Experiences with the HT386s seem to be pretty variable: they work OK for
some folks, and are virtually unusable for others.
A couple of months back, I installed two HT386s side by side. One of
them would lock up on an almost daily basis; I replaced it with an
SPA2002, which has been much
Hi..
Have AAH set up with tdm card. 1 pstn line.
When incoming call initiated hard phone rings almost instantly.
Problem with outgoing calls from sipura spa 941, the call connects etc, but
is very slow to go out onto pstn.
There is a significant lag before the call at other end rings, perhaps
To retort, Digium has ever to my knowledge, stamped an 'Enterprise
Grade' mark on the product. If you are worried about a single point of
failure you may want to replace your toaster.
Asterisk is missing a 'few features' no doubt about it, but it is open
source, it will be a welcome addition if
I had the same issue when I was playing with * @ home and it was the
call waiting feature. I'm pretty sure it's off by default so have a play
with that. *70 to turn it on, *71 to turn it off.
Marc.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rolf
Hi
The H323 patch for [EMAIL PROTECTED] is very out dated. Try
http://www.mbit.com.au/h323/h323.zip
It should have everything you need to get H323 up and running.
Regards
Mark Brooker
T: 02 4959 8670
M: 0415 846 865
F: 02 9882 0947
E: [EMAIL PROTECTED]
W: http://www.mbit.com.au
Could it be Call Waiting Deactived?
On 3/7/06, Rolf Brusletto [EMAIL PROTECTED] wrote:
All - I've been muddling around with this for a few days now.. and I'm
trying to figure out why I am not receiving more than one phone call on
each polycom 501 phone. I can make more than one phone call out,
Hello,
I use both ser/asterisk .
In fact i wish asterisk to forward all the sip
requests which are not handled by domain=domain.tld
in sip.conf
Here is a diagram:
The sip agents use the Sip proxy as an outbound sip
proxy and domain=domain.tld .
When the sip agents dial sip:[EMAIL
Good grief! I posted the message below at 1:17pm... and it appeared on the list
after 8pm.
Nice
-Original Message-
From: Douglas Garstang
Sent: Tue 3/7/2006 1:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Martin Joseph
Sent: 7. ozujak 2006 18:40
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: ON DEMAND call Recording
On Mar 7, 2006, at 2:38 AM,
I'm running 2.4.5 and app_meetme never plays conf-hasleft or
conf-hasjoined with user names. I looked at app_meetme.c, but couldn't
determine the cause. Any suggestions are greatly appreciated.
exten = 600,1,MeetMe(600|i) I get the following:
-- Executing MeetMe(SIP/jon-21f8, 600|aciMps) in
I have 10 different numbers that can come into my asterisk box, but
they all seem to end up as the same extension in my dialing plan.
As far as I can tell the reson is that the INVITE line is always the
same number; but t: shows the correct number. Is there a variable
that I need to check for
Or TRANSFER - BLIND - NUMBER - SEND, for a blind one. Works for me,
no special phone configs.
Moj
[EMAIL PROTECTED] wrote:
Ummm - from memory the sequence is TRANSFER - NUMBER - SEND - chat to
other person - TRANSFER.
PaulH
- Original Message -
*From:* MBIT Technologies
Kerry Garrison wrote:
On a 55 station install onto a Cox PRI with a TE110P (Polycom 501 phones) a
few users are complaiining about echo. According to the users, the echo
seems to be phone number dependant. They claim that certain phone numbers
have echo while others dont. Are there any tuning
Just in case anyone is interested, there is new Mitel SIP firmware out
today. Version 5.00.00.16
http://sipdnld.mitel.com/
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On 1/16/06, Matt Riddell (IT) [EMAIL PROTECTED] wrote:
Simon Faulkner wrote:
Does anyone know of a web based live call monitor for *?
I would have thought this was an ideal application for Ajax?
There's the flash operator panel but nothing much using Ajax. We're
doing some chat room
I used quadBri Junghanns card and I config
zaptel.conf:
ZAPTEL.CONF
loadzone=it
defaultzone=it
span=1,1,3,ccs,ami
span=2,2,3,ccs,ami
span=3,0,3,ccs,ami
span=4,0,3,ccs,ami
bchan=1,2
dchan=3
bchan=4,5
dchan=6
bchan=7,8
dchan=9
bchan=10,11
dchan=12
ZAPATA.CONF
[channels]
language=it
I red that it is possible to send instant messages to the displays of
sip phones. How can I do it using Asterisk?
