On Thursday, August 14, 2014 03:15:16 AM Paul Greenberg wrote:
> Hi Anthony,
>
> That script does not work. My guess is that it is related to the way
> asterisk interacts with CentOS environment.
>
> Best Regards,
> Paul Greenberg, Esq.
>
> On Wednesday, August 13, 2014 12:11:42 PM Carlos Chavez
Hi Anthony,
That script does not work. My guess is that it is related to the way asterisk
interacts with CentOS environment.
Best Regards,
Paul Greenberg, Esq.
Law Office of Paul Greenberg
530 Main Street, Suite 102
Fort Lee, NJ 07024
Tel: 201-402-6777
Fax: 201-301-8876
Web: http://www.greenb
The Asterisk Development Team has announced the releases of:
DAHDI-Linux-v2.10.0
DAHDI-Tools-v2.10.0
dahdi-linux-complete-2.10.0+2.10.0
This release is available for immediate download at:
http://downloads.asterisk.org/pub/telephony/dahdi-linux
http://downloads.asterisk.org/pub/telephony/dahdi-too
On Wed, Aug 13, 2014 at 9:39 AM, Matthew Jordan wrote:
> On Mon, Aug 11, 2014 at 10:46 AM, Farid Fadaie
> wrote:
> > Hello,
> >
> > Full disclosure: my name is Farid Fadaie and I'm in charge of BitTorrent
> > Bleep (a private P2P SIP-based messaging application in early alpha)
> >
> http://blog.
On Wednesday, August 13, 2014 12:11:42 PM Carlos Chavez wrote:
> I installed CentOS 7 on a spare server along with all our Asterisk
> configuration system and the only thing that failed is the asterisk
> startup script included in the asterisk tarball. I guess because the
> startup system
Aaa now I understood better, thanks!
That's the instruction I used originally to write my Kamailio config, but I
wasn't sure on how the sdp was supposed to be altered at which places in
the whole SIP flow. I was thinking the original INVITE with the original
sdp would go all the way to the receivi
On Wed, Aug 13, 2014 at 9:29 AM, Olivier wrote:
> 2014-08-13 15:38 GMT+02:00 Mikael Fredin :
>> On 12 August 2014 16:19, Olivier wrote:
>>>
>>> How can I read RTP ports from CLI (to double check what could be
>>> included in rtp.conf file) ?
>>> "sip show settings" do not provide the answer.
>>
>
On 8/13/14, 11:31 AM, Matthew Jordan wrote:
On Wed, Aug 13, 2014 at 3:10 AM, Ishfaq Malik wrote:
Hi
Is anyone using asterisk on CentOS 7?
If so, is it working fine and as expected?
Random data point: the Asterisk project's build agents are still on CentOS 6.
Your mileage may vary.
I i
On Wed, Aug 13, 2014 at 4:35 AM, Olli Heiskanen
wrote:
> Hi,
>
> Wow, thanks Paul, realizing the problem makes a lot of sense.
>
> So I setup Kamailio as a peer, but if I disable chan_sip module completely,
> I can't do it in sip.conf like I'd otherwise assume to do. I tried to
> rebuild Asterisk
On Wed, Aug 13, 2014 at 12:01 PM, Olivier wrote:
> After installing various packages, here is what I did:
>
> TDIR=/usr/src
> cd $TDIR
> PJOPTIONS="--prefix=/usr --enable-shared --disable-sound
> --disable-resample --disable-video --disable-opencore-amr"
> git clone https://github.com/asterisk/pjp
After installing various packages, here is what I did:
TDIR=/usr/src
cd $TDIR
PJOPTIONS="--prefix=/usr --enable-shared --disable-sound
--disable-resample --disable-video --disable-opencore-amr"
git clone https://github.com/asterisk/pjproject pjproject
cd pjproject/
./configure ${PJOPTIONS}
make de
Hi Matthew,
I am using it. Works like a charm!
Running it for 3 week already and have no issues. However, my system is not
heavily utilized, i.e. 50-150 phone calls a day.
The only thing is I was not able to get asterisk integrated with CentOS
services daemon. So, I am starting asterisk manual
On Mon, Aug 11, 2014 at 10:46 AM, Farid Fadaie wrote:
> Hello,
>
> Full disclosure: my name is Farid Fadaie and I'm in charge of BitTorrent
> Bleep (a private P2P SIP-based messaging application in early alpha)
> http://blog.bittorrent.com/2014/07/30/building-an-engine-for-decentralized-communicat
Can we have concurrent calls via asterisk manager interface, lets say
around 1000 or 1000+ concurrent calls.
