On 10/12/05 13:00 trixter http://www.0xdecafbad.com said the following:
Where I got the data from and all is also on that page if anyone wanted
to make their own lists. I would appreciate any updates or corrections
that anyone happens to notice.
a simple modification which would make this
On Tue, Oct 11, 2005 at 05:37:12PM -0600, Ryan Hulsker wrote:
mine looks like this
#!/usr/bin/perl
# Takes 3 command line args, context, mailbox, password
# updates the mailbox password in mysql
use strict;
use DBI;
my ($Context, $MailBox, $Password) = @ARGV;
my $dbh =
On Tue, Oct 11, 2005 at 10:09:34PM -0400, Andy Goss wrote:
Update -
I made a backup of my entire voicemail directory then deleted it. If I
then try and record a greeting, it works. Asterisk creates the folder
structure and records the greeting. If I try to copy the old file back
into
On Wed, Oct 12, 2005 at 01:44:34AM -0400, Andy Goss wrote:
If anyone could tell me what this error is all about, I would be very
grateful.
Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path: Failed to lock
path '/var/spool/asterisk/voicemail/default/5933/INBOX': Operation not
Dinesh Nair wrote:
On 10/12/05 13:00 trixter http://www.0xdecafbad.com said the following:
Where I got the data from and all is also on that page if anyone wanted
to make their own lists. I would appreciate any updates or corrections
that anyone happens to notice.
a simple
If you haven't seen it already, this will be a lot of help to you.
http://www.voip-info.org/tiki-index.php?page=AreskiCC+CallingCard+Application+The+idiots+guideV2
You should now be on step 12. :)
G
Omar McKenzie wrote:
Hi
I have gone thru the steps of installing AreskiCC, I
I don't think that its the D() dialing before the call is bridged, I
just tested it on Asterisk 1.0.7 and CVS HEAD
Both times I did:
Dial(SIP/[EMAIL PROTECTED],20,D(ww1234))
both times i picked up the phone, it waited about 1 second, dialed 1,
then stopped alltogether.
This might be an
Hi folks,
I've already searched the mailing list but no one else
seems to have my same problem.
I'm using Asterisk with the following configuration:
Fedora Core 4 (but I also tried Fedora 3)
1 Digium TE110P
1 TDM40B
1 HFC-S 'Cologne'
bristuff 0.2.0-RC8o (zaptel 1.0.9.2)
I
Francesco Angi ha scritto:
Hi folks,
I've already searched the mailing list but no one else seems to have
my same problem.
I'm using Asterisk with the following configuration:
Fedora Core 4 (but I also tried Fedora 3)
1 Digium TE110P
1 TDM40B
1 HFC-S 'Cologne'
bristuff
Hello asterisk-users,
Sean
--
+---+
|VOIP= FreeWorldDial: 689482 VOIPBUSTER: thecivvie |
|GPG Key http://thecivvie.fastmail.fm/thecivvie.asc |
+---+
Strange things happen under the midnight sun
Name of the company is MULTI-line GmbH
You can contact mr. Zlatko Medibach at +43 1 78932320 or on cellular +43 676
3220262.
Mail: [EMAIL PROTECTED]
Their HQ is in Wien..
I can not help you with the details, I just know that they implemented SS7
on * for some telcos there.
Goran
-Original
Dear Asterisk Users,
I'm a Japanese and now configuring Voicemail.
Now I need to modify the way cmd VoicemailMain works to fix language
difference and other my conveniences.
What I want to do are...
1) Add words used in message retrieving guidance.
I need to add different suffixes to numeric
On Wed, Oct 12, 2005 at 09:13:44AM +0200, Erik wrote:
Dinesh Nair wrote:
On 10/12/05 13:00 trixter http://www.0xdecafbad.com said the following:
Where I got the data from and all is also on that page if anyone wanted
to make their own lists. I would appreciate any updates or
Hi,
Can anyone recommend a cheap but reliable company to teminate my asterisk sip calls in Israel (mobile/cell)?
