Hi list!
I have a strange echo problem. Two days ago I setup * 1.0.6. at a friend
of mine. Just an * server and for outbound calls wengo.fr was used to
place calls via sip. He had a strange echo on the line I didn't
experience on my setup.
Today I upgraded my asterisk 1.0.5 to 1.0.6 and
If you are familiar with this application i appreciate
any help you can offer me. I downloaded Areski_AGI,
AreskiCC_UI, and AreskiCC.psql and I followed the
installation procedures in INSTALL.TXT, but I don't
know what to do next. I am blank and there is no
user's guide. It would help a lot if you
Other than Broadvoice, are there any VoIP providers (Vonage, Packet8,
etc) that can be hooked into Asterisk directly? I read about a scheme
for Packet8 that involved routing it in through an analog connection
on a FXO port...I'd rather have something I can connect in directly.
Save yourself
Remco,
-Original Message-
I have a strange echo problem. Two days ago I setup * 1.0.6.
at a friend
of mine. Just an * server and for outbound calls wengo.fr was used to
place calls via sip. He had a strange echo on the line I didn't
experience on my setup.
Today I upgraded
Guys.
Which PC softphone (IAX2 or SIP) can support getting a url on the dial cmd
and opening a web page on the users computer?
Any choices?
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Asterisk-Users@lists.digium.com
Today I upgraded my asterisk 1.0.5 to 1.0.6 and suddenly I
have an echo
too on sip calls thru wengo!!
Can you identify which side of the conversation hears the echo ? As you
know, echo is heard on the side opposite of where it is generated.
Yes, it's my side only. The party being called doesn't
Hi,
Which PC softphone (IAX2 or SIP) can support getting a url on the dial cmd
and opening a web page on the users computer?
Try DIAX 0.9.11a (not yet officially available) from the following location:
http://www.laser.com/dante/diax/diax0911a.zip
What's new comparing with 0.9.10f (the latest
Dear all,
I'm trying to use ulaw and alaw with Diax and Asterisk but I'm not able to,
I got the following error message:
Mar 20 11:47:59 NOTICE[7099]: chan_iax2.c:6350 socket_read: Rejected connect
attempt from 192.168.0.55, requested/capability 0x8/0xc incompatible with
our capability 0xfe02.
Release 0.66 of IPSwitchBoard is now available for FREE download at:
http://www.voip-info.org/tiki-index.php?page=IPSwitchBoard+BETA
Enhancements:
Support for Call Parking and retrieve/forward them again.
Last Call on the Queues Page now displays a date-time in human readable
format.
Added
Hi,
I'm trying to use ulaw and alaw with Diax and Asterisk but I'm not able
to, I got the following error message:
Mar 20 11:47:59 NOTICE[7099]: chan_iax2.c:6350 socket_read: Rejected
connect attempt from 192.168.0.55, requested/capability 0x8/0xc
incompatible with our capability 0xfe02.
I do
[EMAIL PROTECTED] is believed to have said:
It dosn't run under the mono framework. There, now you have an answer :-)
Oh, well: sad enough...
Thanks for the answer.
Aldo
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Hi,
I've got an Asterisk system that I've just added an X100P card to.
Incoming calls route to my call group just fine. When I make outgoing calls
by prefixing with 9, they route to the PSTN network okay, get answered but
then drop straight away.
Has anyone seen this before and found a fix?
Solved, thanks
Dan, I'm just performing same tuning testing for echo.
Using Diax I can hear echo in my mobile. It happens for any codec that I use
(G711, GSM, iLbc, speex). Using Firefly with GSM or alaw, echo is very
reduced. (I can hear just a bit of echo). Can you suggest same adjustment
for
Hi,
Solved, thanks
Dan, I'm just performing same tuning testing for echo.
Using Diax I can hear echo in my mobile. It happens for any codec that I
use (G711, GSM, iLbc, speex). Using Firefly with GSM or alaw, echo is very
reduced. (I can hear just a bit of echo). Can you suggest same adjustment
Hi Remco,
-Original Message-
- The methods in which asterisk cancels echo may have been
changed between
1.0.5 and 1.0.6. You would never notice this, because your
setup is all VoIP
(no Zap)
- You might experience echo because the length of the audio
path is now
exceeding
Hi Dan,
disabling AGC echo is reduced.What AGC is and what is it for?
