Hi,
-Original Message-
extension.conf
[incoming]
exten = 955,hint,SIP/22
Thanks. I tried it, but no success. Do you know if the hint extension
does work with 1.0.2 stable?
I've had it working, but I am not sure if the subscribecontext you mention
would work, I've set it up in
Hello,
Just out of curiosity, what E1 card are you using in Asterisk?
Best Regards,
--
Telmo
On Mon Dec 20 8:19 , 'Guillermo Freige' [EMAIL PROTECTED] sent:
MFCR2 is hanging under heavy load in my configuration. It always segfaults
in the same point. This is the coredump debug.
During
Hello folks,
Humrmrm... 2 days, no answers... :-/
Either I made a stupid question (I don't think so: I have *really* tried to
solve
that on my own before asking the list) or this one's just something nobody has
ever tried but me (I also find that unlikely: even the telco here plays a
message
Recently there was talk about NANPA and getting current info from them -
I found this link and thought I should share..
http://bellsmind.net/NANPA/
Matt
--
Matt Gibson
VOIP Administrator
NJ Tech Solutions
1.314.480.4550 ex. 6400
1.877.999.4678 ex. 6400
Hint works on Snom phones with CVS 1.0.2. We have tried different SIP aware
NAT routers with hint, and not all routers pass on the NOTIFY messages used
to turn on and turn off the lamps. Thus, it does not work on the small
Intertex router, but it works OK with the Zyxel Zywall 35 with the last
Hi everybody!
this is third day I'm supposed to work on some telecomunications solution.
We have SIP Express Router to maintain and redirect incomming calls to
asterisk.
The problem is that we (i mean my company) have to run some prepaid
solution with asterisk.
I'm wondering if modified prepaid
That actually tells why it doesn't work. :-) It can't find anything in
[from-sip] that matches the number you are trying to call.
You shouldn't put nat=yes in your sip.conf - * can see the Cisco's
directly, so no need to do that.
It seems you haven't included the [voiptalk] context in the
Hello!
I have a number of IAX clients behind a NAT (on the same LAN) and
asterisk server on the Internet. And that clients doesn't speak directly
to each other, traffic goes through the asterisk server.
What should I configure to make IAX clients on the same LAN to speak
directly, please?
How about this variable? :-)
${SIP_CODEC}: Used to set the SIP codec for a call
That only works for calls going OUT from Asterisk. It does nothing
for incoming calls. By the time the dialplan is called the codec is
already set.
perhaps tha should be changed, then, to allow more control over
Hello Folks,
I'm trying to decide here between a few cards for connecting an Asterisk box to
a
single E1 channel (either PRI or R2 signaling):
- Digium E100P: has been replaced by the TE110P below, but can still be had at
places like digitnetworks.com for $475, and I guess there's always a
Hi
All,
Wehave just
installed [EMAIL PROTECTED]. It was straight
forward as promised. However, I cannot find any guides or tutorials on how to
administer this version of asterisk.
We plan to install a
bunch of Cisco 7960 and 7905 IP phones. I have a test phone which has already
been
Hello, I use TE405 and TE410 for more than one year, they work perfectly!
For me, Digium Cards are the best choice (if they feet the need) for
different reasons:
1. They support Digium that is supporting Asterisk.
2. Digium offer free support to install their card when you buy from them.
3. I
Hi,
I use a simple script to query a 2 table's in a MySQL database. This is called
using AGI.
The first one has got about 200 rows in it each one is a caller ID number and a
name. If I get a match then I set CLI to name number
The second table has over 10,000 rows in it. It has all the
Would you mind posting the script and extensions.conf entry to the list?
On Wed, 22 Dec 2004 11:53:42 -, Peter Braidwood
[EMAIL PROTECTED] wrote:
Hi,
I use a simple script to query a 2 table's in a MySQL database. This is
called using AGI.
The first one has got about 200 rows in it
;;;extensions.conf
[internal] ;;; context used by our internal SIP-phon
include = voiptalk.org ;include context below
exten = 11,1,Dial(SIP/gsbt100,20,tr);calling : dial our office phone
include = invalid_calls;all ext numbers not handled above are
invalid
Since j'ai forgotten to do it, if you see this message, send to me your mall
so qu'on remains in contact. I opposite Marek, where were you with the table
are the preque old man? With soon!
