Hello All,
I have Cisco 7960's, Cisco 2950 Switch, SUSE 9.2,
PIX Firewall. Here is my issue I can dial out no issues but when someone
calls in the phone rings I answer and the phone disconnects the call.
Call from my cell to my house I answer the cisco
phone it then disconnects the call
www.babytel.ca
Does anyone have experience, good or bad with this provider ?
--
Kim Lux, Diesel Research Inc.
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Hi,
I've got the Caller ID name and number working with the application
SetCIDNumber and SetCIDName.
[...]
exten => s,3,SetCIDNumber(4183289901)
exten => s,4,SetCIDName(Frank Black)
exten => s,5,Dial(IAX2/prov01/${DEST})
[...]
You can also use SetCallerID(Frank Black <4183289901>), but no succes
DID not correctly provisioned? Hmm interesting. I seem to be having the same issue with them.
Unfortunately, most every other provider, for my area code, 405, says they require using their equipment and charges a fairly significant setup fee. Too much for a proof of concept. Otherwi
Hi Everybody,
Sure this has been covered a million times on wiki, but couldn't find an
exact answer to my question. I am using * to dial out to peoples phones to
give them alerts of different things. Problem is that the only Caller ID I
can get working is the telephone number. I am unable to
David wrote:
Hello All, I'm trying to dial out with no luck.
I'm using [EMAIL PROTECTED] defaults. I have one X100P card and SJPhone.
*CLI> dial 96985628
No such extension '96985628' in context 'default'
This is kinda self explanatory. A context is the bit between the two
square brackets [ and ].
Hello All, I'm trying to dial out with no luck.
I'm using [EMAIL PROTECTED] defaults. I have one X100P card and SJPhone.
*CLI> dial 96985628
No such extension '96985628' in context 'default'
Here is my exten
[trunklocal]
;
; Local seven-digit dialing accessed through trunk interface
;
exten =>
Hello.
I have set up ASTCC properly, calling it like this:
DeadAGI(${ACCOUNTCODE},${EXTEN})
It seems to be working correctly, but I have two questions:
- Although the cards' credit seems to be maintained correctly, I cannot
see the call details in astcc-admin. When I try to view information on
On Sun, 2005-01-09 at 12:12, Tim Davidson wrote:
> I've looked around so hard for days now for a solution to my problem.
> I'm new to asterisk but I've managed to get an IAX coinnection working
> to voiptalk.org, ive got sip to sipgate. But I just can't get my
> X100P clone working.
Have you don
Here is the discovery that I have made.
* I can Ping my phone w/o any problems.
* Near as I can tell, I have set up the phone correctly in sip.conf
* When I set up sip debug and/or sip debug ip and I reboot my phone, I
do not see any activity from the server from my phone.
Username Secret
Here is the discovery that I have made.
* I can Ping my phone w/o any problems.
* Near as I can tell, I have set up the phone correctly in sip.conf
* When I set up sip debug and/or sip debug ip and I reboot my phone, I
do not see any activity from the server from my phone.
Username Secret
Hi John,
On 8 Jan 2005, at 23:51, John Middleton wrote:
I am tying to clear down an asterisk source directory before CVS'ing a
new version
the --ignore... option is being used but its still not being deleted,
can anyone give me some clues.
When I check out a new CVS version, I move my /usr/src/aste
Quoting from message: 05/01/08 19:12 +0900 sent by Arthur B Olsen:
>I think you need that provisioning tool from digium. And you need a unix
>system to compile and run it. I dont think theres any port to your OS.
Quoting from message: 05/01/08 20:14 +0900 sent by Wilson Pickett:
>Someone you with
Scott,
Right now I am using the Netphone KE1020A.
~Dan
Message: 11
Date: Sat, 08 Jan 2005 13:34:29 -0900
From: Scott Henderson <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Re: Connecting Sip phone to asterisk.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Message-ID: <[E
I've looked around so hard for days now for a
solution to my problem. I'm new to asterisk but I've managed to get an IAX
coinnection working to voiptalk.org, ive got sip to sipgate. But I just
can't get my X100P clone working.
I'm running Redhat 9
It's got IRQ 5 all to itself
ztcfg likes
rmdir will only remove empty folders and --ignore just prevents error
messages being displayed.
Run the command: rm -rf *
in the asterisk root folder and then execute rmdir
Bill Seddon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Middleton
Sent
> I am tying to clear down an asterisk source directory before CVS'ing a
> new version
> the --ignore... option is being used but its still not being deleted,
> can anyone give me some clues.
