system toolbox » voip

ifold rtp.conf

[general]
rtpstart=8000
rtpend=11000

sip.conf

[50]
type=friend
username=50
password=xxxx
host=dynamic
callerid="officeze" 
callgroup=3
nat=yes           <<<<<<<<< promjena default postavki
canreinvite=no    <<<<<<<<<

• alaw je prvi codec
• gsm drugi

• rtp client port 8000 (default je 3000)

• UDP SIP - 5060
• UDP RTP - 8000-11000

i to sve preusmjerio na asterisk.sigma-com.net (192.168.45.4)

http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite

'canreinvite=no' stops the sending of the (re)INVITEs once the call is established. From messages in the archives and the Asterisk handbook one finds out that the Cisco ATA-186 does not handle the (re)INVITE well. This is necessary if the client and the Asterisk server is on opposite sides of a NAT gateway or firewall.

http://www.voip-info.org/tiki-index.php?page=Asterisk+SIP+externip

Synopsis
externip = extern.ip.address

Description
Indicates the IP address (alternatively you can enter a hostname) that will be used as the source IP address for all SIP messages when NAT is specified.

Examples - Use it in [general] section of SIP.CONF
[general]
externip = 200.201.202.203
localnet=192.168.2.0/255.255.255.0

http://forum.voxilla.com/asterisk-support-forum/cant-get-external-sip-phone-register-asterisk-behind-nat-18209.html

ifold*CLI> sip show peers

Name/username              Host            Dyn Nat ACL Port     Status               
pc_hernad/pc_hernad        (Unspecified)    D          0        UNKNOWN              
hernad_zimbra/hernad_zimb  (Unspecified)    D          0        Unmonitored           
52/52                      (Unspecified)    D          0        Unmonitored           
51/51                      (Unspecified)    D          0        Unmonitored           
50/50                      89.146.135.78    D   N      5060     Unmonitored           <<<<<<<<<<<< officeze
34/34                      (Unspecified)    D          0        Unmonitored           
33/33                      (Unspecified)    D          0        Unmonitored           
32/32                      (Unspecified)    D          0        Unmonitored           
31/31                      (Unspecified)    D          0        Unmonitored           
20/20                      192.168.45.151   D          5060     Unmonitored           
13/13                      192.169.45.120   D          5060     Unmonitored           
12/12                      192.169.45.120   D          5060     Unmonitored           
11/11                      (Unspecified)    D          0        Unmonitored           
13 sip peers [Monitored: 0 online, 1 offline Unmonitored: 4 online, 8 offline]

SIP.conf: device configuration - qualify

Syntax:

qualify=xxx|no|yes

where XXX is the number of milliseconds used. If yes the default timeout is used, 2 seconds.

If you turn on qualify in the configuration of a SIP device in sip.conf, Asterisk will send a SIP OPTIONS command regularly to check that the device is still online. If the device does not answer within the configured (or default) period (in ms) Asterisk considers the device off-line for future calls. This status can be checked by the SIPPEER function, and inversely this function will only provide status information for peers which have qualify=yes.

This feature may also be used to keep a UDP session open to a device that is located behind a network address translator (NAT). By sending the OPTIONS request, the UDP port binding in the NAT (on the outside address of the NAT/firewall device) is maintained by sending traffic through it. If the binding were to expire, there would be no way for Asterisk to initiate a call to the SIP device. This can be used in conjunction with the nat=yes setting.

By default chan_sip.c sends the qualify every 60 seconds. There is no way to change this value on a per peer basis. It is compiled into the chan_sip module. The value in qualfiy = represents the timeout after a packet is sent before we consider the peer to be unreachable. If the packet is not responded within 1 second, asterisk will keep trying until 7 packets have failed. At this point, asterisk won't try again until the next 60 cycle period completes. If a packet is lost, which can easily happen with UDP, there are 7 more packets which are transmitted. Additionally asterisk will keep trying every 60 seconds. So even if all 7 packets are lost, asterisk tries again at the next 60 second cycle. The number of retransmits and time between each qualify is defined in chan_sip.c

stavio sam sip qualify="yes" u global sekciju

sada sip show peers kaže:
ifold*CLI> sip show peers

Name/username              Host            Dyn Nat ACL Port     Status               
pc_hernad/pc_hernad        (Unspecified)    D          0        UNKNOWN              
hernad_zimbra/hernad_zimb  (Unspecified)    D          0        UNKNOWN              
52/52                      (Unspecified)    D          0        UNKNOWN              
51/51                      (Unspecified)    D          0        UNKNOWN              
50/50                      89.146.135.78    D   N      1024     OK (240 ms)           
34/34                      (Unspecified)    D          0        UNKNOWN              
33/33                      (Unspecified)    D          0        UNKNOWN              
32/32                      (Unspecified)    D          0        UNKNOWN              
31/31                      (Unspecified)    D          0        UNKNOWN              
20/20                      192.168.45.151   D          5060     OK (78 ms)           
13/13                      192.169.45.120   D          5060     OK (13 ms)           
12/12                      192.169.45.120   D          5060     OK (13 ms)           
11/11                      (Unspecified)    D          0        UNKNOWN              
13 sip peers [Monitored: 4 online, 9 offline Unmonitored: 0 online, 0 offline]

pod ovim podešenjem radi
[general]

srvlookup=yes
context=demo
;allowoverlap=no                        ; Disable overlap dialing support. (Default is yes)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to all)
canreinvite=yes
;srvlookup=yes
language=bs
localnet=192.168.45.0/255.255.255.0
externip=89.146.143.237             <<<<<<<<<<<<<<<< trenutna IP adresa router-wan-sa-1
disallow=all
allow=alaw
allow=gsm
qualify=yes

međutom može se ovo postaviti

 ;externhost=foo.dyndns.net           ; Alternatively you can specify an
                                      ; external host, and Asterisk will
                                      ; perform DNS queries periodically.  Not
                                      ; recommended for production
                                      ; environments!  Use externip instead
 ;externrefresh=10                    ; How often to refresh externhost if
                                      ; used 

podesio ekstenzije za officeze, vranici, beganovici:

[general]
[50]
type=friend
username=50
password=xxxx
host=dynamic
callerid="officeze" 
callgroup=3
nat=yes

[51]
type=friend
username=51
password=xxxx
host=dynamic
callerid="vranici" 
callgroup=3
pickupgroup=3
nat=yes

[52]
type=friend
username=52
password=xxxx
host=dynamic
callerid="beganovici" 
nat=yes

nakon #14426 podesio externhost:

[general]
srvlookup=yes
context=demo
;allowoverlap=no                        ; Disable overlap dialing support. (Default is yes)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to all)
canreinvite=yes
;srvlookup=yes
language=bs
localnet=192.168.45.0/255.255.255.0
;externip=89.146.143.237
externhost=internet.sigma-com.net          
;Specify how often (in seconds) a hostname DNS lookup should be performed for the value entered in 'externhost'. Default 10 seconds
externrefresh=10

disallow=all
allow=alaw
;allow=gsm
qualify=yes