Thanks for your repsonse, perhaps I mis-stated my situation. I have
asterisk up and running with a TDM22B and have two analog phones working
with two analog phone lines. What I can't seem to get started on is the
setup of a SIP phone. I have looked at all the info on voip-info.org
and
On Fri, 2005-06-10 at 22:50 -0700, infra struct wrote:
Download asteriskathome-1.0.iso
This is a CD image that if burnt to a blank CD rom (do not copy the
file you have to use nero or cdrecord or something that way)
and
Download asteriskathome-1.0-md5sum.txt
please anyone
We are developing an IVR application and when I am testing locally on
my machine using a softphone (iaxcomm) the digits I press for GET DATA
work every time. I am testing with a local extension that goes right
into my routine. However when I try to call in to the system using an
analog or
On 6/11/05, Rich Adamson [EMAIL PROTECTED] wrote:
I am wondering if anyone else is experiencing similar issues. I
believe the problem lies with VoicePulse because we are using them for
IAX connections. I don't believe its a bandwidth problem on my
network (cable) because I have tried
I am wondering if anyone else is experiencing similar issues. I
believe the problem lies with VoicePulse because we are using them for
IAX connections. I don't believe its a bandwidth problem on my
network (cable) because I have tried the same exact system/config
everything on
Assalam Alaikum
This is my sip.conf i m using softphones without any problem .. but i m unable to register my netphone IP phone with asterisk plz help a newbie here..
[general]
port=5060bindaddr=0.0.0.0tos=lowdelaydisallow=allallow=ulawcontext=default;trying to register with user id
I have 4 Polycom phones here, two 500s and two 300s. The 500 is a top-shelf
phone with quite a few asterisk-friendly features. I absolutely love the
speakerphone: it has superb tone quality, and its truly full-duplex. The
caller on the speaker does NOT hear him/herself back through the
Hi-
How to configure Asterisk as sip proxy.
Best Regards
Ibrar Ahmed
Project Manager.
Comcept (Pvt) Ltd. Islamabad Pakistan
www.com-cept.com
[EMAIL PROTECTED]
[EMAIL PROTECTED]
Ph # (Off) +92-51-111784784
Ph # (Res) +92-51-2271283
Ph # (Mob) +92-3009543001
Fax # 92-51-111784785
On Friday 10 Jun 2005 22:46, list wrote:
RFC 1912
Every Internet-reachable host should have a name. and then For every IP
address, there should be a matching PTR record in the in-addr.arpa
domain. and Failure to have matching PTR and A records can cause loss
of Internet services similar to
On 6/11/05, Rich Adamson [EMAIL PROTECTED] wrote:
That's entirely possible. Had something similar with livevoip.com (with
the answered iax call issue).
You should be able to determine whether its a voicepulse issue by either
doing a iax debug (look for the dtmf digits), or, using ethereal
altus wrote:
We installed a box a long time ago and they bought g729a licenses
Now we want to upgrade and reinstall,whats going to happen with the
codec,if I give the box the same ip as always will it work?
The Digium g729 license is bonded to the MAC address of all the
interfaces you have.
I seriously doubt that sf.net has any DB access, so its only suitable if
the wiki is flat files or to temp host the cached pages until something
more perm can be done.
sf.net has mysql running.
Just send a mail when you registered a project and they will
give you a servername/user/pass :)
--
-Original Message-
Provider is doing well and giving good service.
Word of mouth increases userbase and service load fo
provider. Provider wants the money obviously and takes on
load in spite of limited resources. Provider becomes
overloaded and is not longer able to provide
Does the manager API have the option of showing timestamps of events?
I am trying to log events into a database and I need timestamps of when the
events actually occurred.
Is the time lag between events occurring and receiving them in the manager api
very low? I suppose it if is I could
When I check the received email, my user name does not appear on the From list.
All it says is To: asterisk-users@lists.digium.com.
Is there something configured wrongly in my mail client, or is it coming from
the mailing list configuration
On Sat, 2005-06-11 at 11:25 +0200, Michiel van Baak wrote:
I seriously doubt that sf.net has any DB access, so its only suitable if
the wiki is flat files or to temp host the cached pages until something
more perm can be done.
sf.net has mysql running.
Just send a mail when you
On Sat, 2005-06-11 at 10:02 +, Obelix wrote:
When I check the received email, my user name does not appear on the From
list.
All it says is To: asterisk-users@lists.digium.com.
