thanks Samy
i have set nat=yes, now getting sound from both side but there is too uch
disturbance. soetime we becoe audible and sometime not.i did not set extern
ip coz my asterisk server is directly configured on public ip. I have
softphones on some where localnets separate from asterisk server
Hi,
Being audible sometime or bad voice quality is only due to internet latency
or bad internet situation.
[Jul 4 19:37:39] WARNING[3054]: chan_sip.c:3660 retrans_pkt:
Retransmission timeout reached on transmission
551b9e744a6b41ee2c00033e22b333c0@122.160.154.189:5060 for seqno 102
(Critical
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
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
