I drop the call

Asterisk Asterisk General
1

good morning

I have a problem outside of my LAN I fall calls.
this is the problem:

- Executing [1000@esterno:1] Playback("SIP/1002-00000001", "demo-echotest") in new stack -- <SIP/1002-00000001> Playing 'demo-echotest.gsm' (language 'it') [Dec 16 10:34:31] WARNING[2226]: chan_sip.c:3856 retrans_pkt: Retransmission timeout reached on transmission NzU3OTc1ZDUwMzM5OGViMjE2YjY2MzU3MGRmODU1Njg. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6399ms with no response [Dec 16 10:34:31] WARNING[2226]: chan_sip.c:3885 retrans_pkt: Hanging up call NzU3OTc1ZDUwMzM5OGViMjE2YjY2MzU3MGRmODU1Njg. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). == Spawn extension (esterno, 1000, 1) exited non-zero on 'SIP/1002-00000001'

This is the configuration of my interior:

[1002] type = friend callerid = interno 1002 secret = ********* host=dynamic canreinvite = no dtmfmode = rfc2833 disallow = all allow = ulaw allow = alaw allow = speex allow = gsm transport = udp context=esterno qualify = yes language=it Directrtpsetup = yes
the router I opened all the doors, I do not know how to solve this problem.
Thanks for your attention given to me

2

I would use the proper name for canreinvite, as it is possible that this has gone from deprecated to removed. In any case it should be the default.

I would also not use directrpsetup, as I don’t believe that that is production quality.

Apart from that, you need to get the relevant traces to see which packet is not producing a response, and confirm all the right addresses are being sent. It might actually be an ACK due the peer sending an unrouteable Contact header value.

In future please ask for support on forums with Support in their name, not ones intended for general discussion.

3

thanks for the quick response, I removed the items that you told me, but I still have the same problem, you can send me your configuration inside?

4

You need to add to your general config in the sip.conf file the proper setting for NAT Support.

Example :

externaddr = 12.34.56.78 (Replace this for your public IP).

; localnet=192.168.0.0/255.255.0.0 ; RFC 1918 addresses
; localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
; localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
; localnet=169.254.0.0/255.255.0.0 ; Zero conf local network

; nat = no ; Do no special NAT handling other than RFC3581
; nat = force_rport ; Pretend there was an rport parameter even if there wasn’t
; nat = comedia ; Send media to the port Asterisk received it from regardless
; ; of where the SDP says to send it.
; nat = auto_force_rport ; Set the force_rport option if Asterisk detects NAT (default)
; nat = auto_comedia ; Set the comedia option if Asterisk detects NAT

5

[quote=“ambiorixg12”]You need to add to your general config in the sip.conf file the proper setting for NAT Support.

Example :

externaddr = 12.34.56.78 (Replace this for your public IP).

; localnet=192.168.0.0/255.255.0.0 ; RFC 1918 addresses
; localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
; localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
; localnet=169.254.0.0/255.255.0.0 ; Zero conf local network

; nat = no ; Do no special NAT handling other than RFC3581
; nat = force_rport ; Pretend there was an rport parameter even if there wasn’t
; nat = comedia ; Send media to the port Asterisk received it from regardless
; ; of where the SDP says to send it.
; nat = auto_force_rport ; Set the force_rport option if Asterisk detects NAT (default)
; nat = auto_comedia ; Set the comedia option if Asterisk detects NAT
[/quote]

many thanks to all problem solved