IAX Trunk issue.

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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…

Asterisk Users 3.1 years ago 1 Answer

Problem with SendDTMF

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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…

Asterisk Users 3.3 years ago 1 Answer

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

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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

Asterisk Users 3.3 years ago 2 Answers

Park and PARKINGDYNAMIC

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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…

Asterisk Users 3.5 years ago 2 Answers

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

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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

Asterisk Users 3.6 years ago 0 Answers

Help_video call not run

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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…

Asterisk Users 3.7 years ago 0 Answers

Help_video call not run

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Hi all In sip.conf i take as [general] videosupport=yes                                ; then UDPTL will flow to the remote device [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…

Asterisk Users 3.7 years ago 2 Answers

Confrence call is not make

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Hi, I am making confrence application. In sip.conf [phone1]
type=friend
host=dynamic
Takes an alphanumeric string.
context= employees [phone2]
type=friend
host=dynamic
context= employees [phone3]
type=friend
host=dynamic
context= employees In extension.conf [employees]
exten => 101,1,Dial(SIP/phone1,20,tT) exten => 102,1,Dial(SIP/phone2,20,tT) exten => 103,1,Dial(SIP/phone3,20,tT) exten => 777,1,MeetMe(777) In meetme.conf [rooms]
conf => 777 when i call 777 from phone1 ,its shows 603 declined. I check in CLI [Dec 6 17:46:58] WARNING[16264]: pbx.c:4088 pbx_extension_helper: No application 'MeetMe' for extension (employees, 777, 1)
== Spawn extension (employees, 777, 1) exited non-zero on 'SIP/phone1-00000000' Plz…

Asterisk Users 3.7 years ago 3 Answers

Still having trouble to configure gxw4108 with asterisk 1.8 need enlightenment

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Dear all,
I'm still having trouble using asterisk with the grandstream gxw4108,
in the gxw4108 I'm using 1 stage dialing in the profile1 I already
type my asterisk server address 192.168.14.80 and my grandstream IP is
192.168.101.184 here's my asterisk config files SIP.CONF
[1401]
type = friend
username = 1401
secret = 1401
host = dynamic
context = kantor-mtx
insecure = port
nat = yes
dtmfmode = rfc2833
canreinvite = yes
notifyringing = yes [1402]
type = friend
username = 1402
secret…

Asterisk Users 3.8 years ago 0 Answers

device state of SIP device is stucked into NOT_INUSE, and cannto be reverted to unavailable

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Hello Have a setup of asterisk with realtime SIP devices.
Trying to organise monitoring of my SIP devices. Once device
registered, its state becomes NOT_INUSE (result of
DEVICE_STATE(SIP/device) function).
Simulating of device breakage - powerdown it.
Waiting for a while (minute or two), retrieving
DEVICE_STATE(SIP/device) again - no changes! Awaited UNAVAILABLE. doing from CLI:
sip qualify peer device load
no result. What I did not configured? My sip.conf
[general]
context = default allowguest = no
bindport = 5060
bindaddr = 0.0.0.0 allowexternaldomains = no
allowoverlap = yes

Asterisk Users 3.8 years ago 0 Answers