Thanks
- Original Message -
From: Dan [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, March 20, 2005 12:07 PM
Subject: Re: [Asterisk-Users] I
Hi,
Ive just been made aware of Asterisk through
the article on Slashdot. Were a small UK
company with a ISDN-30 link to the office and a Meridian (Nortel) PBX. Were expanding
and moving offices soon and Im looking seriously at VoIP technologies.
So starter for ten, can Asterisk
Hi,
disabling AGC echo is reduced.What AGC is and what is it for?
It is Automatic Gain Control for the local microphone.
It can be usefull when is a good isolation between speaker and
microphone and you want to capture low level sounds.
Best regards,
Dan
I ran into the web ui issues with Postgres, but it works fine
with MySQL. I haven't had the time to figure out what was wrong
with the sql statements, so I just switched to MySQL.
Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kris
Edwards
Sent:
Great, just received 9 virus emails in the past 24 hours
from the asterisk list where people have had my address in their address book.
Heads up people, its an attachment, the text looks a
little jinglish why would you open it?
Cheers,
Dean
Hi Bob,
Thanks for the reply. If Asterisk didn't interface with POTS then it was a
non-starter for us. I'll keep a note of the contact name.
Cheers, Rob.
-Original Message-
From: Bob Goddard [mailto:[EMAIL PROTECTED]
Sent: 20 March 2005 12:54
To: Rob Nicholson
Subject: Re:
Peter,
You need to spend some more time reading the wiki, your question is far
to basic for someone who had invested anything more than about 15
minutes looking at asterisk.
http://www.voip-info.org/wiki-VOIP+Service+Providers+Residential
this is a basic list of voip providers some will
Adnan Ahmed [EMAIL PROTECTED] wrote:
can anyone installing/configuring asterisk's on SLES9 if someone can
share his/her views experiences .
The easiest way is to get the lastest bristuff package and let it do the
work. That's what I did last week, took me less than half an hour.
Be sure to
Which script in [EMAIL PROTECTED] are we talking about? I can't say I've
ever seen it!
On Fri, 18 Mar 2005 21:29:48 -0500, Steve Prior [EMAIL PROTECTED] wrote:
Wolfgang S. Rupprecht wrote:
[EMAIL PROTECTED] (Steve Prior) writes:
The recorded prompts by Allison are more in line with the
Take a look on the [EMAIL PROTECTED] sourceforge forum for festival-weather,
it's like the most popular topic over there.
I think like about 50 people have downloaded it since it got put up a
few weeks ago.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Tom Samplonius wrote:
What does progressinband do exactly? Does it disable 180 responses?
I can't find any references to what effect no, yes, and never
have on the SIP exhange. In fact, why is it called inband if it
involves the SIP messages? Wouldn't inband refer to messaging in
the media
Kevin P. Fleming wrote:
If set to 'yes', and '183 Session Progress' has not already been sent,
then '180 Ringing' is sent _and_ audio ringback is also generated
(although I can't seem to figure out how that could work, since if '183
Session Progress' has not been sent, there is no early media
Im in the initial stages of my asterisk
experimentation, and after some messing about, have it working to some
extent. Right now Im in a pure SIP environment with no trunk lines and no NAT, and am
configuring everything via [EMAIL PROTECTED]
My problem is that I am only able to get one
Message: 5
Date: Sun, 20 Mar 2005 03:55:50 +0100
From: bram [EMAIL PROTECTED]
Subject: [Asterisk-Users] RE:Newbie question
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain
It said 'include zapata-channels.conf', where this line wasn't
commented bij the
cmisip wrote:
No path to translate from SIP/fwdpulvercom-dd5a(2) to Phone/phone0(1)
I don't know why the above message is printing codec numnbers, rather
than names. *shrug*
show codecs will tell you what codec number are what codec name.
It appears that your Phone/phone0 is using G723.1. Looks
Tyler wrote:
I think you're looking for the 'ChanSpy' application that seems to have
inexplicably vanished from the asterisk CVS..
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20ChanSpy
If anyone has any info on this, let me know as I'm in a similar
situation.
As far as I know
I've just finished building an ISDN30 system. You will need a TE110P
from Digium which will cost around £350.
