[asterisk-users] SIP trunk and congestion handling

[Mordechay Kaganer]

B.H.

While dialing out i get a lot of AMI responses like this:

Event: Hangup
Privilege: call,all
Channel: SIP/TRK012-000336b0
Uniqueid: S5-1376567634.218719
CallerIDNum: XXXXXXXXX
CallerIDName: YYYYYYYYYY
ConnectedLineNum: XXXXXXXXX
ConnectedLineName: YYYYYYYYYY
*Cause: 19*
*Cause-txt: User alerting, no answer*

Event: OriginateResponse
Privilege: call,all
ActionID: 249867518_255525#YD_UFOzWQx30Wm6PM3USxGE
Response: Failure
Channel: SIP/TRK012/YYYYYYYYYY
Context: YemotDialer_Bridge
Exten: s
*Reason: 8*
Uniqueid: <null>
CallerIDNum: XXXXXXXXX
CallerIDName: YYYYYYYYYY

As mentioned in the previous mails, SIP response code is 480. I would
expect to get reason 3 not 8. Reason 8 is confusing my dialer software so
it wants to redial the number.

I use Asterisk 1.8.22. Is this a bug in asterisk or is a problem with my
SIP trunk provider?



On Wed, Aug 14, 2013 at 9:00 AM, Mordechay Kaganer <mkaganer at gmail.com>wrote:

> 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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>> _____________________________________________________________________
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>> New to Asterisk? Join us for a live introductory webinar every Thurs:
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>>
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>>
>


-- 
משיח NOW!
Moshiach is coming very soon, prepare yourself!
יחי אדוננו מורינו ורבינו מלך המשיח לעולם ועד!
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