Hi Guys,<br>first of all, thanks Danny for your support trying to help is a big help itself.<br>so the thing is:<br>from pbx1 to pbx2 which was able to leave VM, it was set up like this:<br>exten => 8021,1,Dial(IAX2/pbx2/${EXTEN},30,tTWwr)<br>
<br>but from pbx2 to pbx1 which was not able to leave VM, it was setup like this:<br>exten => 8093,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr)<br><br>that seems to me suly, but though i wnet ahead and modified the only difference which is the ring time from 20 to 30, and IT WORKED!!!<br>
i wasted some time going over values, and it seems like it's working for 21 but not for 20, maybe a pro can give us precise explanation, but at least i can leave a VM now :)! <br><br><div class="gmail_quote">2010/5/5 Danny Nicholas <span dir="ltr"><<a href="mailto:danny@debsinc.com">danny@debsinc.com</a>></span><br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div link="blue" vlink="blue" lang="EN-US">
<div>
<p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; 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?</span></font></p>
<p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; font-family: Arial; color: navy;"> </span></font></p>
<div>
<div class="MsoNormal" style="text-align: center;" align="center"><font size="3" face="Times New Roman"><span style="font-size: 12pt;">
<hr size="2" width="100%" align="center">
</span></font></div>
<p class="MsoNormal"><b><font size="2" face="Tahoma"><span style="font-size: 10pt; font-family: Tahoma; font-weight: bold;">From:</span></font></b><font size="2" face="Tahoma"><span style="font-size: 10pt; 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> Wednesday, May 05, 2010 8:36
AM<div><div></div><div class="h5"><br>
<b><span style="font-weight: bold;">To:</span></b> Asterisk
Users Mailing List - Non-Commercial Discussion<br>
<b><span style="font-weight: bold;">Subject:</span></b> Re: [asterisk-users] Code
in extensions.conf to leave a voicemailin another PBX ?!</div></div></span></font></p>
</div><div><div></div><div class="h5">
<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size: 12pt;"> </span></font></p>
<p class="MsoNormal" style="margin-bottom: 12pt;"><font size="3" face="Times New Roman"><span style="font-size: 12pt;">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.</span></font></p>
<div>
<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size: 12pt;">2010/5/4 Danny Nicholas <<a href="mailto:danny@debsinc.com" target="_blank">danny@debsinc.com</a>></span></font></p>
<div link="blue" vlink="blue">
<div>
<p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; font-family: Arial; color: navy;">See if this helps</span></font></p>
<p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; 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></p>
<p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; font-family: Arial; color: navy;"> </span></font></p>
<p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; font-family: Arial; color: navy;"> </span></font></p>
<div>
<div class="MsoNormal" style="text-align: center;" align="center"><font size="3" face="Times New Roman"><span style="font-size: 12pt;">
<hr size="2" width="100%" align="center">
</span></font></div>
<p class="MsoNormal"><b><font size="2" face="Tahoma"><span style="font-size: 10pt; font-family: Tahoma; font-weight: bold;">From:</span></font></b><font size="2" face="Tahoma"><span style="font-size: 10pt; 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</span></font></p>
<div>
<p class="MsoNormal"><font size="2" face="Tahoma"><span style="font-size: 10pt; font-family: Tahoma;"><br>
<b><span style="font-weight: bold;">To:</span></b> Asterisk
Users Mailing List - Non-Commercial Discussion</span></font></p>
</div>
<p class="MsoNormal"><b><font size="2" face="Tahoma"><span style="font-size: 10pt; font-family: Tahoma; font-weight: bold;">Subject:</span></font></b><font size="2" face="Tahoma"><span style="font-size: 10pt; font-family: Tahoma;"> Re:
[asterisk-users] Code in extensions.conf to leave a voice mailin another PBX ?!</span></font></p>
</div>
<div>
<div>
<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size: 12pt;"> </span></font></p>
<p class="MsoNormal" style="margin-bottom: 12pt;"><font size="3" face="Times New Roman"><span style="font-size: 12pt;">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?</span></font></p>
<div>
<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size: 12pt;">2010/4/29 khalid
touati <<a href="mailto:khalidtouati@gmail.com" target="_blank">khalidtouati@gmail.com</a>></span></font></p>
<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size: 12pt;">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}@${ARG1},20,tTrWw) ;Ring
phone for 20 seconds</span></font></p>
<div>
<p class="MsoNormal" style="margin-bottom: 12pt;"><font size="3" face="Times New Roman"><span style="font-size: 12pt;"><br>
exten => s,n,Goto(s-${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>
</span></font></p>
</div>
<div>
<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size: 12pt;">2010/4/29 Peder
<<a href="mailto:peder@networkoblivion.com" target="_blank">peder@networkoblivion.com</a>></span></font></p>
<div>
<div>
<div link="blue" vlink="purple">
<div>
<p class="MsoNormal"><font size="2" color="#1f497d" face="Times New Roman"><span style="font-size: 11pt; color: rgb(31, 73, 125);">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></p>
<p class="MsoNormal"><font size="2" color="#1f497d" face="Times New Roman"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></font></p>
<p class="MsoNormal"><font size="2" color="#1f497d" face="Times New Roman"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></font></p>
<div style="border-style: solid none none; border-color: -moz-use-text-color; border-width: 1pt medium medium; padding: 3pt 0in 0in;">
<p class="MsoNormal"><b><font size="2" face="Times New Roman"><span style="font-size: 10pt; font-weight: bold;">From:</span></font></b><font size="2"><span style="font-size: 10pt;"> <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></p>
<div>
<p class="MsoNormal"><font size="2" face="Times New Roman"><span style="font-size: 10pt;"><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> Asterisk
Users Mailing List - Non-Commercial Discussion</span></font></p>
</div>
<p class="MsoNormal"><b><font size="2" face="Times New Roman"><span style="font-size: 10pt; font-weight: bold;">Subject:</span></font></b><font size="2"><span style="font-size: 10pt;"> [asterisk-users] Code in extensions.conf
to leave a voice mail in another PBX ?!</span></font></p>
</div>
<div>
<div>
<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size: 12pt;"> </span></font></p>
<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size: 12pt;">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</span></font></p>
</div>
</div>
</div>
</div>
<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size: 12pt;"> </span></font></p>
</div>
</div>
<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size: 12pt;">--</span></font></p>
<div>
<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size: 12pt;"><br>
_____________________________________________________________________<br>
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<p class="MsoNormal"><font size="3" color="#888888" face="Times New Roman"><span style="font-size: 12pt; color: rgb(136, 136, 136);"><br>
<br clear="all">
<br>
-- <br>
Abdullah</span></font></p>
</div>
<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size: 12pt;"><br>
<br clear="all">
<br>
-- <br>
Abdullah</span></font></p>
</div>
</div>
</div>
</div>
<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size: 12pt;"><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>
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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: 12pt;"><br>
<br clear="all">
<br>
-- <br>
Abdullah</span></font></p>
</div></div></div>
</div>
<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>
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To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br><br clear="all"><br>-- <br>Abdullah<br>