for help...
I was perhaps using old version of mpg123 and with beta1 and
mpg123-0.59r it working smoothly.
Gurminder
On 9/15/05, Gurminder Arora <[EMAIL PROTECTED]> wrote:
> Hi
> On my FC3 box with asterisk 1.0.9MusicOnHold is not working.
> It starts and stops immedi
Hello all,
We have a situation where our 30 lines are maxing out, but
no one is on a call. We are currently running CVS head downloaded on
15/8/2005 on a Dell Power Edge 2850. Our office mainly functions on a
queue system. At the time this happened all our agents were logged in and
n
Guys. Im using cvs-head from around may and when tring to use automon
(hitting #3) the files are left as .WAV but when trying to open thru winamp
or media player, they complain of bad codecs as if the files werent wavs...
Anybody had issues like this?
__
sweet, i'll play with it tomorrow many thanks
-Original Message-
From: Luki [mailto:[EMAIL PROTECTED]
Sent: Tuesday, September 20, 2005 10:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] MWI indicator HINT on Snom thru IAX?
> Has anyone don
I want to have customers make payments by keying in their cc#'s.
I can see it's possible, I just want to know if anyone out there is doing
this and what financial institutions are supporting Asterisk PBX's.
So far I have found a few leads but would like to check here at the same
time.
Thank
On Wed, 2005-09-21 at 14:36 +1000, Mark Edwards wrote:
> terminating asterices. (Is that the plural of asterisk?)
I propose asterii, while by no means gramatically correct it wont fall
under potential sue happy lawyers who own the unix trademark (after all
the plural there is unices). oh no I sa
Have come to a solution on this, and as I suspected, the issue appears
to be a bit of a version mismatch between terminating asterices. (Is
that the plural of asterisk?) Anyway, to cut a long story short, I
tested with another provider, found that they were running a later
version (nearer CVS-HEAD)
> Has anyone done this, or is there a way I could fake it? tia
Remote MWI notification via IAX works quite well. There is a thread
about this in May -- read the entire thread to see how I got it to
work: http://lists.digium.com/pipermail/asterisk-users/2005-May/109726.html
Same setup (central voi
Have you tried upgrading the firmware? I had several problems with the
outbound volume of these phones until I upgraded them.
On Tuesday 20 September 2005 20:46, Anton Krall wrote:
> Hi Guys!
>
> I have a problems with some sipuras 841 and asterisk 1.0.9.
>
> Im using 841 with asterisk 1.0.9 wit
On Tuesday 20 September 2005 21:45, Technical Support scribbled:
> Has anyone tried HooDaHek with asterisk 1.2b1 ?
> I know the data structures have changed somewhat...
I certainly have not yet -- I'm kinda waiting to see how well 1.2 cooks before
I decide to try the dish. I have enough stuff on
Hi,
I was wondering if I can connect my IAXy (the old blue model) to my
Panasonic Analog PBX???
Something like this:
(PanasonicPBX)<->(IAXy)<->(*)<-->[Internet]<-->(*)<->(IAXy)<->(PanasonicPBX)
Thanks in advance!
Raul
___
--Bandwidth and Colocation
> I want to be able to dial a 'pager' extension from an phone on my
> asterisk server, and have it ring all other extensions *except* the
> extension from which I am calling
Sure. Fair enough.
1) Define a list of all phones
2) Build a dial string by going through the list of all phones and add
al
Hey anyone know once you've bought a DID from DIDx
under the Purchased DIDs tab you can click a link for
one of your numbers that says '0 (SIP)'
and when you click that like there's SIP and a space
or the URL and AIX and a space for the URL
What do I put here?
How do I setup Asterisk to accept
On 9/20/05, Matt Roth <[EMAIL PROTECTED]> wrote:
Patrick,
Thank you for your suggestions.
Our initial runs were recording directly to an NFS mount and they
experienced the same problems as recording to the local disk. In our
final setup, the copy will be done to an NFS mount as long a
Has anyone tried HooDaHek with asterisk 1.2b1 ?
I know the data structures have changed somewhat...
MD
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http://lists.digium.com/mailm
Hi all..
After some further research I have come up with a quick and dirty way to
playback the "longblob" recordings from the ODBC database for those of you
that are running the ODBC storage for voicemail.
Have a look
http://www.itsngroup.com/software/asterisk/downloads/ODBC_VM_1.0.tar
A little
Patrick,
Thank you for your suggestions.
