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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>This is a little over my head, but the
message indicates that you don’t have a fully authorized connection. Can you
post the iax.conf snippets relevant to the call?<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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face="Times New Roman"><span style='font-size:12.0pt'>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>
asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]
<b><span style='font-weight:bold'>On Behalf Of </span></b>khalid touati<br>
<b><span style='font-weight:bold'>Sent:</span></b> Wednesday, May 05, 2010 8:36
AM<br>
<b><span style='font-weight:bold'>To:</span></b> <st1:PersonName w:st="on">Asterisk
Users Mailing List - Non-Commercial Discussion</st1:PersonName><br>
<b><span style='font-weight:bold'>Subject:</span></b> Re: [asterisk-users] Code
in extensions.conf to leave a voicemailin another PBX ?!</span></font><o:p></o:p></p>
</div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-bottom:12.0pt'><font size=3
face="Times New Roman"><span style='font-size:12.0pt'>Thank you Danny, but it
says in the link that it's an iptables issue, though i allowed everything on
this network interface and even stopped iptables but still i have this issue.<o:p></o:p></span></font></p>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>2010/5/4 Danny Nicholas <<a href="mailto:danny@debsinc.com">danny@debsinc.com</a>><o:p></o:p></span></font></p>
<div link=blue vlink=blue>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><font
size=2 color=navy face=Arial><span style='font-size:10.0pt;font-family:Arial;
color:navy'>See if this helps</span></font><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><font
size=2 color=navy face=Arial><span style='font-size:10.0pt;font-family:Arial;
color:navy'><a href="http://www.voipuser.org/forum_topic_3921.html"
target="_blank">http://www.voipuser.org/forum_topic_3921.html</a></span></font><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><font
size=2 color=navy face=Arial><span style='font-size:10.0pt;font-family:Arial;
color:navy'> </span></font><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><font
size=2 color=navy face=Arial><span style='font-size:10.0pt;font-family:Arial;
color:navy'> </span></font><o:p></o:p></p>
<div>
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face="Times New Roman"><span style='font-size:12.0pt'>
<hr size=2 width="100%" align=center>
</span></font></div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b><font
size=2 face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma;font-weight:
bold'>From:</span></font></b><font size=2 face=Tahoma><span style='font-size:
10.0pt;font-family:Tahoma'> <a
href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>]
<b><span style='font-weight:bold'>On Behalf Of </span></b>khalid touati<br>
<b><span style='font-weight:bold'>Sent:</span></b> Tuesday, May 04, 2010 11:35
AM<o:p></o:p></span></font></p>
<div>
<p class=MsoNormal><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma'><br>
<b><span style='font-weight:bold'>To:</span></b> <st1:PersonName w:st="on">Asterisk
Users Mailing List - Non-Commercial Discussion</st1:PersonName><o:p></o:p></span></font></p>
</div>
<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>Subject:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'> Re:
[asterisk-users] Code in extensions.conf to leave a voice mailin another PBX ?!</span></font><o:p></o:p></p>
</div>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><font
size=3 face="Times New Roman"><span style='font-size:12.0pt'> <o:p></o:p></span></font></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;margin-bottom:12.0pt'><font
size=3 face="Times New Roman"><span style='font-size:12.0pt'>Hi Guys,<br>
so when i dial from an asterisk 1.2 to asterisk 1.4 i get the following
warning:<br>
WARNING[640]: file.c:738 ast_readaudio_callback: Failed to write frame<br>
is anyone familiar with?<o:p></o:p></span></font></p>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><font
size=3 face="Times New Roman"><span style='font-size:12.0pt'>2010/4/29 khalid
touati <<a href="mailto:khalidtouati@gmail.com" target="_blank">khalidtouati@gmail.com</a>><o:p></o:p></span></font></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><font
size=3 face="Times New Roman"><span style='font-size:12.0pt'>Hi Guys,<br>
Danny: as i said from pbx1 (1.4) to pbx2 (1.2) it's working fine.<br>
Peder: i just didn't want to put a lot of lines, (by the way it's dialing
talking fine), but here you are:<br>
<br>
[macro-stdexten]<br>
<br>
exten =>
s,n,Dial(SIP/${ARG1}&IAX2/${ARG1}@${ARG1},20,tTrWw) ;Ring
phone for 20 seconds<o:p></o:p></span></font></p>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;margin-bottom:12.0pt'><font
size=3 face="Times New Roman"><span style='font-size:12.0pt'><br>
exten => s,n,Goto(s-${DIALSTATUS},1)<br>
<br>
exten => s-NOANSWER,1,Voicemail(u${ARG1})<br>
exten => s-NOANSWER,2,Goto(default,s,1)<br>
<br>
exten => s-BUSY,1,Voicemail(b${ARG1})<br>
exten => s-BUSY,2,Goto(default,s,1)<br>
<br>
exten => _s-.,1,Goto(s-NOANSWER,1)<br>
<br>
exten => a,1,VoicemailMain(${ARG1})<br>
<br>
<o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><font
size=3 face="Times New Roman"><span style='font-size:12.0pt'>2010/4/29 Peder
<<a href="mailto:peder@networkoblivion.com" target="_blank">peder@networkoblivion.com</a>><o:p></o:p></span></font></p>