Is there something configured wrongly in my mail client, or is it coming from
the mailing list configuration
Jumping in very late to this thread...
Is the solution not to change the voicemail system to enable it to utilise
other entities as the store, e.g. a pop3 server or an imap server rather than
just flat files on disk (which should remain an option).
That way it doesn't matter where they listen
etc. However, I missed the initial part of this thread. Why is this
an
issue? I went to voip-info.org today just to see what was going on
(while writing a googleapi tool to pull all cached docs from a given
domain) and it was running and appears to be there. Nothing on their
main page or
With call manager V4 and above it's extremely easy, just connect a SIP trunk to
*.
BTW Unity is the Cisco voicemail system, Call Manager (CCM) is the actual PBX
so your terminology may be confusing some people.
From: [EMAIL PROTECTED] on behalf of Simone
etc. However, I missed the initial part of this thread. Why is this an
issue? I went to voip-info.org today just to see what was going on
It is not really an issue at all. The thread started due to scheduled
maintenance of the server, which scared a lot of Asterisk users. The
wiki is safely
These questions are probably better sent
to the [EMAIL PROTECTED] sourceforge forum, but I would have answered it over there
as well.
The iso is a type of cd burn (if you use
Nero or Ulead read the instructions there).
You dont need to install Centos
first, it is installed
On Sat, 2005-06-11 at 20:11 +1000, Rob Thomas wrote:
etc. However, I missed the initial part of this thread. Why is this
an
issue? I went to voip-info.org today just to see what was going on
(while writing a googleapi tool to pull all cached docs from a given
domain) and it was running
I had problems and given up with a x100p clone ebay card.
On the asterisk side it was amplifying everything said so loud back
into my ear that it was so uncomfortable it cannot be used.
(sounds something like phones did before a duplex coupler)
not a fix sorry ;p
im quite the asterisk newb too,
Michael,
try relaxdtmf=yes in your iax.conf, or if you are using sip, then in
sip.conf
regards,
Umair bari
Michael Stearne wrote:
On 6/11/05, Rich Adamson [EMAIL PROTECTED] wrote:
That's entirely possible. Had something similar with livevoip.com (with
the answered iax call issue).
What platform should you suggest to use asterisk?
I tried with SUSE now all the time but there are too many
problems with the updates.
On is the development platform on which * is developed ?
Regards,
___
Asterisk-Users mailing
On 13:22, Sat 11 Jun 05, Serge Schumacher wrote:
What platform should you suggest to use asterisk ?
I tried with SUSE now all the time but there are too many problems with
the updates.
On is the development platform on which * is developed ?
Regards,
I love the way the Debian updates
On Sat, Jun 11, 2005 at 02:03:59PM +0200, Michiel van Baak wrote:
On 13:22, Sat 11 Jun 05, Serge Schumacher wrote:
What platform should you suggest to use asterisk ?
I love the way the Debian updates work.
Me too, but has the installation improved with the latest Sarge release?
The
On 08:19, Sat 11 Jun 05, Mike M wrote:
On Sat, Jun 11, 2005 at 02:03:59PM +0200, Michiel van Baak wrote:
On 13:22, Sat 11 Jun 05, Serge Schumacher wrote:
What platform should you suggest to use asterisk ?
I love the way the Debian updates work.
Me too, but has the installation
On Sat, 2005-06-11 at 14:03 +0200, Michiel van Baak wrote:
On 13:22, Sat 11 Jun 05, Serge Schumacher wrote:
What platform should you suggest to use asterisk ?
I tried with SUSE now all the time but there are too many problems with
the updates.
On is the development platform on which
should != must - it is not illegal.
True. However, RFC's are in place to make sure we all play by the same
rules. If we all play by the same rules things on the internet tend to
work as expected. I like things to work as expected, don't you?
The reason most people (myself included) block mail
Mike M wrote:
On Sat, Jun 11, 2005 at 02:03:59PM +0200, Michiel van Baak wrote:
On 13:22, Sat 11 Jun 05, Serge Schumacher wrote:
What platform should you suggest to use asterisk ?
I love the way the Debian updates work.
Me too, but has the installation improved with the
Darren Wiebe wrote:
Replies are inline.
Thanks! I am sure we will solve it ;-)
Below is the source code of the web page of astcc-admin.cgi
bodytable align=center width=100%
trtdimg src=/_astcc/astcc.png/tdtd align=centerfont face=verdana,helvetica size=5Asterisktrade; Calling Card Admin:
I've tried almost
any softphone available on the market with many different PC, soundcard,
headphones combinations.