>
> Sorry I'm new to Linux, as if you havent guessed. Googling hasnt helped so far
The easiest way (leav
I am tying to clear down an asterisk source directory before CVS'ing a
new version
the --ignore... option is being used but its still not being deleted,
can anyone give me some clues.
Sorry I'm new to Linux, as if you havent guessed. Googling hasnt helped so far
Thanks
___
<< Is it possible to set asterisk up as a virtual pbx like in apache and
virtual host?>>
You have provisioning that may address some or all of your needs. Mark
Spencer talks about other possible deployment options in response to
questions at the end of this presentation...
http://graphics.cs.un
Bill,
Are
you sure there's an AGI enviroment variable that gives me that? I couldn't find
any:
-- accountcode = -- callerid = "Assaf Benharoosh"
<21> -- channel = SIP/26-f39a -- context =
extensions -- dnid = 45 -- enhanced = 0.0 --
extension = 45 -- language = en -- priority = 1 --
rdn
Hi,
> substantially lower (pretty much unhearable) on the NTL line compared
> to the BT Line, so you have a fairly good chance (hopefully) of getting
> a decent solution.
I hope so :)
> 00:0a.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
> Modem/ISDN interface
Yeah, looks l
Comments within:
If you stick with fxsks and add busydetect=no and callprogress=no
you'll find those random disconnects go away. They did for me.
I just tried that and I still got the disconnection so I changed it
back
to fxsls but left those 2 lines in zapata.conf for good measure!
Yes. The e
Since a couple of days I using an Asterisk server. I noticed something
obvious to anybody dealing with telephony since any longer time than
myself; echo is nasty...
Is it correct to say that the difference between a conversation between
SIP phone, Asterisk, an ISDN BRI line and a GSM phone is enti
Hi,
> Try echotraining=800 in zapata.conf
Thanks for the suggestion but I tried that and still get the echo :-(
Any other ideas?
--ian
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Hi,
> If you stick with fxsks and add busydetect=no and callprogress=no
> you'll find those random disconnects go away. They did for me.
I just tried that and I still got the disconnection so I changed it back
to fxsls but left those 2 lines in zapata.conf for good measure!
> Yes. The echo dr
Try echotraining=800 in zapata.conf
From: Ian Chilton <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] Echo on Zaptel FXO :(
Date: Sat, 8 Jan 2005 22:02:43 +
To: asterisk-users@lists.digium.com
> Hi All,
>
> I've got an MD3200 modem which is working as a Zapt
Comments within:
Change fxsls to fxsks for the UK
I had fxsks until today but when I was playing with the gains to try
and
fix this echo issue, I found that if I dial out the Zap to one of my
pstn->sip provider's numbers, it answered the cut off in a few seconds.
If I dial that number on the pst
Well, that'd be quite a trip for a cup of coffee! But you'd be very
welcome for sure!
Jim.
Wojciech Tryc wrote:
> well,
> I am in Ottawa...only 50mins by air :)
> Wojtek
> - Original Message -
> From: "Leif Madsen" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>; "Asterisk Users Mailing Li
Hi Phil,
Thanks for the reply - comments below.
> Change fxsls to fxsks for the UK
I had fxsks until today but when I was playing with the gains to try and
fix this echo issue, I found that if I dial out the Zap to one of my
pstn->sip provider's numbers, it answered the cut off in a few seconds.
One more thing, what phones are you using?
[EMAIL PROTECTED] wrote:
The phone is configured as:
IP Phone Number: 1201
Username: 1201
Password:
Service Address: 192.168.0.104
Sip.conf is configured as:
[1201]
type=friend
username=1201
secret=
mailbox=1201
host=192.168.0.99
To keep the redundant dat
I am guessing that the core of your problem here is that the phone isn't
registering with the server (a statement of the obvious). Anyway I
just went through this on a project and if you study the debugs, you
will see where the registration falls down. Mine was a series of typos
on both the
Comments inline:
On 8 Jan 2005, at 22:02, Ian Chilton wrote:
Hi All,
I've got an MD3200 modem which is working as a Zaptel FXO interface for
Asterisk (X100P clone I believe). It seems to work, but on incoming or
outgoing calls I can hear the other party ok but when I speak, I hear
my voice echo bac
On January 8, 2005 04:49 pm, Wojciech Tryc wrote:
> I am in Ottawa...only 50mins by air :)
And 90 minutes in the airport before, and about 30 after. :-)
-A.
