Julien,
Just for the record, you don't need registration to iptel.org - just
plain DIAL(SIP/iptel/music).
On Sun, Jun 6, 2010 at 11:47 PM, Julien Claassen wrote:
> Thanks anyway, Ira. It was very kind of you to help me along as far as you
> could. I appreciate it.
> anyone else here, who might
The op_server.pl is part of the Flash Operators Panel, which isn't
really important to the operation of the PBX, it is just a nice pretty
interface showing what lines and what groups are active. What O/S are
you using? Are there any errors in the asterisk logs? Does asterisk stay
running after
On Sat, 5 Jun 2010, Adil Zaaraoui wrote:
> I want to write an AGI script doing this:
>
> 1-user call a number.
> 2-asterisk call the agi script
> 3-the script dial the peer
> 4-if the call is answered, let the call up for 1min
> 5-then the script hangs up the channel.
On Sun, 6 Jun 2010, Steve Ed
On Sunday 06 June 2010 17:09:33 bruce bruce wrote:
> Thanks for the input but it has nothing to do with the trunk configuration
> as EXACT same configuration works on another server with iptables disabled.
> I disabled iptables on this server as well but it doesn't work.
>
> sip show registery show
Thanks for the input but it has nothing to do with the trunk configuration
as EXACT same configuration works on another server with iptables disabled.
I disabled iptables on this server as well but it doesn't work.
sip show registery shows a Request Sent.
-Bruce
On Sun, Jun 6, 2010 at 4:58 PM, T
Reboot like 10 times and the problem still presists.
Also, upon reboot despite having done "chkconfig --add asterisk" asterisk
still doesn't load automatically. And amportal start fails. So, I have to do
"asterisk -g" first and then amportal start. Wondering if that might be
related?
Thanks for t
On Sunday 06 June 2010 13:46:49 bruce bruce wrote:
> I have tried every single rule I could into iptables but I can't register
> this VPS to a provider Spikko. Finally I did an iptable accept on INPUT,
> OUTPUT, and FORWARD, for ports 0:65000 just to test things and still I
> can't register to the
On 6 June 2010 19:48, bruce bruce wrote:
> Hi Guys,
>
> Just did an Asterisk 1.6.x (repo install) and FreePBX (source install).
> When trying to dial a number, I get this:
>
> tel*CLI> Use of uninitialized value in hash element at /var/www/html/panel/
> op_server.pl line 3367.
> Use of uninitiali
Thanks anyway, Ira. It was very kind of you to help me along as far as you
could. I appreciate it.
anyone else here, who might be able to help me along with my problem?
Warmly yours
Julien
Music was my first love and it will be my last (John Miles)
FIND MY WEB-
At 11:08 AM 6/6/2010, you wrote:
>So where to go now? Is there a test - without asterisk -, that I
> can perform
>to double check that the ports are correctly forwarded? Or would this be
>pointless, seeing that the registration works fine?
I wish I could help. My one and only Linux experience
Hello all!
Hm, I just examined the output of chan_sip's debug again and found this,
might that be the problem:
Warning: 392 213.192.59.75:5060 "Noisy feedback tells: pid=3955
req_src_ip=91.58.9.172 req_src_port=24002 in_uri=sip:sip.iptel.org
out_uri=sip:sip.iptel.org via_cnt==1"
I don't h
Yes i can get the user remaining minutes from my database, the scrips runs; but
when i run exec("Dial","IAX2/400") then geting the channelStatus if is answer
it does not hangup using either getChannel().hangup() or just hangup().
note: when running Dial from my script, it blocks for a period abo
Hi Guys,
I have tried every single rule I could into iptables but I can't register
this VPS to a provider Spikko. Finally I did an iptable accept on INPUT,
OUTPUT, and FORWARD, for ports 0:65000 just to test things and still I can't
register to the provider.
I can easily register to another provi
Hi Guys,
Just did an Asterisk 1.6.x (repo install) and FreePBX (source install). When
trying to dial a number, I get this:
tel*CLI> Use of uninitialized value in hash element at /var/www/html/panel/
op_server.pl line 3367.
Use of uninitialized value in concatenation (.) or string at
/var/www/html
Hi Ira!