None of them prooved
production reliable in a call center environment.
I've also tested
many IP Phones and Gateways. Even the cheapest one supplies much better quality.
Is
Blocking from unknown domains fine, blocking from dynamic ip's that's
just plain bullshit.
This topic has been done to death, move along nothing to see.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Tracy Phillips
Sent: Saturday,
It just doesn't make sence to charge for 800 termination... as the
person you are CALLING pays for the call.If you are strickly VoIP
based then I dunno what to tell you. We have local PRIs that we route
calls across, so we use those for 800 termination... (why pay for it?)
IF you were only
We're getting close to Astricon Europe 2005, the first Asterisk
Community gathering in Europe. Speakers are coming in from all over the
US and Europe, as well as far away as New Zealand, to talk, teach and
discuss Asterisk -the Open Source PBX.
At this time, we're still accepting registrations
I don't believe relaxdtmf is a valid parameter for iax.conf; just sip.conf.
(per the most recent sample configs)
Michael,
try relaxdtmf=yes in your iax.conf, or if you are using sip, then in
sip.conf
regards,
Umair bari
Michael Stearne wrote:
On 6/11/05,
On Sat, Jun 11, 2005 at 02:30:31PM +0200, Michiel van Baak wrote:
I will have a look at it later this week since my
workstation is now replaced by a laptop so I have some
testing hardware :)
You're running from an upgraded Slink? That's the beauty of Debian.
You may need to use Sid if you
I have sent you a copy of my version of astcc-admin.cgi privately.
There are a few things I wanted to point out.
Ronald Wiplinger wrote:
Darren Wiebe wrote:
Replies are inline.
Thanks! I am sure we will solve it ;-)
Below is the source code of the web page of astcc-admin.cgi
bodytable
Darren Wiebe wrote:
Replies are inline.
Ronald Wiplinger wrote:
Thanks for your config file! Adopting it to my settings let me update
the database!!!
I can now list all my cards, ...
Now I got a new problem ;-)
If I call from a phone that is setup to use the ASTCC system via
context,
No path to translate from SIP/615-25c8(256) to SIP/601-27b6(4)
Jun 11 22:24:30 WARNING[17723]: app_dial.c:1324 dial_exec_full: Had to
drop call because I couldn't make SIP/615-25c8 compatible with SIP/601-27b6
???
bye
Ronald
___
Asterisk-Users
On Saturday 11 June 2005 10:36, Mike M wrote:
You're running from an upgraded Slink? That's the beauty of Debian.
What distro *doesn't* let you do this? I've been doing it this way with
Slackware since the 3.x versions for chrissakes.
-A.
___
On Saturday 11 June 2005 09:56, Tracy Phillips wrote:
True. However, RFC's are in place to make sure we all play by the same
rules. If we all play by the same rules things on the internet tend to
work as expected. I like things to work as expected, don't you?
That is *precisely* why the RFC is
Or maybe a couple of us should just get together and start
our own company. One that explicitly places quality above
quantity. Anyone remember when businesses operated this way?!
This is not a bad idea at all -- and something that's been discussed in
off-list emails. I think it's
In the dial
application when configuring the Limit parameter:
L(x[:y][:z]): Limit the call to 'x' ms, warning
when 'y' ms are left, repeated every 'z' ms)
I want to read 'z'
from database, based on the dialed number.
How is this
possible?
Cenk.
On 6/11/05, Rich Adamson [EMAIL PROTECTED] wrote:
I don't believe relaxdtmf is a valid parameter for iax.conf; just sip.conf.
(per the most recent sample configs)
I didn't find it either. I put it in the config anyway but it didn't
seem to make a difference. I also tried changing the call
I am just glad everyone doesn't have that attitude about RFCs.
--Tracy
On 6/11/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Saturday 11 June 2005 09:56, Tracy Phillips wrote:
True. However, RFC's are in place to make sure we all play by the same
rules. If we all play by the same rules
One side is using G729, the other is using ULAW. Asterisk is having to
convert between the two and can not, probably because you do not have the
G729 codec with the proper license ($10/channel from Digium).
- Joshua Colp.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
If a user has created an unvailable message in Comedian mail is there
anyway to delete that message? I know you can record a new message,
but I would like to delete the file as if the user never recorded one.