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Hi All,
I've got an MD3200 modem which is working as a Zaptel FXO interface for
Asterisk (X100P clone I believe). It seems to work, but on incoming or
outgoing calls I can hear the other party ok but when I speak, I hear
my voice echo back at me (quite quietly but it's distracting!) on
everything
well,
I am in Ottawa...only 50mins by air :)
Wojtek
- Original Message -
From: "Leif Madsen" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - Non-Commercial
Discussion"
Sent: Saturday, January 08, 2005 12:37 PM
Subject: Re: [Asterisk-Users] Toronto?
On Sat, 8
Mark just committed a small fix of mine to FastAGI. Previously there was
a script option to the URI that wasnt't used. Now, it's sent to the AGI
server so that one running server can handle multiple AGI functions.
agi://hostname:port/script
is the full syntax for the fastagi option to th
Is it a HP-Compaq DLXXX G4 machine? Because there is a thread here
about the TE410P not generating interrupts with these servers...
On Sat, 8 Jan 2005 05:14:39 -0800 (PST), Sid <[EMAIL PROTECTED]> wrote:
> Hi Scott, and Jack,
>
> --- Scott Stingel <[EMAIL PROTECTED]> wrote:
>
> > Sid-
> >
> > T
Joel,
I know there are some very vocal critics of the TDM cards on this newsgroup,
but we also have many customers who are using them with no problems. There
are many variables involved and I am not personally convinced that it is
fair to say all of the reported problems are the fault of the Digi
The phone is configured as:
IP Phone Number: 1201
Username: 1201
Password:
Service Address: 192.168.0.104
Sip.conf is configured as:
[1201]
type=friend
username=1201
secret=
mailbox=1201
host=192.168.0.99
To keep the redundant data down, here is what the sip debug shows:
Retransmitting #5 (no
Title: Re: [Asterisk-Users] Best gateway to use for *?
Alexander,
Would you care to recomend a Mother
board?
Thanks in advance,
Seth
- Original Message -
From:
Alexander
Lopez
To: asterisk-users@lists.digium.com
Sent: Saturday, January 08, 2005 3:11
PM
Su
Damon,
Thanks for your comments.
This seems like serious problem, how is it done then
in real world? I cannot see any business moving to
voip when you cannot control quality of service.
Can you recomend any free programs which alow you
analyze delay,jitter and packet loss?
Once again thank you.
rob
Title: Re: [Asterisk-Users] Best gateway to use for *?
I have to date NOT had a problem with the Digium HW. You just got to pick the right Mobo.
-Original Message-
From: [EMAIL PROTECTED] <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED] <[EMAIL PROTECTED]>; Asterisk Users Mailing List - Non
Sipura spa-3000, about $100 each, so 4 will take care of your 4 fxo need
and you will also have 4 fxs ports, there is one fxo and one fxo per
unit. Digium cards would be the ideal solutions, but there does seem to
be some issues that Digium is not resolving quickly.
> -Original Message-
>
Hi All,
I'm trying to record a phone call.
I'm using the Monitor command with the "m" flag for a SIP to SIP call.
I'm running:
Asterisk CVS-HEAD-12/17/04-16:55:26
One side of the call is significantly quieter than the other. Am I doing
something wrong??
Thanks,
Brett
___
> Are there any ways to get around this problem? Is there a way
> to timeout if "ringing" doesn't happen in 5 secs (for
> example) and go to the backup provider?
Anyone?
--
Nabeel Jafferali
tel: 416.491.9136 (toronto)
646.225.7426 (new york)
fwd: 46990
email/msn : nabeeljafferali.net
__
> Anyone in the Toronto area interested in getting together to share
> notes and swap war stories?
One of the other guys in Toronto interested in * put together a
meetup.com group. Please join in and we can see where to go from there.
http://opensource.meetup.com/42/
--
Nabeel Jafferali
tel: 41
On Sat, 8 Jan 2005, Michael Graves wrote:
> I once read that by PSTN standards a good connection should be less
> than 150 ms. I think it was an older Network World review of voip
> phones. The real issue is due the callers step on each others speaking
> due to latency.
Once you get up to 50-100
Hi Jim,
On Sat, 2005-08-01 at 05:40 -0500, Jim Van Meggelen wrote:
> Anyone in the Toronto area interested in getting together to share notes
> and swap war stories?
I'm in Brampton.
I don't have war stories per se, but I have been using Asterisk for my
consulting business for about 8 months now
Title: Re: [Asterisk-Users] virtual pbx
Asterisk IS sleady there! Understand the dialplan and the various settings in voicemail.conf and you got it.