Our initial runs were recording directly to an NFS mount and they
experienced the same problems as recording to the local disk. In our
final setup, the copy will be done to an NFS mount as long as it
exists, falling back to local disk only when the NF
Patrick wrote:
On Tue, 2005-09-20 at 18:37 -0400, Matt Roth wrote:
List users,
Over the last few days we have been working with MCI's development lab
to test our Asterisk setup. We were using a piece of hardware called an
Abacus 5000 that is capable of creating and terminating thousands of
On 9/21/05, Armin Schindler <[EMAIL PROTECTED]> wrote:
> Hi all,
>
> it took a while, but on sourceforge.net I added the new release 0.6 of
> chan_capi-cm driver.
Doesn't seem to work with 1.0.8:
Sep 21 10:25:13 WARNING[16435]:
/usr/lib/asterisk/modules/app_capiCD.so: undefined symbol:
get_ast_c
On 9/21/05, Armin Schindler <[EMAIL PROTECTED]> wrote:
> Hi all,
>
> it took a while, but on sourceforge.net I added the new release 0.6 of
> chan_capi-cm driver.
Great work Armin. I'll try to get around to testing it today :-)
-Shaun
___
--Bandwidth a
On Tue, 2005-09-20 at 18:37 -0400, Matt Roth wrote:
> List users,
>
> Over the last few days we have been working with MCI's development lab
> to test our Asterisk setup. We were using a piece of hardware called an
> Abacus 5000 that is capable of creating and terminating thousands of SIP
> ca
Guys.
Anybody running asterisk cvs-head and the latest unicall from steve using
r2mfc (in Mexico by any chance)?
___
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium
Hello all,
I am curious, does anybody know of a way to handle the SIP 404 event?
(ie: is this stored in a variable somewhere, so one can handle it in
the dial plan).
For example, dialing an invalid number on another softswitch on the network:
-- Executing Dial("SIP/sip7110-8118", "SIP/[EMAIL
On Tue, 2005-09-20 at 21:09 +0200, Armin Schindler wrote:
> Hi all,
>
> it took a while, but on sourceforge.net I added the new release 0.6 of
> chan_capi-cm driver.
[snip]
Thanks for the new release Armin. I will test it tomorrow with cvs HEAD.
Regards,
Patrick
Hi Guys!
I have a problems with some sipuras 841 and asterisk 1.0.9.
Im using 841 with asterisk 1.0.9 with a digium card (single e1 span) with
steve's unicall.
Everything compiled fine and in fact I can make and receive calls but I have
a problem with bad sound when the sipuras call the outside
Well, I answer myself (in case someone had the same pb one day... ;).
I didn't find the exact reason of the matter... only that the error
message (relative to "flexibel rate"...) is not from Asterisk, but
from mpg123.
Anyway, I found the solution to solve the micro-cuts and interruptions
of my Mu
I believe it comes with sox. Both my sox and normalize are in /usr/bin.
Elmar Haneke wrote:
NORMALIZE="nice -n 20 /usr/bin/normalize --no-progress -a 1.0 --peak"
Which package comes this "normalize" from?
Elmar
___
--Bandwidth and Colocation spons
Hi ! :)
Does someone have an idea of the reason why my MusicOnHold (with clean
mp3 downloaded from http://aussievoip.com.au/wiki-MOH) is always
having interruptions and micro-cuts ?
No high load of the system, no swap, no hard disk r/w, mpg123 seems
running well... nothing !
Except a little messa
List users,
Over the last few days we have been working with MCI's development lab
to test our Asterisk setup. We were using a piece of hardware called an
Abacus 5000 that is capable of creating and terminating thousands of SIP
calls. Initially, we could not get past 64 simultaneous digitall
Hello hopefully someone can answer this :)
We currently have an asterisk pbx connected to a FXO channel bank to 10 pots lines. Works great. But due to increasing costs and business load, we have ordered a dedicated T1. We plan on transfering the service to the T1 and cancelling the POT lin
I found configuration via MySQL too limiting.
I went back to text files.
I do not know if it was realtime or not, it was the sql in [EMAIL PROTECTED]
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.
---- --- - - - -
I met same problem when dial via zap channel.
Does anyone know how to solve it?
thanks.
2005/9/15, Mark Edwards <[EMAIL PROTECTED]>:
> Hi.
