Softswitch to Asterisk call termication configuration

Asterisk Asterisk SIP
1

Hai ,

Please check my sip.conf for incoming call recieving from softswitch xxx.xx.xx.x

register =>:@xxx.xx.xxx.x

registertimeout=120

type=friend
fromuser=115.112.144.18
host=192.69.136.4
canreinvite=no
qualify=no
dtmfmode=RFC2833
context=intenal
disallow=all
allow=g729
allow=ulaw
allow=alaw
port=5060
canreinvite=no

Its not working…Please help me to do my prorper configuration for recieving and incomin calls from softswitch to Asterisk

2

You most likely need to use the “insecure” option to match based on IP address. Providing the output of “sip set debug on” with a call attempt will confirm this. As well in the future please try to describe further how something isn’t working so that everyone here can better help.

3

No need to use register= as using “callbackextension” in the trunk config is more convenient.
But as soon as you don’t have username/password in the trunk configuration I suspect that you don’t need registration at all.

fromuser=115.112.144.18 seems to be a mistake
context=intenal seems to be a mistake

As jcolp said more detailed description and clear readable sip debug are required.

EDIT: are we talking about Viatalk? They have an example published on their web site:
https://support.viatalk.com/index.php?_a=knowledgebase&_j=questiondetails&_i=123

4

[general]
;register =>extension:@softswitchip/extension
allowguest=no

[viatalk]
context=global
type=peer
fromuser=
username=
authuser=
secret=
host=softswitch ip
fromdomain=Asterisk server IP
nat=no
canreinvite=yes
insecure=very
qualify=yes
dtmfmode=inband
dtmf=inband

This is my sip.conf file and status show
viatalk 192.69.136.4 No No 5060 UNREACHABLE
1 sip peers [Monitored: 0 online, 1 offline Unmonitored: 0 online, 0 offline]

5

This is slightly corrected Viatalk example:

[viatalk]
context=global ;make sure it exists!
type=peer
fromuser=YOUR NUMBER
defaultuser=YOUR NUMBER
secret=YOUR PASSWORD
host= 192.69.136.4 ; PROXY IP/name
fromdomain= 192.69.136.4 ; PROXY IP/name
callbackextension =  ; any working extension here
nat=no
directmedia=no
insecure=invite
qualify=yes
dtmfmode=rfc2833

Complete configuration, reload, check your sip debug during the registration process.
Once registered - make a call and check your sip debug for incoming INVITE, etc.

6

Hai Andrewz,

I am getting Error like this
[Dec 15 12:31:13] NOTICE[41326]: chan_sip.c:15754 sip_reg_timeout: – Registration for ‘1800919544678732@192.69.136.4’ timed out, trying again (Attempt #2)
Really destroying SIP dialog ‘5a4cd10a02fcaf651f7ee85d047c62c0@[fe80::d6ae:52ff:fee5:6521]’ Method: REGISTER
divoxCLI> sip set debug off
SIP Debugging Disabled
[Dec 15 12:31:13] WARNING[41326]: chan_sip.c:15925 transmit_register: Probably a DNS error for registration to 1800919544678732@192.69.136.4, trying REGISTER again (after 120 seconds)
[Dec 15 12:31:13] NOTICE[41326]: chan_sip.c:15754 sip_reg_timeout: – Registration for ‘1800919544678732@192.69.136.4’ timed out, trying again (Attempt #2)
divoxCLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
123/123 115.112.144.18 Auto (No) No 5060 UNREACHABLE
viatalk/1800919544678732 192.69.136.4 No No 5060 UNREACHABLE

I set registerattempts=1
registertimeout=120

But Still exist this above error

7

Again: check your sip debug, not a regular log which is useless for client-to-provider troubleshooting. Speak SIP! We need to see clear readable SIP message exchange.
Make sure you’re using the right proxy IP or name, check with your provider!

8

This error:
Dec 15 12:31:13] NOTICE[41326]: chan_sip.c:15754 sip_reg_timeout: – Registration for ‘1800919544678732@192.69.136.4’ timed out, trying again (Attempt #2)

means that You still want to register, so You didn’t remove these lines:
register =>:@xxx.xx.xxx.x
registertimeout=120

If You have static IP for peer 192.69.136.4, You don’t have to register. It’s only used when You have dynamic IP, and asterisk have to know what IP it should accept from as an authorized peer. Secondly, if You don’t really need sip login, and pass, so Your softwsitch is a server/peer not a sip client Your config can look like this:

[your_peer]
host=xxx.xxx.xxx.xxx
port=5060
nat=force_rport,comedia
canreinvite=no
insecure=port,invite
dtmfmode=rfc2833
disallow=all
allow=g729
allow=alaw
allow=elaw
type=peer
context=internal - this context must be in extensions.conf

9

Hai maathoh&Andrew,

Thank You maath and Andrew

Now My Asterisk call is working let me explaing again my structure

caller Dialed from a Dialer(Softswitch)12345—>call comes in Asterisk—>Dialed 1122 extension (callee)

Now Again some issues i am facing

  1. call 1122 Answerd the call caller getting the from calee voice
    2.While caller speaks its returns to its own speaker ,callee not getting the voice
10

I don’t understand what You wrote. Please be more precise.

11

issue,

extension1 2000
extemsion2 2010

caller–2010----------------------------------------------------------->calle(2000)
call answered
voice from calle to caller is ok
but voice from caller to calle is not getting
its hearing itself not not getting in calle speaker

– SIP/2000-00001226 answered SIP/192.69.136.4-00001225
– Channel SIP/2000-00001226 joined ‘simple_bridge’ basic-bridge
– Channel SIP/192.69.136.4-00001225 joined ‘simple_bridge’ basic-bridge
> 0x7eff3803bab0 – Probation passed - setting RTP source address to 192.69.136.89:12628
> 0x7efff8006750 – Probation passed - setting RTP source address to 137.97.197.188:10000
from caller to callee sound is not paasing
but from calle to caller is passing
while taking in caller its getting an echo