-Original Message-
From: [EMAIL PROTECTED] <[EMAIL PROTECTED]>
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Sat
There is an extra field in the database called 'userfield' whose value
you can set with the SetCDRUserField() command in your dialplan.
Personally, I use it for recorded calls; I write the recording file name
into that field so I can display it for each recorded call.
-Original Message-
Fr
On January 8, 2005 12:59 pm, Eric Wieling wrote:
> With a PRI you just use the commands to set the CallerID in the
> dialplan. "show applications". You can set the number, but not the
> name. The name will show up on the other end as whatever the telco
> has on file for that number.
Not entirel
Following is sharelessly copied from one of the newsgroups I read on
grc.com..
/Soren
NIST issues recommendations for secure VOIP
http://www.gcn.com/vol1_no1/daily-updates/34747-1.html
http://csrc.nist.gov/publications/nistpubs/800-58/SP800-58-final.pdf
*
Does anyone know some free MGCP
softphone?
Nowadays I'm using one from eyeP Media, but it is
trial for 30 days and it's expiring...
Any ideas?
Thanks,
Leonardo
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Mark Halverson wrote:
My local Telco uses B8ZSESF and does support PBX customizing ANIs on a per
call basis.
What I need to know is, can I use the SetCallerID command in extensions.conf
to transmit the DID# of the extension making the call with the TE410P or is
there a different one that does supp
On Sat, 2005-01-08 at 11:17 -0500, Andres E. Moya wrote:
> Is it possible to set asterisk up as a virtual pbx like in apache and
> virtual host? If so can someone point me to the right direction.
> I would also like to setup asterisk with some type of redundancy, I have
> searched the lists and
I'm setting up an asterisk system between a couple locations.
Questions:
a) What is the correct term for for a connection with a "phone company"
where by we connect to them via H323 and they connect us to PSTN,
allowing us to make PSTN calls and allowing others to call us via PSTN ?
I've call
On Sat, 8 Jan 2005 05:40:02 -0500, Jim Van Meggelen <[EMAIL PROTECTED]> wrote:
> Anyone in the Toronto area interested in getting together to share notes
> and swap war stories?
I'm in Oakville, right across from Sheridan College. So I guess I can
be considered part of the GTA at least.
But you
Hello
Whenever i tried to call through my dialer, im getting this error message there:
"484 Address Incomplete"
While i am using MySQL database for user authentication etc.
; extensions.conf
switch => Realtime/[EMAIL PROTECTED]
Enteries in my extensions_table are like this:
Field-Name: id
> Seriously, I've tried to read everything I could find (& search for) on
> voip-info.org and other sites about this problem, but have been unsuccesful.
>
> Equipment:
> xten lite
> X100P
> Whitebox linux running Asterisk / AMP
> D-Link DI-804HV (VPN router)
>
> I have installed another DI-804H
I guess, but just as you dont have to really worry about howto send out
pstn, unless u plug in the E1 into your box, you can just handoff the
traffic to a sms providers, they interconnect and send sms worldwide
just as voip is done.
Iqbal
David Boyd wrote:
On Wed, 2005-01-05 at 10:23, Jay Milk
Hi All
I am working on setting up a * system to replace our current voicemail box. I
may
also end up using it for a few Voip calls. Anyway, I have heard some people
complaining about the new Digium Fxo cards and having problems with them. I do
not
yet have the computer so if certain issues ar
On January 8, 2005 11:51 am, Eric Wieling wrote:
> That is DISCONNECT supervision, not ANSWER supervision. Asterisk has
> no support for answer supervision using only audio. If you want that
> get a PRI.
Well with a PRI it's not using audio for that purpose. :-)
-A.
___
the only way is to set callprogress=yes but it's very experimental
and makes many wrong alarms.
by the way this feature is really missing in *.
On Sat, 08 Jan 2005 17:42:42 +0200, Gilad Ben-Yossef
<[EMAIL PROTECTED]> wrote:
> Samudra E. Haque wrote:
> > hello, using Asterisk, is there any clever
Gilad Ben-Yossef wrote:
Samudra E. Haque wrote:
hello, using Asterisk, is there any clever way to provide answer
supervision based upon the received audio only from the FXO interface
(from a public PSTN switch that does not have battery reversal, or CPC).
In zapata.conf use either
busydetecgt=
Hi list,
I'm new with asterisk PBX and try to do what is described below :
A B
Asterisk
|
PhoneA1 --- Internet Phone B1
|- |
PhoneA
Hi all,
Seriously, I've tried to read everything I could find (& search for) on
voip-info.org and other sites about this problem, but have been unsuccesful.
