* You are viewing Posts Tagged ‘dynamic context’

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

Problem with SendDTMF

Hello,

I am having a problem with SendDTMF – it is not sending the numbers
properly during the phone call.. I want the numbers key to to be
pressed/sent automatically after 3 seconds during a phone call.

I use software phone to test it… when I dialed 501, I cant hear anything
for about 10 seconds (this is because of SendDTMF) and then I can hear
the operator saying to enter the numbers but SendDTMF already did it?!

Asterisk server are connected to voip.ms provider.

I have spent many hours trying to get to work, how to fix this issue?

See the configuration and debug log below:

extensions.conf
================
[test]
exten => 501,1,Set(CALLERID(num)=004471XXXXXXX)
exten => 501,n,Dial(SIP/+44797XXXXXX@voipms,30,M(sendnumber)t)
exten => 501,n,Hangup()

[macro-sendnumber]
exten => s,1,Wait(3)
exten => s,n,SendDTMF(www0w7w8w8wXwXwXwXwXwX)

sip.conf
==========
[general]
context=default
tcpbindaddr=0.0.0.0
dtmfmode = rfc2833
register => xxxxx:vxxxxx@london.voip.ms:5060

[test]
type=peer
secret=2xxx
host=dynamic
context=test

[voipms]
canreinvite=no
host=london.voip.ms
secret=xxxxxx
type=peer
username=135xxx ;your account
disallow=all
allow=gsm
; allow=g729 ; Uncomment if you support G729
fromuser=135xxx
insecure=invite
trustrpid=yes
sendrpid=yes
nat=yes
dtmfmode=rfc2833

debug:
=====
== Using SIP RTP CoS mark 5

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 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (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

Park and PARKINGDYNAMIC

I have been trying to get the dynamic parking working.

For some reason when I park a call using this method the console says it is
using the default parking context not the one I am trying to specidfy. It
also is playing the parked extension to the caller. I am transfering the
call to an extension that is doing a goto to the context below. Any ideas
or examples on how to make this work. What I need to be able to do is have
multiple parking lots using the same extension pools but seperated by a
dynamic context of ${account}-Lot. So that each office suite cant cross
pickup another groups parked calls while using the same number pool of
110-120. I need the dynamic option as all of our calls are database driven
and we can’t add a seperate entry per customer to the feautres.conf.

[MSIP-DynPark]
exten => s,1,NoOp(Dynamic Parking)
exten => s,n,NoOp(Return Parked Call)
exten => s,n,GoTo(${CUT(${l_ndeContext}-ndeArgs,~,1)},1)

exten => _XXX,1,Set(PARKINGDYNAMIC=parkinglot_small)
exten => _XXX,n,Set(PARKINGDYNEXTEN=110)
exten => _XXX,n,Set(PARKINGDYNPOS=111-120)
exten => _XXX,n,Set(PARKINGDYNCONTEXT=${account}-Lot)
;exten => _XXX,n,Set(PARKINGEXTEN=99)
exten => _XXX,n,Park()

[MSIP-DynParkPickup]
exten => _NXX,1,ParkedCall(${EXTEN},${account}-Lot)
exten => _NXX,hint,park:$EXTEN@${account}-Lot

Thanks

Bryant

Help_In Voicemail , vedio play but voice is not here out.

Hi all,

I am using to Xlite to save video voice mail.

when i retreive it, then only video show , no voice is here out.

Plz tell me where ,i am wrong , and how i can able to see video plus here audio in voice mail box.

I did following configuration

In Sip.conf

videosupport=yes

[phone1]
type=friend
host=dynamic
context= employees
mailbox=101@default
callerid=”phone1<101>”
disallow=all
allow=ilbc
allow=g729
allow=gsm
allow=g723
allow=ulaw
allow=alaw
allow=adpcm
allow=h263p
allow=h264
allow=h263

[phone2]
 type=friend
host=dynamic
context= employees
mailbox=102@default
callerid=”phone2<102>”
disallow=all
allow=ilbc
allow=g723
allow=gsm
allow=g723
allow=ulaw
allow=alaw
allow=adpcm
allow=h263
allow=h263p
allow=h261

In extension.conf

exten => 102,1,VoiceMail( 102@default,u )

exten => 102,n,Hangup()

exten => 704,1,VoiceMailMain()

In voicemail.conf

format=h263|alaw|gsm

101 =>1234,phone1
102 =>1234,phone2

I call to 102 and save video voice mail

check in 102/INBOX

    267 Dec 29 13:22 msg0020.txt
   61503 Dec 29 13:22 msg0020.h263

    330 Dec 29 13:22 msg0020.gsm
  1600 Dec 29 13:22 msg0020.alaw

My Xlite phone codecs

aLaw,uLaw,H.263

Thanks&Regards

Durgesh Mishra

Help_video call not run

Hi all

In sip.conf

i take as

[general]

videosupport=yes

                        

[phone1]
type=friend
host=dynamic
context= employees
disallow=all
allow=ilbc
allow=g729
allow=gsm
allow=g723
allow=ulaw
allow=alaw
allow=adpcm
allow=h263p
allow=h264
allow=h263

[phone2]
 type=friend
host=dynamic
context= employees
disallow=all
allow=ilbc
allow=g729
allow=gsm
allow=g723
allow=ulaw
allow=alaw
allow=adpcm
allow=h263p
allow=h261
allow=h263

in extension.conf

[employees]

exten => 101,1,Dial(SIP/phone1,10)

exten => 102,1,Playback(song2_check)

in /var/lib/asterisk/sounds/en

i store song2_check file(which is video file ,which has audio format   MPEG Layer 3)

i dial 102 from 101 —-
 phone 101(xlite)  has following codec support for H623 H623+

check log as

[Dec 20 18:38:01] WARNING[10533] file.c: File song2_check does not exist in any format
[Dec 20 18:38:01] WARNING[10533] file.c: Unable to open song2_check (format 0x180400 (ilbc|h263|h263p)): No such file or directory

phone1 goes just hung up. no vedio play

I want to play video file. Plz tell me ,where i am wrong ,and how i can do it.

thanks