Hello List,
I'm trying to get a PrimuX Card (www.primuxisdn.de) working. The
Manufacturer says that chan_capi (the older one) used to work.
Now I'm trying with chan_capi-cm and have got the following problems:
Outgoing calls "ausgehend_ueber_ntba_primux.txt" the other phone rings
but when I answ
Ronald Ramos a écrit :
Hi All,
Any solution on how I can implement prepaid billing on asterisk?
But not the calling card type, just a simple Custome rwill buy credit,
consume then buy again.
You mean like a rechargable calling card =)
Also, is there a solution for that when you combine ast
In article <[EMAIL PROTECTED]>,
[EMAIL PROTECTED] says...
> Also: What are the SIP CanReinvite settings for these phones?
This shuldn't be important because he have w and W in his dial plan. *
doesn't allow reinvite if you have t, T, w or W.
--
Tomislav Parcina
[EMAIL PROTECTED]
___
Hi All,
Any solution on how I can implement prepaid billing on asterisk?
But not the calling card type, just a simple Custome rwill buy credit,
consume then buy again.
Also, is there a solution for that when you combine asterisk with ser?
Regards,
Ronald
___
On Fri, January 13, 2006 5:15, Jennifer Hales said:
> Hello all,
>
>
>
> I am unable to get automon recording to work; can someone advise me what I
> am doing wrong? When I do *1 all I see in the CLI screen is "attempting
> native bridge of SIP/3006-291b and SIP/3153-6fdd, and there is no call
> r
No an option. Too slow and too resource intensive.
-Original Message-
From: Gonzalo Servat [mailto:[EMAIL PROTECTED]
Sent: Thu 1/12/2006 9:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: [Asterisk-Users
Actually, I was thinking about that...
I've managed to whip up a simple python fast agi script. It starts a new thread
when it gets a connection and uses python MySQLdb. I couldn't create a pool of
database connections for the threads because the docs for MySQLdb say that
isn't thread safe. It
Not approved for sale in Australia though. Curse our draconian
telecommunications laws!!!
Thanks
James
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Jean-Michel Hiver
> Sent: Friday, 13 January 2006 16:50
> To: Asterisk Users Mail
I've since been thinking that the best way to accomplish this would
actually be a TDMoE PRI device, which would take the PRI signalling in
one side and send TDMoE out the other. Software heartbeat and failover
would decide which Asterisk box talked to it. You then have the TDMoE
PRI device as th
probably you need to modprobe zaptel hardware and the tdm card first.
On 1/13/06, Diseyi Diffa <[EMAIL PROTECTED]> wrote:
Hello I am trying to install the zap channel on asterisk and I get thiserror== Parsing '/etc/asterisk/skinny.conf': Found
Jan 12 15:10:22 WARNING[23463]: chan_skinny.c:2587 relo
No problem - we all know what it's like to be a newbie...
PaulH
- Original Message -
From: <[EMAIL PROTECTED]>
To:
Sent: Thursday, January 12, 2006 7:48 PM
Subject: Re: [Asterisk-Users] Dial application newbie help
> Dear Paul H.,
>
> Thanks my dear friend, that worked.
>
> Thanks a l
I got all my whorts burnt off with liquid
nitrogen - and it really hurt.
later,
PaulH
- Original Message -
From:
blackgecko
To: asterisk-users@lists.digium.com
Sent: Friday, January 13, 2006 1:26
AM
Subject: [Asterisk-Users] dCAp
HI, theres a lot of controve
Hi all,
I have configured
Asterisk using Mysql database. The peers that I have mentioned in the database
are successfully registered to Asterisk, But I am getting a warning stating
“Mysql realtime: Failed to query the database” Below I am pasting
error. Could anybody please let me
On 1/13/06, Mike Fedyk <[EMAIL PROTECTED]> wrote:
[..snip..]
> Also an ugly hack would be to call the perl bytecode instead of the text
> script. That would allow for the ease of AGI (everything is cleaned up
> when the process exits) with lower overhead.