>
> I'm dialling two numbers - one that's unobtainable, one that's busy.
>
> ${DIALSTATUS} is coming back ANSWER each time right before the channels hang
>
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm
> Sent: Monday, September 19, 2005 4:52 PM
> To: Asterisk Users
> Subject: Re: [Asterisk-Users] OT: Hardware Interrupts; Who is it?
>
>CPU0 CPU1
> 0: 85 1703809
Ive already set up the cdr mysql.
Now im trying to add realtime now but stuck on how to do it. those links didnt really help much. and the cli doesnt provide much info on what is going on.
any help would be appreciated.
Thanks
Dan
On 9/20/05, Nathan Pralle <[EMAIL PROTECTED]> wrote:
Dan Jour
Anyone know why I would be getting this error? All calls go
through without problem but I get the following message:
fixlocalprefix: Could not parse
/etc/asterisk/localprefixes.conf
___
--Bandwidth and Colocation sponsored by Ea
You can always take a PCAP (Ethereal) trace from the phone's web page
and analyze it with the RTP Statistics tool in Ethereal. That should
give you a hint whats up with jitter & Co.
CS
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Darren Elli
On 22:28, Tue 20 Sep 05, Sander wrote:
> We have tested this phone with a Asterisk system and deliver the phone with
> pre installed SIP-firmware without License
>
> What about the license?? And do you have to buy a license and changing the
> phone to sip protocol looks scary :( and time consumin
Dan Journo wrote:
Is there a guide anywhere which runs through how to set up asterisk with
mysql?
I've looked and almost all the document misses out relevant information.
Thanks
Dan Journo
What do you want to do with mysql? Did you read on the wiki? There is
tons of info there.
-Matthe
Dan Journo wrote:
Is there a guide anywhere which runs through how to set up asterisk with
mysql?
What, exactly, are you trying to do with MySQL and *?
Access MySQL from the DialPlan:
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+MYSQL
CDR record keeping in MySQL:
http://www.voip-
Steven,
Do you think the below dialplan would be typical for almost any
[EMAIL PROTECTED] setup? If so, I'll add it as supplimental documentation
for HooDaHek for those wanting to use it on [EMAIL PROTECTED]
Thanks,
Nathan
Steven wrote:
I played around with finding the right place to call
usecallerid=yes
hidecallerid=no
callerid=asreceived
usecallingpres=yes
callwaiting=no
callwaitingcallerid=yes
threewaycalling=no
transfer=no
cancallforward=yes
callreturn=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
group=1
callgroup=1
pickupgroup=1
immediate=no
busydetect=yes
busyc
On Tue, Sep 20, 2005 at 04:33:12PM -0400, Joan Bautista wrote:
> Did you make any special configuration with the switch on the card? I have 2
> TE400P that I haven't being able to use on 1 server.
IIRC, the T400Ps and E400Ps had a few problems with multiple cards together...
Unless you're mistaken
Thou hast confused the present tense with the present participle.
Thou couldest have written "" but perchance it
is better to write "there was much head-smacking and gnashing of
teeth" in this case, if thou so desirest to express thyself in the old
tongue. The "eth" suffix is oft abused, and oft
On 9/20/05, Nathan Pralle <[EMAIL PROTECTED]> wrote:
> database on an incoming call?" Much head smacketh ensued, and as I made
Thou hast confused the present tense with the present participle.
Thou couldest have written "" but perchance it
is better to write "there was much head-smacking and gna
I have a snom 360 installed but the woman that is operating it complains
about it all the time i looked at it and sometimes when sh transfers a
phonecall it will just hang and stays in the phone the snom does not have
connection to the line you can only see the line is still there in the
display i
Did you make any special configuration with the switch on the card? I have 2 TE400P that I haven't being able to use on 1 server.
jb
On 8/28/05, Asterisk <[EMAIL PROTECTED]> wrote:
I have 2 TE410P's and a TDM400P in same machine without issuesBart-Original Message-
From: [EMAIL PROTECTED][
We have tested this phone with a Asterisk system and deliver the phone with
pre installed SIP-firmware without License
What about the license?? And do you have to buy a license and changing the
phone to sip protocol looks scary :( and time consuming with 100 phones not
all suppliers will do it fo
Hello,
On Tue, 20 Sep 2005, Darren Ellis wrote:
> Hello,
>
> I just bought a Snom-320 from ATAComm. I plugged it into my LAN,
> registered it with *, etc. All my other SIP gear is Sipura and works
> fine, both on the LAN and over the Internet.