Took a couple of hours - the only thing you have to be careful of is
the zaptel.conf file and the zapata.conf file. My configs are for a 12
channel ISDN30 on BT in London. If you would
yes, this also known as an E-1 PRI
- Original Message -
From:
Rob Nicholson
To: asterisk-users@lists.digium.com
Sent: Sunday, March 20, 2005 6:20
AM
Subject: [Asterisk-Users] ISDN-30 in
UK
Hi,
Ive just been made aware of
Asterisk through the
Avoid iax.cc -- no customer service. There's also livevoip and gafachi.
-Original Message-
From: Jean-Michel Hiver [mailto:[EMAIL PROTECTED]
Sent: Sunday, March 20, 2005 3:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] VoIP service
It's only one infected user so far -- all messages are coming from
67.122.114.23 - registered to Pacific Bell/SBC. Probably someone with
DSL who wasn't smart enough to NOT OPEN attachments.
-Original Message-
From: dean collins [mailto:[EMAIL PROTECTED]
Sent: Sunday, March 20, 2005 7:17
I started with Broadvoice recently but I am constantly having problems.
I got everything configured and now it is dropping outgoing calls after
40 seconds and incoming calls are going direct to voicemail. Getting
customer service is proving very difficult.
I've started to look at some of
Interesting virus though, changing the senders details each time from
either deleted emails or worse from the persons address book.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk
Sent: Sunday, March 20, 2005 10:51 AM
To: 'Asterisk Users
Livevoip.com is another one that's pretty reliable and only charge 1.27
cents per minute. You can add a toll-free number for an extra $1 per
month. Setup was a breeze!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of MF Hulber
Sent: Sunday, March 20,
Good morning all,
I have been trying to research of to change the ring frequency for the
TDM400 FXS port. I have several newer phones that will start to ring and
then quit intermittently. I have tried boosting the voltage using
boostringer=1 and that has not helped. I did verify in dmesg that
To have asterisk ignore incoming calls on an X100P fxo interface, do I
have to just not configure it in zapata.conf, or is there a way to
have the call ignored in a dialplan?
Just define a context like this :
[home-incoming]
exten = s,1,Wait,1 ; Wait 2 seconds, to get callerid
exten =
On Sun, 2005-03-20 at 08:17 -0500, dean collins wrote:
Great, just received 9 virus emails in the past 24 hours from the
asterisk list where people have had my address in their address book.
Heads up people, its an attachment, the text looks a little
jinglish why would you open it?
:) yes Steven we really should keep asterisk a secret - ban all newbies,
all they bring is credibility, revenue and a reason for asterisk to
exist in the first place.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent:
On Sat, March 19, 2005 15:10, Steven Critchfield said:
Funny that you complain about the virus when my spam levels go up
considerably every time another slashdot article goes out about
asterisk. Stupid users who haven't learned safe computing littering the
net with all their trash.
Since I
That's not interesting. That's pretty much par for the course when it
comes to trojans and viruses these days.
-Original Message-
From: dean collins [mailto:[EMAIL PROTECTED]
Sent: Sunday, March 20, 2005 9:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
It seems to me silly to have a T1/E1 card to connect to a channel bank
when you could just have a 24/30 way FXS card in the slot in the first
place.
Does such a thing exist?
Wouldn't Digium have a lot of customers if they could produce one for
say $1000 retail?
Trouble is
I was just checking them out. I notice that their regular DID only
allows for 2 simultaneous calls. They have commercial DIDs that allow
for more. Although I don't anticipate to have more right away, when I
start forwarding calls to all of my other devices I can see actually
using more than 2
Rich Adamson wrote:
Likewise for a pc card supporting 24 fxs lines. The probability of three
or more lines ringing at exactly the same time are very small. With at
least a little engineering forethought, its not that difficult to
create ring cycles where ports 1 through 6 ring during some period,
Other than Broadvoice, are there any VoIP providers (Vonage, Packet8,
etc) that can be hooked into Asterisk directly? I read about a scheme
for Packet8 that involved routing it in through an analog connection
on a FXO port...I'd rather have something I can connect in directly.
Save
[EMAIL PROTECTED] wrote:
Hello,
I have recently bought a X100P card. I installed asterisk succesfuly.
When I plug my operator's line in the line jack of the card the result
is that my line is not working any more. When I put it back to the phone, everything OK.
