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 3.2 years ago 1 Answer

Vitelity Setup

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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

Asterisk Users 3.3 years ago 12 Answers

IP address of remote SIP host

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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

Asterisk Users 3.4 years ago 0 Answers

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 3.4 years ago 2 Answers

CBeyond SIP/connect and Asterisk 1.8

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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

Asterisk Users 3.5 years ago 0 Answers

Sporadic one way audio problem

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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

Asterisk Users 3.7 years ago 1 Answer

tcp version of toronto - osaka doesn't work

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I'm trying to setup a simple tcp sip connection based on the toronto
osaka example in the Asterisk book. On the remote box (osaka) (1.8.9.0-rc1): [toronto]
type=friend
transport=tcp
secret=welcome
context=toronto_incoming
host=dynamic
disallow=all
allow=ulaw sip show peer toronto
* Name : toronto
Secret :
MD5Secret :
Remote Secret:

Context : toronto_incoming
........
Useragent : Asterisk PBX 10.1.0-rc1
Reg. Contact : sip:osaka@:5060;transport=TCP
On the home box (toronto) (10.1.0-rc1): register => tcp://toronto:welcome@officePBX/osaka
[osaka]
type=friend

Asterisk Users 3.7 years ago 4 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 3.8 years ago 3 Answers