>
> FastAGI is of course what you want f
Andreas Sikkema wrote:
Is it possible to have nested MySQL queries in extensions.conf?
Ie, perform a query, grab a value, and then jump to another
location in the dialplan and do another query based on that
original value. I'm having problems with the result and
fetchid's and I'm not sure if
I agree with you on one point - with PRI, you can answer before any rings...
PaulH
- Original Message -
From: "James Harper" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Friday, January 13, 2006 9:42 AM
Subject: RE: [Asterisk-Users] Why can Ast
Jennifer Hales wrote:
I am unable to get automon recording to work; can someone advise me what I
am doing wrong? When I do *1 all I see in the CLI screen is "attempting
native bridge of SIP/3006-291b and SIP/3153-6fdd, and there is no call
record generated in /var/spool/asterisk/monitor/.
I c
Simone Cittadini wrote:
Douglas Garstang ha scritto:
So I really wish there was some way to measure how well the worst
case scenario would perform. This would be 120 simultaneous calls
(don't know how many per second) on a Dual 3.8Ghz Dell PowerEdge 1850
with 2GB RAM. Asterisk would call an
Hello all,
I am unable to get automon recording to work; can someone advise me what I
am doing wrong? When I do *1 all I see in the CLI screen is "attempting
native bridge of SIP/3006-291b and SIP/3153-6fdd, and there is no call
record generated in /var/spool/asterisk/monitor/.
Here are my
Yo!
I changed callprogress to no, and in wcfxo.c source around line 334 i changed
the value 32000 and -32000 to 1 and -1 because it had something to do
with the DC voltage when it was ringing.
I found reference here (http://www.voipuser.org/forum_topic_1791.html) with an
interesting diag
I am hoping some of you can help me troubleshoot this problem I am
having with my home asterisk machine. I have incoming POTS service
using a SPA-3000 (extension 119). Calls on that line go to an
attendant recording that offers a menu choice: press 1 for Nancy,
press 2 for the rest of us. In rea
If it is a toll free number, it may be related to
http://bugs.digium.com/view.php?id=5266 .
-- -- Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.--- -
--- - - - -
- - - -- - - - --- -
-- - - --- - - --
Is there a way to add directory entries for people that do not have VM or local
VM?
I still have a PBX behind my Asterisk and would like to add the PBX extensions
to my to the Asterisk directory.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of having
a
Sorry if this is slightly off topic but it does pertain to Asterisk
Users as well as the biz list
Hello all,
I have created a beta site for "Asterisk Gurus" or Consultants to bid on
projects posted by customers needing to have work done. It is very
similar to scriptlance or any of those ot
On Friday 13 January 2006 15:59, James Harper wrote:
> Can anyone recommend a PRI-to-TDMoE device? Does such a thing exist?
Have you seen the Redfone foneBRIDGE? I have no experience of it but it seems
to be what you are after.
HTH
hads
--
I WILL TRY TO RAISE A BETTER CHILD
I WILL TRY TO RAIS
>
> Geoff Manning wrote:
> > Rich Adamson wrote:
> >
> >> What happens if you set the 3com AND Speedtouch to "full duplex"?
> >>
> >
> > Setting both to Full Duplex (10 or 100) shuts the port on the 3COM
> switch
> > down!
>
> Sounds like you're getting closer to the root cause. Now try using a
>
>
> Matt wrote:
> > On 1/12/06, Tomislav Parcina <[EMAIL PROTECTED]> wrote:
> >> In article
> <[EMAIL PROTECTED]>,
> >> [EMAIL PROTECTED] says...
> >>> First,
> >>> Something seems to be wrong with the list. I'm not the only
person
> >>> who has expressed seeing their messages either arrive late,
A very good day to you all,
We can't get the phones to pick up on an incoming call on analog trunks.
We're using the digium products in the box, with snom phones internally.