<div>
<div>
<div link=blue vlink=purple>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><font
size=2 color="#1f497d" face="Times New Roman"><span style='font-size:11.0pt;
color:#1F497D'>In PBX1, where are you actually dialing the phone? The
first line of the macro just says “goto dialstatus” with no Dial statement.</span></font><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><font
size=2 color="#1f497d" face="Times New Roman"><span style='font-size:11.0pt;
color:#1F497D'> </span></font><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><font
size=2 color="#1f497d" face="Times New Roman"><span style='font-size:11.0pt;
color:#1F497D'> </span></font><o:p></o:p></p>
<div style='border:none;border-top:solid windowtext 1.0pt;padding:3.0pt 0in 0in 0in;
border-color:-moz-use-text-color'>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b><font
size=2 face="Times New Roman"><span style='font-size:10.0pt;font-weight:bold'>From:</span></font></b><font
size=2><span style='font-size:10.0pt'> <a
href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>]
<b><span style='font-weight:bold'>On Behalf Of </span></b>khalid touati</span></font><o:p></o:p></p>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><font
size=2 face="Times New Roman"><span style='font-size:10.0pt'><br>
<b><span style='font-weight:bold'>Sent:</span></b> Thursday, April 29, 2010
2:03 PM<br>
<b><span style='font-weight:bold'>To:</span></b> <st1:PersonName w:st="on">Asterisk
Users Mailing List - Non-Commercial Discussion</st1:PersonName></span></font><o:p></o:p></p>
</div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b><font
size=2 face="Times New Roman"><span style='font-size:10.0pt;font-weight:bold'>Subject:</span></font></b><font
size=2><span style='font-size:10.0pt'> [asterisk-users] Code in extensions.conf
to leave a voice mail in another PBX ?!</span></font><o:p></o:p></p>
</div>
<div>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><font
size=3 face="Times New Roman"><span style='font-size:12.0pt'> <o:p></o:p></span></font></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><font
size=3 face="Times New Roman"><span style='font-size:12.0pt'>Hi Guys,<br>
i spent some time to figure this out (since i love how dialplan is written) but
i decided to ask for your help guys.<br>
<br>
i have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1) to 1.2
(pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1 it just
hang up.<br>
<br>
in pbx2 extensions.conf:<br>
i am using: exten => 8029,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr)<br>
<br>
in pbx1, i have:<br>
exten => 8029,1,Macro(stdexten,8029)<br>
and in stdexten macro:<br>
<br>
exten => s,n,Goto(s-${DIALSTATUS},1)<br>
exten => s-NOANSWER,1,Voicemail(u${ARG1})<br>
exten => s-NOANSWER,2,Goto(default,s,1)<br>
<br>
exten => s-BUSY,1,Voicemail(b${ARG1})<br>
exten => s-BUSY,2,Goto(default,s,1)<br>
<br>
exten => _s-.,1,Goto(s-NOANSWER,1)<br>
exten => a,1,VoicemailMain(${ARG1})<br>
<br>
when calling from 8021(pbx2) to 8029(pbx1) i get on CLI pbx1:<br>
<br>
-- Executing [s@macro-stdexten:6] Goto("IAX2/pbx2-15464",
"s-NOANSWER|1") in new stack<br>
-- Goto (macro-stdexten,s-NOANSWER,1)<br>
-- Executing [s-NOANSWER@macro-stdexten:1]
VoiceMail("IAX2/pbx2-15464", "u8029") in new stack<br>
<b><span style='font-weight:bold'>[Apr 29 14:36:35] WARNING[7307]: file.c:738
ast_readaudio_callback: Failed to write frame</span></b><br>
-- <IAX2/pbx2-15464> Playing
'/var/spool/asterisk/voicemail/default/8029/unavail' (language 'en')<br>
== Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on
'IAX2/pbx2-15464' in macro 'stdexten'<br>
== Spawn extension (default, 8029, 1) exited non-zero on
'IAX2/pbx2-15464'<br>
-- Hungup 'IAX2/pbx2-15464'<br>
<br>
any other ideas how to be able to leave a voice mail from 1.2 to 1.4 or fix the
issue I'm having, thanks a lot! <br clear=all>
<br>
-- <br>
Abdullah<o:p></o:p></span></font></p>
</div>
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<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><font
size=3 face="Times New Roman"><span style='font-size:12.0pt'> <o:p></o:p></span></font></p>
</div>
</div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><font
size=3 face="Times New Roman"><span style='font-size:12.0pt'>--<o:p></o:p></span></font></p>
<div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><font
size=3 face="Times New Roman"><span style='font-size:12.0pt'><br>
_____________________________________________________________________<br>
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target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
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<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><font
size=3 color="#888888" face="Times New Roman"><span style='font-size:12.0pt;
color:#888888'><br>
<br clear=all>
<br>
-- <br>
Abdullah</span></font><o:p></o:p></p>
</div>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><font
size=3 face="Times New Roman"><span style='font-size:12.0pt'><br>
<br clear=all>
<br>
-- <br>
Abdullah<o:p></o:p></span></font></p>
</div>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><br>
--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com"
target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a
href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><br>
<br clear=all>
<br>
-- <br>
Abdullah<o:p></o:p></span></font></p>
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