* You are viewing Posts Tagged ‘HOST’

IAX Trunk issue.

I’m testing a few IAX trunk scenarios in a controlled lab. From server2 extension 5000 (Server IP Address 172.16.200.212) I dial 6001 which goes across the IAX trunk to server 1 (IP address 172.16.200.210). Instead of ringing the 6001 phone, it plays tt-weasels (the s extension). When I dial 6099 it also plays tt-weasels as it’s supposed to, but it’s not the tt-weasels under its extension. It also dials the s extension.

I only placed the s extension in the dial plan to verify that the traffic was going across the IAX trunk and hitting the correct context.

Any help would be greatly appreciated.

Thanks Mitch

Asterisk-1

IP Address 172.16.200.210

SIP.CONF

[6001]
type=friend
host=dynamic
context=internal_users
secret=xxxxxxx
nat=yes

[6002]
type=friend
host=dynamic
context=internal_users
secret=xxxxxxx
nat=yes

extensions.conf

[internal_users]
exten => 6000,1,Answer()
exten => 6000,2,Playback(hello-world)
exten => 6000,3,Hangup()
exten => 6001,1,Dial(SIP/6001)
exten => 6002,1,Dial(SIP/6002)
exten => 6099,1,Playback(tt-weasels)
exten => 6099,n,HangUp
exten => _5XXX,1,Dial(${IAXTrunk}/${EXTEN})
same => n,Hangup()
exten => s,1,Answer()
exten => s,n,Playback(tt-weasels)
exten => s,n,Hangup()

IAX.conf

[trunk-1]
type=friend
username=trunk-1
trunk=yes
requiretoken=no
secret=password
host=172.16.200.212
context=internal_users
auth=plaintext
disallow=all
;allow=ulaw
;allow=alaw
allow=gsm

Asterisk-2

IP Address 172.16.200.212

sip.conf

[5000]
type=friend
context=phones
host=dynamic
disallow=all
allow=ulaw
secret=xxxxxxx

extensions.conf

[phones]
exten => _60XX,1,Dial(IAX2/trunk-1)
exten => _X.,1,Dial(IAX2/trunk-1)
exten => 5000,1,Dial(SIP/${EXTEN})
exten => 5000,n,Hangup
same => n,Hangup()
exten => 5099,1,Playback(tt-monkeys)
exten => 5099,n,HangUp

iax.conf

[trunk-1]
type=friend
username=trunk-1
trunk=yes
requiretoken=no
secret=password
host=172.16.200.210
context=phones
auth=plaintext
disallow=all
;allow=ulaw
;allow=alaw
allow=gsm

Vitelity Setup

On Wed, May 23, 2012 at 7:57 AM, Gopalakrishnan N
wrote:
> Hi,
>
> I am unable to register vitelity SIP trunk, where its keep on sending
> registration request, and I am using Asterisk 1.4.39.2, my registration
> procedure as follows,
>
> sip.conf
>
> register => username:secret@sip41.vitelity.net:5060
>

We use viteity w/o registration like so:

[vitel-inbound]
type=friend
dtmfmode=auto
host=inbound24.vitelity.net
context=vitelity-inbound
allow=all
insecure=very

[vitel-outbound]
type=friend
dtmfmode=auto
host=outbound.vitelity.net
context=vitelity-outbound
allow=all
insecure=very

IP address of remote SIP host

Hi,

is it possible to get the SIP IP address of the remote (calling) party,
in the dialplan or (preferrably) in an AGI script?

(This sounded like a rather basic question to me, but I could not find
an answer…)

TIA & regards
Jakob

Can’t make Asterisk send authentication to remote peer on INVITE

This is a really simple problem that I just can’t get to work. There
are two Asterisk servers with the following sip user and peer. When a
call is attempted, Asterisk is not sending authentication details in
response to the 401. Note, if the secret is blank on 172.16.0.2 test,
the INVITE succeeds.

on 172.16.0.2:

[test]
type=friend
secret=abcde
host=dynamic
context=demo

on 172.16.0.1 :

[natty]
type=peer
host=172.16.0.2
fromuser=test
secret=abcde

originate SIP/natty/1234568 extension 200
== Using SIP RTP CoS mark 5
Audio is at 172.16.0.1 port 19486
Adding codec 0×2 (gsm) to SDP
Adding codec 0×4 (ulaw) to SDP
Adding codec 0×8 (alaw) to SDP
Adding non-codec 0×1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 172.16.0.2:5060:
INVITE sip:1234568@172.16.0.2 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.1:5066;branch=z9hG4bK59f50e30;rport
Max-Forwards: 70
From: “asterisk” ;tag=as1689b2b6
To:
Contact:
Call-ID: 2353cf0e59596e285c684b44220f8915@172.16.0.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2.1
Date: Sat, 14 Apr 2012 09:10:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1594270426 1594270426 IN IP4 172.16.0.1
s=Asterisk PBX 1.6.2.9-2ubuntu2.1
c=IN IP4 172.16.0.1
t=0 0
m=audio 19486 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

CBeyond SIP/connect and Asterisk 1.8

We have many installs running Asterisk 1.4 on CBeyond SIP/connect service, but recently did an Elastix 2.2 with Asterisk 1.8.7. This is behind a pfSense 2.0.1 firewall. With the settings from CBeyond, we don’t get a stable registration. With the settings we are using now, it works but incoming calls don’t always make it to Asterisk and calls over 10 minutes often drop. Seems we are sending multiple registration requests and occasionally we are getting a failure. Does anyone have an Asterisk 1.8 system on CBeyond with settings that work?

Here is what we are using now for Peer:

host=sipconnect.atl0.cbeyond.net
username=XXXXXXXXX
secret=XXXXXXX
srvlookup=no
type=peer
qualify=no
session-timers=refuse
insecure=port,invite
fromdomain=sipconnect.atl0.cbeyond.net
outboundproxy=sip-proxy.atl0.cbeyond.net
dtmfmode=inband
disallow=all
context=from-trunk
allow=ulaw
canreinvite=no
sendrpid=yes
trustrpid=yes

User:

type=user
host=sip-proxy-public.atl0.cbeyond.net
dtmfmode=auto
context=from-pstn
session-timers=refuse

Sporadic one way audio problem

Hi all again,

I’ve got a problem with sporadic one way audio calls, which means
sometimes I can’t hear the calling party (call is established, but audio
is missing). Today I received ~90 calls, one of them got this problem.

I’ve got two networks involved, without NAT:

- 192.168.1.X, in there one nic of my server and all the phones
- a private net to my provider, in there a nic of my server and the voice
switch of my provider

My sip.conf:
[general]
port=5060
bindaddr=0.0.0.0
language=de
allowguest=no
;echocancel=yes
;echotraining=yes
alwaysauthreject=yes
disallow=all
allow=alaw
deny=0.0.0.0/0.0.0.0
permit=XXX.XXX.X.X/29
permit=192.168.1.0/24
;jbenable=yes
;jbforce=yes
;jbmaxsize=20
;jbresyncthreshold=1000
tos=0×10
directmedia=no
nat=no
directrtpsetup=no

[provider]
type=peer
host=XXX.XXX.X.X
insecure=port,invite
context=XXXXXXXXX
dtmfmode=rfc2833
directmedia=no
nat=no
directrtpsetup=no
;qualify=300

[one-phone]
[10]
type=peer
context=XXXXXXXXX
secret=XXXXXX
host=dynamic
;qualify=300
directmedia=no
nat=no
directrtpsetup=no
dtmfmode=inband

Any help greatly appreciated!

Thanks,
Georg