Asterisk shutdown automatically when trying to make a call

Asterisk Asterisk Support
1

Whenever I try to make a call from MicroSIP to Zoiper, the server automatically disconnects. I am testing this on my local PC, and my configurations are as follows.

[167]
auth_type=userpass
type=auth
username=167
password=167

[167]
type=aor
qualify_frequency=60
max_contacts=100000
remove_existing=yes
qualify_timeout=3.0
authenticate_qualify=no

[167]
context=internal
auth=167
aors=167
type=endpoint
language=en
deny=0.0.0.0/0.0.0.0
trust_id_inbound=yes
send_rpid=no
transport=tcp_transport
rtcp_mux=no
call_group=
pickup_group=
disallow=all
allow=ulaw,alaw,gsm
mailboxes=300
permit=0.0.0.0/0.0.0.0
ice_support=no
use_avpf=no
dtls_cert_file=
dtls_private_key=
dtls_ca_file=
dtls_setup=actpass
dtls_verify=no
media_encryption=no
message_context=
subscribe_context=
allow_subscribe=yes
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
direct_media=no
media_use_received_transport=no
callerid=“linuxhelp” <167>

[168]
auth_type=userpass
type=auth
username=168
password=168

[168]
type=aor
qualify_frequency=60
max_contacts=100000
remove_existing=yes
qualify_timeout=3.0
authenticate_qualify=no

[168]
context=internal
auth=168
aors=168
type=endpoint
language=en
deny=0.0.0.0/0.0.0.0
trust_id_inbound=yes
send_rpid=no
transport=tcp_transport
rtcp_mux=no
call_group=
pickup_group=
disallow=all
allow=ulaw,alaw,gsm
mailboxes=300
permit=0.0.0.0/0.0.0.0
ice_support=no
use_avpf=no
dtls_cert_file=
dtls_private_key=
dtls_ca_file=
dtls_setup=actpass
dtls_verify=no
media_encryption=no
message_context=
subscribe_context=
allow_subscribe=yes
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
direct_media=no
media_use_received_transport=no
callerid=“linuxhelp2” <168>

[tcp_transport]
type=transport ; Must be of type ‘transport’
protocol=tcp
bind=0.0.0.0
tos=cs3
cos=3
allow_reload=false
[udp_transport]
type=transport ; Must be of type ‘transport’
protocol=udp
bind=0.0.0.0
tos=cs3
cos=3
allow_reload=false

2

You need to provide logs.

Your subject suggests a crash, for which we would want backtraces obtained and submitted as part of a bug report.

Your body suggests the call was simply terminated early, for which we would require logs, here, to identify which side closed and the reason they did.

There are no valid reasons for such a high figure.

This is pointless with a large max_contacts.

You haven’t included the dialplan.

3

[internal]
exten => 167,1,Dial(PJSIP/167)
exten => 168,2,Dial(PJSIP/168)

my dialplan

4

[Feb 19 10:39:31] NOTICE[4141]: res_pjsip_session.c:4030 new_invite: 168: Call (TCP:192.168.18.225:51921) to extension ‘607’ rejected because extension not found in context ‘internal’.

when i make call from 608 from my device to another it provide me this error
i update the dialplan look like this
[external]
exten => 167,1,Dial(PJSIP/167)
[internal]
exten => 168,2,Dial(PJSIP/168)

my pjsip configuration now

[167]
auth_type=userpass
type=auth
username=167
password=167

[167]
type=aor
qualify_frequency=60
max_contacts=1
remove_existing=yes
qualify_timeout=3.0
authenticate_qualify=no

[167]
context=external
auth=167
aors=167
type=endpoint
language=en
deny=0.0.0.0/0.0.0.0
trust_id_inbound=yes
send_rpid=no
transport=tcp_transport
rtcp_mux=no
call_group=
pickup_group=
disallow=all
allow=ulaw,alaw,gsm
mailboxes=300
permit=0.0.0.0/0.0.0.0
ice_support=no
use_avpf=no
dtls_cert_file=
dtls_private_key=
dtls_ca_file=
dtls_setup=actpass
dtls_verify=no
media_encryption=no
message_context=
subscribe_context=
allow_subscribe=yes
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
direct_media=no
media_use_received_transport=no
callerid=“linuxhelp” <167>

