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The_KOPACb2014-02-04 10:53:04
Asterisk
The_KOPACb, 2014-02-04 10:53:04

Why, when a client connects from the outside (through the gateway 192.168.1.1), Asterisk registers, but cannot call?

Good afternoon.
There is an Asterisk server and a Beeline provider.
The provider issues sip only through its local address (10.78.210.26), the traffic flows to the external ip (213.33.248.28).
That is, the system has 2 interfaces, one for Beeline, one for our clients.

ifconfig
eth0      Link encap:Ethernet  HWaddr 90:2b:34:ce:ae:71
          inet addr:192.168.1.30  Bcast:192.168.1.255  Mask:255.255.255.0

eth0:0    Link encap:Ethernet  HWaddr 90:2b:34:ce:ae:71
          inet addr:10.78.210.26  Bcast:10.78.210.27  Mask:255.255.255.252

I wrote routes:
#       ### static routing ###
        route add default gw 192.168.1.1
        route add -net 213.33.248.28 netmask 255.255.255.252 gw 10.78.210.25

And everything seems to be fine, but there is one snag: when a client connects from the outside (through the 192.168.1.1 gateway), it registers, but cannot call.
rtp debug doesn't show any rtp traffic.
rtp debug
*CLI> rtp set debug on
RTP Debugging Enabled
  == Using SIP RTP CoS mark 5
    -- Executing [xxxxxxx@main:1] Dial("SIP/xxx-0000001c", "SIP/[email protected],30,tr") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/[email protected]
    -- SIP/beeline20-0000001d is making progress passing it to SIP/xxx-0000001c
    -- SIP/beeline20-0000001d answered SIP/xxx-0000001c
[Feb  4 11:48:33] WARNING[15035]: chan_sip.c:3656 retrans_pkt: Retransmission timeout reached on transmission W-Y.GiIfKanBTJx79PwCxxa24.q4vU6E for seqno 14764 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[Feb  4 11:48:33] WARNING[15035]: chan_sip.c:3685 retrans_pkt: Hanging up call W-Y.GiIfKanBTJx79PwCxxa24.q4vU6E - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
  == Spawn extension (main, xxxxxx, 1) exited non-zero on 'SIP/xxx-0000001c'


If you specify externip then I can call locally (because of NAT), but I won't be able to call via beeline...

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2 answer(s)
T
The_KOPACb, 2014-02-04
@The_KOPACb

pointed out

localnet=192.168.1.0/255.255.255.0 ; RFC 1918 addresses
       localnet=213.33.248.28/255.255.255.252

        externaddr = address.on.192.168.1.1          ; use this address.

and everything worked as it should.

K
Konstantin, 2014-02-05
@derwin

keep in mind - the beeline is a strange provider
. In my city, they do not make a full-fledged proxy, but require my asterisk to look into their network as into the main gateway. Why is this dangerous - calls do not go inside Beeline clients. I tried to specify only the necessary subnets - so it turned out that they do not disdain random addresses in gray networks (gray and even WHITE!). For example, even 1.1.1.1/30 was used as a gray address. And also, from time to time, they smoke "something there" and change the ACL so that either the voice does not go, or the signaling within the network. Oh yes, Barnaul.

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