It seems that the card leaves
Robert Webb wrote:
Good morning all,
I have been trying to research of to change the ring frequency for the
TDM400 FXS port. I have several newer phones that will start to ring and
then quit intermittently. I have tried boosting the voltage using
boostringer=1 and that has not helped. I did
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Lyle Giese
Sent: Sunday, March 20, 2005 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] wctdm fxs ring frequency
Robert Webb wrote:
Yesterday I was using one of the cheap Radio Shack phone polarity on various
phone outlets in my house and ended up plugging it into my IAXY. While the
regular phone jacks tested OK, the IAXY tested as being reverse polarity.
The tester was plugged directly into the IAXY so there is no chance of
I took a look at teliax. The pay as you go plan appears not to include
international dialing and the commercial plan is fixed price of $44.99
per month capped at 500 international minutes a month. Are you aware
if they have international rates based on usage?
MARK.
Rich Adamson wrote:
Here is what should work for you.
In your Cisco
dial-peer voice x voip
huntstop
destination-pattern x - Extension number you want to dial
progress_ind setup enable 3
session protocol sipv2
session target ipv4:y.y.y.y - Your * IP
session transport udp
dtmf-relay rtp-nte
codec
On Sun, 2005-03-20 at 09:54 -0600, Rich Adamson wrote:
It seems to me silly to have a T1/E1 card to connect to a channel bank
when you could just have a 24/30 way FXS card in the slot in the first
place.
Does such a thing exist?
Wouldn't Digium have a lot of customers if
I am looking at using a dual Xeon Dell 1850 with a PERC 4e/Si raid
controller.
Is anyone using these in production right now? If so can you share some of
your experiences?
Thanks,
Geoff
___
Asterisk-Users mailing list
The echo is quite slow, I would estimate about half a second
or even more!
Wow, that's enormous - However your ears can easily deceive you on this. The
only way to know for sure is to record and analyse. Half a second would
imply its accoustic coming from the other end. Is it on all calls or only
Two minutes seems like a long time to initialize a Cisco 7960 IP phone.
What times are others seeing for the load when you reboot a phone? We are
running the SIP 7.4 load. Our * 1.0 stable is also our http, dhcp and tftp
server.
During boot, the display shows:
Configuring VLAN 100 seconds
Am I supposed to create an admin and user menu context that I get sent
to when I press * from the conference?
That's what I decided to do after having similar problems and looking
at the source.
I only compared the source long enough to realize that the menu
functions were coded differently
--On Sunday, March 20, 2005 1:41 PM +0400 Jean-Michel Hiver
[EMAIL PROTECTED] wrote:
re is:
- iax.cc (haven't tried them)
- connect.voicepusle.com (haven't tried them)
connect.voicepulse.com: Very good for incoming. Too expensive for outgoing.
- nufone.net (they're meant to be quite reliable - i
Tom wrote:
Configuring VLAN 100 seconds
TFTP SIP loads a few seconds
back to Configuring VLAN the rest of the time.
Roughly the same there here as well. 7940 boots faster, but not by much.
Doug
___
Asterisk-Users mailing list
I'm running a pair of these. Both run Vmware ESX and one virtual machine
runs * using only ztdummy. Seems to run just fine. It's not not used in
production, just test/development.
Don't know how much that helps you.
Geoff Nordli wrote:
I am looking at using a dual Xeon Dell 1850 with a PERC
Tom wrote:
Configuring VLAN 100 seconds
TFTP SIP loads a few seconds
back to Configuring VLAN the rest of the time.
That's about normal; I wish Cisco would let us turn off CDP in these
phones, it would help tremendously.
___
Asterisk-Users mailing list
Ah! Thanks for the pointer. I was suspicious that I was unable to find
a page like that in the wiki, but I apparently was just using the wrong
keywords.
I looked at [EMAIL PROTECTED], but it seems to require a disk format, and I
don't have a spare box. I'm going to be running Asterisk
For the *Brave At Heart* it might run under wine/xoveroffice.
ms net framework 1.1 appears to run in xoveroffice, don't know about
2.0beta: http://www.interex.org/hpworldnews/hpw310/01lab.jsp
[EMAIL PROTECTED] is believed to have said:
It dosn't run under the mono framework. There, now you
Hello everyone,
Asterisk-addons 1.0.7 was actually a snapshot from CVS HEAD instead of
the 1.0 branch. It has been fixed now.
If you tried to use the original tar of -addons, just download it again
and you should be fine.
Thanks,
Russell Bryant
drumkilla
Stoopid question 1:
I see how to make a call but for the life of me I can't see how to DROP
a call.
