<div class="gmail_quote">On Wed, Dec 28, 2011 at 3:32 PM, Gilberto Verástegui <span dir="ltr"><<a href="mailto:gilbertovm@ti-m.com.mx">gilbertovm@ti-m.com.mx</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Calls to long distance get disconnected before answer.<br>
Telco: Alestra<br>
Country: Mexico<br>
System: Elastix 2.2<br>
Digital Card: Digium TE122<br>
<br>
Log:<br>
<br>
[Dec 28 14:37:44] VERBOSE[4586] pbx.c: -- Executing <a href="tel:%5B%2B525552622900" value="+525552622900">[+525552622900</a>@default:1] Set("SIP/OCS_TRUNK-000001bf", "EXT=015552622900") in new stack<br>
[Dec 28 14:37:44] VERBOSE[4586] pbx.c: -- Executing <a href="tel:%5B%2B525552622900" value="+525552622900">[+525552622900</a>@default:2] Dial("SIP/OCS_TRUNK-000001bf", "DAHDI/g1/015552622900,60") in new stack<br>
[Dec 28 14:37:44] VERBOSE[4586] app_dial.c: -- Called DAHDI/g1/015552622900<br>
[Dec 28 14:37:44] DEBUG[4586] chan_dahdi.c: bits changed in chan 1<br>
[Dec 28 14:37:53] DEBUG[4586] chan_dahdi.c: disconnecting MFC/R2 call on chan 1<br>
[Dec 28 14:37:53] DEBUG[4586] chan_dahdi.c: ast cause 0 resulted in openr2 cause 6/Normal Clearing<br>
[Dec 28 14:37:53] VERBOSE[4586] chan_dahdi.c: -- Hungup 'DAHDI/1-1'<br>
[Dec 28 14:37:53] VERBOSE[4586] pbx.c: == Spawn extension (default, <a href="tel:%2B525552622900" value="+525552622900">+525552622900</a>, 2) exited non-zero on 'SIP/OCS_TRUNK-000001bf'<br>
[Dec 28 14:37:53] VERBOSE[9190] chan_dahdi.c: MFC/R2 call end on channel 1<br>
<br>
Found this email list, but I think is too old.<br>
<br>
<a href="http://www.mail-archive.com/asterisk-users@lists.digium.com/msg205765.html" target="_blank">http://www.mail-archive.com/asterisk-users@lists.digium.com/msg205765.html</a><br>
<br>
</blockquote><div><br></div><div>You would be better off asking this questions in asterisk-r2 mailing list.</div><div><br></div><div>I will answer the same way that I answered back then. You need to enable protocol debugging. Without protocol debugging there is no way to tell what is happening to the call.</div>
<div><br></div><div>Read the sample chan_dahdi.conf included with Asterisk and search for mfcr2 logging options.</div><div><br></div><div>Having said that, it is possible in international calls you need to specify a different caller category.</div>
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