[Asterisk-Users] inter asterisk

David J Carter david.carter at codepipe.com
Sun Feb 6 05:18:12 MST 2005


One thing I do on remote sites is set up a soft phone so I can call myself,
this proves out the link and quality before anything else. DIAX id good for
this as you can connect to multiple sites, also good to see if you have
problems before anyone else calls you to say there is a problem.

It also helps in cases like this, if your return quality is good then the
possible fault lies with the ZAP interface.

Process of elimination, works for me every time.

Regards

Dave

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Ousmane Doukara
Sent: 06 February 2005 08:46
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] inter asterisk


Hi,
I am trying to forward calls to another * server with IAX
Here is What I want to Do
1- Call SERVER1, let say at 51412345678
2- SERVER1 should transfer the call to SERVER2 in a remote location
3- SERVER2 Receive the call and transfer it to the PSTN number.

I have one X100P  card on each machine. What is happening is that when the
remote party picks up the phone, all he can hear
is a weird sound.

CONFIGS:

 SERVER1:
  zaptel.conf
   ---------------------------------------------
   ~ [channels]
   ~ language=fr
   ~ context=montréal
   ~ signalling=fxs_ks
   ~ usercallerid=yes
   ~ callwaiting=yes
   ~ threewaycalling=yes
   ~ transfer=yes
   ~ cancellforward=yes
   ~ echocancel=yes
   ~ echocancelwhenbridged=yes
   ~ echotraining=yes
   ~ relaxdtmf=yes
   ~ busydetect=yes
   ~ busycount=4
   ~ callprogress=yes
   ~ group=1
   ~ channel=>1
   -------------------------------------------------- (same for SERVER2)

  IAX.conf
   ------------------------------------------------
   ~ [general]
   ~ bindport=4569
   ~ delayreject=yes
   ~ language=fr
   ~ allow=all
   ~ jutterbuffer=no
   ~ register => username:password at server2.domain.com
   ~ tos=lowdelay
   ~ autokill=yes
   ~
   ~ [quebec]
   ~ type=friends
   ~ username = username
   ~ password=password
   ~ context=montréal
   ~ host=Dynamic
   ~ secret = password
   ~ disallow = all
   ~ allow=ulaw
   ~ allow=gsm

  extensions.conf
   ------------------------------------------(Same for SERVER2 but no
registration)
   ~ [general]
   ~ static=yes
   ~ writeprotect=yes
   ~ autofallthrough=yes
   ~ [montréal]
   ~ exten=>s,1,Answer
   ~ exten=>s,2,Playback(message-transfer)
   ~
exten=>s,3,Dial(IAX2/username:password at SERVER2.DOMAIN.COM/51412345678 at montré
al) ; always the same number
   ~ exten=>s,4,Hangup



My remote server receive the call, answer the line and then
Dial(ZAP/1/51412345678). So far so good. But when 51412345678 pickup the
phone,
all she can hear is a weird sound.
What am I doing wrong ?




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