Thorben Jensen wrote:
Release 0.66 of IPSwitchBoard is now available for FREE download at:
http://www.voip-info.org/tiki-index.php?page=IPSwitchBoard+BETA
Enhancements:
Support for Call Parking
You will be very disappointed at the call quality if you try and run
other software on an asterisk box, pc interrupts and processing glitches
just don't 'play well' with voice.
For $200 a an old P3/P4 pc it's worth buying a separate box.
Cheers,
Dean
-Original Message-
From: [EMAIL
Hmm. Can you point me to some more info on that topic? I understand the
concepts, I'm just after some more quantitative data.
I really don't have the room to run another machine, and I'm trying to
limit my power consumption.
Thanks.
-Pete
On Mar 20, 2005, at 10:41 AM, dean collins wrote:
You
On Sun, 20 Mar 2005, Rob Gillan wrote:
Hi,
Having all sorts of troubles getting mysql cdr support under OS X.
Mysql, DBI and DBD all installed and running ok, privileges all set
correctly (I think). Latest asterisk-addons checked out of cvs. Keep
getting error on make install
You will be very disappointed at the call quality if you try and
run other software on an asterisk box, pc interrupts and processing
glitches just don't 'play well' with voice.
Call me crazy, but that's what I've been doing for months and the
service quality has been great. I'm running * on a
That's not a 'Stoopid' question, because you can't drop a call. I will add
that feature soon.
Thorben
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] På vegne af John Breeden
Sendt: 20. marts 2005 19:33
Til: Asterisk Users Mailing List -
Jay Milk wrote:
Avoid iax.cc -- no customer service.
Second on that one. Also codec/protocol weirdnesses.
B.
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Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update
I have searched the list and the wiki and have seen references to
changing this in the wcfxs.c file but I am not using that. Likewise, I
have not founf anything in by looking into the wctdm.c file. I am no
programmer but can somewhat follow the code.
This is one of the best kept secrets of
It seems to me silly to have a T1/E1 card to connect to a channel bank
when you could just have a 24/30 way FXS card in the slot in the first
place.
Does such a thing exist?
Wouldn't Digium have a lot of customers if they could produce one for
say $1000 retail?
Guys. I know this might be a long shot but wanted to check with the gurus.
I have outlook 2003 on my computer and wanted to check if there is a way of
connecting outlook with asterisk so that caller id name could be set based
on my outlook address or contacts? Each time a call comes in for me,
Quoting Rich Adamson [EMAIL PROTECTED]:
It seems to me silly to have a T1/E1 card to connect to a channel bank
when you could just have a 24/30 way FXS card in the slot in the first
place.
Does such a thing exist?
Wouldn't Digium have a lot of customers if
Release 0.67 of IPSwitchBoard will be available tomorrow and that release
can Drop calls, as well as look-up Web Pages when a call comes in. (CRM
function).
Thorben
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] På vegne af Thorben Jensen
Top posting for consistency
I don't know what teliax has for international services/rates. I didn't
have a need for those and didn't ask. Send them an email and ask.
It's fairly common knowledge that several of the itsp's are trying
to profit from consolidating long distance, and some will
Anton Krall wrote:
I have outlook 2003 on my computer and wanted to check if there is a way of
connecting outlook with asterisk so that caller id name could be set based
[..]
Go to http://www.voip-info.org and do a search for TAPI.
--
Best regards
Peer Oliver Schmidt
PGP Key ID: 0x83E1C2EA
Hrmm.. do you happen to have the URL off hand? I looked at the
[EMAIL PROTECTED] site but see no forum...
On Sun, 20 Mar 2005 09:46:44 -0500, dean collins [EMAIL PROTECTED] wrote:
Take a look on the [EMAIL PROTECTED] sourceforge forum for festival-weather,
it's like the most popular topic over
Hi,
I'm using a sipura SPA-841... Asterisk seems to be silence away (in
that it doesn't send data if it's silent)... I've set the sipura
device to be silence aware... but it still seems to send data even
when I hit mute.. anyone have any experience with this device or any
thoughts?
Ahh n/m found it:
http://sourceforge.net/forum/?group_id=123387
There definatley should be a link for that on the main [EMAIL PROTECTED] site!
On Sun, 20 Mar 2005 15:29:42 -0500, Matt [EMAIL PROTECTED] wrote:
Hrmm.. do you happen to have the URL off hand? I looked at the
[EMAIL PROTECTED]
Yesterday I was using one of the cheap Radio Shack phone polarity on various
phone outlets in my house and ended up plugging it into my IAXY. While the
regular phone jacks tested OK, the IAXY tested as being reverse polarity.