I wondered where Marek came from until I saw the typo in the original! :)
ok, thanks
Walid Azab wrote:
Hi All,
We have just installed [EMAIL PROTECTED] It was straight forward as
promised. However, I cannot find any guides or tutorials on how to
administer this version of asterisk.
We plan to install a bunch of Cisco 7960 and 7905 IP phones. I have
a test phone which has
I tried type=friend and it is registering now... I'm happy with it this
time, but why can't I have the phone as user only (only to make calls)
and not as peer (to receive calls)??
Thanks,
RODOLFO
Rodolfo Grave wrote:
Hi again. I cant get my Budgetone registered in Asterisk, and I cant
find
Inline...
Humrmrm... 2 days, no answers... :-/
Well, let me see if I can take a stab at this one.
After working with * for about a year now, I'd suggest the toughest
part of the learning curve is truly understanding how to take advantage
of the various 'context' statements to accomplish an
To help clearify for those that don't have a good understanding of the
tcp-ip protocol, consider this:
- sip and rtp use the udp protocol (not tcp)
- all sip boxes (regardless of whether the box is a phone or *) _listen_
on udp port 5060. (What udp port is used as the _source_ port when
On Tue, Dec 21, 2004 at 11:05:27PM -0500, Alex Brecher said:
I still don't get why we don't move over to a web based forum ?
Because web-based forums suck. The only people who seem to like web
forums are those that don't know how to use their email client (or use a
client so braindead that it
Why is it that newcomers always feel like inserting 'Answer' is a
necessary step in their extension.conf entries?
[voiptalk.org]
;forwards any calls starting with an 8 thru voiptalk.org
exten = _8.,1,Answer
exten = _8.,3,SetCIDNum()
exten = _8.,4,SetCIDName(My Name And Surname)
exten =
Matt, You are my new best friend...Thanks for the post. Jon
- Original Message -
From: Matt Gibson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, December 22, 2004 4:55 AM
Subject: [Asterisk-Users] Daily
can anyone using/integrating modified-prepaid-application avaiable on wiki .
if anyone kindly guided me.
Thanks.
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-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Wednesday, December 22, 2004 8:12 AM
To: Asterisk-a-users-list
Subject: [Asterisk-Users] Why use 'Answer'?
Why is it that newcomers always feel like inserting
Seems I was looking in all the wrong places.
The problem was that I was stripping the leading '1' off of the outbound
IAXTEL phone number.
exten = _91700NXX,1,Dial(${IAXNET}/${EXTEN:[EMAIL PROTECTED]) will not work
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Hi Benoit,
On Wed Dec 22 3:47 , 'B. Vallet - www.acropolistelecom.net'
[EMAIL PROTECTED] sent:
Hello, I use TE405 and TE410 for more than one year, they work perfectly!
Ah! Thanks for the info.Can you tell me whether you are using them with E1 or
T1,
and with what signaling?
For me, Digium
Can someone please tell me what has happened to this app ? It was a patch
(http://bugs.digium.com/bug_view_page.php?bug_id=0002379), which was closed
pending changes to CVS head. That was back in October.
Any news, anyone ? I need to be able to monitor my SIP channels :(
Julian
I was ZapBarging an extension the other day (sorry, CVS Head as of last
week) and forgot to put down the phone when the conversation ended. To my
consternation, I then started to listen to a totally different conversation
- I think that the zap channel got re-used.
Is there any way to force the
Hello,
On Wed Dec 22 5:16 , Walt Reed [EMAIL PROTECTED] sent:
On Tue, Dec 21, 2004 at 11:05:27PM -0500, Alex Brecher said:
I still don't get why we don't move over to a web based forum ?
Because web-based forums suck. The only people who seem to like web
forums are those that don't know how to
On Wed, 22 Dec 2004 17:12:56 +1100, Eric Bishop [EMAIL PROTECTED] wrote:
Hi All,
Is it possible to match caller ID on incoming calls against say text
file of know numbers and diaplay the name rather than the numerical
caller ID?
Eric,
Check out the following.
Hi,
I have to link an Asterisk Box with a PBX Matra 6501.
