Using Asterisk 18.12.0, a little confused on how to configure my pjsip.conf
file to determine the codec to use for a call
I have 2 endpoints:
[Alice]
disallow:all
allow:ulaw,alaw,g729
[Bob]
disallow:all
allow:ulaw,alaw,g729
Alice calls into Asterisk on ext 100 and then we dial Bob
I want to w
Asterisk receives a 180 Ringing with SDP from the called side, then it sends
180 without SDP to the calling side.
We would like asterisk to sends to the calling side the same response that was
received from the called side.
This is Asterisk cert 13.1, is that a new behavior, is there a setting t
In Version 1.8 asterisk introduced this parameter preferred_codec_only, when
set to "yes" the 200 OK to the INVITE contains 1 codec only from the available
ones in the user sip profile.
But in version 13.1 (I think version 11.2 also) is not working like that , it
keeps sending all the codecs an
thank you, we are using the same configuration files in 13, same setup, just
different asterisk version. we just dont see the msgs in the console/logs, it
is the same exact voice traffic on both asterisk versions
is that something that you set on/off? if that is the case how can it be done?
wha
Starting with Asterisk 13.1 we are seeing this WARNING
messages a lot in our logs and console:
WARNING[25164][C-0004865e]: chan_sip.c:7364 sip_write: Can't send 10 type
frames with SIP write)
We found that line in function "sip_write" inside "chan_sip.c".
In our previous
Hi Kevin, I created the issue on the https://issues.asterisk.org/jira web site,
posted the description of the prob and submitted asterisk console logs [sip and
udptl debug on] and a wireshark capture taken on the asterisk machine showing
both legs with signaling and media.
PLease let me know wh
Txs a lot Kevin.
I had just created and account on https://issues.asterisk.org/jira
Let me know if this is the right place to post both the pcap capture and the
sip logs. If not please help me out creating the account in the right place so
that I can provide all the information you need.
The sip
txs a lot for your explanation steve
so, it should work w/o spandsp fairly fine if we do not have a bad connection.
I see that this version has a lot of fixes related to t.38
but is the implementation already mature enough to guarantee a decent success
rate with fax calls?
From: fbo...@hot
will installing spandsp help with t.38 pass-through?
From: fbo...@hotmail.com
To: asterisk-users@lists.digium.com
Subject: RE: T.38 passthru on 1.8.5
Date: Tue, 30 Aug 2011 11:42:41 -0400
both endpoints use public Ips, I just changed the real ones for the privates
ones to protect our
both endpoints use public Ips, I just changed the real ones for the privates
ones to protect our ips but made a mistake and left the dest as a pub and the
orig as private, my bad.
but for the record, both are public IPs, there is no nat and iptables is off
also, I see that the quintum sends a
yes, same thing
From: fbo...@hotmail.com
To: fbo...@hotmail.com
Subject: RE: same sip peer as user and provider
Date: Tue, 30 Aug 2011 10:35:01 -0400
yes my friend. same thing
From: fbo...@hotmail.com
To: asterisk-users@lists.digium.com
Subject: same sip peer as user and provider
Hello
Up to version 1.6.0 we have been able to configure the same SIP device as a
user [inbound trunk] and as a peer [outbound trunk] w/o issues.
After we switched to version 1.8 this setup wont work, apparently one can not
have the same IP on 2 different trunks anymore. The trunk that is conf
Hello
We want to implement T.38 passthru with asterisk 1.8.5 [Asterisk 1.8.5.0 built
by root @ asterisk1-8.labdomain.com on a x86_64 running Linux on 2011-08-26
21:31:22 UTC]
The call flow is:
quintum gateway --> asterisk --> Dialogic IMG 1010
the call starts as a voice call, the remote fax
n you please try to increase the 'Timeout'? What happens when you
set it to this:
Timeout: 5000
Fabian Müller
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look at
http://www.voip-info.org/tiki-print.php?page=Asterisk+auto-dial+out
Fabian Müller
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<[EMAIL PROTECTED]> wrote:
> How do I download the development branch of asterisk 1.4. I am
> eagerly waiting for it.
svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk
See http://www.asterisk.org/download for further information.
F
sources of course.)
Fabian Müller
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or.org/rfc/rfc3428.txt). Have a look at the
comment of the function receive_message() in chan_sip.c.
Fabian Müller
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can help you.
Fabian Müller
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e you can configure in its administration interface
which dtmf mode the telefone should use. If your IP phone is
configured to use rfc2833 for example then you would write
dtmfmode=rfc2833 in your sip.conf. If all users use the same
dtmfmode it should be ok to write this to the general se
Hello
I want to to know if the motherboards VIA are fully supporte by asterisk.
And also, some of those motherboars say that with 1 pci slot , using a
special riser card you can connect 2 pci cards. Will that work to have 2 pci
cards (FXS or FXO ) on asterisk?
thank you
Fabian
onnhold
etc?
4- If I have a
SIP device behind a firewall the supports SIP transformations (sonicwall pro230)
and the * is outside the firewall, do I have to open ports 5060
anyway?