--
Alejandro Vargas
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Hello!
On 3/7/06, Martin Joseph [EMAIL PROTECTED] wrote:
On Mar 7, 2006, at 7:02 AM, artifex maximus wrote:
I have the following setup:
Phone lines - traditional PBX - Welltech 3802
- VPN -
Asterisk - Linksys PAP2/Welltech ATA-151 - phone
There is 2 pieces of Welltech 3802 (2 port
Hi Bradley,
Yes, I experienced quite a lot of echo with my IAXy, until I switched
analog handsets - in my case, it was severe acoustic coupling in a
cheap handset.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed:
This is a SER/Polycom question, but I hoped we may have some SER guru's here...
I have a series of Polycom phones that are tying to register with OpenSER. The
phone sends a REGISTER message, and OpenSER replies with Unauthorised (all
normal). The phone re-sends the REGISTER with the
What exactly do you need? A digium card could be anything from one
pstn line, to multiple t1 lines, to who knows what else. And serial
number authentication...what's this for? Does a user dial in, enter
in a serial, then get access to something? Like a calling card, or
something completely
Thanks Moj.
But i need to connect to MySQL. Could this be a problemwith C libraries that i am using.
Regards,
Sharath
On 3/8/06, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote:
This may not be the applicable solution, but if you're not using themysql config capabilities, add noload =
Douglas Garstang wrote:
Pardon my candour, but for a product Digium calls 'enterprise grade' it sure
seems to be missing a few features.
You can't resist digging at Digium every time something doesn't work
just the way you expect it to, can you?
Someday you'll be bleating in the ether all
Hi Joe!
Thank you for your mail. The thing is that I have never
program anything so it will take a lot of my time, which I don't have right now.
Hopefully, when I finish started projects I'll be able to play with this
stuff.
In the meantime if anybody solves this problem, please let
the
Hello all,
I am having an issue with a BT-101 and * . When dialing a number from the
BT-101, upon the remote side answering, the call is established but no
audio is passed in either direction. I have tcpdump'd this session and
found this:
(192.168.30.1 is * - 192.168.30.32 is BT-101)
Hi Sharon!
This is pretty difficult, i was not able to implement it so far(though
my ser-skills are pretty basic).
At http://www.voip-info.org/wiki-Asterisk+at+large you'll find some
howto's, method 2 seems to be the most promising to me...
regards
christian
On Tue, 7 Mar 2006 15:36:57 -0600
Hi Phil.
So, you are just like I am...
You can´t make a conclusion about this problem, and nobody has a clue about
what could be the problem...
I will change more than 20 to sipurar 2002... A pay for this
I will never buy again grandstreeam hardware,And I´ve bought more than 100
phones(bt
Can someone help me with upgrading to the lastest version. I am using the
same sip.conf file, but the headers have changed and registration fails.
Has something change in the conf file that would cause this?
Notice 1.2.5 has no Authoization at all...
Regards,
Jason
Version 1.0.9
We had the same issue but we found that it was really the MS proxy server
that the phone was going though. Set it up to use a different route out to
the server and everything worked fine.
Had to prove it to the admin at the location too, that was fun!
Rick
-Original Message-
From:
On Tue, 7 Mar 2006 18:49:58 +0100, Olle E Johansson [EMAIL PROTECTED]
said:
With SIP phones, the phone, not Asterisk, generates all the indications.
Check with Aastra.
In some cases, like during a call transfer, Asterisk may generate a
tone.
Hi Olle,
Thanks for clarifying things for
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Mojo with Horan Company, LLC wrote:
This may not be the applicable solution, but if you're not using the
mysql config capabilities, add noload = res_config_mysql.so to
modules.conf
Moj
Sharath Chandra wrote:
Hi all,
I installed the
pali ismail wrote:
i have do some touch tones registration system in asterisk .
know i hae some problem i my extensions.conf ,,,because the script there
cannot run yet
so i hope some budy have codeing in check password plase give to me
i can check that my code its right or wrong
:)
I may be mistaken (the first time) but I thought someone once told me it
was possible to set the volume on a per phone basis. All users have
7960's running 7.4 or 7.5; server running 1.2.5; one tdm04b. Is this
indeed possible?, and, if so, would someone be so generous as to point
me at a URL.