Thanks,
Michael
___
Asterisk-Users mailing
Hey guys,
I would like to do some very basic Caller ID transforms on incoming
PSTN calls, traversing via SIP on my Cisco 1760V router to *. What is
the best place to do them, and could you specify an example? I've
browsed the Wiki quite a bit, and I know how to act on certain calls -
but I
I have had several issues flashing between SCCP/SIP/MGCP on those phones
where it will eventually cause the handset to bleed through the
speakerphone. Once that happens, the phone is basically trash - it
never stops...
-Greg
I'm having a problem with one of our 7960. They all run latest
On Saturday 11 June 2005 11:35, Tracy Phillips wrote:
That is *precisely* why the RFC is worded should -- it is optional. If
the RFC said must then it is required. RFCs are worded very carefully
as a general rule.
I am just glad everyone doesn't have that attitude about RFCs.
I'm not
William Waites [EMAIL PROTECTED] writes:
So this is a version of Asterisk that is released by Digium but
is not released under the GPL. Correct?
Yes, because digium has a dual license, you have to give up your
copyright if you submit code to the project. This makes it possible to
release a non
i think the time between sent event from Asterisk and catch the event
with some other application is not important for most applications, so
you may save the timestamp from your own application.
And of course you have other option, modify the function:
int manager_event(int category, char
I don't believe relaxdtmf is a valid parameter for iax.conf; just sip.conf.
(per the most recent sample configs)
I didn't find it either. I put it in the config anyway but it didn't
seem to make a difference. I also tried changing the call codec to
ulaw but that had no significant
Andrew Kohlsmith [EMAIL PROTECTED] writes:
I don't know, I've got no problem with them dual-licensing it.
It means the project will receive less contribution from free software
developers. I certainly would not give up my copyright on free
software so that someone else could release it as non
Joshua Colp wrote:
One side is using G729, the other is using ULAW. Asterisk is having to
convert between the two and can not, probably because you do not have the
G729 codec with the proper license ($10/channel from Digium).
You are right! I removed g729, but I still wonder, why it did
Matthew T. O'Connor matthew@zeut.net writes:
I have looked at all the info on voip-info.org
It would be nice if this was a public wiki, meaning requiring no
registration to edit. I think we would get more activity there, then.
--
Esben Stien is [EMAIL PROTECTED] s a
Codecs are negotiated between asterisk and the device, not device to
device... So since you specify G729, one side negotiated to G729 first...
Then when you dialed the other device, that one negotiated at ULAW... And
then when they attempted to be bridged together - voila, failure.
- Joshua
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Andrew Kohlsmith
Sent: Saturday, June 11, 2005 11:58 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] ATTN: Keith
On Saturday 11 June 2005 11:35, Tracy Phillips wrote:
That
On Saturday 11 June 2005 12:12, Esben Stien wrote:
It means the project will receive less contribution from free software
developers. I certainly would not give up my copyright on free
software so that someone else could release it as non free software.
Only to those who agree with your views.
One side is using G729, the other is using ULAW. Asterisk is having to
convert between the two and can not, probably because you do not have the
G729 codec with the proper license ($10/channel from Digium).
You are right! I removed g729, but I still wonder, why it did not go to
the
Hello,
Maybe I'm missing something here. What is the proper way to use RTC
with ztdummy now?
I'm using -HEAD from a day or two ago on Linux 2.6.11.11.
In zaptel/Makefile, I changed CFLAGS to:
CFLAGS+=-I. -O4 -g -Wall -DBUILDING_TONEZONE -DUSE_RTC
#-DTONEZONE_DRIVER
and I get..
make -C
On Sat, 2005-06-11 at 12:44 -0400, Andrew Kohlsmith wrote:
On Saturday 11 June 2005 12:12, Esben Stien wrote:
It means the project will receive less contribution from free software
developers. I certainly would not give up my copyright on free
software so that someone else could release it
On 6/11/05, Rich Adamson [EMAIL PROTECTED] wrote:
I don't believe relaxdtmf is a valid parameter for iax.conf; just
sip.conf.
(per the most recent sample configs)
I didn't find it either. I put it in the config anyway but it didn't
seem to make a difference. I also tried changing
Hello All
I'm settup my asterisk as belows:
sangoma card, connected with E1, CAS Signalling.
I have two problem.
1. The asterisk don't received any DTMF when caller input to
2. when i dial to system, the caller hear bad sounds. monitor on console. asterisk show error.