This is the output from the asterisk console:
linux*CLI> zap show channels
Chan Extension Context Language
Matt wrote:
On 1/12/06, Tomislav Parcina <[EMAIL PROTECTED]> wrote:
In article <[EMAIL PROTECTED]>,
[EMAIL PROTECTED] says...
First,
Something seems to be wrong with the list. I'm not the only person
who has expressed seeing their messages either arrive late, or not at
all.
I'm sure that I'm
Geoff Manning wrote:
Rich Adamson wrote:
What happens if you set the 3com AND Speedtouch to "full duplex"?
Setting both to Full Duplex (10 or 100) shuts the port on the 3COM switch
down!
Sounds like you're getting closer to the root cause. Now try using a
different switch to see if the is
Try something like this in sip.conf
nat=yes
externip=210.77.45.46
localnet=192.168.1.125/255.255.255.0
canreinvite=no
Cheers,
jergas
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Zeeshan
Enviado el: Jueves, 12 de Enero de 2006 13:46
Para: 'Asterisk Use
Have you settled on a calling card application yet? There are a host of
different options. I, of course, recommend astpp. :-) The wiki will
have much of the info you will need.
Darren Wiebe
[EMAIL PROTECTED]
Dirgan Putra wrote:
Iam new in asterisk user, can helpme to install asterisk for
Iam new in asterisk user, can helpme to install asterisk for applications callingcard ?current ialready install asterisk with mysql db and already connected, and next i dont know how to create as calling card applications, adn how other how to setup using SIP sofphone if iam using expresstalk softp
Jorge nosotros tenemos problemas de fax solo con lineas de telemex con
lineas de axtel no. Y no hemos podido tampoco detectar el problemas.
SI logras resolverlas hasnolo saber porfabor.
On 12/27/05, Martinez Felix <[EMAIL PROTECTED]> wrote:
no podria decirte, porqe tengo problemas con los scripts
On Thu, 2006-01-12 at 16:53 -0800, William Boehlke wrote:
> We train ten to fifteen people every month in our three day course, and
> tried offering the exam for a while.
>
> The practical exam is pretty easy. The written exam, in our opinion, is too
> hard and not very relevant to the life of an
Hi,
You can try changing your section name ([UTStarcomF1000]) to the user
name, i.e. [anonymous]. I also noticed that you had a typo in the
'dtmfmode' line; it should be 'rfc2833' and not 'rfca2833'.
-kokmeng.
Christoph Merk wrote:
Hi there,
I am trying desperatly to register my WiFi Phone
hi All iam a new one in asterisk, can somebody guideme to install asterisk for prepaid calling card, now iam already install asterisk + mysql DB addons and i dont know next install other addons or no ?, and how to setup sipphone with , if iam install Asterisk management portal, current i see in Mt
We train ten to fifteen people every month in our three day course, and
tried offering the exam for a while.
The practical exam is pretty easy. The written exam, in our opinion, is too
hard and not very relevant to the life of an Asterisk expert.
Our take is, wait until the exam is revised so th
Ahhh, sorry send you the supplementary services. This is the equal to
Q.931 - Layer 3.
http://www.ecma-international.org/publications/files/ECMA-ST/Ecma-143.pdf
Jan
[EMAIL PROTECTED] wrote:
Anyhow, I'm now thinking about using QSIG, I've configured two spans
of a TE405P as switchtype =
I think he was using a buggy TIFF viewer. There are a *lot* of buggy
TIFF viewers.
Steve
James Sizemore wrote:
Shawn, you ever get a fix for this problem?
> samples are at
> http://tumtum.no-ip.com/faxes/1128432831.3.tif
> http://tumtum.no-ip.com/faxes/853107320051004-150908.tif
> Both of
Well, Philipp .. as it is a development of DLINK Germany, they can name it
whatever they like, even if it sounds strange ;-) ...