Docs? Polycom has docs? Where would one find this fabled land of... err I mean
Polycom does stating what ftp servers are supported?
Doug.
-Original Message-
From: Ken D'Ambrosio [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 07, 2006 12:12 PM
To: Asterisk Users Mailing List -
Douglas Garstang wrote:
Pardon my candour, but for a product Digium calls 'enterprise grade' it sure
seems to be missing a few features.
Um...it's Open Source. Why don't you add the features you require
yourself or pay someone to add them for you...
This is your third similar post in as many
I have an ZAP extension number 222 which is connected instead to a phone
to a FXS/FXO converter and from there to a CDMA gateway.
To dial my mobile phone I use:
222 (wait 2 seconds) 09123456789
I cannot figure out how to write this into the dialplan as a default number!
222 as above I will
Hey
Everyone,
We are in the works
of planning a new * installation for our company. We have 20 users in our
main office and 5 users in a remote office a couple of states away. Our
call volume for the main office will be anywhere from 5-10 concurrent
calls. The remote office will have
Hi,
You have get the openh323 and pwlib release supported
by asterisk 1.2.1. You can check it on the README file
located at the /path/of/asterisk/channels/h323/
--- Viktor Tatianin [EMAIL PROTECTED] wrote:
Hi
I use Asterisk 1.2.1
-Original Message-
From: [EMAIL PROTECTED]
Brian Roy wrote:
On 3/3/06, Gary Richardson [EMAIL PROTECTED] wrote:
I'm running 1.2.4 and just about every call is cut short. I'm using Cisco
IP phones as end points. All the outbound calls are routed via SIP through a
PRI line attached to a Cisco 2811..
I'm running 1.2.1 and most of
Hey all,
I have a situation where I have 8 lines from the phone company in a hunt
group coming in to my asterisk box. These are the same lines I'm using
for outgoing calls ( named g0 ).
The problem arises when someone dials our number at the same time
asterisk tries to put a call out on
Right, Avaya can do that. Use Avaya.
On 3/7/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Pardon my candour, but for a product Digium calls 'enterprise grade' it sure
seems to be missing a few features.
-Original Message-
From: Alexander Lopez [mailto:[EMAIL PROTECTED]
Sent:
Hey everyone,
I have noticed a few questions close to the issue I am having but I
haven't seen any that quite match the problem I am seeing.
I have 3 queues. Some members share one queue and some are completely
separate. Some members have a higher penalty then others. I am using
Ken D'Ambrosio wrote:
HTTP's nice, but FTP does the job. Check the docs for supported FTP
servers -- many of the stock Linux FTP servers will give the exact problem
you discussed, below. I should know -- took me almost a week before
trying proftpd, and WHAMMO, worked like a champ.
-Ken
Sina, hi;
Let's just do a little recap.
You've downloaded zaptel-1.2.4 and done the
make linux26
make install
make config
thing on it. If you don't uncomment anything, the builds complete
without error and modules are installed in
/lib/modules/`uname -r`/extra.
You've performed the 2.6
Hi all,
I've [EMAIL PROTECTED] with Ztdummy running on VMWare, and i've adjust
already three times the date and it seems to me it is running clock
faster... After a while Asterisk clock greater than my windows clock
time
Isn't this strange?
I'm just waiting for a Digium card to change this
Hi,
Anybody got an idea of how many SIP calls I can run through asterisk on
a Dual 3.6 Xeon (Dell 2850)
if:
- It doesn't perform any transcoding
- Calls are g.729a
- It doesn't have an interface to reg phone network (MGW in an other
box) ie. everything is SIP to SIP.
- Re-Invite is not allowed (I
With the help of one of the providers we terminate on, I've found the
source of the problem of getting busy even when the called isn't really
busy in the absence of ANI codes in sip headers generated by asterisk.