If its against the rules to discuss this on the list, please email it me directly.
Thanks
Dan
___
--Bandwidth and Colocation sponsored by
Dear Asterisk users,
Other than VoicemailMain, which Im asking in the other mail, I have another
thing to fix.
That is low recording volume of Voicemail. Compared with sound files, volume
in other phone devices that pick up the same kind of phonecall, obviously
the sound level of sound file
Hi there
I am using IAX2 softphones dialing into meetme conferences.
I also have jitterbuffer=yes, with typical jitterbuffer settings. The problem I
am having is that as soon as there is a delay from a participant, then the delay
continues until the participant hangs up and dials in
Hello,
I need to detect availability of SIP phone before dialing. I need to
know if phone is
BUSY, CHANUNAVAIL before dialing. If phone is free, then I will dial
it.
I need for automatic callback (.call files), but I need to know if it
is available both
SIP phones before calling.
Title: Message
Hi,
We
have phones registered at another soft switch, and would like to use Asterisk as
a Voicemail system.
Is it
possible and how to configure Asterisk to send NOTIFY messages (for MWI) to the
endpoints that are not registered to the Asterisk?
Regards,
Stojan
Sljivic
Is there a way to
1) disable asterisk from writing in the full log ? (
/var/log/asterisk/full )
or
2) implement a log rotation or similar of the full log ?
I see the full log grows a lot (about 100 MB per Month)
thanks in advance,
Andrea
Chi ricevesse questa mail per errore e'
Hi,
I found E400P quad PRI card quite cheap (749USD):
http://www.govarion.com/product_info.php?cPath=1products_id=2osCsid=68cdd6e3d08754
in comparison to te410p (approx 1500 USD )
http://www.digium.com/index.php?menu=product_detailcategory=hardwareproduct=TE410P
Now newer generation with HW
Hi, everybody.
I don´t know if it is an * or an AAH issue - I can´t get the
Snom-Phone-hints working under AAH 1.5 running * 1.0.9. I tried with the
Snom 360 softphone and it just doesn´t work.
Is there any known issue?
Is there a AAH mailing list available?
Thank you in advance.
Best
Hi,
[EMAIL PROTECTED] wrote:
Is there a way to
1) disable asterisk from writing in the full log ? (
/var/log/asterisk/full )
Take a look at /etc/asterisk/logger.conf
or
2) implement a log rotation or similar of the full log ?
I see the full log grows a lot (about 100 MB per
[EMAIL PROTECTED] wrote:
Is there a way to
1) disable asterisk from writing in the full log ? (
/var/log/asterisk/full )
Have a look at /etc/asterisk/logger.conf
2) implement a log rotation or similar of the full log ?
I see the full log grows a lot (about 100 MB per Month)
Have a
On Wed, Oct 12, 2005 at 10:18:15AM +0200, Simone Cittadini wrote:
Same problem with debian sarge on a dell and asterisk 1.0.7 from
packages, unloading the module freezes the system, (rebooting the
machine worked right), I installed zaptel 1.2beta and it seems to work,
but I haven't really
In article [EMAIL PROTECTED],
Steven Langley [EMAIL PROTECTED] wrote:
I am using IAX2 softphones dialing into meetme conferences. I also have
jitterbuffer=yes, with typical jitterbuffer settings. The problem I am
having is that as soon as there is a delay from a participant, then the
delay
Sorry, I could not find it there. I found only version for *-1.1.0.
Could You send right URL to me.
Thanks,
Bob.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Roman
Sent: Friday, October 07, 2005 4:17 PM
To: Asterisk Users Mailing List -
We're using Asterisk
with IAX soft phones to provide communication back to our central office when
our people travel.
We have configured our
firewall to allow UDP 4569 forwarded to Asterisk, and have tested this, works no
problem.
