On 01/10/09 00:57, Kirill 'Big K' Katsnelson wrote:
> Asterisk 1.6 behind a firewall and NAT. We are terminating multiple DID
> calls, originating and transferring.
>
> A provider offers both SIP and IAX trunking. Cateris paribus, what is
> the preferred solution to choose? What points to consider?
Steve Edwards wrote:
>> My understanding was that IAX encapsulates the same RTP traffic, or, and
>> the very least, same stream of data encoded by a codec. Is that not true
>> in case of IAX? How can a transport protocol affect volume--or quality
>> (lest it is dropping packets)?
>
> My (limite
Hi All,
I have a problem, when I was doing a performance testing using an asterisk
server: Quadcore processor, 4GB RAM, CentOS5.2, after 150-151 calls all the
other calls are giving busy, I tried to do ulimit related stuff, like
increasing the soft and hard limits to 10 but no luck, Any ideas
My system is linked to a legacy PBX via Q-SIG and most of my tests so
far have been from that PBX. I created a number of MeetMe conference rooms
and they work fine when called from the legacy PBX. However, when there's
a MeetMe room with a legacy caller and a SIP phone, the SIP phone can
hear th
> Steve Edwards wrote:
>> Some say the audio quality is better with SIP. My experience has been
>> with "low volume" (xx) calls across the internet and "high volume"
>> (xxx) within the same cabinet.
On Wed, 30 Sep 2009, Kirill 'Big K' Katsnelson wrote:
> My understanding was that IAX encapsul
That's nice. At least now peopel that want to do call recording can do
so without having to keep Asterisk in between the circuits.
However all other applications like added voicemail, conferencing,
followme etc ... still needs Asterisk in between unless "they" have a
spare port on the PBX and do th
Steve Edwards wrote:
> Some say the audio quality is better with SIP. My experience has been with
> "low volume" (xx) calls across the internet and "high volume" (xxx) within
> the same cabinet.
My understanding was that IAX encapsulates the same RTP traffic, or, and
the very least, same stream
Steve Edwards wrote:
>> Steve Edwards wrote:
>>> Is the manager or are the agents using disa()?
>>>
>>> How about:
>>>
>>> exten = *,n,set(SPYGROUP=ALLOW-SPYING)
>>>
>>> for the agents and:
>>>
>>> exten = *,n,chanspy(,g(ALLOW-SPYING))
>>>
>
>
>
> Is your code vendor locked to Sangoma ???
>
>
Hello Martin, not at all. The code is intended to be part of chan_dahdi
Asterisk channel driver and as such any card capable of using the dahdi
interface can benefit from it.
--
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIn
On Wed, 30 Sep 2009, Kirill 'Big K' Katsnelson wrote:
> Asterisk 1.6 behind a firewall and NAT. We are terminating multiple DID
> calls, originating and transferring.
>
> A provider offers both SIP and IAX trunking. Cateris paribus, what is
> the preferred solution to choose? What points to cons
Asterisk 1.6 behind a firewall and NAT. We are terminating multiple DID
calls, originating and transferring.
A provider offers both SIP and IAX trunking. Cateris paribus, what is
the preferred solution to choose? What points to consider?
I can name the provider if this is not against this list
I want to use IPKall with Asterisk.
Now, I want my calls to land at 2 different locations , not connected with
each other.
If I want to configure IPKall DID number in Asterisk , I need to specify IP
on IPKall.
How can I make it enable so that calls can land up at both locations ?
___
>> On Wed, 30 Sep 2009, cov...@ccs.covici.com wrote:
>>
>>> there is an undocumented feature in meetme using the kick option called
>>> all, which kicks everyone off if you want to be sure and end the
>>> conference.
> Steve Edwards wrote:
>
>> Are you referring to the documented 'K' option for t
Hello I have a CentOS OS that have asterisk installed, also zaptel, but when
I use the:
lspci command
I have the next asnwer:
03:80.0 Ethernet controller: Unknown device d161:8000 (rev 11)
I also check the zaptel file that contain the modules that can support and
the wcte12xp module is not in th
Moises,
You forgot to add that in order to monitor one T1 or E1 circuit you
need two ports on your card...