If I put a NoOp(${CALLINGANI2}) in the dialplan before the dial I can
see it
Your long pause complaint is the
timeout on the PAP2 before it thinks you're done dialing. The voicemail issue
sounds like the dialplan on the PAP2, what do you use to connect? if it's a
star-code (*), you need *#. in the plan to pass *+any
numbers
SKM
From: [EMAIL PROTECTED]
Hi all,
I didn't find yet any info about this. Is there any way for a
Conference Room Owner to change his own password? A kind of Menu like
calling his conference room:
example:8200
And an IVR option to change password.
That seems to me interesting, because i may not want the same users
To add to the other post... aah or amp actually has a DB that contains
call waiting information. It may have the default setup such that call
waiting is disabled. You should be able to dial *70 and enable it.
Sean
On Tue, 2006-03-07 at 11:33 -0700, Rolf Brusletto wrote:
All - I've been
ThanksI've got the SEPMAC files that I use successfully with SCCP.
-D
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg
Oliver
Sent: Tuesday, March 07, 2006 4:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
Hello,Somebody has managed to make Mitel SX-2000 and Asterisk integration work. http://www.voip-info.org/wiki-Asterisk+legacy+integrationCan you please post your zaptel.conf and zapata.conf for T1/PRI config?I will be configuring a TE210P to connect to an SX-2000 PBX.Thanks.
Hello!
Does TE110P (a 32-bit PCI) fit into PCI Express x8 slot? I'm thinking
of buying a Sun X2100 and it has a PCI Express x8 slot.
Or perhaps, does Digium produce PCI Express E1 cards?
Thanks in advance!
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Hi all,
The metermaid patch allows you to use the programmable buttons and
LEDs on phones (like the Snom 2xx or 3xx) to view the status of parking
slots and transfer to them. This should be really useful for
small-office environments.
Anyway, the patch seems to work with Snom phones (and
Hello all,
I am new in asterisk configuration. I want to configure a Radius server
to authenticate the sip users of asterisk. I have trying to use the next
document:
http://www-rocq.inria.fr/who/Philippe.Sultan/Asterisk/asterisk_sip_external_authentication.html
Can you help me?
I have the exact same problem. SetMusicOnHold between two sip phones always
returns the default class.
Any ideas?
Joe
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Does anyone have this working on 1800MHz eg TMobile or Orange in the UK
and does CLID work or not?
THanks
Harvey
- Original Message -
From: Conrad Wood [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, February
I have installed asterisk @ home 2.6. I am using a
Telasip VOIP account. When I make outbound or inbound calls the calls seem
to connect and then get hung up. I was wondering if there was something
that I am misisng. I have tried several different sip.conf
configurations. Here is what they are
I can't be bothered looking for the link right now, but it's definitely stated
somewhere on Digium's website.
-Original Message-
From: Alexander Lopez [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 07, 2006 3:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
This might be a better question for the dev list, but does anyone know
the status of a jitter buffer for SIP channels?
I know they created a generic jitter buffer and implemented it for IAX
channels. Does it work yet for SIP? Like is it there and disabled or
not there at all?
Hi all,
Do anyone have experience www.Chinaroby.com VOIP
phones?
I am very interestedfor models:PY-60 and PB-35
Phones.
Good or bad
experience with sip and IAX2, please comment.
Regards
Darko
Sundek
eLink
Group
Kotor-Montenegro
___
Hi guys,
I want to setup a environment where asterisk load all information from a
Postgresql database. So here goes my questions:
1) Is real time asterisk stable enough?
2) Where can I found documentation about using it with Postgresql? (
including meet me conferences)
Thanks in advance.
There is a patch to chan_sip on voip-info.org that I use. It seems to
work very well. I believe it is on the Astrisk at large page on the
voip-info.org wiki.
regards,
Darvid
On 3/7/06, Sharon [EMAIL PROTECTED] wrote:
I have my peers registered to SER.asterisk seems to be sending mwi for
the
I havent got any mails since 2:42 this morning..usually i get at least
the normal 10-15 a hour, if someone gets this can they reply?
Thanks!