Jun 11 12:15:45
Joshua Colp wrote:
Codecs are negotiated between asterisk and the device, not device to
device... So since you specify G729, one side negotiated to G729 first...
Then when you dialed the other device, that one negotiated at ULAW... And
then when they attempted to be bridged together - voila,
Another alternative is to get another connection in addition to DSL for
example Cable Connection.
That is what we have, our main connection is DSL and we have a backup
Cable connection, if one connection goes down you switch to another.
It had happened to us in a past DSL went down, 10min. and we
Hi all.
Could someone point me an example to use SIP_HEADER function!? I want
to read the To: and send this INVITE to an internal extension.
Tks.
Denis Galvão
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
That made no sense to me. Please try again. If you mean why did it not go to
the next line when it tried to bridge it's because you can't switch codecs
in the middle of a call.
- Joshua Colp.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronald
In our experience, the total cost of softphones(money, reduced sound quality
and lower reliability) in a large call center environment is actually
greater over time than the cost of a channelbank and cheap analog
headphones. We've tried 2 softphones, 2 kinds of SIP VOIP hardphones, 2
kinds of SIP
Hi,
I've an Asterisk box acting as firewall with
Shorewall, yet I can't get a SIP client (Sipura 2000)
to connect remotely (behind a firewall). My Shorewall
Config as follows:
interfaces
#ZONE INTERFACE BROADCAST OPTIONS
net eth0 detect
dhcp,routefilter,norfc1918,tcpflags
loc eth1
I don't believe relaxdtmf is a valid parameter for iax.conf; just
sip.conf.
(per the most recent sample configs)
I didn't find it either. I put it in the config anyway but it didn't
seem to make a difference. I also tried changing the call codec to
ulaw but that had no
Hi,
There is one asterisk server, and there are several locations. On each
location there are 100 (SIP) extensions. The idea is to set up a limit
of 10 concurrent calls for each location (because of bandwidth issues on
each location). How can I do that?
Thanks!
I think you're looking for RFC 2119
http://www.ietf.org/rfc/rfc2119.txt
-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
I'm not sure I understand -- I'm not making this up, RFCs use
must and
should very carefully. The latter is a guideline, and the
former
Codecs are negotiated between asterisk and the device, not device to
device... So since you specify G729, one side negotiated to G729 first...
Then when you dialed the other device, that one negotiated at ULAW... And
then when they attempted to be bridged together - voila, failure.
On 6/11/05, Juan Pablo Abuyeres [EMAIL PROTECTED] wrote:
Hi,
There is one asterisk server, and there are several locations. On each
location there are 100 (SIP) extensions. The idea is to set up a limit
of 10 concurrent calls for each location (because of bandwidth issues on
each location).
make -C /lib/modules/2.6.11.11/build SUBDIRS=/usr/src/zaptel modules
make[1]: Entering directory `/usr/src/linux-2.6.11.11'
Building modules, stage 2.
MODPOST
*** Warning: rtc_unregister [/usr/src/zaptel/ztdummy.ko] undefined!
*** Warning: rtc_control [/usr/src/zaptel/ztdummy.ko]
All I am saying it that it won't work if the user is using POP3. I don't
think it is at all possible to overcome this. And as I said before this is
not the use case we are talking about.
The solution simply does not work for users retrieving e-mail via POP3 and
I don't see a way that it would.
On Saturday 11 Jun 2005 14:56, Tracy Phillips wrote:
[...]
I wonder if there is an RFC from top posting? I doubt it... seems the
rest of the world can get along fine reading top posts...
rfc1855 details the netiquette guidelines.
From paragraph 3.1.1
If you are sending a reply to a message
trixter http://www.0xdecafbad.com wrote:
Further his point seems to be anti BSD license. If I write software and
give it away free what difference does it make to me if someone sells
it. They still have to find someone who is willing to pay for it when
they could get it from me for free.
that looks pretty much like it... thanks!
Brian Roy wrote:
On 6/11/05, Juan Pablo Abuyeres [EMAIL PROTECTED] wrote:
Hi,
There is one asterisk server, and there are several locations. On each
location there are 100 (SIP) extensions. The idea is to set up a limit
of 10 concurrent
On Sat, 2005-06-11 at 15:09 -0400, Aidan Van Dyk wrote:
Most people haven't had a problem with that, because, in the past, Digium
has been a benevolent keeper-of-the-code, not a direct competitor to the
contributors. But that Digium is directly competing with what others are
trying to
Curious as to why there is any problem in general, I went to google and
started hunting the license information. I found a couple of resources
they all say basically the same thing, all are on digiums site.
