-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir
Mikhelson
Sent: Thursday, January 16, 2014 8:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 11 an
On 17-01-14 01:57, Dan Austin wrote:
Patrick Lists wrote:
On 16-01-14 21:37, Gergely Kiss wrote:
Dear List,
I'm about to build an Asterisk 11.7 based PBX from scratch for our
company. I'm in the middle of the planning phase and it turned out that
our VoIP provider prefers H.323 protocol for ha
I'm used to seeing fraudulent attempts to authenticate, But now I'm
getting them from the server itself.
I have an asterisk server behind a firewalled router. The local subnet
is 10.10.10.0/24, the server is 10.10.10.100.
Now I'm seeing in the log lots of:
Failed to authenticate device
<*>
On 1/16/2014 6:57 PM, Dan Austin wrote:
> Patrick Lists wrote:
>> On 16-01-14 21:37, Gergely Kiss wrote:
>>> Dear List,
>>>
>>> I'm about to build an Asterisk 11.7 based PBX from scratch for our
>>> company. I'm in the middle of the planning phase and it turned out that
>>> our VoIP provider prefe
Patrick Lists wrote:
> On 16-01-14 21:37, Gergely Kiss wrote:
>> Dear List,
>>
>> I'm about to build an Asterisk 11.7 based PBX from scratch for our
>> company. I'm in the middle of the planning phase and it turned out that
>> our VoIP provider prefers H.323 protocol for handling voice calls (while
On 16-01-14 21:37, Gergely Kiss wrote:
Dear List,
I'm about to build an Asterisk 11.7 based PBX from scratch for our
company. I'm in the middle of the planning phase and it turned out that
our VoIP provider prefers H.323 protocol for handling voice calls (while
SIP is also supported as "plan B")
On 14-01-16 03:37 PM, Gergely Kiss wrote:
Dear List,
I'm about to build an Asterisk 11.7 based PBX from scratch for our company.
I'm in the middle of the planning phase and it turned out that our VoIP
provider prefers H.323 protocol for handling voice calls (while SIP is also
supported as "plan
On Thu, Jan 16, 2014 at 3:17 PM, Andres wrote:
> On 1/16/14, 2:23 PM, Michael L. Young wrote:
>
>> - Original Message -
>>
>> From: "Andres"
>>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>>>
>>> Sent: Wednesday, January 15, 2014 7:51:28 PM
>>> Subject: Re: [asterisk
- Original Message -
> From: "Andres"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Thursday, January 16, 2014 4:17:53 PM
> Subject: Re: [asterisk-users] Asterisk ignoring nat settings
>
> > I am curious why you would say that "nat=yes" might work over
> > "n
On 1/16/14, 2:23 PM, Michael L. Young wrote:
- Original Message -
From: "Andres"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, January 15, 2014 7:51:28 PM
Subject: Re: [asterisk-users] Asterisk ignoring nat settings
Why don't you try with nat=yes.
Dear List,
I'm about to build an Asterisk 11.7 based PBX from scratch for our company.
I'm in the middle of the planning phase and it turned out that our VoIP
provider prefers H.323 protocol for handling voice calls (while SIP is also
supported as "plan B").
As I never worked with H.323 channels
- Original Message -
> From: "Andres"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Wednesday, January 15, 2014 7:51:28 PM
> Subject: Re: [asterisk-users] Asterisk ignoring nat settings
> Why don't you try with nat=yes. It should be equivalent to what you
On Thu, Jan 16, 2014 at 04:58:14PM +0100, Olivier wrote:
> Thanks for replying.
>
> So as python-starpy requires asterisk in Debian Wheezy repo, for a Debian
> setup the alternatives are either :
> - to install it from source
> - tto build my own custom package removing this asterisk dependency (i
exten => 179,1,SIPAddHeader(Call-Info:\;answer-after=0)
exten => 179,2,Page(SIP/180&SIP/181&SIP/182&SIP/184)
The asterisk11 page() application works great, but I've just learned
that the person who initiated the page can transfer or conference the
page if they don't hang it up
And we just figured that sound quality issues were not due to tcpdump ..
anyway sorry to troll this feed, and thank you for your sugestion
On 16 January 2014 16:57, Tiago Geada wrote:
> Gareth,
>
> I had to disable the tcpdump process, has we were having sound quality
> issues.
>
> :-(
>
>
> O
Gareth,
I had to disable the tcpdump process, has we were having sound quality
issues.
:-(
On 16 January 2014 15:35, Gareth Blades wrote:
> On 16/01/14 15:29, Kevin Larsen wrote:
>
> Not to derail the conversation, Gareth, but do you leave this running full
> time on your asterisk boxes or ju
On Thursday 16 January 2014, Olivier wrote:
> Thanks for replying.
>
> So as python-starpy requires asterisk in Debian Wheezy repo, for a Debian
> setup the alternatives are either :
> - to install it from source
> - tto build my own custom package removing this asterisk dependency (is it
> easy o
Thanks for replying.
So as python-starpy requires asterisk in Debian Wheezy repo, for a Debian
setup the alternatives are either :
- to install it from source
- tto build my own custom package removing this asterisk dependency (is it
easy or even possible ?)
