[asterisk-users] SIP trunk and congestion handling

Wed Aug 21 02:39:33 CDT 2013

You got to set event off while connecting to AMI to get rid of AMI responses on each event. There are ways you can suppress the events

http://www.voip-info.org/wiki/view/Asterisk+manager+API

Ask your provider to send 180 instead of 183.

From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Mordechay Kaganer
Sent: Thursday, August 15, 2013 5:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP trunk and congestion handling

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<mailto: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<mailto: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> [mailto: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<mailto: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> [mailto: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.

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