FW: [Serusers] ser and * voicemail

Mark Aiken aiken.mark at gmail.com
Sat Oct 1 05:59:27 CEST 2005


Sorry about that error. Your quite right the 487 are always after the 200
response to a CANCEL.

No idea what sort of strange problem your having there. I would have guessed
that SER is timing out due to the INVITE timer and your ser.cfg is not
properly sending that to voicemail.

We always use a feature server, of one form or another, for that sort of
thing but I've seen ser configs at onsip.org <http://onsip.org> that do what
your trying to do with SER.

Have a look at www.onsip.org <http://www.onsip.org>

Mark

On 9/30/05, Rick Thompson <rthompson at vir2com.com> wrote:
>
>    I have tethereal on my ser server but didn't see a 408.
>   ------------------------------
>
> *From:* Leon Sun [mailto:leon.sun at keywestcommunications.com]
> *Sent:* Friday, September 30, 2005 5:41 PM
> *To:* 'Leon Sun'; rthompson at vir2com.com
> *Cc:* 'Mark Aiken'
> *Subject:* RE: [Serusers] ser and * voicemail
>
>  Hi, guys,
>
>  I used Ethereal to trace SIP and found code 408 when it's time out. I
> believe we need focus 408 rather than 487.
>
>  Sorry Rick, for last e-mail.
>
>  Regards
>
>  Leon
>
>   ------------------------------
>
> *From:* Leon Sun [mailto:leon.sun at keywestcommunications.com<http://nications.com>]
>
> *Sent:* Friday, September 30, 2005 2:33 PM
> *To:* 'rthompson at vir2com.com'
> *Cc:* 'Mark Aiken'
> *Subject:* RE: [Serusers] ser and * voicemail
>
>  From Mark:
>
>  SER normally sends a 487 when the INVITE timer runs out so you would need
> to trigger the voicemail on that event. We have our feature server handing
> timeouts not SER (our SER timeout is set to a very large value) so I'm not
> sure the best way to proceed.
>
> Does SER call the failure_route for a locally generated 487 timeout? If so
> then rather than the 'break' you have there now just forward to vm. I would
> set a different t_on_failure instead of reusing "1" though, so you dont keep
> forwarding if the vm fails.
>
> Mark
>
>    ------------------------------
>
> *From:* Rick Thompson [mailto:rthompson at vir2com.com]
> *Sent:* Friday, September 30, 2005 2:27 PM
> *To:* 'Leon Sun'
> *Subject:* RE: [Serusers] ser and * voicemail
>
>  Leon
>
>  Thanks for the reply to my post
>
> I see what you are doing here, but that part has been taken care of. When
> I get a failure on "INVITE" of 486, 404, 408 or 480 I look in the mysql
> database and find the voicemail box for that uri and replace the sip contact
> then prefix it with a "V" and relay it to asterisk. Over on the asterisk
> side, I route the call to the voicemail context and delete the "V" and send
> it to voicemail. All this works except when ser times out on the "INVITE"
> and for some reason it doesn't route to asterisk. That's the problem I'm
> having.
>
>  Rick
>
>      ------------------------------
>
> *From:* Leon Sun [mailto:leon.sun at keywestcommunications.com<http://nications.com>]
>
> *Sent:* Friday, September 30, 2005 4:36 PM
> *To:* rthompson at vir2com.com; serusers at lists.iptel.org
> *Subject:* RE: [Serusers] ser and * voicemail
>
>  Rick,
>
>  I had same problem before and I gave it up since I didn't get any answer
> from list. I am using another way(tricky but working) to do voice mail. Hope
> it can help you if you can't fix it.
>
>  1. Check location in your routing parts before relay, if not, forward to
> Asterisk.
>
> 2. set up unconditional forward in ATA as 00 + ATA'DID. Make a route in
> SER and point 00* to Asterisk.
>
> 3. Strip 00 in Asterisk and send it to voicemail2(${EXTEN})
>
>  Regards
>
>  Leon Sun
>
>   ------------------------------
>
> *From:* serusers-bounces at iptel.org [mailto:serusers-bounces at lists.iptel.org] *On
> Behalf Of *Rick Thompson
> *Sent:* Friday, September 30, 2005 12:58 PM
> *To:* serusers at lists.iptel.org
> *Subject:* [Serusers] ser and * voicemail
>
>  Hi All
>
>  I'm working with a ser script, written by someone that's gone now, that
> routes inbound calls to an asterisk server for voicemail. The
> failure_route[1], sends calls to asterisk and the IVR plays if the ua is
> unreachable (not in location) "404", "408" or the ua is busy "486" but it
> doesn't when the inv time exceeds 30 sec (rings for 30 sec or more). The
> call just stops ringing and 10 sec later gets a fast busy. Any ideas from
> anyone would be greatly appreciated. Here is the code I'm working with.
>
>   failure_route[1] {
>
> xlog("L_ALERT", "%Tf %mf ****** Failure Route 1: <%rm> <%rr> <%rs>\n");
>
> if(t_check_status("487")) {
>
> break;
>
> };
>
>  if(method=="INVITE" && (t_check_status("486|408|404|480"))) {
>
> if(avp_db_load("$ruri", "s:mailbox"))
>
> avp_pushto("$ruri/username", "s:mailbox");
>
> prefix("V");
>
> rewritehostport("A.B.C.D:5060");
>
> append_branch();
>
> xlog("L_ALERT", "****** Transfering to Voicemail\n");
>
> t_on_reply("1");
>
> t_relay();
>
> };
>
> }
>
> Thanks
>
>  Rick
>
>
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