System look like this :
E1--Te110P Asterisk Te110P-E1Matra 6501-Phones
|
|
Ip Phones
Incoming call from E1 will enter on asterisk, if incoming number is
_800n then go
Mine is working fine so far. Issues, #1 make sure the mailbox is just a
numerical value without any context defined in your /etc/asterisk/sip.conf.
In my polycom sip.cfg file I have:
msg msg.bypassInstantMessage=1
mwi msg.mwi.1.subscribe=8500 msg.mwi.1.callBackMode=registration
Thanks for lan for your reply can you share your extention.conf setting .
this is my extention.conf lines
[globals]
OUTGOING = Zap/1
exten = _9,1(${OUTGOING}/${EXTEN:1})
i am trying to dial pstn through firefly using 9-55212323 ( Suppose this
my Pstn number ) i get these error :
Yeah thanks. I got that already from the webpage. The problem is that I need
a guide to help us know the correct sequence of adding phones and doing the
first call.
Walid
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Soren Rathje
Sent: Wednesday,
Running cvs of last night, I cannot anymore put a call on hold and then
pick it back up? This is using the hold button on the phone and then the
resume button.
On the phone it looks like it is picked up, but the call contiues to
hear music on hold.
Any ideas?
--
respectfully, Joseph
Hello,
We have a DID partner sending traffic to Asterisk via SIP, but we are not
hearing ringtones. When we call the same extension via SIP, we can hear
that's its ringing (virtually)..
Is is something related with call-progress not recognized by DID provider
?
Thanks,
I have been seeing some strange problems with our in-house Asterisk system.
Each of them have slightly different circumstances but I want to focus on
one in particular.
Here is how the call flowed:
1- Came in via iax.cc from our DID with them to our Asterisk system
2- The caller dialed Zero for
Hi,
I purchashed a Telular Phonecell Fixed Cellular
Terminal. I hooked it up to my wildcard fxo card. I
can receive calls and these calls are passed on to the
Asterisk Calling Card application. My problem is that
i can't get DTMF to work properly. If a pin number is
484443543639 i get
Its a way of storing ur sip stuff in a database rather than using the
flat files. Sip friends - extensions.conf stuff. Sip_buddies -
sip.conf stuff
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Hi All,
I am wondering if I will be breaking the GPL,
if I write for example, a channel driver or
make some modifications to the astrisk source code,
to interface at RUN TIME, through sockets, with
a proprietary system.
Eg.
1. I write chan_xxx + modify asterisk source
(make changes + new code
Are there any IAX speaking hardphones out there?
If so, can anyone offer comment on their quality?
Thanks!
-Dorn
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Hi all,
What is the best commercial soft phone out there that can be used with
Asterisk?
Is acoustic echo cancellation at the softphone level a real good feature
to have? I do have a bit on an echo problem, with X-lite and Iaxclient.
Regards,
Francois
Random Thought:
---
There
Alternate Certification
For those of you who can't (or won't) shell-out the $3000+ for the 5 day
certification class,
here's a quicker way AND IT'S HALF THE MONEY!
www.metrotel.net/asterisk.htm
Asterisk is a good product.
Some people need certification.
A mature product needs certified
How much time did you waste on that?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
Taylor
Sent: Sunday, August 22, 2004 10:24 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: [Asterisk-Users] Another
On Wed, 2004-12-22 at 08:41, John Hill wrote:
Question:
Do you need to answer to detect a fax?
Yes. You need to answer the line so the calling fax will start sending
the fax tones and * can detect them.
-Seth
--
Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
I'm sure it took several hours, but, hey, he only has to sell one to get
his money back (:
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Wed, 22 Dec 2004, Luke Catranis wrote:
How much time did you waste on that?
-Original Message-
From: [EMAIL
The place it has for me is that my work is hesitant to pay for additional
training if there's nothing to show for it, like a certification.
They can't tell if I go to a $2000 training course and just goof off,
there's no tangible goods that they can see. If I got them to shell out
$3000 for my
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Seth Remington
Sent: Wednesday, December 22, 2004 10:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Why use 'Answer'?
On Wed,
El 22/12/2004, a las 1:51, Eric Wieling aka ManxPower escribió:
No. Hint is not supported in 1.0.x. Only in CVS-HEAD developement
version of Asterisk.