Regards
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live i
On Wed, Aug 13, 2014 at 3:10 AM, Ishfaq Malik wrote:
> Hi
>
> Is anyone using asterisk on CentOS 7?
>
> If so, is it working fine and as expected?
>
Random data point: the Asterisk project's build agents are still on CentOS 6.
Your mileage may vary.
--
Matthew Jordan
Digium, Inc. | Engineering
You would probably have better results from using a specific frequency tone
(or dual tones) as the beep and then using a tone detection algorithm to
locate it, in the same way that DTMF works.
On Tue, Aug 12, 2014 at 2:25 AM, Satish Barot
wrote:
> Hi All,
>
> I have been working on a project wh
2014-08-13 15:38 GMT+02:00 Mikael Fredin :
> On 12 August 2014 16:19, Olivier wrote:
>>
>> How can I read RTP ports from CLI (to double check what could be
>> included in rtp.conf file) ?
>> "sip show settings" do not provide the answer.
>
>
> On way would be to activate SIP debugging:
> sip set d
Nick Olsen wrote:
Hey everyone,
Currently, I've got a PBX that is emailing me on call failures to an
international SIP provider of ours.
I'm doing this with exten => 1,1,System(mail -s "Call from
${CALLERID(num)} to ${DNID} Failed with DialStatus ${DIALSTATUS}"
n...@flhsi.com < /dev/null)
This wo
asterisk-users-boun...@lists.digium.com wrote on 08/13/2014 08:31:01 AM:
> From: "Nick Olsen"
> To: ,
> Date: 08/13/2014 08:31 AM
> Subject: [asterisk-users] Better info on call failure
> Sent by: asterisk-users-boun...@lists.digium.com
>
> Hey everyone,
>
> Currently, I've got a PBX that is e
On 12 August 2014 16:19, Olivier wrote:
>
> How can I read RTP ports from CLI (to double check what could be
> included in rtp.conf file) ?
> "sip show settings" do not provide the answer.
>
On way would be to activate SIP debugging:
sip set debug on
Then check the INVITE body/SDP for port on a
Hey everyone,
Currently, I've got a PBX that is emailing me on call failures to an
international SIP provider of ours.
I'm doing this with exten => 1,1,System(mail -s "Call from
${CALLERID(num)} to ${DNID} Failed with DialStatus ${DIALSTATUS}"
n...@flhsi.com < /dev/null)
This works f
Leandro Dardini wrote:
Hello,
Kia ora,
I have my provider dropping the calls after 41 seconds of not receiving
any RTP from my asterisk. Obviously there is no RTP back when the caller
is leaving a message in the voicemail. Is it possible to have asterisk
generate some RTP packet back?
There
Hello,
trying to implement srtp with already working tls i somehow stuck with
srtp. If the extension has successfully registered a call from asterisk
to that extension works fine. But the other way round nothing happens.
[Aug 13 14:54:16] WARNING[31053]: chan_sip.c:3906 __sip_xmit: sip_xmit
of 0x
i'm using asterisk with tls but always get
WARNING[17634]: chan_sip.c:3906 __sip_xmit: sip_xmit of 0x7fe394006590
(len 609) to 83.78.150.198:60709 returned -2: Success
whats wrong there?
Best Regards Jakob
signature.asc
Description: OpenPGP digital signature
--
__
As we are top posting I will continue this.
Please have a look at:
https://wiki.asterisk.org/wiki/display/AST/Early+Media+and+the+Progress+Application
I hope this answers your questions.
Regards,
Michel.
op 13-08-14 01:34, Rafael Visser schreef:
> I am talking about sip on asterisk 11.10.2
>
Hi,
Wow, thanks Paul, realizing the problem makes a lot of sense.
So I setup Kamailio as a peer, but if I disable chan_sip module completely,
I can't do it in sip.conf like I'd otherwise assume to do. I tried to
rebuild Asterisk without chan_sip, but I guess that's not quite the way to
go? Asteri
Hi
Is anyone using asterisk on CentOS 7?
If so, is it working fine and as expected?
Regards
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIM
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