Nevertheless, the box looks like quite competitive thing, with ISDN FXO and
FXS, 3 POTS FXS, WLAN etc ... Hard to compete for an embedded system like
the Fritz!box
Anyhow, I'm now thinking about using QSIG, I've configured two spans
of a TE405P as switchtype = qsig and connected them back to back for
testing and calls are ok. However, I cant find any documentation
relating to what messages I might see come across the link once I
connect it to the IND
Hey, I tested this today, and it's working!
Thanks!
> I think, the Group-Function in Asterisk 1.2 is what You are looking
for.
> In older versions the Group()-Function was implemented as application
> SetGroup.
> More information can be found in the wiki:
> (http://www.voip-info.org/wiki-Asteris
Hi,
try a different PCI slot or contact your vendor to replace the card.
Plex
2006/1/13, Diseyi Diffa <[EMAIL PROTECTED]>:
Does anyone one know why I am getting this after a install the zapteldriversZT_SPANCONFIG failed on span 1: No such device or address (6)
_
Found this today:
D-Link has apparently positioned itself to challenge the AVM Fritz!Box
(which is an analog & ISDN home PBX including DSL modem, router, LAN &
WLAN). The PBX service of this device called "HorstBox Professional" DVA-
G3342SB will be provided by Asterisk, and it should become avail
On Thursday 12 January 2006 17:24, Dakota wrote:
> I have an Asterisk installation, however whever someone calls into it, they
> hear it ring two times, before the Auto Attendant comes on.
>
> Is there a way to get the system to answer on the first ring?
Turn off caller ID detection. With it on,
Does anyone one know why I am getting this after a install the zaptel
drivers
ZT_SPANCONFIG failed on span 1: No such device or address (6)
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To UNSUBSCRIBE or updat
On 1/12/06, Jean-Michel Hiver <[EMAIL PROTECTED]> wrote:
>
> What interface is your current PBX using?
The current PBX has a T1 PRI.
To clarify - the current PBX isn't going away - I'm just adding an
asterisk box to the mix.
___
--Bandwidth and Colocati
hugolivude wrote:
I checked out the wiki. Just to confirm, if I issue the command:
# svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk
I'd get the most recent _working_ version of Asterisk. But if I issue
the command:
# svn checkout http://svn.digium.com/svn/asterisk/branches
Erik Anderson a écrit :
Greetings all -
I'm interested in using an asterisk box to supplement and add VoIP
capabilities to our legacy InterTel Axxess PBX. After searching
through the list archives and through google, it seems that the best
way to go about this is to connect the two systems via
On Thu, 12 Jan 2006, Christoph Merk wrote:
> Pls, where do I find asterisk-capi
> I am using now asterisk 1.2.1 with a SuSE 9.3
> in SuSE 9.3 there was the old version for 1.0.6 ... can I use that old
> asterisk-capi for the current and on my system installed version 1.2.1 ???
http://sourceforge.n
I have a fast agi python script that reads some numbers from MySQL, and then
instructs asterisk to try those numbers in sequential order.
ie:
def run(self):
agi = AGI(self.client)
db =
MySQLdb.connect(host="192.168.10.15",user="user",passwd="password",db="somedb")
c
Someone will probably tell you with more certainty, but (you don't say
but I assume you are talking about FXO) the Caller ID normally comes in
between the first and seconds rings, I think you can tell asterisk not
to get the CID but if you don't, it waits for it.
Also, I remember reading in a mode
ver 1.17.2
[EMAIL PROTECTED] openh323_v1_17_2]# make opt
/usr/src/openh323_v1_17_2/openh323u.mak:192:
usr/src/pwlib_v1_9_1/make/ptlib.mak: No such file or directory
make: *** No rule to make target `usr/src/pwlib_v1_9_1/make/ptlib.mak'.
Stop.