Ron
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Hi,
I setup a SIP trunk in a brand new Cisco Call Manager and I
try to place the calls using Asterisk but I get error:
-- SIP read from 192.168.11.10:5060:
SIP/2.0 400 Bad Request - 'Malformed/Missing URL'
Via: SIP/2.0/UDP
192.168.10.199:5060;branch=z9hG4bK2e7ca9c9;rport
From:
Hello,I am using pickup, i can pickup an extension from outside of the queue, but i cannot pickup any call comes to queue.queue strategy=ringallWhat is the problem with queue?Is there anyway to pickup last ringing phone?-erkaN
Yahoo! Mail
Bring photos to life! New PhotoMail makes sharing a
Hi Martin,
I have 3 choices on my ATA webpage and I chose SIP INFO:
/Send DTMF: / in-audio via RTP (RFC2833) via SIP INFO
This is the only point I can make changes since it is connected to my
asterisk box through a TDM400P:
asterisk box ---TDM400P -(telephone cable)- HT-288 --- LAN ---
In Asterisk 1.2.4 is love being able to recording conferences. However,
using the default variables, the files are being written to
/var/lib/asterisk/sounds instead of /var/spool/asterisk/meetme.
If I change MEETME_RECORDINGFILE variable to something different in works,
bit I lose the ability to
Youre almost right.
The PAP2 has some features
that are factory default. I dont remember the section in the web
interface, but heres what you going to do:
Find the section that
contains a lot of features name with values like this *56 or *78.
Erase all of them. Letting
this filled
I have a Linksys PAP2. Identical setups for the two channels in both
the unit and in Asterisk. In particular, both channels enable g729 and
set it as the preferred codec, and have disallow=all and allow=g729 in
sip.conf.
If we make a call on one channel, it works (and uses g729), but if we
I've an Asterisk 1.2.4 installation, where I've also installed the G729
codec license. I'd like to upgrade that installation to 1.2.5, but I'm
not sure if I'll lost the license in the process (and if I'll be able to
recover it later!!!).
Is there any special consideration I've to keep in mind
Theres the
SetCallerID cmd that you should read about.
http://www.voip-info.org/wiki-Asterisk+cmd+SetCallerID
It has others links to clarify
your ideas.
Tell us if you get
something.
Filipe Mordhorst
Brazil-SC
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Hi,
What port does mpg123 uses to play music on when it starts MoH after
asterisk has put called on hold?
Zach A
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Hi,
Is it possible to do this in extensions.conf to put a caller
in queue and dial an agents extension so that he knows that somebody is
in queue waiting to be answered. This agent will be a remote agent and
extension will dial his cell phone.
Thanks
Zach A.
I'd like to know if it's possible to set the REINVITE on or off dynamically,
based on the extension being dialed.
Define two peers in sip.conf, one with canreinvite=yes and the second
with canreinvite=no. Then you can route your calls with or without
reinvites depending on the dialed number.
Do you have the phone specific config file for the polycom set to something
like this?
?xml version=1.0 encoding=UTF-8 standalone=yes?
phone1
reg reg.1.displayName=default reg.1.address=27 reg.1.label=27
reg.1.type=private reg.1.auth.userId=27 reg.1.auth.password=
reg.1.lineKeys=3/
Anyone have any information on the performance impact of
using qualify=yes for hundreds (500ish) of SIP UAs?
I have seen tidbits on qualifyspreading=yes, but not enough
to understand what it does. I assume lessens the peak load of qualify sip
options queries?
Thx!
Do you have call-limit parameter set to 3 in sip.conf or possibly
sip_additional.conf on AAH?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rolf
Brusletto
Sent: Tuesday, March 07, 2006 1:34 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
Hello,This is not a question directly related to asterisk.I am currently rinning ansterisk on a Debian server and i just upgraded my memory from 1GB to 2GB. However, my linux OS does not recognise the memory upgrade. The BIOS does, but the Debian Linux refuses to use the entier memory, currently,
Are you guys perchance using Local/[EMAIL PROTECTED] in your installations?