> The new Snom seems like it can't process the audi
Yes Darren. The problem is the same using Zap or SIP. I had no
oportunity to verify that using IAX or E1/T1.
Rgds, Ricardo Poppi.
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h
I'm not expert on this matter,but base on experience that is a normal situation on SIP/IAX channels since the call made is answered by the other end regardless of the situation you might found.
I'm on PRI ISDN and for me dialstatus and hangupcause works pretty good.
Regards
Jb
On 9/15/05, Mark Edw
Is there a guide anywhere which runs through how to set up asterisk with mysql?
I've looked and almost all the document misses out relevant information.
Thanks
Dan Journo
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users
I played around with finding the right place to call the agi.
Since my config started as [EMAIL PROTECTED], there are a lot of macros that
complicate things.
I put the agi in the macro-dial and it is working as expected. (just the
CLID record and change)
Thanks for the new tool.
ref:
[macro-dia
Anybody know anything about using "Asterisk vertical service
activation codes" as described in the wiki:
http://www.voip-info.org/tiki-index.php?page=Asterisk+vertical+service+activation+codes
Specifically I'm interested in *0 that (apparently) flashes an
external trunk on bridged channel. Nothi
On 21:30, Tue 20 Sep 05, Anders Svensson wrote:
> Have you tested Aastra. Works great with * and reasoable pricing
Nope, haven't seen any phone of them in real life yet.
Right now we deploy snom's for the price/quality rate they
deliver. I find them very stable and nice phones.
--
Michiel van Ba
This kind of mistakes are very common, I made them myself a couple of times,
that's is why instead of going around removing and coping and symlinking
files I prefeer to use the packages:
emerge spandsp
would do the trick.
On Tuesday 20 September 2005 15:38, Alexander Lopez wrote:
> Try
>
> rm -r
Steven wrote:
I did find this other option
http://archives.free.net.ph/message/20050309.013714.2d1bf446.en.html , but I
do not really want to imbed this info in the asterisk database if I can have
it external. (note: this other option did work when tested)
Yeah, I tried that when I first star
On Tuesday 20 September 2005 15:36, Michael Welter wrote:
> What version of libtiff are you using. Has anyone tried 3.7.x with
> spandsp?
My setup:
tiff-3.7.3 *
spandsp-0.0.2_pre20 *
Asterisk HEAD with app_[rt]xfax-0.0.2_pre20
* These are Gentoo packages.
It compiled, it started, it worked, send
i checked the document about indicator.conf nd it is used to generator
the tone of busy, ringing, congestion or dialtone. Bt how can I detect
it in extension.conf?
I hope to know whether the callee is answered the call, or know the
duration of answered time. but even the callee doesnt picked the c
Hello,
I just bought a Snom-320 from ATAComm. I plugged it into my LAN,
registered it with *, etc. All my other SIP gear is Sipura and works
fine, both on the LAN and over the Internet.
The new Snom seems like it can't process the audio from the handset
mic. A steady tone is garbled, even o
Matt Fredrickson wrote:
On Sun, Sep 18, 2005 at 11:32:00AM -0500, Brian Capouch wrote:
Senad J wrote:
If you are looking for the maximum number of cheap flights from around
the world, and plenty of convention and room space, the answer is Las
Vegas :-)
I would definitively agree!
Yes, b
OK Great, I'll give it a shot.
I did find this other option
http://archives.free.net.ph/message/20050309.013714.2d1bf446.en.html , but I
do not really want to imbed this info in the asterisk database if I can have
it external. (note: this other option did work when tested)
--
--
Steven
May
On 20 Sep 2005, at 12:12, Baris Simsek wrote:Status: Provisioned, In Alarm, Down, Active Call your provider and ask them what they see. I guess they haven't enabled it yet.Tim.___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mai
Have you tested Aastra. Works great with * and reasoable pricing
Anders
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michiel van
Baak
Sent: den 20 september 2005 20:57
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Cisco Ip phones
O
Figured it out. I didn't have tT in my dial command:
Dial(ZAP/1${ARG3},10,tT)
Thanks for posting your problem and solution. It sure helped me out...
Hugh
On 9/20/05, hugolivude <[EMAIL PROTECTED]> wrote:
> I'm having the same problem you had Frank, so I'm pleased you came up
> with a fix. No
1) how to config callprogress=yes ? in extensions.conf?
could you give me an example?