The tester was plugged directly into the IAXY so there is no chance
Thx!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peer Oliver
Schmidt
Sent: Domingo, 20 de Marzo de 2005 02:17 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] asterisk and outlook
Anton Krall wrote:
I
I have searched the list and the wiki and have seen references to
changing this in the wcfxs.c file but I am not using that. Likewise,
I
have not founf anything in by looking into the wctdm.c file. I am no
programmer but can somewhat follow the code.
This is one of the best kept
[EMAIL PROTECTED] (dean collins) writes:
You will be very disappointed at the call quality if you try and run
other software on an asterisk box, pc interrupts and processing glitches
just don't 'play well' with voice.
This isn't that much of a problem if you structure your phone system
to be
We have 2x BRI's connected to Asterisk which give us a total of 4 lines
(using the bristuffed package). We would like to limit the number of
incoming calls to 2 calls and if a 3rd call comes in, we would like this
to go to another extension (voicemail or similar). Is this possible in
Asterisk?
Hello everyone,
Does anyone out there have actual experience with running * on a
mini-itx board from VIA? They look good, but I have some reserves
because of VIA's problems with PCI latency in recent years (audio
dropouts, wierd things happening). I am looking at the EPIA CL-1.
For
Hello,
Is it possible to initiate/receive calls from a url (that is without
having to install and configure a PC soft phone) using asterisk?
If yes, may I please get some sites, pointers, HOWTOs on how its done?
Thanks,
Julius.
___
Asterisk-Users
Guys.
Im having a big problem transfering incoming calls thru zap channels to some
other extension. If the call is made by me to the outside via zap channels,
no problem, hitting # gets me the transfer prompt, but if the call comes in
thru zap and eventhough I am sending the call from the zap
Julius Kidubuka wrote:
Is it possible to initiate/receive calls from a url (that is without
having to install and configure a PC soft phone) using asterisk?
If yes, may I please get some sites, pointers, HOWTOs on how its done?
I think you need asterisk call manager, that can initiate calls for
I just installed tapi and some app called identapop pro. I havent tested
incoming calls yet but so far, I cant get calls out using outlooks.
I configured TAPI for asterisk inside outlooks and I set TAPI to these
configs:
TAPI connects using the manager to asterisk without problems.
As channels
Check out the setgroup checkgroup commands on the wiki.
The wiki is located here:
www.voip-info.org
On Mon, 21 Mar 2005 09:16:59 +1200, James Doherty [EMAIL PROTECTED] wrote:
We have 2x BRI's connected to Asterisk which give us a total of 4 lines
(using the bristuffed package). We would like
Put the new bootrom.ld and bootrom.ver files on the config server (FTP
or TFTP) that your phones load from, and they will upgrade automatically.
Easy. Too easy. ;-) Seriously, though: I'd've never thought of that.
Thanks much!
-Ken
___
How to compile additional module to asterisk?
I have app_nv_backgrounddetect.c file and followed instructions below,
but make did not generate app_nv_backgrounddetect.so or
app_nv_backgrounddetect.o
(1) Drop the code in your /usr/src/asterisk/apps directory
(2) Edit the Makefile in the apps
OK, the outbound problem is fixed... Now, my other question is, anybody
using identapop for popup CID on your screen?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Domingo, 20 de Marzo de 2005 03:34 p.m.
To: 'Asterisk Users Mailing
Another solution would be the FWD web-based phone, where youd call a
FWD number, that is linked to Asterisk:
http://www.freeworlddialup.com/content/view/full/332/
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Kristof Hardy
Sent: Sonntag, 20.
I am having trouble with this.
I can dial 1800 numbers fine
as well as FWD service numbers but not Vonage.
I can be called from ipkall and fwd and can call aixtel numbers.
I use aix2 with Fwd.
My extensions.conf for Vonage:
; vonage numbers
;
; +2431
exten =
looks like an dtmf mode setting problem, make sure you have it set to
dtmfmode=rfc2833 or dtfmmode=info in sip.conf, the same goes for your
ata.
On Sun, 20 Mar 2005 15:29:18 -0600, Anton Krall
[EMAIL PROTECTED] wrote:
Guys.
Im having a big problem transfering incoming calls thru zap channels
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