--
Cheers,
Matt Riddell
___
Is there a known issue when using the Local/[EMAIL PROTECTED]
thanks,
This is how I would read it.. but yes.. can someone give
Some one was on my server making changes to my sip.conf files. I am now having trouble with DTMF. No matter what I use (inband,auto,rfc2833) the dtmf tones seem to not come thru. I compared it to the wiki and all the configs seem to be in order.Here is my sip.conf
Figured it out. It was simple had to add Answer and hangupDovid Bender [EMAIL PROTECTED] wrote: Helo List,First I would like to apologize for my bad spelling aswell as that I did not search the wiki first. I onlyhave email access at the moment.I am having trouble setting both variables and
Hey
Everyone,
We are in the works
of planning a new * installation for our company. We have 20 users in our
main office and 5 users in a remote office a couple of states away. Our
call volume for the main office will be anywhere from 5-10 concurrent
calls. The remote office will have
Hi,
I've found several softphones for Windows Mobile 2003, but does anyone
know of a softphone (or older version of a current softphone) that
will run on Windows CE 3.0?
~ Matt
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I am having a problem when trying to send a receive faxes on an E1
running with unicall on an asterisk 1.2.4 x64 server. The same server has a
TDM02 card and if I send or receive faxes through there there is usually no
problem. I am afraid that my customer insists that he wants to use the
Tellabs looks a little too up-scale for what I need :). $1k for a
single port orion unit might be worth considering for really stubborn
installs though.
Why? they go for around $100.00 on eBay.
What goes for $100 on eBay? Tellabs? or Orion? I can't find any
Orion equipment on eBay.
I have decided to move on from [EMAIL PROTECTED] and start
compiling asterisk myself now. I got a dedicated box and put my X100P in it. I
installed the server version of CentOS 4.2 (2.6.9-22.EL kernel) barebones. The
box is a dell GX270 workstation with 1GB of RAM. I got a fresh copy of
Does anybody have any experience with capabilities
here? I need to know if IAX is able to handle more than that. I
think I might just benchmark this with a bunch of .call files between servers to
see how they are handled.
Any input?
- Gabriel Afana
- Original Message -
From:
Is there a way to display the time of the 7960 running
firmware 7.4? Im unable to find any information.
Thanks,
Ben Blakely
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Does anyone know why putting an outbound proxy in the SIPmac.cnf file
causes the phone to not pull it's logo from logo_url?
Aaron
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Hello,
I'm trying to do call forwarding based on this:
http://www.voip-info.org/wiki/view/Asterisk+call+forwarding
In the extensions.conf file I have several context defined (local,
longdistance, mobile, international and so on). Each user can be
associated with different context (so can
Is anyone with a yahoo account having problems
recieving emails from the list. I have not recieved
any emails in about 8 hours and I posted something
about 3 hours ago. If anyone knows please email to
asteriskdigium _AT_ yahoo.com
Thanks
__
Do You
I don't have the most reliable internet connection in the world.
Whenever it goes out, I can't receive any incoming calls at all, not
even from pstn. When it first goes out I can still make outgoing
calls through pstn, but eventually that fails too (as does voicemail,
everything's out). Yes,
Sorry, This is a mistake, sip.conf:
[302]canreinvite=no[301]canreinvite=noAny idea?Thanks
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Good to know I'm not the only one...
I thought perhaps I had been expelled from the list...
Bob McDowell
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Tuesday, March 07, 2006 10:44 PM
To: Asterisk Users Mailing List -
we mirror all the files our selves so our scripts work
flawlessly.
--- Alistair Cunningham [EMAIL PROTECTED]
wrote:
This is a request to the website manager for
asterisk.org.
The build scripts for our ITSP product include the
URLs to download the
Asterisk files, such as:
wget
why not use astcc ? it comes with asterisk and does
all that you have requested. we have scripts running.
one that works via CID and one the user enters the
number.
--- leonimar cape [EMAIL PROTECTED] wrote:
Hi group,
I am currently looking for a prepaid application
that
can do the
I have an ZAP extension number 222 which is connected instead to a phone
to a FXS/FXO converter and from there to a CDMA gateway.
To dial my mobile phone I use:
222 (wait 2 seconds) 09123456789
I cannot figure out how to write this into the dialplan as a default number!
222 as above I will
Tomislav Parcina wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Martin Joseph
Sent: 7. ozujak 2006 18:40
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: ON DEMAND call Recording
On Mar 7, 2006,
Are the Polycoms doing this on a different network than the Polycoms
not doing this?
On 3/7/06, Douglas Garstang [EMAIL PROTECTED] wrote:
This is a SER/Polycom question, but I hoped we may have some SER guru's
here...
I have a series of Polycom phones that are tying to register with OpenSER.
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