Hi,
I was wondering if someone could shed some light on the issue im having. Currently having working perfectly asterisk with my Trunk Sip VOIP provider (UNE based on colombia south America) Currently trying to use zoiper using SIP for the users to call the office, the issue is that they cant hear each other. after hours of researching found out in the sip.nat config i would put this
after reloaded it starts to working the zoiper calls both sides can hear BUT after few mins users started to say that the calls weren't coming in to the Trunk Sip of my VOIP provider so i had to remove those lines on the sip nat config any idea how to troubleshoot this i know its a nat thing but dont know where to look.
This is my SIP TRUNK
This is what i get when i hear no sound
Thank you
I was wondering if someone could shed some light on the issue im having. Currently having working perfectly asterisk with my Trunk Sip VOIP provider (UNE based on colombia south America) Currently trying to use zoiper using SIP for the users to call the office, the issue is that they cant hear each other. after hours of researching found out in the sip.nat config i would put this
Code:
nat=yes
externip=mywanip
externhost=mywanip
localnet=192.168.1.0/255.255.255.0after reloaded it starts to working the zoiper calls both sides can hear BUT after few mins users started to say that the calls weren't coming in to the Trunk Sip of my VOIP provider so i had to remove those lines on the sip nat config any idea how to troubleshoot this i know its a nat thing but dont know where to look.
This is my SIP TRUNK
Code:
type=friend
context=from-trunk
host=172.17.179.166
dtfmode=rfc2833
allow=ulaw&alaw&gsm&g729
qualify=yes
insecure=yes
nat=yesThis is what i get when i hear no sound
Code:
WARNING[2780]: chan_sip.c:4024 retrans_pkt: Retransmission timeout reached on transmission zhi5gw5iezxz3w2xsvii5jv0egzjhexw@SoftX3000 for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[2018-06-14 19:31:03] WARNING[2780]: chan_sip.c:4053 retrans_pkt: Hanging up call zhi5gw5iezxz3w2xsvii5jv0egzjhexw@SoftX3000 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).Thank you
