dear
i have configured properly asterisk. At the one end i am using x-lite soft
ph and another end twinkle. call is going properly from both end but after
picking the phone not able to listen other one.
when i checked the port 5060 on the asterisk server it is always showing
closed while i have
Port 5060 when used with the sip protocol is used witj UDP protocol. Telnet
is using TCP.
I am typing from my mobile phone...
Il giorno 01/lug/2012 09:35, alok srivastava alok...@gmail.com ha
scritto:
dear
i have configured properly asterisk. At the one end i am using x-lite soft
ph and
No voice means you have to look at the rtp ports.
You can find more via google firewall rtp ports asterisk
B.
Op 1-7-2012 9:34, alok srivastava schreef:
dear
i have configured properly asterisk. At the one end i am using x-lite soft
ph and another end twinkle. call is going properly from
Hello,
On http://pastebin.com/AP5GBWUR you may see an excerpt from asterisk
full log that includes a failing fax sending session. As you can see in
line 328, the transmission fails with error Received bad response to
DCS or training. It seems that something goes wrong along the lines 315
to 320,
On 07/01/2012 01:46 AM, Petros Moisiadis wrote:
Hello,
On http://pastebin.com/AP5GBWUR you may see an excerpt from asterisk
full log that includes a failing fax sending session. As you can see in
line 328, the transmission fails with error Received bad response to
DCS or training. It seems that
On 12-06-30 03:43 PM, Chris Gentle wrote:
On Wed, Jun 13, 2012 at 5:12 AM, Administrator TOOTAI ad...@tootai.netwrote:
someone knows when asterisk binary packages will be available on
asterisk.org for Ubuntu precise (aka 12.04)?
I did a fresh install of Ubuntu Server 12.04 LTS (precise) on
On Sun, 2012-07-01 at 13:04 +0530, alok srivastava wrote:
dear
i have configured properly asterisk. At the one end i am using x-lite
soft ph and another end twinkle. call is going properly from both end
but after picking the phone not able to listen other one.
when i checked the port 5060 on
if you check out your sip.conf.
On Jun 29, 2012, at 5:54 PM, gincantalupo wrote:
Hi all,
after upgrading my Asterisk 1.4.26.2 to 1.8.11.0 I cannot register to my VoIP
provider because it says I'm trying to connect to port 55150 (that's what the
call center guy told me)...but I'm not. In
