To close the loop on this I have found that this appears to no longer be
an issue since I moved to 1.6rc6.
Mark Michelson wrote:
MFH wrote:
I'm noticing in 1.6 Beta 9 that on outgoing calls I get a brief audio
drop when the audio starts on the other end of the call. So basically I
The best way I can think of is:
wget http://ftp.digium.com/pub/asterisk/asterisk-1.4.21.2.tar.gz
tar -zxvf asterisk-1.4.21.2.tar.gz
cd asterisk-1.4.21.2
./configure
make menuselect (You don't have to select anything)
make
make install
make samples
Pascal Bruno wrote:
I am about to
for the asterisk addonds and sounds? Can you
provide me the commands to get, build and install for the 1.4.21
version? Thanks a lot guys.
On Fri, Sep 12, 2008 at 6:07 AM, MFH [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
The best way I can think of is:
wget http
As a result of:
http://lists.digium.com/pipermail/svn-commits/2008-July/036122.html
I am seeing these messages when upgrading for 1.6b9 to 1.6rc4. Is there
something I should be doing to address this warning?
[Sep 4 13:31:34] WARNING[2954]: chan_iax2.c:1200 __send_lagrq: I was
supposed to
12:59:33 MFH wrote:
As a result of:
http://lists.digium.com/pipermail/svn-commits/2008-July/036122.html
I am seeing these messages when upgrading for 1.6b9 to 1.6rc4. Is there
something I should be doing to address this warning?
[Sep 4 13:31:34] WARNING[2954]: chan_iax2.c:1200
Depending on how smart your router configuration is, you can leave both
boxes A and B on port 4569 and then just set up two rules on your router:
Port 4569 - Box A port 4569
Port 4570 - Box B port 4569
If your router is not complex enough to allow a port mapping such as the
second line above
Asterisk supports conferencing without using meetme. In this case you
don't have a central dial in number but a single extension can initiate
the conference call. Generally this is done the same way as with
traditional PSTN service which is that while on a call between two
parties, flash the
Can anyone recommend decent quality as close to pay-as-you-go SIP
wholesale termination providers in both Singapore and Sydney,
Australia? I will be in both places and want a local carrier while I'm
there. It needs to be easy in and easy out and if it's not $0 base or
close I'll need to be
Dean asked for it so he can decide if it's worth it to him but that
sounds like the price someone would pay for flatrate and probably not
what one would want to pay for 5 calls per day.
MARK.
Sam Tam wrote:
We have got that for $10 USD setup and $25 USD per month
If you are interested please
I'm noticing in 1.6 Beta 9 that on outgoing calls I get a brief audio
drop when the audio starts on the other end of the call. So basically I
hear the first word, miss the second word and then hear the rest fine.
I've noticed this after calling multiple locations and getting some
recording
It depends on which type of SIP device you have that determines on how
you signal a distinctive ring. You need to change the SIP Header like:
exten = s,n,SIPAddHeader(Alert-Info:Bellcore-r8)
where the number after the 'r' signifies a different ring tone but some
devices uses different names
My internal calls start in an entirely different context than calls
coming in externally. There's never any confusion about where the call
is coming from and I don't use prefixes.
Allann Jones wrote:
Internal and external calls can be distinguished generally by the
phone number. A prefix or
I was curious so I took a look at my sounds directory. Most of the
files are 644 except the g729 which are 444. I also noticed that the
ownerid/groupid are a non-existent 1000/1000. I take it that the sound
installer uses something like tar with the option to keep the original
owner and
Very cool, you've piqued my interest. Since I haven't launched an
instance before, where's the best place to learn to do that? What's the
approximate monthly cost of hosting an Asterisk PBX on EC2?
Ronald Lewis wrote:
I've just added a PREVIEW release of my upcoming how-to guide for
This is what I use. The Read does have a default timeout but you should
be able to put your own.
extensions.conf:
exten =
s,n(dial),Dial(SIP/sipura2_1SIP/sipura1_1SIP/sipura2_2SIP/spa942_3SIP/aastra480_3,20,mtTM(screen))
exten = s,n(vmail),Voicemail([EMAIL PROTECTED])
[macro-screen]
From what I can tell Read allows for a floating point input which uses
ast_waitfordigit that accepts milliseconds as input.
Douglas Garstang wrote:
Admittedly I have not used the ExternalIVR app. Is it any good?
I'm not sure I agree that Asterisk is GOOD for building IVR's. Sure,
it can do
Are asterisk and the phone on the same lan? I see you have nat=no. Do
you see the phone adapter registered?
Emmanuel Favre-Nicolin wrote:
Hi,
I'm having a problem to receive inbound call from my sip provider. I used to
be OK, I may I have change something (for example I switched from
:12:44
callcentric.com:5080177729x 46 Registered
Wed, 09 Jul 2008 10:13:29
I use line2 of my pap2t (line 1 is not enabled). Here is the conf :
http://emmanuelfavrenicolin.free.fr/Public/Divers/Snapshots1/20080709_pap2t.jpg
On 7/9/08, MFH [EMAIL PROTECTED] wrote
It looks like it's 19:
http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause
Nhadie wrote:
Hi,
How do i send proper message when hanging up?
[from-trunk]
exten = _1234,1,Dial(SIP/${EXTEN}|30|t)
exten = _1234,n,Hangup
With that, the other end receives a call reject if
You didn't give details of your networking setup but do you have the
3102 and then X-Lite client connected to the same switch or router? It
not, one switch could be dropping packets or slow. Do you qualify both
devices in Asterisk? Do they have the same ping times?
I haven't done any audio
I've been having a problem with Asterisk MWI notification on my SIP
phones since going to version 1.4 a long time ago. Since going to this
version, I have needed to go into chan_sip.c and do the following:
/*! \brief Check whether peer needs a new MWI notification check */
static int
21 matches
Mail list logo