2) you means record the call (via zaptel) into a file and analyze it
with audio tool?
thanks..
2005/9/20, Alchaemist <[EMAIL PROTECTED]>:
> Hi there,
>
>Basically, youare supposed to play arround with i
could you add it into cvs head?
thanks..
2005/9/20, Dan Littlejohn <[EMAIL PROTECTED]>:
> On 9/20/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> > Ok.
> >
> > I was sucessful in installing ODBC storage
> >
> > I'm using MySQL in the backend as it is. but all my drivers are now ODBC.
> >
> > I
Hi all,
it took a while, but on sourceforge.net I added the new release 0.6 of
chan_capi-cm driver.
Note: dial string and capi.conf has changed.
The main changes are:
- added 'relaxdtmf'.
- more BSD compatibility
- correct PROGRESS handling
- added verbose text for capi info/reason error message
(trimmed)
http://info.yellowpages.com/asp/partner/whitepages/reversephone.asp
http://www.whitepages.com/10001/reverse_phone
http://directory.google.com/Top/Reference/Directories/Address_and_Phone_Numbers/
and lets not forget google itself (residential only aparently)
phonebook:QUERY (smith, ca
On 20:38, Tue 20 Sep 05, Florian Overkamp wrote:
> Hi Sander,
>
> Sander wrote:
> >Hi there does any of you use ip phones from cisco on asterisk and how is
> >the quality of this configuration ? i have to make a price of an
> >asterisk server with 100 ip phones but i need stable phones snom is n
Michael Welter wrote:
What version of libtiff are you using. Has anyone tried 3.7.x with
spandsp?
I was running 3.7.2 without issues, but reverted to 3.5.7 because of issues I was trying to track down. Didn't do any better or worse then 3.5.7.
Doug
--
Ben Franklin quote:
"Those who
On 9/20/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> Ok.
>
> I was sucessful in installing ODBC storage
>
> I'm using MySQL in the backend as it is. but all my drivers are now ODBC.
>
> I am running asterisk-cvs head as of last night 9/19/05
>
> My question is this... the old voicemail.cg
use g711u for fax not 729
- Original Message -
From: <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Monday, September 19, 2005 4:21 PM
Subject: [Asterisk-Users] T.38 & Canreinvite (yes, again)
I know this has been asked before, but I've chec
Hi there,
Basically, youare supposed to play arround with indications.conf
To have the extensions configured with callprogress=yes but, be
carefull because it is quite experimental.
Also, what I did was to get an audio program (Cooledit, Adobe
audition, or other), and you
On Tue, 2005-09-20 at 14:31 -0400, Jonathan k. Creasy wrote:
> Yellowpages.com has a reverse lookup on it.
>
> http://info.yellowpages.com/asp/partner/whitepages/reversephone.asp
>
> As does whitepages:
>
> http://www.whitepages.com/10001/reverse_phone
>
>
http://directory.google.com/Top/Refer
Hi Sander,
Sander wrote:
Hi there does any of you use ip phones from cisco on asterisk and how is
the quality of this configuration ? i have to make a price of an
asterisk server with 100 ip phones but i need stable phones snom is nice
but still i have trouble with echo on them and budgetone i
What version of libtiff are you using. Has anyone tried 3.7.x with spandsp?
Doug Lytle wrote:
Alexander Lopez wrote:
I have used the pre20 package, with the latest CVS-head. COmpile goes
cleanly, NO ERRORS.
then I get this when I try to load asterisk -cvv
[app_rxfax.so]Sep 2
Try
rm -rf /usr/include/spandsp*
rm -rf /usr/lib/libspandsp*
Then do a make install in the spandsp directory..
It may make you smile!
It made me!!
Alex
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Doug Lytle
> Sent: Tuesday, Sep
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> José Pablo Ezequiel Fernández
> Sent: Tuesday, September 20, 2005 2:17 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] RxFax/TxFax - Compile Problem
>
Alexander Lopez wrote:
I have used the pre20 package, with the latest CVS-head. COmpile goes
cleanly, NO ERRORS.
then I get this when I try to load asterisk -cvv
[app_rxfax.so]Sep 20 14:00:23 WARNING[5924]: loader.c:325
__load_resource: /usr/lib/asterisk/modules/app_rxfax.so: un
Yellowpages.com has a reverse lookup on it.
http://info.yellowpages.com/asp/partner/whitepages/reversephone.asp
As does whitepages:
http://www.whitepages.com/10001/reverse_phone
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nathan
Pralle
S
On Tuesday 20 September 2005 15:10, Alexander Lopez wrote:
> I have used the pre20 package, with the latest CVS-head. COmpile goes
> cleanly, NO ERRORS.
