<DIV>Hi,</DIV>
<DIV> </DIV>
<DIV>I had already come across some of that stuff (forgot to post that part of my sip.conf). Here is what I'm using right now:-</DIV>
<DIV> </DIV>
<DIV>FROM SIP.CONF GENERAL</DIV>
<DIV>disallow = all<BR>allow=ULAW<BR>allow=ALAW<BR>allow=GSM</DIV>
<DIV>canreinvite=no</DIV>
<DIV> </DIV>
<DIV>[001] ; Budgetone<BR>disallow = all<BR>allow=GSM<BR>allow=ULAW<BR>allow=ALAW<BR>allow=ilbc</DIV>
<DIV>...</DIV>
<DIV> </DIV>
<DIV>The Budgetone has the latest firmware.. As I mentioned - I'm not having issues internally or with SIP calls at all - only inbound PSTN calls..</DIV>
<DIV> </DIV>
<DIV>Many thanks,</DIV>
<DIV> </DIV>
<DIV>Michael.<BR><BR><B><I>"Yiannis Costopoulos, Web2Net Solutions Ltd." <yiannis@w2ns.com></I></B> wrote:</DIV>
<BLOCKQUOTE class=replbq style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid">
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<DIV><SPAN class=647032808-26072004><FONT face=Arial color=#0000ff size=2>Hi,</FONT></SPAN></DIV>
<DIV><SPAN class=647032808-26072004><FONT face=Arial color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=647032808-26072004> <FONT face=Arial color=#0000ff size=2>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*.</FONT></SPAN></DIV>
<DIV><SPAN class=647032808-26072004><FONT face=Arial color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=647032808-26072004><FONT face=Arial color=#0000ff size=2>Yiannis.</FONT></SPAN></DIV>
<DIV><SPAN class=647032808-26072004></SPAN> </DIV>
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<DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma size=2>-----Original Message-----<BR><B>From:</B> asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]<B>On Behalf Of </B>mpwspam-digiumlist@yahoo.com<BR><B>Sent:</B> 26 July 2004 01:52<BR><B>To:</B> asterisk-users@lists.digium.com<BR><B>Subject:</B> [Asterisk-Users] X100P Inbound Issue<BR><BR></FONT></DIV>
<DIV>Hello,</DIV>
<DIV> </DIV>
<DIV>After much searching of voip-info.org and google, I'm finally giving in and asking the list.</DIV>
<DIV> </DIV>
<DIV>The setup I have is this:-</DIV>
<DIV> </DIV>
<DIV>Single X100P card in a Debian system</DIV>
<DIV>Inbound/Outbound POTS line connects to the X100P</DIV>
<DIV>Sipura 2000 and Budgetone 100 on the LAN</DIV>
<DIV>1 Cordless and one conventional phone connected to the sipura</DIV>
<DIV>Account on Stanaphone.com for eitherbound SIP calls.</DIV>
<DIV>(I have other SIP accounts as well - all work flawlessly)</DIV>
<DIV> </DIV>
<DIV>I have a simple dialplan - an incoming call rings all phones and goes to voicemail if not answered.</DIV>
<DIV> </DIV>
<DIV>When I dial '8' followed by a number - the call routes out via Stanaphone fine. No issues.</DIV>
<DIV>When I call the Stanaphone number - all phones ring as expected, I can answer the call and talk fine. no issues at all.</DIV>
<DIV> </DIV>
<DIV>When I dial '9' followed by a number - the call routes out via the POTS line just fine. No issues.</DIV>
<DIV> </DIV>
<DIV>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' !</DIV>
<DIV> </DIV>
<DIV>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!</DIV>
<DIV> </DIV>
<DIV>Michael.</DIV>
<DIV> </DIV>
<DIV>EXAMPLE FROM: zapata.conf</DIV>
<DIV>[channels]</DIV>
<DIV>busydetect=1<BR>busycount=7<BR>callprogress=yes<BR>relaxdtmf=yes<BR>callwaiting=yes<BR>callwaitingcallerid=yes<BR>threewaycalling=yes<BR>transfer=yes<BR>cancallforward=yes<BR>usecallerid=yes<BR>echocancel=yes<BR>echocancelwhenbridged=yes<BR>rxgain=0.0<BR>txgain=0.0<BR>group=1<BR>pickupgroup=1-4<BR>immediate=no<BR>context=from-bell<BR>signalling=fxs_ks<BR>callerid=asreceived<BR>channel=1<BR></DIV>
<DIV> </DIV>
<DIV>EXAMPLE FROM: extensions.conf</DIV>
<DIV>[from-bell]<BR>exten => _.,1,Dial(SIP/001&SIP/002&SIP/003,30,t)<BR>exten => _.,2,Answer<BR>exten => _.,3,Wait(1)<BR>exten => _.,4,Voicemail(u099)<BR>exten => h,1,Hangup<BR></DIV>
<DIV>EXAMPLE FROM: sip.conf</DIV>
<DIV>[002] ; Line 1 on adapter<BR>type=friend<BR>username=002<BR>secret=<some password></DIV>
<DIV>host=dynamic<BR>context=extensions<BR>mailbox=099<BR>incominglimlit=2<BR>canreinvite=no<BR></DIV></BLOCKQUOTE></BLOCKQUOTE>