<DIV>
<DIV>
<DIV>Steve and all,</DIV>
<DIV>Did you ever get resolution on the issue of the timeout and t_on_failure commands killing or routing PSTN calls to voicemail?</DIV>
<DIV> </DIV>
<DIV>I looked through the mail archives and it isn't clear what the resolution was. I'm running into the same problem. If I set t_on_failure to occur after a certain timeout, outgoing PSTN calls fail after that timeout as well.</DIV>
<DIV> </DIV>
<DIV>In fact, it seems that calls fail after the timeout even if t_on_failure isn't set.</DIV>
<DIV> </DIV>
<DIV>I've successfully gotten outgoing PSTN calls being handled by a different t_relay than incoming or internal network calls.</DIV>
<DIV> </DIV>
<DIV>What did you do to resolve the issue?</DIV>
<DIV> </DIV>
<DIV>My config.</DIV>
<DIV> </DIV>
<DIV>
<DIV>
<DIV>RedHat 9.0<BR>kernel 2.4.20-20smp</DIV>
<DIV>ser 0.8.11 (i386/linux)</DIV>
<DIV>main.c, v 1.162.2.5</DIV></DIV></DIV>
<DIV> </DIV>
<DIV>Also, when a Status: 486 (busy) is encountered on the recieving party side, or a timeout occurs due to fr_inv_timer, I'm getting this error in my log</DIV>
<DIV><FONT size=2>
<P>Sep 25 14:07:56 jiffypop /usr/local/sbin/ser[23164]: ERROR: t_newtran: transaction already in process 0x422c0b38</FONT></P></DIV>
<DIV>Anyone have any ideas on what the problem is?</DIV>
<DIV>ser.cft and ngrep output attached.</DIV>
<DIV> </DIV>
<DIV>Thanks,</DIV>
<DIV>G<BR></DIV>
<DIV> </DIV>
<DIV><BR><B><I>Steve Dolloff <sdolloff@noc.dls.net></I></B> wrote:</DIV>
<BLOCKQUOTE class=replbq style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid"><BR><BR>>> I did place this portion inside the myself check<BR>>>and it still tries to transfer to vm after the time expires. <BR><BR>>I'm puzzled -- did not you want to transfer to vm after the time<BR>expires?<BR><BR>I will try to make this clearer. I am behind an ATA with a SIP proxy of<BR>209.242.10.153. If I call someone else registered on my domain and they<BR>are not available, I want to go to voice mail. If I call 1-800-555-1212<BR>from my phone, I do not want my sip proxy to reroute the call to<BR>voicemail after 10 seconds if no one answers(or ever for that matter).<BR>Right now if I dial 18005551212 from my handset, I see the destination<BR>as sip:18005551212@209.242.10.153 on the server which matches to myself<BR>and ser tries to send it to voicemail. <BR><BR>Someone calling into the network is not a problem. They will never hit<BR>our server unless the
destination is local.<BR><BR>>>This is the part that I really need help with! When the call timer<BR>>>fails, the call goes to the route[1]. How do I get it into voice mail<BR>>>from that point?<BR><BR>>See bellow, I think that should work.<BR><BR>This is what I had originally, and I get the following syslog.<BR><BR><BR>Sep 10 16:36:36 voip2 ser: parse error (127,37-38): Command cannot be<BR>used in the block<BR>Sep 10 16:36:36 voip2 ser: ERROR: bad config file (1 errors)<BR>Sep 10 16:36:36 voip2 ser: ser startup failed<BR><BR>Is says that vm is not valid in the block. According the admin guide,<BR>only certain commands can be used within a failure block. I assume that<BR>is the problem here. If not, please let me know as this is exactly what<BR>I want to do.<BR><BR>>THE SAME STUFF LIKE ABOVE, YOU DON'T WANT TO t_relay ANYTHING<BR><BR>>if(!vm("/tmp/am_fifo","voicemail")){<BR>> t_reply("500", "SEMS<BR>>error");<BR>> };<BR>>
break;<BR><BR><BR>_______________________________________________<BR>Serusers mailing list<BR>serusers@lists.iptel.org<BR>http://lists.iptel.org/mailman/listinfo/serusers</BLOCKQUOTE></DIV></DIV><p><hr SIZE=1>
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