--Eric
running fine for my in 1.0.3 release and snom 190
adrià
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adria vidal wrote:
El 22/12/2004, a las 1:51, Eric Wieling aka ManxPower escribió:
No. Hint is not supported in 1.0.x. Only in CVS-HEAD developement
version of Asterisk.
--Eric
running fine for my in 1.0.3 release and snom 190
adrià
I sit corrected.
Hi all,
I am wondering if PassThru mode can work in NAT environment.
Many Thanks.
Vincent
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No disrespect here but... YOU HAVE GOT TO BE SHITTING ME!!!
This is a joke right? I has to be. :P
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of James Taylor
Sent: Sunday, August 22, 2004 9:24 AM
To: [EMAIL PROTECTED];
Eric,
A few days ago when you got pissed at me because I was not subscribed to
the mailing lists, you had asked me what PCI revision my motherboard had
in it. I have PCI revision 2.3 (it's an Intel 6300ESB ICH controller). I
have opened up a trouble ticket with Digium, and was told use another
At 07:30 AM 12/22/04, you wrote:
I tried type=friend and it is registering now... I'm happy with it this
time, but why can't I have the phone as user only (only to make calls) and
not as peer (to receive calls)??
Thanks,
RODOLFO
Rodolfo Grave wrote:
Hi again. I cant get my Budgetone registered
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of James Taylor
Sent: Sunday, August 22, 2004 9:24 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: [Asterisk-Users] Another Asterisk
The scripts and details are here http://www.networks.org.uk if anyone is
interested
Regards
Peter
-Original Message-
From: Eric Bishop [mailto:[EMAIL PROTECTED]
Sent: 22 December 2004 12:25
To: Peter Braidwood; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re:
Here, here,
I agree. Each channel should be able to decide which codec to try first.
This will cause more call setup messages.
1. Hi, do you have GSM?
No
2. Hi again, do you have G.723
No, go Fish.
3. Hmmm, do you have (cringe) G.729?
Yes, do you have License for your
This is working in my phone:
msg msg.bypassInstantMessage=1
mwi msg.mwi.1.subscribe=299 msg.mwi.1.callBackMode=contact
msg.mwi.1.callBack=8 /
/msg
299 is both * extension and mailbox. 8 is special extension to dial from
local phone to check voicemail.
You may want to check
Once you've been ordained, do you have to wear black robes and a white
collar while working on Asterisk? :-)
Steve
Brian West wrote:
No disrespect here but... YOU HAVE GOT TO BE SHITTING ME!!!
This is a joke right? I has to be. :P
bkw
-Original Message-
From: [EMAIL PROTECTED]
But wait, that's not all! I, too, have a laser printer!
If you send me $50, I'll fire you off a certificate too!
You can be a Certified Asterisk Certification Certificate Buyer!
Enough. It is what it is. Don't like it? Don't pay for it.
Think it's a joke? Sure, but it's the same sick joke
Are there any IAX speaking hardphones out there?
There will be soon
If so, can anyone offer comment on their quality?
Not yet but soon.
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At 09:44 AM 12/22/04, you wrote:
Its a way of storing ur sip stuff in a database rather than using the
flat files. Sip friends - extensions.conf stuff. Sip_buddies -
sip.conf stuff
___
this is the database to flat file storage I take it.
Regards
Greg
thing call dies after 3 minutes or so.
Any AbsoluteTimeout(180) lines in extensions.conf ?
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I am not sure if I am doing the right things to figure out Caller-ID in
Taiwan.
Usually Taiwan follows everything what USA says (was not meant as political)
zaptel.conf:
loadzone=tw
defaultzone=tw
fxoks=1-2
fxsks=3-4
zapata.conf:
usecallerid=yes
progzone=tw
cidsignalling=dtmf
Try
exten= _X.,1, Dial(Zap/g2) If your 2nd TE110 has defined a group.
Regards,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Jeremy SALMON
Enviado el: miércoles, 22 de diciembre de 2004 15:01
Para: asterisk-users@lists.digium.com
Running cvs of last night, I cannot anymore put a call on hold and then
pick it back up? This is using the hold button on the phone and then the
resume button.
