Error while making a call

[Albo]

I have an Asterisk instance inside a Docker machine, the machine’s network is set to network_mode: host.
I connect to the telephony provider with an IP address, and manage to receive calls, but when I try to make calls from the switchboard, I get an error.

  == Using SIP RTP CoS mark 5
    -- Called out-call/<phone-number>
[Mar  9 14:33:28] WARNING[78]: chan_sip.c:4140 retrans_pkt: Retransmission timeout reached on transmission 1ef6442f0be524a22c4ed4d61b31091a@<isp-ip> for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response
[Mar  9 14:33:28] WARNING[78]: chan_sip.c:4164 retrans_pkt: Hanging up call 1ef6442f0be524a22c4ed4d61b31091a@<isp-ip> - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).

The phone provider claims that the request does not reach him at all, and it has probably already been blocked by me.
How can I tell what is causing this block?

Below are my configuration files

sip.conf

[general]
disallow=all
allow=ulaw
allow=alaw
allow=gsm

language=he
context=from-trunks
allowguest=no
canreinvite=no
qualify=yes
insecure=port,invite

nat=force_rport,comedia
externip=<my-ip>
localnet=192.168.1.0/255.255.255.0
transport=udp

#include sip_trunks.conf

sip_trunks.conf

[out-call]
type=friend
insecure=port,invite
host=<isp-ip>
host=<isp-second-ip>
fromdomain=<isp-ip>
context=from-trunks
allow=all

I should mention that I am new to Asterisk, if more information is needed I would be happy to receive guidance.

Thanks in advance

[david551]

@Albo

I should mention that I am new to Asterisk

You should not be using chan_sip, as it will be completely dropped in the next release.

You would need to monitor the network outbound of Asterisk to find out where it was last seen.

You could also use tcpdump, probably producing a pcap, for wireshark, to look for ICMP destination unreachable packets, or other errors.

@Albo

host=<isp-ip>
host=<isp-second-ip>

You can’t have multiple host lines.

@Albo

allow=all

This could result in the request being too large and a fragmentation required ICMP being returned. In some versions of Asterisk, it will result in a request without SDP, which Asterisk won’t be prepared to handle. You should always limit your codecs to the ones you intend to use. I’d assume PSTN In Israel would use alaw, so that is the only one I’d probably set.

Specifying:

type=friend (rather than peer);
insecure=port
canreinvite (rather than directmedia)
nat=force_rport,comedia (on everything)

tend to suggest you have copied an example without looking at what it means.

[Albo]

Indeed, you are right…
Probably the duplication of the host and allow all caused the problem.
Thank you very much, you helped me a lot.