[168]
auth_type=userpass
type=auth
username=168
password=168

[168]
type=aor
qualify_frequency=60
max_contacts=1
remove_existing=yes
qualify_timeout=3.0
authenticate_qualify=no

[168]
context=internal
auth=168
aors=168
type=endpoint
language=en
deny=0.0.0.0/0.0.0.0
trust_id_inbound=yes
send_rpid=no
transport=tcp_transport
rtcp_mux=no
call_group=
pickup_group=
disallow=all
allow=ulaw,alaw,gsm
mailboxes=300
permit=0.0.0.0/0.0.0.0
ice_support=no
use_avpf=no
dtls_cert_file=
dtls_private_key=
dtls_ca_file=
dtls_setup=actpass
dtls_verify=no
media_encryption=no
message_context=
subscribe_context=
allow_subscribe=yes
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
direct_media=no
media_use_received_transport=no
callerid=“linuxhelp2” <168>

[tcp_transport]
type=transport ; Must be of type ‘transport’
protocol=tcp
bind=0.0.0.0
tos=cs3
cos=3
allow_reload=false
[udp_transport]
type=transport ; Must be of type ‘transport’
protocol=udp
bind=0.0.0.0
tos=cs3
cos=3
allow_reload=false

5

So, extension 167 does a Dial() at priority 1 and then stops when the call
ends.

Extension 168 has no priority 1, so it cannot work.

Change that 2 to a 1 and try again.

Antony.

6

The error message is self explanatory.

The other problem also exists, but the incoming call isn’t for 168.

7

Thank you, the problem has been solved, but I want to ask one more thing. I’m making a call with the same IP address. If I want to make a call from one IP address to another, how is it possible?

[167]
auth_type=userpass
type=auth
username=167
password=167

[167]
type=aor
qualify_frequency=60
max_contacts=1
remove_existing=yes
qualify_timeout=3.0
authenticate_qualify=no

[167]
context=internal
auth=167
aors=167
type=endpoint
language=en
deny=0.0.0.0/0.0.0.0
trust_id_inbound=yes
send_rpid=no
transport=tcp_transport
rtcp_mux=no
call_group=
pickup_group=
disallow=all
allow=ulaw,alaw,gsm
mailboxes=300
permit=0.0.0.0/0.0.0.0
ice_support=no
use_avpf=no
dtls_cert_file=
dtls_private_key=
dtls_ca_file=
dtls_setup=actpass
dtls_verify=no
media_encryption=no
message_context=
subscribe_context=
allow_subscribe=yes
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
direct_media=no
media_use_received_transport=no
callerid=“linuxhelp” <167>

[168]
auth_type=userpass
type=auth
username=168
password=168

[168]
type=aor
qualify_frequency=60
max_contacts=1
remove_existing=yes
qualify_timeout=3.0
authenticate_qualify=no

[168]
context=internal
auth=168
aors=168
type=endpoint
language=en
deny=0.0.0.0/0.0.0.0
trust_id_inbound=yes
send_rpid=no
transport=tcp_transport
rtcp_mux=no
call_group=
pickup_group=
disallow=all
allow=ulaw,alaw,gsm
mailboxes=300
permit=0.0.0.0/0.0.0.0
ice_support=no
use_avpf=no
dtls_cert_file=
dtls_private_key=
dtls_ca_file=
dtls_setup=actpass
dtls_verify=no
media_encryption=no
message_context=
subscribe_context=
allow_subscribe=yes
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
direct_media=no
media_use_received_transport=no
callerid=“linuxhelp2” <168>

[tcp_transport]
type=transport ; Must be of type ‘transport’
protocol=tcp
bind=0.0.0.0
tos=cs3
cos=3
allow_reload=false
[udp_transport]
type=transport ; Must be of type ‘transport’
protocol=udp
bind=0.0.0.0
tos=cs3
cos=3
allow_reload=false

8

I want to make the IP address dynamic, so that a call can be made from one extension to another, regardless of the IP address. For example, if extension 1 is connected to an IP address, suppose 123.456.51, and the second extension is connected to another IP address, suppose 145.543.68, I want to be able to make a call between them. Currently, the call is only possible if both extensions have the same IP address. How can this be made possible?

9

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