>
> then I get this when I try to load asterisk -cvv
>
> [app_rxfax.so]Sep 20 14:00:23 WARNING[5924]: loader.c:325
> __load_resource:
Rene Kluwen wrote:
Some websites allow you to look up a phone number and return a name/address.
As a possible add-on to this, I have an agi script that looks up caller ID
information on a few of these websites.
It is written in C/C++.
I'm not aware of websites like this in the USA or other coun
I have used the pre20 package, with the latest CVS-head.
COmpile goes cleanly, NO ERRORS.
then I get this when I try to load asterisk
-cvv
[app_rxfax.so]Sep 20 14:00:23 WARNING[5924]: loader.c:325
__load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol:
fax_
when a call file is used to place a call FROM an agent the agent is flagged as
busy/unavail even
if the call is subsequently transfered.
call file has..."Channel: AGENT/blah"...
Any way to stop the agent channel being flagged as busy?
Cheers
__
Hi,
On Wed, Sep 14, 2005 at 04:53:54PM +0200, Roger Schreiter wrote:
>
> I have some experience in sending SMSs using smsclient.
> I call the german Vodafone SMSC (01722278020),
> and smsclient takes approx 20 secs to send a SMS.
> The hardware is an Sedlbauer ISDN card.
smsclient seems to be si
Ok.
I was sucessful in installing ODBC storage
I'm using MySQL in the backend as it is. but all my drivers are now ODBC.
I am running asterisk-cvs head as of last night 9/19/05
My question is this... the old voicemail.cgi script that allowed checking
voicemail no longer works etc, and never did
I have a TDM card in a asterisk machine.
I found that once I used it to call out, the call status changed to
"connected" even the callee is still ring.
How could asterisk distinguish the "ringing" and "connected" in zap channel?
thanks.
___
--Bandwidth a
Some websites allow you to look up a phone number and return a name/address.
As a possible add-on to this, I have an agi script that looks up caller ID
information on a few of these websites.
It is written in C/C++.
Currently these scripts are limited to Dutch numbers, since those are
basically th
Paul wrote:
Nathan Pralle wrote:
HooDaHek 0.6 has been released.
As always, information and download linkage available here:
http://www.nathanpralle.com/software/hoodahek.html
Does that mean I could use it with no instant messaging? I would like to
have a local callerID database.
Absol
Hi all,
I hate to ask such a simple question, but it has stumped me over the
past couple of days.
I have 2 asterisk servers connected to the house lan and also via a
crossover ethernet cable. The original purpose of the crossover was to
create a private lan for TDMoE.
I have a TE410P in e
I'm having the same problem you had Frank, so I'm pleased you came up
with a fix. No luck for me yet!
Incoming & outgoing calls work fine using X-Lite, I just cannot transfer.
It's the first time I've ventured in to features.conf so I'm likely
doing something silly. I'd be grateful if you could
What is this "sip-nat-helper" thing, is there a website were we can get
some info on it, partly thinking as the question before was relating to
open source software, I would assume that I could download this thing.
Dan
On Wed, 14 Sep 2005 [EMAIL PROTECTED] wrote:
If you have a linux box, the
Nathan Pralle wrote:
HooDaHek 0.6 has been released.
So soon, you say? Well, the best laid plans of mice and men...
Steven BerkHolz is a pretty sharp stick and said to me, "Why don't you
have HooDaHek change the CallerID when it looks up the name in the
database on an incoming call?" Much
HooDaHek 0.6 has been released.
So soon, you say? Well, the best laid plans of mice and men...
Steven BerkHolz is a pretty sharp stick and said to me, "Why don't you
have HooDaHek change the CallerID when it looks up the name in the
database on an incoming call?" Much head smacketh ensued, a
> Hi there does any of you use ip phones from cisco on asterisk and how is the
> quality of this
configuration ? i have to make a price of an asterisk
> server with 100 ip phones but i need stable phones snom is nice but still i
> have trouble with
echo on them and budgetone is cheap and feels
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