My question is, will
this support more than 1
On Wed, 2005-10-12 at 00:59 -0400, Cory Andrews wrote:
[snip]
* SuperMicro 1U rackmount server chassis
* Intel *P4 3.2GHz* Processor
* *1GB PC3200* SDRAM (Single DIMM)
* (2) Maxtor *80GB* SATA Hot Swap HDD's (Support 4 Hot Swap Drives)
* Dual Onboard *Gigabit Ethernet*
Thank you very much
I decided not lo lower the log information (leaving all : full =
notice,warning,error,debug,verbose)
I started a weekly-rotation of the full log.
Andrea
gincantalupo
Hi I have encountered a problem with my asterisk. Here is my set-up , I
am using E1_PRI as signalling over a Nortel PABX. What i intended on
doing is sending a call rejected signal . I have it set-up as
PRI_CALLED=21 , it sends the signal but then it hangs up the channel , i
need help sending
Hi List
Im getting this notification from my one and
only SNOM 360 every time a number button is pushed.
I know that its only a notification, but it really
irritates me. Is it anything I can/should do anything about ??
Oct 12 10:34:33 NOTICE[3566]: chan_sip.c:10530
handle_request:
Another trivial question:
Is there a place where all the parameters are documented ?
In example (my example!) I would like to know the meaning of a lot of
parameter that can be used in sip.conf,
A lot of these keywords are intuitive keywords (i.e. NAT=YES/NO;PORT=5060;
context=x) but other
Has anybody tried something like this:
http://www.arca-technologies.com/datasheets/arcaplexhorizon.pdf
It will be interesting to have ability to make systems like:
SCENARIO 1 (2 incoming BRI lines and 12 analog extensions with ability
to connect additional isdn devices to s0
On Wed, 2005-10-12 at 00:59 -0400, Cory Andrews wrote:
[snip]
* SuperMicro 1U rackmount server chassis
* Intel *P4 3.2GHz* Processor
* *1GB PC3200* SDRAM (Single DIMM)
* (2) Maxtor *80GB* SATA Hot Swap HDD's (Support 4 Hot Swap Drives)
* Dual Onboard *Gigabit
drwxr-xr-x6 root root 1024 Oct 12 01:10 .
drwxr-xr-x6 root root 1024 Oct 12 01:15 ..
-rwxr-xr-x1 root root12801 Oct 11 21:28 busy.wav
drwxr-xr-x2 root root 1024 Oct 11 21:28 cust3
-rwxr-xr-x1 root root 3051 Oct 11
I have check document , still not very clear on default html or php
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Garth Summey
Sent: Wednesday, October 12, 2005 2:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
On Tuesday 11 Oct 2005 23:57, Lee Howard wrote:
Bob Goddard wrote:
On Tuesday 11 Oct 2005 22:41, Lee Howard wrote:
Tom Rymes wrote:
Use the right tool for the job!!!
Use a hardware based DSP for faxing not software based.
Why is a soft-DSP to be considered any less-capable than hardware
Hi,
I found E400P quad PRI card quite cheap (749USD):
http://www.govarion.com/product_info.php?cPath=1products_id=2osCsid=68cdd6e3d08754
in comparison to te410p (approx 1500 USD )
http://www.digium.com/index.php?menu=product_detailcategory=hardwareproduct=TE410P
Now newer generation
www.voip-info.org
Another trivial question:
Is there a place where all the parameters are documented ?
In example (my example!) I would like to know the meaning of a lot of
parameter that can be used in sip.conf,
A lot of these keywords are intuitive keywords (i.e. NAT=YES/NO;PORT=5060;
-bash-2.05b# ls -la /var/spool/asterisk/voicemail/default/5933/
total 288
drwxr-xr-x6 root root32768 Oct 12 01:18 .
drwxr-xr-x 19 root root32768 Oct 12 01:17 ..
-rwxr-xr-x1 root root12936 Oct 12 01:14 busy.gsm
drwxr-xr-x2 root root32768
1) Add words used in message retrieving guidance.