Sorry, can't use any softphone, to my knowledge. They all come with GUIs or
don't support JACK or have so limited ALSA support, that they don't fit my
card (which has a lot of channels and some other HD-recording stuff).
Still I did try the sip call with app playback as well. That's
On Sun, 6 Jun 2010, Adil Zaaraoui wrote:
i do not need absolute timeout, i have to get from my database how many
minutes can the caller communicate; so in my script run the dial command
(fire the call), controlling the elapsed time if the channel is
answered, then hanging up the channel.
Can
At 08:43 AM 6/6/2010, you wrote:
>So now I found someone to forward the ports 5060 and 16000-16100 on my
>router and made sure to enter these ports 16000-16100 in rtp.conf.
>Still I get
>no calls going.
I should point out, that I just realized I've not a clue what app
jack is. I use sip and
Thank you Tzafrir
Adolphe Cher-aime
From my Iphone
On Jun 6, 2010, at 11:28 AM, Tzafrir Cohen
wrote:
> On Sun, Jun 06, 2010 at 11:27:45AM -0500, Adolphe Cher-aime wrote:
>>
>> Hello guys,
>>
>> I was wondering if it's possible to assign a dahdi channel to two
>> diferent groups.
>
> Sure. N
Thanks again,
i do not need absolute timeout, i have to get from my database how many minutes
can the caller communicate; so in my script run the dial command (fire the
call), controlling the elapsed time if the channel is answered, then hanging up
the channel.
Any help.
Regards
--
_
On Sun, Jun 06, 2010 at 11:27:45AM -0500, Adolphe Cher-aime wrote:
>
> Hello guys,
>
> I was wondering if it's possible to assign a dahdi channel to two
> diferent groups.
Sure. No problem:
group = 1,2,3,5,8,13,21,34,55
channel => 15
--
Tzafrir Cohen
icq#16849755
Un-top-posting...
On Sat, 5 Jun 2010, Adil Zaaraoui wrote:
> I want to write an AGI script doing this:
>
> 1-user call a number.
> 2-asterisk call the agi script
> 3-the script dial the peer
> 4-if the call is answered, let the call up for 1min
> 5-then the script hangs up the channel.
On Sun
Hello guys,
I was wondering if it's possible to assign a dahdi channel to two
diferent groups.
Thanks
Adolphe Cher-aime
From my Iphone
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Aste
Hello everyone!
So now I found someone to forward the ports 5060 and 16000-16100 on my
router and made sure to enter these ports 16000-16100 in rtp.conf. Still I get
no calls going.
The call is initiated. "sip show channels" shows the call with status ACK
and then the dialog with method in
Thank you for the reply.
1- yes i need to call my agi script; because i have to process some tasks with
my DBMS on the caller.
2- yes it is my first script, "While very simple, the AGI protocol is easy to
violate" i did not get your meaning.
5-yes i agree with you, is there an other solution?
S
Thank you for the reply.
1- yes i need to call my agi script; because i have to process some tasks with
my DBMS on the caller.
2- yes it is my first script, "While very simple, the AGI protocol is easy to
violate" i did not get your meaning.
So do you have a perfect solution?
Best regards
Richard Kenner wrote:
>> Is this bug alive in 1.6.2.9-rc1? I'm getting segfaults from chan_dahdi.
>> If it does effect 1.6.2.8-rc1, I'll just wait for rc2 to see if this
>> is my problem, instead of filing.
>
> I reported another instance of this today and it was fixed in the SVN a few
> hour
On Sun, Jun 6, 2010 at 11:10 AM, Kevin P. Fleming wrote:
> The message is labeled WARNING, which means it is not an error. This can
> be ignored, unless you are actually experiencing a problem.
What dedication, Kevin! First, it's Sunday. Second you're enjoying
AMOOCON with other lucky attendees.
On 05/28/2010 08:24 PM, Noah Pugsley wrote:
> I am having a small problem with asterisk-1.6.2.7 + app_fax on OpenBSD
> 4.7 -release. Everything seems to work fine. I have a macro which
> answers, receives the fax to a tiff, and then runs a script (mailfax) to
> convert that to pdf and email it.
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