So that might be getting expensive with Sangoma cards You can do
the same with cheap Tormenta
cards that sell for ~$350 (I did that some time ago)
Anyways all zaptel/dahdi cards can be s
Steve Edwards wrote:
> On Wed, 30 Sep 2009, cov...@ccs.covici.com wrote:
>
> > there is an undocumented feature in meetme using the kick option called
> > all, which kicks everyone off if you want to be sure and end the
> > conference.
>
> Are you referring to the documented 'K' option for the
On Wed, 30 Sep 2009, cov...@ccs.covici.com wrote:
> there is an undocumented feature in meetme using the kick option called
> all, which kicks everyone off if you want to be sure and end the
> conference.
Are you referring to the documented 'K' option for the meetmeadmin()
dialplan application o
Tom Browning wrote:
> I use SIPAddHeader today to put some proprietary info into the SIP
> header of an outbound call. Now I'd like to add some proprietary info
> to the SDP portion of an outbound call. Can this be done with
> SIPAddHeader?
Nope; there is no way to make modifications to the SD
there is an undocumented feature in meetme using the kick option called
all, which kicks everyone off if you want to be sure and end the
conference.
Ivan Stepaniuk wrote:
> Anahi Ludueña wrote:
> > Hi people, I want to make a meetme between 2 numbers.
> > First I enter the number1 into the meetm
Howdy,
I've spent a couple of days writing a new feature for Asterisk that allows
to record calls in T1 or E1 PRI lines using Asterisk connected to tapped
lines. This means that you don't have to install anything in the PBX's/telco
equipment that is going to be monitored, all you need is to install
Bayardo Sanchez wrote:
> I have a user but I need to give that user only kill and disable all
> connection cut calls what is the command in the CLI
Please rephrase your question. I've just read your message 5 times and I
still don't understand what do you want to do. Regards.
PS: A 15+ line signa
Anahi Ludueña wrote:
> Hi people, I want to make a meetme between 2 numbers.
> First I enter the number1 into the meetme. It is waiting for the other
> number, but the other number never entered, so, how can I finish the
> meetme from the dialplan?. Is it posible by using MeetmeAdmin and kick
>
Try 'pri intense debug span 1'
Used it last night.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Thursday, 1 October 2009 4:09 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussio
I use SIPAddHeader today to put some proprietary info into the SIP header of
an outbound call. Now I'd like to add some proprietary info to the SDP
portion of an outbound call. Can this be done with SIPAddHeader?
Thanks in advance,
Tom
___
-- Bandwid
Are you using a VPM module? The dahdi changelog mentions some recent work
related to VPM modules and HDLC aborts.
https://issues.asterisk.org/view.php?id=15498
https://issues.asterisk.org/view.php?id=15529
I just rebuilt a server this weekend for the same problem on a single span
card with a V
How do these extensions show up on a "core show channels verbose"? I do my
hints like this
[internal]
- exten => 501,hint,SIP/100
- exten => 502,hint,DAHDI/1
- exten => 503,hint,ZAP/1
you should be able to register a hint based on the cscv output.
_
Hi
I've a queue which has generic zap extensions (of my legacy PBX which is
connected to asterisk via cross over on span 4 ) logged in ..The legacy pbx
extensions are able to logon to queue perfect.. but Whenever a call comes in
queue the status of that extension in "queue show " always shows
as
On Wednesday 30 September 2009 11:54:11 Jerry Geis wrote:
> > pri intense debug span
>
> Just pointing out that was not clear from the HELP command.
>
> I thought span was the span number
>
> not span
>
> Thanks for the direction.
At the list level, we only provide the keywords. If you had expl
I've got PBXNSIP running on a windows box and it is trying to register with
my Asterisk box. I can set up one trunk and it works fine, however if I try
to setup a second trunk from the same box, there is some sort of
authentication issue where Asterisk appears to be confusing which trunk is
which.
>
> pri intense debug span
>
Just pointing out that was not clear from the HELP command.
I thought span was the span number
not span
Thanks for the direction.