I need to add different suffixes to numeric words due to Japanese way of
mentioning time. (e.g. in English, you can say Five forty-five for 5:45,
but in Japanese, we have to put hour and minute for respective time
unit (meaning, VoicemailMain
Yeah I should have picked up on that, single PCI Riser in this one, so 1
card. I don't know of any 1U solution out there that would give you 3
PCI slots to work with, I think you'll have to go to a 2U at least to
achieve this.
Cory Andrews
Senior Partner
+++
VOIPSupply.com
454 Sonwil
Hello all. I am new to Asterisk as well as this group so please excuse me for
a bit as I learn the
ropes of Asterisk. Anyway, I currently am using a pap2-na adapter with Teliax
and Mesa Networks (my isp) and
was wondering what I will need to get Asterisk running correctly. I am
wondering
For your information.. if someone get in
the same trouble.. problem is solved, but not with the software
We just changed our BRI NT device with a
different one.. from now on it works very well
We had Elcon NT1+2a/b and now it is
replaced with Santis ISDN NT1+2ab
Here is pri
Same problem with debian sarge on a dell and asterisk 1.0.7 from
packages, unloading the module freezes the system, (rebooting the
machine worked right), I installed zaptel 1.2beta and it seems to
work,
but I haven't really tested it, only loaded/unloaded/loaded and
placed a
couple of
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Bob Goddard
Sent: Tuesday, October 11, 2005 6:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Which asterisk-friendly cards
are fax-capable?
On
Dear All,
I am a newbie about asterisk. I have 1x X100P card 3x Sip phone
I got aware of problem, after I saw the caller id on my sip phone. I noticed that if I receive a call from GSM Operator A, I can see caller id. But any other operator, I gotno caller id, even my direct PSTN service
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
trixter http://www.0xdecafbad.com
Sent: Tuesday, October 11, 2005 5:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Which asterisk-friendly cards
are
Really strange answer. I am non used to search on playboy.com.
Anyway, if you try to search
insecure=very
on www.voip-info.org, you find 742 links , a bit more for me. (I just want
to know what it means)
Moreovere, the first 20 links are non accessible at all
We are trying to debug a connection between Asterisk and a legacy PBX (Mitel
SX200). We turned on the Zaptel debugging and we get the following message
quite frequently:
Oct 12 07:14:09 localhost kernel: T1: Lost our place, resyncing ( 28 )
Oct 12 07:14:09 localhost last message repeated 3 times
I really hope this project will be implemented, without documentation
evrything is too hard
Not for the thousands of people that have figured it out.
3Com NBX might be more your speed and plenty of documentation.
Really strange answer. I am non used to search on playboy.com.
Anyway, if
On 12/10/05, Stojan Sljivic - Pamet [EMAIL PROTECTED] wrote:
Hi,
We have phones registered at another soft switch, and would like to use
Asterisk as a Voicemail system.
Is it possible and how to configure Asterisk to send NOTIFY messages (for
MWI) to the endpoints that are not registered to
I have not seen the output of modprob zaptel in this thread, which has
to take place before loading the other kernel drivers.
Lyle
so
mesh s wrote:
Hi,
I changed the mother board (MB) but it is giving still
the same problem.
On Wed, 2005-10-12 at 08:42 -0400, Steve Totaro wrote:
On Wed, 2005-10-12 at 00:59 -0400, Cory Andrews wrote:
[snip]
* SuperMicro 1U rackmount server chassis
* Intel *P4 3.2GHz* Processor
* *1GB PC3200* SDRAM (Single DIMM)
* (2) Maxtor *80GB* SATA Hot Swap HDD's
I come from a NBX100
No documentation available.
1 day it starts saying: syslog full and voicemail stop working
No one was able to tell me what was the meaning of that alert
.
3COM NBX anyway is a good product, but the price is too high, especially 4
years ago, and especially the price of the
[EMAIL PROTECTED] wrote:
Anyway, if you try to search
insecure=very
on www.voip-info.org, you find 742 links , a bit more for me. (I just want
to know what it means)
I think the search is broken there. Just go in under Asterisk and
look for where the configuration files are documented.
Doug
--
Thank you very much for your answer.