I cant understand why there is any sort of problem. There are 2
licenses they sell, one
Hello There,
I *think* i've setuped the AreskiCC2 Calling Card system right , but
i've yet to make any calls out of it , i added a rate card , trunk
and defined some rates , generated some users , added 10 dollars in
them , okay , now i call any number , it asks me to enter my pin , i
do , it
Digium is taking a some more equal than others sort of approach to
Asterisk. They figure that since they developed the base code, they
deserve a privileged position in the food chain, where they can do
things with the code that others can't. That is absolutely their right,
but I've never liked
On Sat, 2005-06-11 at 13:10 -0700, trixter http://www.0xdecafbad.com
wrote:
Look at 'big evil corporations' like apple. They did in a year with
mach what the FSF/GNU wants to do with HURD and still cant (to quote
stallman 'its really hard' while explaining why after 10 years HURD
still doesnt
In article [EMAIL PROTECTED],
Kevin Bockman [EMAIL PROTECTED] wrote:
make -C /lib/modules/2.6.11.11/build SUBDIRS=/usr/src/zaptel modules
make[1]: Entering directory `/usr/src/linux-2.6.11.11'
Building modules, stage 2.
MODPOST
*** Warning: rtc_unregister [/usr/src/zaptel/ztdummy.ko]
Also, I do not have RTC support in the kernel since the headers are
included from ztdummy, I thought that Tony said that it is not
required. Do I need RTC support compiled into the kernel?
I was going to reply to your first message, but then I thought I'd see
if you'd figured it out yourself.
Greetings,
I have one PSTN line connected to
my Asterisk@ Home box with call waiting. I also have an SPA-2002
connected to an analog phone. When I am calling on the PSTN and a call
waiting beep comes through, I can hear it, but when I press the flash key,
nothing happens. It is as if the
I am curious to what your loading was/is with 100 extensions. How many
concurrent calls should be planned - in an extensions to line ratio? I
had heard that 10 to 1 was a pretty good metric. Thoughts?
-Steve
There is one asterisk server, and there are several locations. On each
My VOIP carrier is using G723.1 Codec, so I have set my SIP softphone to G723.1, but I have also set up a Prepaid Calling Card application, which requires a number of sound files to be played. Due to licensing issues sound files on GSMcan not be played because the SIP softphones are on G723.1
Greetings to the list:
this is my problen when I make a call from my asterisk towards a nortel
PBX , the call is made but in my telephone sip I do not listen the dial tone
or the busy tone but the call it is completed normally.
just a small sidenote: digium does not sell ss7 licenses, thats someone
else doing that.
trixter http://www.0xdecafbad.com wrote:
On Sat, 2005-06-11 at 15:09 -0400, Aidan Van Dyk wrote:
Most people haven't had a problem with that, because, in the past, Digium
has been a benevolent
I just signed up and configured a SIP connection from BroadVoice. It
works great. This issue I have is that it seems after a couple calls
(or a certain amount of time) Asterisk doesn't seem to be receiving
these calls anymore. It seems as if BroadVoice is not redirecting the
call to my
Quoting Obelix [EMAIL PROTECTED]:
Is this question too difficult, or is it simply one that only a few users have
experienced?
My CDR is displaying wildly inaccurate results.
When I make a call the CDR records the time between connecting into the
server and hanging up, instead of recording
in one of the two defines configs (where you set the database up)
(sorry cant recall which one and im out of the office) there is a min
call value, its set by default around the 10 unit mark. if the cards
credit is below this it stops you going any further. I can only assume
this was to end the
Hi Tony,
You do need RTC support in the Kernel, because it is the hooks in the
rtc.c driver that the new ztdummy requires.
That's what I thought. That was going to be my next step but I hate
messing with the kernel remotely. I just made it as a module like you
did and it worked. Thanks.
I'm
I have two asterisk servers connected using IAX. Server A has a
TE410P running on a Xeon 2.4Ghz with 2GB RAM and 36G IDE HD on Debian
2.6.11-1-686 and Asterisk CVS-Nv1-0-7-06/01/05-01:27:25.
Server B does not have any Digium board, but has ztdummy and zaptel
loaded. It's runnin on a P4
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