Good Day
Find attached the relevant portions of the asterisk CLI.
Please,which portion of the extension .conf should i send ?
It is connected via RJ 45 connector to an E1 modem to the telco company.
I use E1 link.
I will appreciate your reply.
Best Regards
On Dec 18, 2007 4:02 PM, dave
Thanks
Please am using putty to again access to my Linux asterisk box.
How can i use tcpdump to get your request on the exact Ethernet port and
port number.
I will appreciate your reply.
Best Regards
On Dec 18, 2007 4:02 PM, dave cantera [EMAIL PROTECTED] wrote:
lolu,
sounds more like a
Hi Steve
Am connected to the telco through an E1 link using modem(Watson 5 modem
SDHSL 1 PAIR schmid telecommunications).The MODEM is connected to the
asterisk box through RJ 45 to the asterisk box end and serial connector to
the modem end .
Which portion of the extension conf should i post ?
lolu,
while you are making the call., capture and post your CLI output
... this is easy to do since you are using putty.
login to your pbx and start asterisk, use the below command:
# asterisk -vvvr
then make the call. hilite the text on the putty terminal and paste it
into the body of the
What is the output of ztconfig from the Linux command line? What does
your zaptel.conf and zapata.conf look like? What is the relevant part
of extensions.conf (the dialout section that fails). Also from the CLI,
it would be most helpful to post the output you get when dialing out
fails. I
Hi all,
I am grateful for our contribution so far .
I followed dave advise and i have the attached file using the aterisk -r
when a call is made.
I attached two files.
One of the attached file is for the external call,which replied with the
PROBLEM all trunks are busy now,please try your
Hi All
I FOUND OUT THAT THE ATTACHMENT WAS NOT SENT WITH THE MAIL.
FIND BELOW THE OUTPUT USING asterisk -vvvr command for EXTERNAL calls that
gave the ouput ALL TRUNKS ARE BUSY PLEASE TRY YOUR CALL LATER.
Verbosity is at least 3
-- Executing Macro(SIP/7871-f813, dialout-trunk|1|018774957||)
lolu
I reformated the output so it was easier to understand. I attached the
word document for you.
on the below line:
--
Executing Dial("SIP/7871-f813", "ZAP/1/8774957|120|W") in
new stack
-- Requested transfer capability:
0x00 - SPEECH
-- Called 1/8774957
-- Zap/1-1 is proceeding passing
Good Day all
Please I am having some issues on my voip asterisk server
I make internal calls on extensions configured ie extension 192 can
call extension 195 etc
But each time i try to make calls outside the extension ie calling a
GSM or an external line ,i always hear this response all trunk
Post:
Asterisk CLI : sip show peers
Asterisk CLI : zap show channels
Asterisk CLI: zap show status
As well as your extensions.conf
Are you able to ping you GSM gateway? is connected via SIP or Telephony
interface card?
Best regards,
Mouta
On Dec 18, 2007 10:47 AM, Lolu Gbenga [EMAIL
Lolu Gbenga wrote:
Good Day all
Please I am having some issues on my voip asterisk server
I make internal calls on extensions configured ie extension 192 can
call extension 195 etc
But each time i try to make calls outside the extension ie calling a
GSM or an external line ,i always hear
lolu,
sounds more like a telco/itsp problem then *.
I would
tcpdump -i eth0 port 5060
to make sure it is actually going out... change 5060 if you have changed
your port to your itsp, of course.
to see what is going on as well as the other debugging notes mentioned
in this thread.
daveC
Lolu
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