I searched the wiki using your criteria, and I found
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf
which seems to be the answer to my question
thank you again
Andrea
Hey that works great, only problem is how do i configure an outbound
i.e. from asterisk to firefly extension?
Is this possible?
Cheers
Bails
Michael Graves wrote:
I do this all the time. I simply use the Firefly IAX2 soft phone and
don't bother with SIP at all. I forward port 4569 to my *
There is plenty of documentation online for both the 3com and *. You have
to have good search skills I guess.
3com has the best knowledge base I have seen.
http://knowledgebase.3com.com/ and there are tons of 3com dealers that can
help.
I think you may need to learn some basic networking before
Does anybody have any experience using a Patton SmartNode as a SIP/Telco
gateway with Asterisk? They seem really inexpensive and appear to
support all of the necessary features, but I don't have any experience
with their products, so I don't know if they are any good. We are
currently using
Tom Rymes wrote:
(I would like to be able to receive faxes reliably
over our PRI)
Until then, however, I still recommend HylaFAX.
If your PRI comes in to a TE405P or somesuch then you can pass fax DIDs
out through another port on the TE405P and out to a T1 faxmodem (such as
a Patton
Hello!
I'm having an echo problem with a TDM
card. The TDM card is being fed by a channel bank just 12 or so feet
away. When you put an analog handset on the line, both the RX and
TX volume seem to be just fine. However, when I use the TDM card,
I have to have an rxgain of 13.5, and even then,
Saw an ad in the latest Linux Journal for Sangoma's FXO/FXS analog cards.
it was part of an ad for a reseller. I can't find anything on the
resellers site or Sangoma's site either. Did the ad jump the gun or
someting? Is this for real?
Paul
--
Paul Dugas, Computer Engineer Dugas
I need to find a way to have the Polycom phones automatically park
calls. Right now my users hit #70# (I know the last # is optional but
it speeds it up.) to park a call. Personally I think this is easy, but
my users would like one button to do this for them. The Polycom has
buttons we
Matthew, when I tried this, I couldn't get the soundpoints to dial
in-call. They thought there were picking up a new line for a new call.
I created a speed-dial entry (in MACADDRESS-directory.xml,
itemfnPark/fnct#70#/ctsd3/sd/item) and then in ipmid.cfg:
keys
They will be announced formally soon.
-Original Message-
From: Paul Dugas [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 12, 2005 10:41 AM
To: Asterisk Mailing List
Subject: [Asterisk-Users] Sangoma FXO/FXS cards?
Saw an ad in the latest Linux Journal for Sangoma's FXO/FXS analog
Hi everyone,
I have a PRI conection on an * system running Asterisk 1.0.9, libpri 1.0.9 and
zaptel 1.0.9.2 connected to an AXE 10 (APZ 21220 System 64) in the network
side. I know the system and the wildcard I´m using are ok because I´ve used
them before with other PRI connections (to a
Dear folk,
You are right, seems sangoma is going to produce FXO/FXS cards but its still in the lab and not released yet but will do it in near future.
Regards,M. Shokuie Nia,CEO,SENA Co.
From:"Nathan C. Smith" [EMAIL PROTECTED]Reply-To:Asterisk Users Mailing List - Non-Commercial
Hi,
I've been trying to use the set_callerid function in the AGI. It sets
the CallerIDname properly but I can't figure out how to set the
CallerIDnumber.
Is it at at possible ?
Cheers.
SL
___
--Bandwidth and Colocation sponsored by Easynews.com
I have very loud sound through IAX2 and SIP channels, even very
saturated in some moments.
Why? How to change sound level (on IAX2 and SIP channels)?
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Curse,
Look at this php script ...
