Why does asterisk drop outgoing call?

A

Alexey Korolev2015-01-21 11:05:11

Asterisk

Alexey Korolev, 2015-01-21 11:05:11

Good day,
I installed an asterisk, configured trunks, extensions, and immediately ran into a problem, the asterisk dials the subscriber and immediately drops the call.
in the CLI the following:

== Using SIP CoS mark 4
-- Unregistered SIP '101'
-- Registered SIP '101' at 194.xx.xx.178:11621
== Using SIP RTP CoS mark 5
-- Executing [[email protected]:1] NoOp ("SIP/101-0000000c", "") in new stack
-- Executing [[email protected]:2] Dial("SIP/101-0000000c", "SIP/zadarma/7963xxxxxxx") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/zadarma/7963xxxxxxx
-- SIP/zadarma-0000000d is ringing
-- SIP/zadarma-0000000d is making progress passing it to SIP/101-0000000c
== Spawn extension (office, 87963xxxxxxx, 2) exited non-zero on 'SIP/101-0000000c'

The server is behind NAT, ssh and sip ports are forwarded to it (22.5060)

Reply

Answer the question

In order to leave comments, you need to log in

[[+comments_count]] answer(s)

V

Vladimir, 2015-01-21
@rostel

perhaps there is no common codec, and the remaining ones cannot be recoded due to lack of a license

*CLI> core show translation
*CLI> sip set debug peer 101
*CLI> sip set debug peer zadarma

when making a call
, you also need to forward the range of UDP ports for RTP, look in /etc/asterisk/rtp.conf from where to where

�

Михаил Лялин, 2015-01-21
@mr_jok

habrahabr.ru/post/157545
voxlink.ru/kb/asterisk-configuration/asterisk-nat-...