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has anyone tested the vst1000<font size="5"><strong></strong></font>
SIP phone from pcphoneline ?<br>
<a class="moz-txt-link-freetext" href="http://www.pcphoneline.com/">http://www.pcphoneline.com/</a><br>
<br>
Doug<br>
<br>
Jerry Roy wrote:<br>
<blockquote type="cite"
cite="midC0E23A9B1942314FAEC0134C189CF79BC5B8B9@elm.ent.gric.com">
<pre wrap="">Hi All,
Looking for a recommendation. I was hoping to purchase a * "KIT" for a
small office. I have 4 lines and 4 extensions need phones so I need 4
phones. What phones would many of you recommend? Can you refer me to any
companies that have built a kit I can plugin and configure?
Thanks,
Jerry Roy
RemoteHand, Inc.
562-305-9545
-----Original Message-----
From: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>
[<a class="moz-txt-link-freetext" href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Rich
Adamson
Sent: Friday, August 27, 2004 7:15 AM
To: Asterisk-a-users-list
Subject: [Asterisk-Users] sip change?
Just upgrade from July 12th cvs to last night CVS-HEAD-08/27/04-00:00:09
When call comes in and is sent to a Cisco 7960, I see:
-- Executing Dial("SIP/3008-9a9b", "SIP/3000|15") in new stack
-- Called 3000
Aug 27 08:13:25 WARNING[1092070192]: chan_sip.c:676 retrans_pkt: Maximum
retries
exceeded on call <a class="moz-txt-link-abbreviated" href="mailto:033f41c2187409b13ca364502ea9434e@206.222.193.101">033f41c2187409b13ca364502ea9434e@206.222.193.101</a> for
seqno 102
(Critical Request)
== No one is available to answer at this time
-- Executing VoiceMail2("SIP/3008-9a9b", "u3000") in new stack
-- Playing 'voicemail/default/3000/greet' (language 'en')
-- Playing 'vm-isunavail' (language 'en')
but the phone doesn't ring. The 7960 is registered and can place
outbound calls. Same with multiple 7960's.
Did I miss a mandatory config change, or is sip broken?
Rich
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</pre>
</blockquote>
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