Any ideas?
yep http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0003113
waiting for the fix from oej
It's happened before, cleared up and now it happened again.
The IAXy, working for a total of about 6 months.
Symptoms:
Registered with asterisk and even receives calls (the LED shows it's
ringing) but phones connected to it are dead.
Same phones work connected directly to the phone line.
This just in:
Centrality has released Version 1.4 for PA168x based phones.
Firmwarefiles for different protocols like SIP, IAX etc. can be
downloaded from Centrality Website. Firmware for different brands is
already available. I've tried out with my HOP-1002 which is actually a
WuChuan with
At 12:43 AM 12/22/04, you wrote:
Seeding occurs if there is still a persistent record (in astdb) of a preceding
location registration of a peer after a restart of asterisk or the sip
channel.
If Asterisk goes down and the peer has a long registration refresh time,
the phone maybe inaccessible
It would be a real kick to get one of these to run Asterisk. :)
http://www.gumstix.com/
Tim
--
Tim SailerCoastal Internet, Inc.
Network and Systems OperationsPO Box 726
http://www.buoy.com Moriches, NY 11955
Hello Guys,
I think this is not bad (Certification) While is a real certification
like Cisco , Novell, etc.
how many of us have a cisco certification or even Micro-$hit.
In my point of vew this 3K buck are well spended if I want to have the
skills quick to put hand-on. and as per Brian comment ,
jafar mohammed wrote:
Hi,
I purchashed a Telular Phonecell Fixed Cellular
Terminal. I hooked it up to my wildcard fxo card. I
can receive calls and these calls are passed on to the
Asterisk Calling Card application. My problem is that
i can't get DTMF to work properly. If a pin number is
On Tue, 21 Dec 2004 23:21:50 -0800, Chris Travers wrote:
Fractional T1's don't have the 4 to 2 wire conversions at your phone
switch of you are doing VOIP or other digital telephony technologies.
As a result, you should not have to deal with echo cancellation as much
if you are on a fully
Tracy R Reed wrote:
I was running a recent CVS version of * but have gone back to
CVS-v1-0-12/21/04-16:31:58 just to make sure it isn't some sort of problem
with latest cvs.
There are SIP transfer problems in CVS HEAD at the moment, although a
fix is on the way (monitor bug 3113 in Mantis).
Hi All,
Im sure this is something simple that I have
missed somewhere. When I make a call using BT100 over IAX2 with Voipjet
terminating I dont get a ringing sound whilst Im waiting to be
connected. The destination party can answer the call (they do get
ringing) and conversation can
Sounds like a thermal problem -- which most intermittent problems are.
Had this happen with a network switch in my home office. Pull it out,
disconnect it and put it in a cool spot for a few hours. If the problem
goes away, see whether you can stabilize the environmentals.
-Original
I'm not familiar with Mepis so I'm not sure whow best to tell you to
proceed but you can certainly download the kernel source from
www.kernel.org. The kernel will need to be configured in order for the
module build to work however and I'm not sure how best to help you with
that.
Adam
Imran
What started out as a good thing for the community has veared it ugly
head and will come back to bite us in the ass. I give my respect to the
two companies that decided to put themselves 'out there' and attempted
to bring 'real world' certifications of knowledge in an area that is
unregulated,
I have an xtraphone number connected to my FWD account which is in turn connected to asterisk. When someone dials that number they are successfully connected to asterisk and hear the message asking them to enter an extension number, when they enter the number it is ignored and just times out.
Hi,
We currently have an Asterisk box (P4 2.4 Ghz, 512Mb RAM, 1 T100P,
Zhone Channel Bank and 1 E100P) connected to an ISDN running without
any problems. That machine is working for about 1 year.
Two days ago, we decided to switch that machine for two PowerEdge 600SC
(HA) and we got some
Did anyone here use the * forums over at asterisk.xvoip.com? I've been
unable to connect for a few days now and was wondering if anyone knew if
they're down for good.
It'd be a shame if they are since * newbs like me need every resource we can
find.
Joel Moore
We get this all the time. We use Cisco
7960s connected to a local asterisk server via SIP which pushes it via
IAX2 to another Asterisk server which then pushes the calls out a SIP PSTN
gateway.