Contactlookup.agi
#!/usr/local/bin/php -q
?php
ob_implicit_flush(true);
set_time_limit(6);
$in = fopen(php://stdin,r);
$stdlog = fopen(/var/log/asterisk/my_agi.log, w);
// toggle debugging output (more verbose)
$debug = true;
// Do function
thanx, i did it.
i just instaled the debian packages for asterisk and
asterisk-app-fax and its working this way
fax--fxs--ipnetwork--fxs--fax
obviously the fxs ports are digium cards in linux
machines running asterisk using sip. the fax quality
is perfect
now i am trying with the following
Hello :)
For example, once I have the rxgain calibrated for all of the lines,
could I then call into, say, Zap/3 from Zap/4 and run Milliwatt() on
Zap/3 and use ztmonitor on Zap/4 to calibrate it? I'm sure it's not
perfect, but would it be close enough?
That's exactly what you do. Once I
On Wed, Oct 12, 2005 at 12:05:32PM -0400, [EMAIL PROTECTED] wrote:
I am in the middle of trying to get a milliwatt test line to calibrate the
rxgain properly. However, this won't help me with the txgain, will it?
How can I properly calibrate the txgain? By ear? Or is there a more
Hi there,
Does anyone know how to
setup an overflow queue? When a call rings on the queue A, if all agents were
busy, the call goes to the queue B.
If all agents in queue B were
busy, then the call stays on both queues until somebody answers it.
I think this is a basic
ACD
Take a look at sipsak. http://sipsak.org/
Or at the wiki on how to use it.
http://www.voip-info.org/wiki-Asterisk+at+large
I think you will still need to be able to look up the IP address that
corresponds to your sip client though.
Ryan Hulsker
On Wed, 2005-10-12 at 08:21, Peter Bowyer wrote:
Can you send me those scripts to calperinatsenecacom.net.?
Thanks in advance.
Carlos Alperin
Senior System Engineer
Seneca Communications, LLC
[EMAIL PROTECTED]
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Technical Support
Sent: Friday, October 07,
Ok, that would be helpful for me with some other problems, however I
don't see keys anywhere in my sip.conf or my phoneXX.conf files. I'm
using the 1.5.2 Sip firmware the the conf files that came with that, so
I don't have an ipmid.cfg file. Is this something I can just add to my
sip.conf?
Please send them to my email [EMAIL PROTECTED]
Thanks,
ThameemOn 10/12/05, Carlos Alperin [EMAIL PROTECTED] wrote:
Can you send me those scripts to calperinat
senecacom.net.?
Thanks in advance.
Carlos Alperin
Senior System Engineer
Seneca Communications, LLC
[EMAIL
Do you already have an ipmid/ipmid block in your sip.cfg? add the
keys ... / in there:
Try putting:
ipmid
...
...
keys key.IP_500.31.function.prim=SpeedDial
key.IP_500.31.subPoint.prim=3/
/ipmid
Moj
Matthew T. O'Connor wrote:
Ok, that would be helpful for me with some other problems,
This is a pretty popular question. IIRC SIP phones can't tell you their
statuses, you need to send a call to them and determine whether or not
they're Busy Now...
[EMAIL PROTECTED] wrote:
Hello,
I need to detect availability of SIP phone before dialing. I need to
know if phone is
BUSY,
Greetings fellow list members,
It seems like a lot of
peoplehave been having trouble getting indicators workingon the Snom
phones, myself included. Recently I was able to get the "desktop"
functionality of sipsak to work on my Snom320, and I thought I would share what
I could with the
Hi,
Check out http://store.pbxhardware.com = it has better prices on the
E400P / T400P cards. There are also 2 port versions of these.
The difference between the TE4XX cards is there is no echo canceller
and the PCI chipset doesn't handle the master mode - that eats a
little bit of CPU time.
Hey folks,
Anyone know of companies selling bulk SIP adaptors (phones, adaptors,
etc.) or has the list ever considered doing something like a bulk buy?
I was just curious...I'm looking to get another 5-6 Grandstreams or
similar and I figured I'd ask the list. If we found something that lots
What is the best solution? I dont want to have modify firewall's at all or do port fowarding. Ideally I would like a solution that with either a softphone or wireless hardphone one could connect via friends, family, or hotspots without reconfiguring their devices.
What are people using? STUN?