Jerry
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Astri
Jerry Geis wrote:
> Running asterisk 1.4.26.2
>
> help pri
>pri debug span Enables PRI debugging on a span
>pri intense debug span Enables REALLY INTENSE PRI debugging
> pri no debug span Disables PRI debugging on a span
>pri set debug file Sends PRI debug outp
Because you need to type "pri intense debug SPAN 1"
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Wednesday, September 30, 2009 10:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discus
"pri intense debug span Enables REALLY INTENSE PRI debugging"
add span keyword
or use a tabulator that will do that for you
Martin
On Wed, Sep 30, 2009 at 10:08 AM, Jerry Geis wrote:
> Running asterisk 1.4.26.2
>
> help pri
> pri debug span Enables PRI debugging on a span
> pri
Running asterisk 1.4.26.2
help pri
pri debug span Enables PRI debugging on a span
pri intense debug span Enables REALLY INTENSE PRI debugging
pri no debug span Disables PRI debugging on a span
pri set debug file Sends PRI debug output to the specified file
On Wed, Sep 30, 2009 at 10:03:34AM +0200, jonas kellens wrote:
> Thanks for your response. I have modified asterisk.conf as follow :
>
> [directories]
> astetcdir => /opt/etc/asterisk
> astmoddir => /usr/lib/asterisk/modules
> astvarlibdir => /opt/var/lib/asterisk
> astdatadir => /opt/var/lib/aste
I'm probably wrong, but IMO CDR{start} is the SIP Invite time and
CDR(answer) is the time that the 183 signal was received. You can probably
tweak sip.conf to make this so (or not).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digiu
Thanks,
It worked, it seems there was something wrong. The following is working now:
Action: UpdateConfig
srcFileName: voicemail.conf
dstFileName: voicemail.conf
Action-00: Append
Cat-00: default
Var-00: 2000
Value-00: >,Jhon
ActionID: 1234
Bye,
Anahi Ludueña
> Date:
Hi people, I want to make a meetme between 2 numbers.
First I enter the number1 into the meetme. It is waiting for the other number,
but the other number never entered, so, how can I finish the meetme from the
dialplan?. Is it posible by using MeetmeAdmin and kick all the users?
Thanks,
An
Shaun Ruffell wrote:
> On 09/29/2009 06:52 AM, Jerry Geis wrote:
>> A user report that this issue:
>>
>> https://issues.asterisk.org/view.php?id=15429
>>
>>
>> Has resolved their problem with a TDM card.
>>
>> My card is a T1/PRI card. Different module to load.
>> I have the same issue.
>>
>> Does
On Wed, 2009-09-30 at 14:57 +0300, Cyprus VoIP wrote:
>
> > You see the wav files but do you see the files encoded for the codecs
> you are using?
> There's only one wav file there. No encoded files, but on asterisk 1.2
> we have, it's the same file and it works.
Hmm . . only one wav file. W
> I'm afraid I can't be much help as I am both a newbie and it works just
> fine for me on 1.6.1.6. Of course, mine was a fresh installation.
Thanks for your help, John. Mine is also a fresh installation, but now
at least I know it's not a version issue.
> Is there anything in the logs to gi
I'm afraid I can't be much help as I am both a newbie and it works just
fine for me on 1.6.1.6. Of course, mine was a fresh installation.
Is there anything in the logs to give you a clue? You see the wav files
but do you see the files encoded for the codecs you are using? I think
Asterisk will tr
Dear all
is it possible to handle a queue using a programmed IVR?
As i understood, is possible to:
- do some IVR in the dialplan BEFORE to queue the call
- put a timeout to exit from the call and then do some IVR in the dialplan
- intercept a single dialtone to exit the queue and performe some I
I've set transfer = no for all channels in chan_dahdi.conf, but I still
have the same
[channels]
context=from-pstn
signalling=fxs_ks
rxwink=300 ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
three
Hello,
We posted the question below yesterday, but got no answer from the
community.
When we checked the same behavior with Asterisk 1.2, we got the "Started
music on hold, class..." message on the console, but in 1.6, we get
absolutely nothing.
I tried to unload and reload the moh module and
Il giorno 29/set/09, alle ore 17:46, Danny Nicholas ha scritto:
> I believe that this information is at least indirectly in the CDR.
>
> [...]
> If you subtract the 92 from the 97, you get the 5 second number
> you’re looking for. These fields have actual names, but they aren’t
> relevant to
Thanks for your response. I have modified asterisk.conf as follow :
[directories]
astetcdir => /opt/etc/asterisk
astmoddir => /usr/lib/asterisk/modules
astvarlibdir => /opt/var/lib/asterisk
astdatadir => /opt/var/lib/asterisk
astagidir => /opt/var/lib/asterisk/agi-bin
astspooldir => /opt/var/spool
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