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  • Committer: Package Import Robot
  • Author(s): Mark Purcell
  • Date: 2012-05-19 21:46:37 UTC
  • mfrom: (1.1.7)
  • Revision ID: package-import@ubuntu.com-20120519214637-la8rbrford5kj6m3
Tags: 1.1.0-1
http://bugs.debian.org/663282
http://bugs.debian.org/665789
* New upstream release 
  - Fixes "FTBFS with libccrtp-dev/2.0.2 from experimental" (Closes: #663282)
* NEW Maintainer: Debian VoIP Team - Thanks Francois for your work.
  - (Closes: #665789: O: sflphone -- SIP and IAX2 compatible VoIP phone)
* Added Build-Depends: libdbus-c++-bin
* Add gcc47-fixes.patch

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tools/sippxml/account_uas_recv_transfered.xml
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<?xml version="1.0" encoding="ISO-8859-1" ?>
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<!DOCTYPE scenario SYSTEM "sipp.dtd">
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<!-- This program is free software; you can redistribute it and/or      -->
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<!-- modify it under the terms of the GNU General Public License as     -->
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<!-- published by the Free Software Foundation; either version 2 of the -->
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<!-- License, or (at your option) any later version.                    -->
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<!--                                                                    -->
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<!-- This program is distributed in the hope that it will be useful,    -->
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<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of     -->
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<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the      -->
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<!-- GNU General Public License for more details.                       -->
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<!--                                                                    -->
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<!-- You should have received a copy of the GNU General Public License  -->
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<!-- along with this program; if not, write to the                      -->
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<!-- Free Software Foundation, Inc.,                                    -->
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<!-- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             -->
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<scenario name="accountcall_client">
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  <recv request="INVITE">
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  </recv>
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  <send>
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   <![CDATA[
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     SIP/2.0 180 Ringing
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     [last_Via:]
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     [last_From:]
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     [last_To:];tag=[call_number]
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     [last_Call-ID:]
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     [last_CSeq:]
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     Contact: <sip:[local_ip]:[local_port];transport=[transport]>
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     Content-Length: 0
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   ]]>
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  </send>
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  <send retrans="500">
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   <![CDATA[
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     SIP/2.0 200 OK
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     [last_Via:]
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     [last_From:]
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     [last_To:];tag=[call_number]
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     [last_Call-ID:]
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     [last_CSeq:]
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     Contact: <sip:[local_ip]:[local_port];transport=[transport]>
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     Content-Type: application/sdp
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     Content-Length: [len]
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     v=0
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     o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
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     s=-
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     c=IN IP[media_ip_type] [media_ip]
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     t=0 0
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     m=audio [media_port] RTP/AVP 0
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     a=rtpmap:0 PCMU/8000
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   ]]>
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  </send>
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  <recv request="ACK">
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  </recv>
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  <recv request="INVITE">
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  </recv>
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  <pause milliseconds="200">
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  </recv>
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  <send retrans="500">
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   <![CDATA[
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     SIP/2.0 200 OK
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     [last_Via:]
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     [last_From:]
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     [last_To:];tag=[call_number]
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     [last_Call-ID:]
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     [last_CSeq:]
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     Contact: <sip:[local_ip]:[local_port];transport=[transport]>
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     Content-Type: application/sdp
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     Content-Length: [len]
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     v=0
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     o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
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     s=-
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     c=IN IP[media_ip_type] [media_ip]
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     t=0 0
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     m=audio [media_port] RTP/AVP 0
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     a=rtpmap:0 PCMU/8000
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   ]]>
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  </send>
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  <recv request="ACK">
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  </recv>
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  <!--
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  <recv request="BYE">
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  </recv>
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  <send retrans="500">
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   <![CDATA[
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     SIP/2.0 200 OK
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     [last_Via:]
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     [last_From:]
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     [last_To:];tag=[call_number]
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     [last_Call-ID:]
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     [last_CSeq:]
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     Contact: <sip:[local_ip]:[local_port];transport=[transport]>
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     Content-Type: application/sdp
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     Content-Length: [len]
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     v=0
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     o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
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     s=-
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     c=IN IP[media_ip_type] [media_ip]
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     t=0 0
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     m=audio [media_port] RTP/AVP 0
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     a=rtpmap:0 PCMU/8000
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   ]]>
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  </send>
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  -->
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</scenario>