- to use another solution such as pyst.
On 16/01/14 15:29, Kevin Larsen wrote:
Not to derail the conversation, Gareth, but do you leave this running
full time on your asterisk boxes or just turn it on when you are
trying to track problems?
On average, how far back can you go for looking at problems?
Its normally running full time
Looking at his tcpdump command it keeps 500 files of 10 MB each? (not sure)
On 16 January 2014 15:29, Kevin Larsen wrote:
> asterisk-users-boun...@lists.digium.com wrote on 01/16/2014 08:55:31 AM:
>
> > From: Gareth Blades
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > ,
>
Hi,
I transfered the capture to my local machine and opened it in wireshark, I
can search from there:
--> SIP Display info:
"Sapo:0:243709253:1389884558.292163:SIP/covilha-pstn-000201f3"
but I will add your comment to my notes.
I've already searched the asterisk FULL log, and seen the Set() lin
asterisk-users-boun...@lists.digium.com wrote on 01/16/2014 08:55:31 AM:
> From: Gareth Blades
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> ,
> Date: 01/16/2014 08:55 AM
> Subject: Re: [asterisk-users] Weird issue with
Set(CALLERID(name)=string);
> Sent by: asterisk-users-bo
The SIP trace will give you an idea is perhaps something is becoming
corrupted. If you keep a log of the asterisk console output (asterisk
-rvvv) then you will see what it attempts to set the callerid to and any
errors.
Another tip. When you have a look at the sip trace you will see the
call-
Yes you can. This what starpy is for. It's build around Python twisted
which allow you to write non blocked socket servers. You can use starpy as
a fastagi server.
Both AMI and FASTAGI can be configured from a .conf file as follow:
[AMI]
username=ami_user
secret=ami_pass
server=asterisk_ami_ip
po
Thank you Gareth
I will try that :)
On 16 January 2014 14:55, Gareth Blades wrote:
> Very little as the amount of data being captured is quite small. We have
> it running on our production servers which routinely handle a couple of
> hundred concurrent calls.
>
> This is the script we use to s
Hello,
Is it possible to run Starpy and Asterisk on different machines ?
A quick glance at http://www.vrplumber.com/programming/starpy/ seems to
tell it is possible but Debian's python-starpy package installs Asterisk.
What do you think ?
Regards
--
___
Second thought, that would only allow me to know if there is a problem on
asterisk or softphone.
Because the old callerid(name) that was presented on the softphone,
belonged to a call made to a different peer, I doubt that it would be a
softphone problem.
Our softphones are fixed with the same pe
Very little as the amount of data being captured is quite small. We have
it running on our production servers which routinely handle a couple of
hundred concurrent calls.
This is the script we use to start off the capture. It uses rolling
capture files so we will always have the last X number
You're right, seems like a nice way to debug. Regarding that, how would the
impact be affected running it on asterisk box? I guess only port 5060 is
not too bad
On 16 January 2014 14:09, Gareth Blades wrote:
> On 16/01/14 10:47, Tiago Geada wrote:
>
> Hi folks,
>
> We've been having a weird i
On 16/01/14 10:47, Tiago Geada wrote:
Hi folks,
We've been having a weird issue... It is happening more often in the
last few months...
Most inbound calls, we have in our dialplan before Queue():
Set(CALLERID(name)=${PARTNER}:0:${CALLERID(num)}:${UNIQUEID}:${CHANNEL});
So when the call ring
Yup. That's what i do. The CLI version of linphone set to autoanswer, with
the audio jacks tied to our exernal sound system. Works well. The echo
cancellation in linphone helps a lot for speakerphones.
On Jan 16, 2014 7:51 AM, "Administrator TOOTAI" wrote:
> Hi list,
>
> I have a customer which w
Hi list,
I have a customer which will organize a conference in a big meeting room
which has a sound system. He would like to connect this sound system to
a MeetMe room so participant in the MeetMe can act as if they where on site.
My idea is to take a barbone or Notebook, connect it to the so
Hi folks,
We've been having a weird issue... It is happening more often in the last
few months...
Most inbound calls, we have in our dialplan before Queue():
Set(CALLERID(name)=${PARTNER}:0:${CALLERID(num)}:${UNIQUEID}:${CHANNEL});
So when the call rings a member, softphone will show this strin
Hi everyone.
Having experimented a but with a prototype of a system I described in
an earlier thread (Reading DTMF sent by callee during a SIP call), I
decided to implement my requirement by transferring the call to
another extension. This way, the callee can open the door by pressing
#1, and the
Yes, thank you. Maybe I have found the problem. The asterisk server is
behind a nat and the RTP port range was not redirected to the asterisk box,
so the Symmetric RTP cannot work because the asterisk is not receiving any
RTP packet from the remote phone.
Leandro
2014/1/16 Ishfaq Malik
> Is di
Is directmedia set to no?
On 15 January 2014 23:11, Leandro Dardini wrote:
> Hello,
> I have an asterisk box with a peer configured with
> nat=force_rport,comedia, but asterisk keeps sending the audio to the
> private IP address and ignoring the client peer nat settings.
>
> If I check the "sip
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