What about the
audio?
Regards
Fabian
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Paulo <[EMAIL PROTECTED]> writes:
> Hi all,
Hi Paulo,
> I was wondering if there is any documentation of the Asterisk C
> source code.
A good point to start is the file apps/app_skel.c which shows the
minumum you need in an application.
Regards,
i don't think there are channel driver's for dialogic cards yet...
On Thu, 23 Dec 2004 09:49:54 +0200, Tasos Daskalopoulos <[EMAIL PROTECTED]>
wrote:
> Hi there
>
> I have a Dialogic VFX/40 ESC plus installed on Redhat Linux 8.0
> and looking for Channel drivers for this Card.
> where cann i
les about nat (and other things as
well). Here is one of them:
http://lists.digium.com/pipermail/asterisk-users/2004-October/068275.html
You probably should read the whole thread. Click the link "thread" on
that page and search in the new page for postings with the the s
Every time I asterisk to retrieve voice mail, or dial to
the menu extension there is choppy sound coming out. When that happens asterisk
reports Received bad packet with bad udp checksum. I can have conversation to
other people just fine, but any voice exchange with asterisk is horrible. W
I understand asterisk invokes sendmail in order to send
email notifications of messages left. Is there another application less
complicated than Sendmail, I already got mail servers else where and they are
the ones I want to use.
Any light in this matter will be appreciated.
Hi,
Is there something you suggest to have the mail server working? Do I need to
setup sendmail?
Thanks.
Fabian
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Leif Madsen
Sent: Monday, October 18, 2004 3:17 PM
To: Asterisk Users Mailing List - Non
From: Fabian Garcia
[mailto:[EMAIL PROTECTED]
Sent: Friday, October 15, 2004
7:25 PM
To: [EMAIL PROTECTED]
Subject: How to make asterisk send
email notification of voicemessages?
Hi,
I’ve been trying to have Asterisk to email user each
time a voice message is left
mple) on the Asterisk box or does it work without a SIP proxy
as well? Do I have to register the Asterisk box on the softswitch?
(Should this be possible at all?)
Thanks very much in advance for any kind of help.
Regards,
Fabian Müller
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onf.
Wow, thanks a lot Oleg. I overlooked that :-(
Regards,
Fabian Müller
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is in the queue and sends a DTMF signal I see this
message:
DEBUG[327698]: chan_zap.c:3955 zt_read: DTMF digit: 5 on Zap/4-1
Regards,
Fabian Müller
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We've no problems with this...
Single E1 Line from Colt Telecom.
We use a quad p3 xeon * server with a TE405P
The only difference between the TE405P and the TE410P is the pci bus voltage...
Regards
Fabe
On Tue, 07 Sep 2004 18:41:40 +0200, Roger Schreiter
<[EMAIL PROTECTED]> wrote:
> Henrik Pflug
i think your problem is above the pasted error message.
i compiled chan_capi on fedora 2 just yesterday.
only problem was that some isdn4k devel pkg was missing.
install it and it'll probably work fine.
On Thu, 26 Aug 2004 13:32:06 +0200 (CEST), [EMAIL PROTECTED]
<[EMAIL PROTECTED]> wrote:
> Hello
zaptel.conf
span=1,0,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
loadzone=nl
zone=de doesen't work correctly for me :( but nl does...
zapata.conf
switchtype=euroisdn
pridialplan=unknown
signalling=pri_cpe
group = 1
channel => 1-15,17-31
context=incoming
this is the base config that works with colt
n => _X.,4,HangUp
exten => h,1,Macro(record-cleanup)
Try it ;) It's working for me.
Fabian Stelzer
Gigacodes GmbH
On Fri, 9 Jul 2004 13:54:15 -0500, Mark Johnston <[EMAIL PROTECTED]> wrote:
> I'm trying to record a Meetme conference to disk, but the Monitor application
> do
In Latest CVS HEAD MeetMe returns the variable MEETMESECS which is
the number of seconds the user was connected to the conference.
But you can also do some scripting of your own and can make it more specific
to you application.
(only billing with multiple users ... and so on)
- Original Messa
Title: Nachricht
It's
already in the bugtracker...
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ben
MerrillsSent: Tuesday, May 25, 2004 10:26 AMTo:
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Meetme
Options (new one)
Seems
ff multi processing in the hardware.
But that's not a real solution and wasn't possible as well on this
mainboard.
Regards
Fabian Stelzer
Gigacodes GmbH
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Steve
Underwood
Gesendet: Mittwoch, 31.
it's not very accurate.
I'm not aware of any other technique to discover an answering machine /
voicemail...
Regards
Fabian Stelzer
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Serge
Mankovski
Sent: Tuesday, March 16, 2004 5:45 PM
To: [EMAIL
m you can buy a lot more of
processing power.
and even more servers! So perhaps it would be better to distribute your app
to several * servers and you still would be a lot cheaper than using
dialogics hardware...
mho.
Regards
Fabian
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