[EMAIL PROTECTED] openh323_v1_17_2]#
thx in advan
Has anyone successful setup up asterisk with cisco router with FXO
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http://lists.digium.com/mailman/listinfo/asterisk-use
ver 1.9.1
-I. -shared sound_oss.cxx -o ../pwlib/device/sound/oss_pwplugin.so
sound_oss.cxx: In member function virtual BOOL
PSoundChannelOSS::Read(void*, PINDEX):
sound_oss.cxx:766: error: cast from void* to unsigned int loses
precision
make[3]: *** [../pwlib/device/sound/oss_pwplugin.so
I have an Asterisk installation, however whever someone calls into it, they
hear it ring two times, before the Auto Attendant comes on.
Is there a way to get the system to answer on the first ring?
---Dakota
___
--Bandwidth and Colocation provided
300 calls depends on your mix of work.
If you are just switching calls and have hardware echo cancellation you can
put 300 calls across a fast server with adequate RAM. With that much
traffic, we would use two servers so you can fail gracefully when one of
them goes down.
If you're transcoding,
David C. Nicosia wrote:
You can search by either:
Directory(vm-context[|dial-context[|options]])
The 'f' option causes the directory to match based on the first name in
voicemail.conf instead of the last name
That I didn't know, thanks!
Doug
I've studied several aspects of asterisk's logging capabilities, and I'd
like to know if there is any built in functionality that could trigger an
event based on a warning or an error message. I'm thinking along the lines
of sending an email, or making an outbound call to my cell to TTS the error.
You can search by either:
Directory(vm-context[|dial-context[|options]])
The 'f' option causes the directory to match based on the first name in
voicemail.conf instead of the last name
-Original Message-
From: Doug Lytle [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 12, 2006 10:55 A
Hello,
Is the hardware specification is enough to get 300 simultaneous calls?
What should be the Bandwidth to get 300 simultaneous calls?
--
Thank You,
Code Lover
___
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T
I asked the [EMAIL PROTECTED] for the documents and the tools that
are referenced in the admin guides and was told that I had to become
a registered user in the support section of the ww.sipura.com website.
They wanted name, title, phone # and type of support I provide for
the devices.
I think I
Hi all,
I'm planning on connecting our Asterisk to our legacy PBX (an Avaya INDeX).
I was originally going to sit it between our ISDN connection and the INDeX
(tried it, worked ok) but now I intend to hang it off a spare PRI card just
so should the * fail we keep our ISDN's at the INDeX, yes I
Hello I am trying to install the zap channel on asterisk and I get this
error
== Parsing '/etc/asterisk/skinny.conf': Found
Jan 12 15:10:22 WARNING[23463]: chan_skinny.c:2587 reload_config: Unable to
get
our IP address, Skinny disabled
== Registered channel type 'Skinny' (Skinny Client Cont
I have the 941 admin guide in zipped PDF format, I
have sent it to a lot of people on the list, if you need it, email
me.
Cory AndrewsPurchasing
Manager++VOIPSupply.comA Division of b2
Technologies454 Sonwil DriveBuffalo, NY
14225direct - 716.250.3402mo
I don't think they have a specifig Provisioning Guide for each device. They
have a general provisionning guide and you can generate an example from the
Sipura Profile Compiler for the available options though
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Take a look at http://www.sipura.com/support/index.htm. They have an IP Phone administrators Guide that talks about provisioning. It is for the 841 phone but it might give you some hints.
Richard Radcliffe
Owner, Kondor Waffenamt
[EMAI
I've been experiencing some crashes in Asterisk in the past few weeks. I haven't been able to find out why as gdb shows it's in a different function every time. But, in the meantime, I've been using safe_asterisk hoping that it would simply restart Asterisk by itself. It doesn't seem to do that.
Shawn, you ever get a fix for this problem?