On 10/12/05 15:41 Corey Frang said the following:
Interestingly, I started playing with the numbers on my phone after the
dial messed up, and I could get the DTMF tones stuck playing one tone
for a long time. If i took the D() out of it It didn't have that problem.
On Aug 25, 2005, at
While I don't have it working yet, I think I have it figured out. I
have to add keys / entries to my sip.conf Based on your example I was
able to find the relevant info in the Polycom SIP 1.5 Admin Guide
section 4.6.1.15.
My next question, which I haven't found in the admin guide (at least
Hi all,
I'm looking for a way with any asterisk-version with TE410P (cpe
EuroISDN, Q931)
for sending an INFORMATION ELEMENT KeypadFacility,
eg. *87, during a connected call to the PSTN switch.
Are there existing functions in asterisk to generate send such IE ?
If not what existing modules
Nathan Pralle wrote:
Hey folks,
Anyone know of companies selling bulk SIP adaptors (phones, adaptors,
etc.) or has the list ever considered doing something like a bulk buy?
Give a call to VoipSupply.com 800-398-VOIP (8647)
I was just curious...I'm looking to get another 5-6 Grandstreams or
Hi,
This is what I have in extensions_custom.conf:
; Time of Day functionality:
exten = *60,1,Answer
exten = *60,2,Wait(1)
exten = *60,3,SayUnixTime(,,IMSP)
exten = *60,4,Hangup
It works on a Cisco 7940 IP Phone, but on
analog phones, when I dial *60, I just
get a dial tone. If I dial *60#,
Mensaje citado por: Blake Krone [EMAIL PROTECTED]:
What is the best solution? I dont want to have modify firewall\'s at all or
do port fowarding. Ideally I would like a solution that with either a
softphone or wireless hardphone one could connect via friends, family, or
hotspots without
Use application ChanIsAvail with the s option. This option only exists
in CVS-HEAD version, the 1.0.x versions don't have this option.
from documentation:
If the option 's' is specified (state), will consider channel unavailable
when the channel is in use at all, even if it can take another
They are in the 1.5 admin guide, pages 22-25
Matthew T. O'Connor wrote:
While I don't have it working yet, I think I have it figured out. I
have to add keys / entries to my sip.conf Based on your example I was
able to find the relevant info in the Polycom SIP 1.5 Admin Guide
section
Hi
Is anyone using Asterisk for PPP over PRI ISDN. Any example would be
appreciated. I saw ZAPRAS and PPPD commands. The documentation has
zaptel.conf example for PPP using T1/E1 clear channel.
Regards
Goutam
___
--Bandwidth and Colocation sponsored
It is on page 22 and 23 in my admin guide.
Andy
--
H. Andy Goss
Network Engineer
Network Advocates Inc.
Main: 502.412.1050
DID: 502.992.5933
Mobile: 502.387.8216
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of
Hello,
I'd like to know if it is possible to get * to listen and respond on more
than just one single udp port.
I've run into several situations where I'd like IAX to work on an alternate
port as well as be able to work on the standard port.
I'm wondering if there is a way to do this?
Thanks!!
My apologies for the cross-posting.
If you are a business or individual providing Voice over IP services in
Canada then we encourage you to read this email carefully otherwise
please disregard.
-
As you are most likely aware, the CRTC has undertaken the roll of
regulating VoIP services in
The fax2mail and
mail2fax scripts can be found on www.generationd.com
Michelle
DupuisTechnical Support SpecialistOxford Consulting Group Ltd.Making IT work for your
business...
T: (519) 672-8238E:
[EMAIL PROTECTED]W:
www.ocg.ca
___
The people who have been documenting Asterisk have been working on a book
for the last few months, it has been published by O'reilly (Asterisk-The
Future of Telephony)and is just now finding it's way into the major
bookstores, listed under Open-Source at BarnsNoble.
While it will not answer
On Wednesday 12 Oct 2005 14:53, Tom Rymes wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Bob Goddard
Sent: Tuesday, October 11, 2005 6:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
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