I have an app which handles a Mitel's command port to change the MWI
lights. It detects dial tone, plays some DTMF digits, listens for
dialtone-or-busy, does a manager event on what it finds, and returns.
Since the Mitel command port does not give answer supervision (looks like
it's ringing),
Joseph wrote:
I always had a problem with SIP and DTMF, I'm using old sipura adapters and
have one digium iaxy FXS unit which works almost perfectly, never had any
problem
with DTMF on this unit.
However, all phones connected to Sipura don't work very well especially when
I setup speed
On Fri, Oct 30, 2009 at 04:54:46AM -0700, Vieri wrote:
Hi,
I have a PRI euroisdn link between an Alcatel PBX and Asterisk.
I'm having some trouble with overlap dialing.
Suppose I dial '874053' from an Alcatel extension ('7034') where '87' is an
Alcatel prefix of type ARS Prof.Trg Grp
Sorry for the off-topic, but perhaps this will be of interest to other
asterisk based ITSPs.
We are starting service in a rural area where the ILEC has the rural
monopoly. From what we have read in the FCC docs this does NOT exempt
them from number portability, but what does it take for us
On Sat, Oct 31, 2009 at 5:27 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
I'm not sure if handling of overlap hasn't changed since.
But can you provide a trace of how Asterisk sees things? e.g. 'pri
intense debug span 1'
the intense debug is overkill we only need messages of layer 3 ...
Your chances are likely slim to none. But good luck.
First to port numbers you have to be a recognized carrier, which for the
most part means getting numbers from NANPA : North American Numbering Plan
Administration. To do that you have to be certified by your state PUC or be
a CMRS (cell
Two more comments.
Yes, to join the PSTN call distribution system you must have SS7.
While rural ILECs are not exempt from number portability, there is a court
injunction that saves them from having to transport the call out of their
local rate center, so getting calls from a distant RILEC to a
Thanks,
I tried these options with no luck. I have an RMA in place for the
card. Tried loading a fresh install of Centos with no change. Will try
another card and hopefully try this card in another machine.
On Oct 30, 2009, at 5:49 PM, Mariano Lecuona wrote:
Take a look at this document.
On Sat, 31 Oct 2009, Cary Fitch wrote:
Two more comments.
Yes, to join the PSTN call distribution system you must have SS7.
While rural ILECs are not exempt from number portability, there is a court
injunction that saves them from having to transport the call out of their
local rate
Hi All;
Asterisk version is 1.6.1.8
Dahdi version is: dahdi-linux-complete-2.2.0.2+2.2.0
Since long time, and I am facing this problem and I did all the trouble
shooting that I know without any success.
The problem that while we are talking with someone through the FXO (connected
to the PSTN
Hello,
My client customers complaining that their calls suddenly get hung-up, I am
just investigating if the problem from my side, I had a log of a hang-up
case,
Does it help to know if there is a problem that can be resolved from my
side?
elastix*CLI
-- Hungup
Hi,
Is it possible to run a !command from Manager connection?
Thanks in advance!
CB
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On 10/31/09 10:24, Ivan Stepaniuk wrote:
Joseph wrote:
I always had a problem with SIP and DTMF, I'm using old sipura adapters and
have one digium iaxy FXS unit which works almost perfectly, never had any
problem
with DTMF on this unit.
However, all phones connected to Sipura don't work
Was'nt sure if this mail got through earlier:
I have been having a weird issue with my telco's ISDN PRI occasionally
resetting on a incoming call, i suspect it to possibly be a timing issue
since some of the incoming call work. This problem happens very frequently.
I am using asterisk-1.6.0.1
Hi
actually, i test a new Asterisk Server and i want add Mysql Realtime SIP.
I read on the wiki:
===
Database Config
put the following in res_mysql.conf
[general]
dbhost = 127.0.0.1
dbname = asterisk
dbuser = myuser
dbpass = mypass
dbport = 3306
Where is the log for the actual hang up of the call?.. can you do a sip
debug?