Equipment:
xten lite
X100P
Whitebox linux running Asterisk / AMP
D-Link DI-804HV (VPN router)
I have installed another DI-804HV at a second
Is it possible to set asterisk up as a virtual pbx like in apache and
virtual host? If so can someone point me to the right direction.
I would also like to setup asterisk with some type of redundancy, I have
searched the lists and googled but havent really found anything, I would
be willing to
> On January 8, 2005 10:03 am, David Cook wrote:
>> I'm Toronto (well Pickering). I think that could prove helpful.
>
> I'm in Listowel, which is about 1.5hrs WNW of Pearson (think 40 minutes
> from
> Kitchener/Waterloo).
I'm from Niagara, I know Wade Weppler on the list is from Burlington as we
That "program" will be detected by your ISP within a day or so,
determined to be a virus, and your service will get disconnected...which
n turn will not help your latency or jitter at all.
VoIP can tolerate a fair amount of latency; latency over about 100ms is
heard as a perceptible delay resultin
On January 8, 2005 10:03 am, David Cook wrote:
> I'm Toronto (well Pickering). I think that could prove helpful.
I'm in Listowel, which is about 1.5hrs WNW of Pearson (think 40 minutes from
Kitchener/Waterloo).
-A.
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Aster
Shoval Tomer wrote:
Hi all.
Can anyone comment why shouldn't we use FC 3 for an * production system?
For the same reason you should not use Fedora Core line for ANY
production system, as it designers intend it to be an experimental
branch. In particular, FC3 has the NSA's SELinux patches integrat
Samudra E. Haque wrote:
hello, using Asterisk, is there any clever way to provide answer
supervision based upon the received audio only from the FXO interface
(from a public PSTN switch that does not have battery reversal, or CPC).
In zapata.conf use either
busydetecgt=yes
busycount=6
(it will
Jay Milk wrote:
That's a known, yet not feasible work-around over accessing an
SMS-center directly. But the question remains how to accept IMCOMING
messages with *.
It's very simple - you register withj your Telco to receive SMS messages.
When an SMS message arrives, your Telco line will ring and
I noticed the following came into cvs head yesterday:
> Update of /usr/cvsroot/zaptel
> In directory mongoose.digium.com:/tmp/cvs-serv2118
>
> Modified Files:
> fxotune.c wctdm.c wctdm.h
> Log Message:
> More TDM card echo API modifications. Making the fxotune program
> automatically
> f
I'm Toronto (well Pickering). I think that could prove helpful.
--
David Cook
Quoting [EMAIL PROTECTED]:
> Anyone in the Toronto area interested in getting together to share
> notes
> and swap war stories?
> --
> Jim Van Meggelen
> [EMAIL PROTECTED]
__
Well there is nothing much you can do if you don't own all the routes. But in
concept you can, and this is purely just theoritical and a very unhealthy
thing for the Internet, is to write a program running on your router that
constantly streams traffic to your end point, this will maintain a const
While we have not been able to get our hands on this model yet (we have
the other Linksys models) the Linksys voice products are based on the
sipura technologies ATA and will work. I was under the impression that
the only shipping WRT54GP2's are provisioned and locked to ATT (for
now).
Also keep i
Hi,
I'd just like to confirm compatibility of Linksys router WRT54GP2 as local
extensions to Asterisk.
Can it register to local Asterisk behing him ? How stable/good is analog
interface ?
Any experience would be more than welcome.
Thanks in advance,
regards,
Rob.
___
Very good point.
So what can you do ( if anything ) to control the load
on the network outside of your control?
robert
--- David Liu <[EMAIL PROTECTED]> wrote:
> Assuming the network loading is fairly constant,
> 300ms latency is actually not
> noticeable unless you put both phones next to your
>
On Sat, 8 Jan 2005, chawki hammoud wrote:
> I have Asterisk running on Linux Redhat9 dstr. I
> subscribed to a third party sip providers to make LD
> calls. Can I initiate a call sessions from asterisk
> CLI> command prompt after I configure extensions.conf
> and iax.conf?
Hi Chawki,
yes you can
I once read that by PSTN standards a good connection should be less
than 150 ms. I think it was an older Network World review of voip
phones. The real issue is due the callers step on each others speaking
due to latency.
