[Asterisk-Users] X100P Inbound Issue

Yiannis Costopoulos, Web2Net Solutions Ltd. yiannis at w2ns.com
Mon Jul 26 01:31:43 MST 2004


Hi,

    I think that the problem is with the codecs. Search the Wiki and the
list archives (through Google) to find what settings in sip.conf you need
for Budgetone and Sipura. The settings you need are *allow* and/or
*disallow*.

Yiannis.

  -----Original Message-----
  From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of
mpwspam-digiumlist at yahoo.com
  Sent: 26 July 2004 01:52
  To: asterisk-users at lists.digium.com
  Subject: [Asterisk-Users] X100P Inbound Issue


  Hello,

  After much searching of voip-info.org and google, I'm finally giving in
and asking the list.

  The setup I have is this:-

  Single X100P card in a Debian system
  Inbound/Outbound POTS line connects to the X100P
  Sipura 2000 and Budgetone 100 on the LAN
  1 Cordless and one conventional phone connected to the sipura
  Account on Stanaphone.com for eitherbound SIP calls.
  (I have other SIP accounts as well - all work flawlessly)

  I have a simple dialplan - an incoming call rings all phones and goes to
voicemail if not answered.

  When I dial '8' followed by a number - the call routes out via Stanaphone
fine.  No issues.
  When I call the Stanaphone number - all phones ring as expected, I can
answer the call and talk fine.  no issues at all.

  When I dial '9' followed by a number - the call routes out via the POTS
line just fine. No issues.

  However, inbound calls on the POTS line are the issue.  When a call comes
in, * detects it and starts ringing all of the extensions.  However, when I
pickup the extension - it gets immediately disconnected.  Other SIP
extensions keep ringing - and the caller still hears the ring tone.  Caller
hangs up - SIP extensions keep ringing. Phone I picket up I now return to
the hook.  * then 'calls me back' !

  Does anybody have any idea what's going on?  I have put some snippets from
the configs below..  Any insight would be very much appreciated!

  Michael.

  EXAMPLE FROM: zapata.conf
  [channels]
  busydetect=1
  busycount=7
  callprogress=yes
  relaxdtmf=yes
  callwaiting=yes
  callwaitingcallerid=yes
  threewaycalling=yes
  transfer=yes
  cancallforward=yes
  usecallerid=yes
  echocancel=yes
  echocancelwhenbridged=yes
  rxgain=0.0
  txgain=0.0
  group=1
  pickupgroup=1-4
  immediate=no
  context=from-bell
  signalling=fxs_ks
  callerid=asreceived
  channel=1


  EXAMPLE FROM: extensions.conf
  [from-bell]
  exten => _.,1,Dial(SIP/001&SIP/002&SIP/003,30,t)
  exten => _.,2,Answer
  exten => _.,3,Wait(1)
  exten => _.,4,Voicemail(u099)
  exten => h,1,Hangup

  EXAMPLE FROM: sip.conf
  [002]                  ; Line 1 on adapter
  type=friend
  username=002
  secret=<some password>
  host=dynamic
  context=extensions
  mailbox=099
  incominglimlit=2
  canreinvite=no
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040726/912dbaad/attachment.htm


More information about the asterisk-users mailing list