> samples are at
> http://tumtum.no-ip.com/faxes/1128432831.3.tif
> http://tumtum.no-ip.com/faxes/853107320051004-150908.tif
> Both of these were faxed from a Brother intellifax 750 through a ring-it
> single-line simulator into my asterisk box (thro
does anyone get a hold of the SPA-941 Provisioning Guide?
i tried call Sipura's tech support, seems like none of
them heard of the term "remote provisioning". they kept
refering me to their web site which i've check thoroughly,
and could not find any documentations on the SPA-941. finally
they gav
Greetings all -
I'm interested in using an asterisk box to supplement and add VoIP
capabilities to our legacy InterTel Axxess PBX. After searching
through the list archives and through google, it seems that the best
way to go about this is to connect the two systems via a T1. Is this
correct? T
well roughly 80 calls on g729 or 120 on g711, figures may differ in
realtime, 100 gb bandwidth may not be sufficient, you will have to know the
actual throughput too
you should check this tool for bandwidth calculation
http://www.asteriskguru.com/bandwidth_calculator.php
Diyanat
From: Ab
Connection pooling doesn't require threading.
You can also use a pool of processes which are quite cheap on Linux.
Douglas Garstang wrote:
Do you have a link to where it says this? The DBI docs that I looked at
(perldoc dbi) said that it isn't thread-safe.
-Original Message-
From: Le
[New Features]
1. Added focus and tab-order to all input fields
2. Dynamic generation of date/month/year listboxes
a. It is no longer possible to schedule an invalid
date.
3. Added 'Extend' and 'End Now' buttons to the monitor
page.
Hi All,
I was making plan to set an VoIP Gateway in India. And
found some copanies who offered me to host my Asterisk
server.
I will be appriciated if anyone can suggest me how
much simultaneous calls can be handeled with the
following server specification?
CPU : Dual Intel® Xeon® Processor at 2
Hi, is there any good calculator/table/reference about proper dimensioning?
I read the wiki and they basically say "xx users run fine in yy hardware"
http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning.
SO far I read that:
-Run up to 4 E1s per CPU (which one? an i386 or a dual core?
On Thu, 2006-01-12 at 13:48 -0600, Schochet, Wes wrote:
> Hi All-
>
> I am trying to create a post call survey application. I would like to:
>
> 1. ask the caller if they want to take a survey after their call completes
> 2. If no, just transfer the call
> 3. if yes,
> 4. bridge up anothe
I have been getting the following messages on Asterisk for a couple of my
client's SNOM phones:
7881 Jan 12 14:52:22 NOTICE[6538] channel.c: Dropping incompatible voice
frame on Local/[EMAIL PROTECTED],2 of format slin since our native format has
changed to ulaw
...
8185 Jan 12 14:52:28 NOTICE[65
Really interesting.
Thanks Hannes!!
Hannes Vogel wrote:
I've written myself a easy to use telephone directory
which I use at home and thought it may be of interrest
to others.
The purpose of this agi script is to provide an online
telephone directory that can be easily accessed using
the numb
Hey all again, I'm wrestling with echo problems on our sip extensions. I've
set these items in zapata.conf but tweaking these values doesn't seem to
make much difference
echocancel=yes
echocancelwhenbridged=yes
echotraining=2500
rxgain=8.0
txgain=1.0
are there other settings that can help me ta
I've written myself a easy to use telephone directory
which I use at home and thought it may be of interrest
to others.
The purpose of this agi script is to provide an online
telephone directory that can be easily accessed using
the numbers on the phone dial pad.
You select entries by spelling o
Hi All-
I am trying to create a post call survey application. I would like to:
1. ask the caller if they want to take a survey after their call completes
2. If no, just transfer the call
3. if yes,
4. bridge up another extension
5. wait for that extension to hang-up
6.
Zeeshan a écrit :
Hi everybody,
One of my client's Asterisk box is behind NAT. They have only one public
IP on which they have their router. I can access the Asterisk server
using port forwarding (port 22) for SSH. Now this client wants to
connect two SIP phones to this Asterisk box from two re
But for us?
From: William Boehlke
[mailto:[EMAIL PROTECTED] Sent: Wednesday, January 11,
2006 2:24 PMTo: 'Asterisk Users Mailing List - Non-Commercial
Discussion'Subject: RE: [Asterisk-Users] Second edition of my * book
has been released
$39.95 retail.
