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<font face="Helvetica, Arial, sans-serif">Danny,<br>
<br>
thank you for you feedback.<br>
<br>
I have the following setting in sip.conf :<br>
<br>
limitonpeer = yes<br>
<br>
and for every sip peer definition I have :<br>
<br>
asterisk*CLI> sip show peer test1<br>
<br>
* Name : test1<br>
Realtime peer: Yes, cached<br>
Secret : <Set><br>
MD5Secret : <Not set><br>
Context : from-TEST<br>
Subscr.Cont. : <Not set><br>
<snip><br>
Transfer mode: open<br>
CallingPres : Presentation Allowed, Not Screened<br>
Callgroup : <br>
Pickupgroup : <br>
Mailbox : <br>
VM Extension : asterisk<br>
LastMsgsSent : 32767/65535<br>
Call limit : 4<br>
<snip><br>
<br>
<br>
With a call limit of 4, I think it must be possible to transfer a call,
no ?!<br>
<br>
<br>
Jonas.<br>
<br>
</font><br>
On 07/02/2010 03:02 PM, Danny Nicholas wrote:
<blockquote cite="mid:201007021248.o62Cm9Ud008551@mail.debsinc.com"
type="cite"><o:smarttagtype
namespaceuri="urn:schemas-microsoft-com:office:smarttags"
name="PersonName">
<div class="Section1">
<p class="MsoNormal"><font size="2" color="navy" face="Arial"><span
style="font-size: 10pt; font-family: Arial; color: navy;">A good
possibility is that you have an
over-restrictive call-limit (or whatever it’s called in your branch)
that
is “filling the bucket” on the incoming call and not allowing a
transfer.<o:p></o:p></span></font></p>
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