Anyways, sometimes when we call people, we
dont hear ringing. Its not always, its 1 out of 10
Mepis is debian based, so you should be able to apt-get
install your sources that you need, specifically the kernel sources.
Google apt-get kernel source and see what you
get...
sean
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
FinebergSent:
certify this
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If everything on the Aterisk side of the socket is GPL'd
I think you are OK.
Look at the example of a Linux system running Netscape
browing the web when there is a Microsoft HTTP server.
Then we have a GPL's system conected to a closed
comercial system over a socket (port 80).
The grey area is
At 12:21 PM 12/22/04, you wrote:
Did anyone here use the * forums over at asterisk.xvoip.com? I've been
unable to connect for a few days now and was wondering if anyone knew if
they're down for good.
It'd be a shame if they are since * newbs like me need every resource we can
find.
Joel Moore
On Wed, Dec 22, 2004 at 09:58:53AM -0700, Kevin P. Fleming spake thusly:
There are SIP transfer problems in CVS HEAD at the moment, although a
fix is on the way (monitor bug 3113 in Mantis).
Thanks. Last night I downgraded to CVS-v1-0-12/21/04-16:31:58 just in case
this was the issue but I
Seeding is important, more important then most people think.
If you are running a gateway and people can't call a phone behind the
gateway because Asterisk has done a reboot, then the call from the pay
phone cannot get in because the phone address is not know.
While the 30-45 second or 5 minute
The problem is * not supporting or handling early media. I have looked through the sniffer traces and I see the RTP stream being setup between
* and the gateway during the invite and or 183 message, but * does not setup a corresponding
stream to the client until it sees an OK (200)
On Wed, Dec 22, 2004 at 09:58:53AM -0700, Kevin P. Fleming spake thusly:
There are SIP transfer problems in CVS HEAD at the moment, although a
fix is on the way (monitor bug 3113 in Mantis).
Not only in HEAD but also in STABLE, oej just informed me. I thought I was
losing my mind when I went
hi
I keep getting these sometimes
RFC3389: 1 bytes, level 8...
does this mean silence suppression is turned on on the client?
is there a way to print the SIP user id in this error message?
roy
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Steve Murphy wrote:
--On Tuesday, December 21, 2004 9:37 AM -0700 Steve Murphy
[EMAIL PROTECTED] wrote:
Howdy--
I'm playing with different IAX softphones. I've got DIAX and
IAXPHONE on
a windows (XP) machine on my network, and I'm
Nope, I searched the extensions.conf, sip.conf and iax.conf for 180 and
found nothing.
--
Start Your Own ISP!
http://www.YourOwnISP.com
- Original Message -
From: Wilson Pickett [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
What started out as a good thing for the community has veared it ugly
head and will come back to bite us in the ass. I give my respect to the
two companies that decided to put themselves 'out there' and attempted
to bring 'real world' certifications of knowledge in an area that is
Man, this is sick! :-)))
Isn't there a law against unclearly-marked jokes (except on April 1st, of
course)? Some people could even take you seriously! :-))
Most relevant points in the web page:
Starting a telephone company or consulting business is easy.
We authorize you to perform all
I'm looking at finding a way for my Macro(dundi-dundi-test,${ENTEN})
when I dial out on the dundi-test network to return a +101 to my
[dundi-test-out] context, if the number being dialed on the dundi-test
network does not exist, then I will route the call out using my pstn
or voip connection i
I, being one of the original Microsoft Certified guys, back then you
sent them $150 and you got the certificate and some logos (think 1980's
certification.)
In 1996 I was told by the company I was working for the certification
was needed if I was to keep my current salary.
What I saw was morons
[EMAIL PROTECTED] wrote:
Man, this is sick! :-)))
Isn't there a law against unclearly-marked jokes (except on April 1st, of
course)? Some people could even take you seriously! :-))
I'm rolling on the floor here :-
Regards,
Telmo.
Then I guess you haven't seen this one:
I am new to the list, and Asterisk itself, but I am currently planning to test
a uCLibc build of Asterisk this weekend on a Gumstix. I have two of them
sitting in my bag next to me here in my office.
I am not currently sure how I will go about doing this. The setup of Asterisk
on it
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