From:
[EMAIL PROTECTED]
Christopher-Nothing like defining a complicated environment. I do have some experience in this arena- but unfortunately, not with the OH323 driver- I generally stick to the Nufone driver, as I find it more reliable overall. YMMV. One thing that might help is if you could tell us if it ever worke
You may be having an issue with the arguments.
Try 'wraping the script within another (ie runbr)
Also login as asterisk user, you may need to change /etc/passwd to give
asterisk a shell, I am not sure as I don't do @ Home.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAI
I had a very similar problem some months ago, was using a Sangoma A101 card though. The problem was something related to the card's memory and was able to solve it by updating the driver. It was caused due to I was using a brand new card with a not so updated driver (I was using one that I thought
Rich Adamson wrote:
> What happens if you set the 3com AND Speedtouch to "full duplex"?
>
Setting both to Full Duplex (10 or 100) shuts the port on the 3COM switch
down!
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It is running as asterisk ([EMAIL PROTECTED] default). I have tried 'chown
asterisk:asterisk br' as well as 'chmod 775 br' but nothing seems to help.
Kaleb
-Original Message-
From: Alexander Lopez
Sent: Thursday, January 12, 2006 11:59 AM
Subject: RE: [Asterisk-Users] Using an extens
I currently do this for about 30 different cisco 79xx's connecting to
some hosted Asterisk servers.
Asterisk listens by default for any SIP connection on UDP port 5060.
And will use RTP UDP port 1 to 2
The phones use UDP Port 5061 for incoming connections (from Asterisks or
other SIP Dev
Sorry, I don't know how to forward a range of ports. To forward a single port, use something like: ip nat inside source static udp 192.168.1.2 5060 x.x.x.x 5060 extendable where x.x.x.x is your public IP. just add the range ports tih a ":" e.g 192.168.1.2 1 : 10007 > (4)Please,I know alot of
What user is yoru asterisk service running as?
It is probably a permissions or path issue.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Kaleb L. Kunzler
> Sent: Thursday, January 12, 2006 1:43 PM
> To: asterisk-users@lists.digium.com
> Subj
I am a newbie to asterisk and am trying to send a linux command using
extensions in asterisk, for example when I dial I want to run the linux
command "/usr/local/bin/br -c -n 1" (obviously without the quotes). If I
SSH into my asterisk box and enter that command, it works, however I can't
s
On Jan 12, 2006, at 8:33 AM, Kevin P. Fleming wrote:
BJ Weschke wrote:
It has just been fixed in /trunk
That fix should be merged over to branch-1.2 as well then, if needed.
It shouldn't be; it's something left over from a merge into HEAD that I
made the other day.
Matthew Fredrickso
On Thu, January 12, 2006 19:18, Ben Ferguson said:
> Hello all. I've been searching and can't quite find what I'm looking
> for...
>
> I've gotten AMP installed and up and running quite decently on an Asterisk
> box and am now in the process of tweaking it to my needs. My company
> currently has
Title: Message
Hello
all. I've been searching and can't quite find what I'm looking
for...
I've
gotten AMP installed and up and running quite decently on an Asterisk box and am now in the
process of tweaking it to my needs. My company currently has around 70
employees and we are runnin
Just out of curiosity, how many of you are using trunk in a production environment? Are you performing regular compilations of the code as well? Do you explicitly prefer trunk over stable, or vice versa?Ronald Lewis
www.ronaldlewis.com
___
--Bandwidth and
I would be useful if you could post your config files and the pri debug as well. check your zapata.conf or paste it here so we can take a look.AlyedReturn-Path: <[EMAIL PROTECTED]> Thu Jan 12 10:04:28 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcent
We've got this configuration :
Cisco as5400 --- asterisk main server asterisk for cells gsm
gateway
cisco and the gsm gateway are connected to asterisk via sip, the two
asterisk servers are connected via iax.
On a succesful call the cisco (not always, 60% of the times) will keep
send
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