Although there can be many reasons, my first suspect is always a nat issue,
which manifest as the inability of asterisk to receive the incoming packets.
In that case, you should be getting a message saying hanging
To insert long pause during dialing and submitting multiple DTMF tones, is
there better solution then below:
exten =
_51,1,Dial(SIP/18778794...@pstn-5665,300,D(www1www),D(005893884053811#))
I think submitting multiple DTMF tones is not allow from one command line. The
first
On 10/30/09 12:55, Vincent wrote:
Hello
Since SIP/RTP is a pain to use with road warriors who need to connect
from any location over the Internet, I'd like to get them some IAX
phones instead.
For those of you using this protocol instead of SIP, what would you
recommend as IAX hardphones and
For anyone interested, this is an HP ML115 Proliant server using AMD.
We put in a PCI Digium card and all was bliss. Also found the PCI
Express card works fine in a Dell T100 with Xeon processors.
On Oct 30, 2009, at 5:49 PM, Mariano Lecuona wrote:
Take a look at this document. This may
On 10/31/09 08:20, bilal ghayyad wrote:
Hi All;
Asterisk version is 1.6.1.8
Dahdi version is: dahdi-linux-complete-2.2.0.2+2.2.0
Since long time, and I am facing this problem and I did all the trouble
shooting that I know without any success.
The problem that while we are talking with someone
My server use public ip, so no nat issues, here is the out of sip debug:
-
--- (10 headers 0 lines) ---
Sending to 213.165.32.100 : 5060 (no NAT)
--- Reliably Transmitting (no NAT) to 213.165.32.100:5060 ---
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP
On 10/31/09 12:56, Joseph wrote:
To insert long pause during dialing and submitting multiple DTMF tones, is
there better solution then below:
exten =
_51,1,Dial(SIP/18778794...@pstn-5665,300,D(www1www),D(005893884053811#))
I think submitting multiple DTMF tones is not allow
On 10/30/09 10:32, Carlos Chavez wrote:
On Fri, 2009-10-30 at 08:37 -0500, Tilghman Lesher wrote:
On Thursday 29 October 2009 12:32:48 Carlos Chavez wrote:
On Thu, 2009-10-29 at 12:23 -0500, Tilghman Lesher wrote:
On Thursday 29 October 2009 11:49:30 Carlos Chavez wrote:
On Wed,
Hello list,
How can you obtain the default mailbox for a SIP extension (as stored
in sip.conf and shown with sip show peer ext)? Is there a
function to extract it?
Why? Some extensions have shared mailboxes and others do not and I
don't want to duplicate logic, just use the extension's default
On Sat, Oct 31, 2009 at 12:04:18PM -0400, cbulist wrote:
Hi,
Is it possible to run a !command from Manager connection?
No. You can implement it yourself.
'!' is not sent to the asterisk daemon. Rather, the local client runs a
command.
For instance:
# id -a
uid=0(root) gid=0(root)
The only informative part are the 2 paragraphs of the sip debug, but can't
tell much since you only show a very small portion of the sip log. There is
a 487 Request terminated there screaming at you but can't tell if meaning
that provider is not handling the ACKs. That section of the
Hello,
I have grabbed again a whole call when it hangs up debug, I dono what else I
can read??
What exactly you want me to look for?
And assuming there is a firewall at my ISP, how to diagnose it?
Thanks for the advise,
Here is another log:
-- Called 9/0557202919
-- Call
Steve Johnson schrieb:
How can you obtain the default mailbox for a SIP extension (as stored
in sip.conf and shown with sip show peer ext)? Is there a
function to extract it?
Why? Some extensions have shared mailboxes and others do not and I
don't want to duplicate logic, just use the
I have a DID but for some reason is not working in asterisk-1.6
The same sip connection in asterisk-1.4 is working OK, but it doesn't work with
asterisk-1.6
Here is my sip.conf section:
...
[actio-out]
type=friend
secret=password
user=48746612254
username=48746612254
fromuser=48746612254
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