[asterisk-users] SIP trunk and congestion handling

[Mordechay Kaganer]

B.H.

But if the final response is 480 doesn't it mean that the call was placed
but there was no reply?
 On Aug 13, 2013 10:30 PM, "Shishir Pokharel" <Shishir.Pokharel at on24.com>
wrote:

>   *21.1.5* <http://tools.ietf.org/html/rfc3261#section-21.1.5>* 183
> Session Progress*
>
> ** **
>
> ** **
>
>    The 183 (Session Progress) response is used to convey information****
>
>    about the progress of the call that is not otherwise classified.  The**
> **
>
>    Reason-Phrase, header fields, or message body MAY be used to convey****
>
>    more details about the call progress.****
>
> * *
> 21.1.2 <http://tools.ietf.org/html/rfc3261#section-21.1.2> 180 Ringing****
>
> ** **
>
> ** **
>
>    The UA receiving the INVITE is trying to alert the user.  This****
>
>    response MAY be used to initiate local ringback.****
>
> * *
>
> http://tools.ietf.org/html/rfc3261#section-21.1.2**
>
> ** **
>
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Mordechay Kaganer
> *Sent:* Tuesday, August 13, 2013 10:55 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] SIP trunk and congestion handling****
>
> ** **
>
> B.H.****
>
> Asterisk 1.8.22****
>
> Thanks****
>
> On Aug 12, 2013 8:05 PM, "Shishir Pokharel" <Shishir.Pokharel at on24.com>
> wrote:****
>
> Which version of asterisk are you using ? ****
>
>  ****
>
>  ****
>
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Mordechay Kaganer
> *Sent:* Sunday, August 11, 2013 8:59 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] SIP trunk and congestion handling****
>
>  ****
>
> B.H.****
>
>  ****
>
> Hello, all. We have a dialer software that runs outgoing telephony
> campaigns. We have been using it successfully with PRI cards, now we're
> evaluating it's use also with a SIP trunk. Most of the things run perfectly
> good without a need to change anything except for dial string, but there's
> some strange problem with asterisk interpreting SIP result codes. ****
>
>  ****
>
> Our software is written in Java using asterisk-java library. It is using
> Asterisk's reason code from OriginateResponseEvent to determine if it
> should redial the number. Our consideration is that if Asterisk returns
> reason code 8 (Congestion) this means that the call has never actually
> reached the destination number, and it's OK to try to redial again.****
>
>  ****
>
> But with SIP trunk, many times i can see a really strange sequence of
> events:****
>
>  ****
>
> After INVITE i get the following responses (example from a real
> conversation)****
>
> [17:01:40] SIP/2.0 100 Trying****
>
> [17:01:40] SIP/2.0 183 Session Progress****
>
> [17:01:51] SIP/2.0 480 Temporarily not available****
>
>  ****
>
> As far as i understand, this means that the remote phone was ringing for
> 10 seconds and then the call failed due to a timeout. As far as i
> understand, i'm supposed to get reason code 3, but actually the java
> application gets OriginateResponseEvent with failure reason code 8.****
>
>  ****
>
> This behavior is hard to reproduce. I was trying with my own phone number
> and then i get the expected reason code 3, but i constantly get this
> situation running our customer's campaigns.****
>
>  ****
>
>  ****
>
> -- ****
>
> משיח NOW!****
>
> Moshiach is coming very soon, prepare yourself!****
>
> יחי אדוננו מורינו ורבינו מלך המשיח לעולם ועד!****
>
>
> --
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