Michael
On Sat, 8 Jan 2005 21:41:37 +0800, David Liu wrote:
>Assuming the n
Assuming the network loading is fairly constant, 300ms latency is actually not
noticeable unless you put both phones next to your ears to compare.
Latency affects delay while network loading affects voice quality (e.g. break
ups) If the either end of your network is experiencing very bursty traf
Hi every one:
I appreciate the conrtibution every one is making and
please forgive me for my question.
I have Asterisk running on Linux Redhat9 dstr. I
subscribed to a third party sip providers to make LD
calls. Can I initiate a call sessions from asterisk
CLI> command prompt after I configure exte
> I would like to know what would be acceptable latency
> on a connection to the termination server( but still
> having good quality voice )
It's pretty hard to answer that except with a subjective appreciation.
I have had very good one hour talks with the figure being 250-350 for
the distant user
Hi Scott, and Jack,
--- Scott Stingel <[EMAIL PROTECTED]> wrote:
> Sid-
>
> Try connecting one port to another. Note that one of the ports must be
> set up as "cpe" and the other as "net" in zapata.conf when you loop them
> together like this.
>
> A suitable crossover cable for testing can b
If so please let me know off list and I will try to coordinate.
Dave
[EMAIL PROTECTED]
703-727-1312 Mobile
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> Is there a real registration server on len1.host.wengo.fr instead of the
> proxy on proxy1.host.wengo.fr ?
The 2 ips are different, but if I use only len1 on X-Lite it still works.
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I cannot get the sound quality very good, it is very very
broken.
Any hints?
I also use the SJphone on my laptop with the same wireless
access point, and have no problems.
Thank you in advance
Thorben
___
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On Sat, 8 Jan 2005, Wilson Pickett wrote:
> The service is using a GPL SIP client, which after a few sniffs from
> various Usenet denizons gave us what was needed to make the service
> work with X-Lite. However, attempts to port the account over to
> asterisk (which have always worked with all oth
For people in Melbourne Australia there'll be a get together for people
intrested in VoIP of all persuasions on the 22nd of January in Preston.
If you're not a LUV member e-mail me privatly and I'll forward on the
invite.
Thanks,
Julien
pgpcSjgKQTPUA.pgp
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Asterisk must have a reasonably large community here in France judging
from the number of people who came out to meet Mark. Either that or we
were ALL there :)
Something I've been waiting for, a voIP carrier on the models we are
used to (low monthly or pay as you go, web account) has just set up
t
Hi folks
Seems really simple but run out of ideas:
1. Used Redhat 9 Packages for Asterisk
2. Zaptel one didn't work so recompiled zaptel from cvs against my kernel
2.4.20-8
3. Compile is fine no depmod problems/ugly message etc.
4. lsmod:
wcusb 20064 0 (unused)
wcfxo
> So... how can i get the IAXy to work?
Someone you with a linux server can probably provision your IAXy
building the provision prog from downloadable source. They would need
to know a few things about your network and also the network you need
to connect to.
Just about anyone with a working aste
Anyone in the Toronto area interested in getting together to share notes
and swap war stories?
--
Jim Van Meggelen
[EMAIL PROTECTED]
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I think you need that provisioning tool from digium. And you need a unix
system to compile and run it. I dont think theres any port to your OS.
Sorry...
On Saturday 08 January 2005 08:08, Daiku wrote:
> Hi,
>
> hoping that experienced hands will quickly show me the right way: after a
> fruitless
/etc/group
/etc/passwd
/etc/shadow
The line looks like a yp line. It tells the pam module to search the NIS
server for users, groups and password
On Saturday 08 January 2005 05:28, Michael Levenson wrote:
> Can someone help me answer this question?
>
> Where would you most likely find a file wit
Try debugging sip with "debug sip" at the CLI. This will provide you
with more detail on what is going wrong.
Scott
[EMAIL PROTECTED] wrote:
I am having a major dillema here, I have been trying to get my sip phone (hard
phone) to communicate with the asterisk server. Below is my configuration:
<>
Assaf, I don’t know if there is such
an ID available. However if there is not, the value you want is pushed
out in one of the events that Asterisk publishes to AGI connections when a call
is constructed. As it result it ought to be possible to write an AGI
script using, say, Perl t
Thanks for your replies. I'd read somewhere that the n priority was "after
1.0" but I now understand that this means "after 1.0 and subsequent stable
patches". Thanks again and I shall download HEAD.
Bill Seddon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Beh
Is it possible for someone to post the UK CID patch for the latest CVS. The
one I have fails to patch channels/chan_zap.c?
Many thanks
Vassilis
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