Inappropriate voice prompts

devplan · **Joined:** Sat Apr 07, 2007 5:01 pm **Posts:** 48

I am running YellowJacket on TB 1.2.24 and I am trying to make some sense out of various voice prompts:

For instance, on some providers, when I am out of credits, I get the "prepaid-isbusy.gsm" prompt while on some others, I get the "prepaid-noanswer.gsm" prompt.

When I dial a vacant number (not existing number) most provider continue ringing and after some time I get the "prepaid-isbusy.gsm" prompt.

Some few, ring only a couple of times on a vacant number, and then I get the prepaid-dest-unreachable.gsm prompt.

In all above instances, the prompts are inappropriate to the user.
However - as far I see from the logs - it depends how the provider returns the "fail-event".

What can I do from my side to bring some order into this jungle?

rgds, devplan

middletn · **Posted:** Sat Jan 05, 2008 9:23 pm

Have a look at this

http://forum.asterisk2billing.org/viewtopic.php?t=2982

Might Help, or not as the case ,may be
regards

devplan · **Joined:** Sat Apr 07, 2007 5:01 pm **Posts:** 48

Hi middletn

Yes, that helped a bit.
Only thing is that on some numbers, A2B would report "The number you have dialed is currently unavailable" while when tested with PSTN phones on POTS lines, the number was valid and ringing and was picked-up.

I guess that some call-centers must have a big problem with these inappropriate voice prompts, because numbers get reported and archived as "vacant" when they aren't.

I looked a bit into this and found that the dial status "congestion" is often too general.

sample:

Code:

Jan 7 00:37:49 VERBOSE[14823] logger.c: -- SIP/myprovider-09f673e8 is circuit-busy
Jan 7 00:37:49 DEBUG[14823] chan_sip.c: update_call_counter(001245678000) - decrement call limit counter
Jan 7 00:37:49 VERBOSE[14823] logger.c: == Everyone is busy/congested at this time (1:0/1/0)
Jan 7 00:37:49 DEBUG[14823] app_dial.c: Exiting with DIALSTATUS=CONGESTION.

This can make voip lines compared to PSTN unreliable.

What are good guide-lines and configs to remedy this?

rgds, devplan

devplan · **Joined:** Sat Apr 07, 2007 5:01 pm **Posts:** 48

It can go even more astray:

My sip debug shows: (sample)

Quote:

Jan 8 20:02:43 VERBOSE[15390] logger.c: -- Got SIP response 402 "Payment Required" back from 80.180.180.180
Jan 8 20:02:43 DEBUG[5129] chan_sip.c: update_call_counter(001234567878) - decrement call limit counter
Jan 8 20:02:43 VERBOSE[5129] logger.c: == No one is available to answer at this time (1:0/0/0)
Jan 8 20:02:43 DEBUG[5129] app_dial.c: Exiting with DIALSTATUS=NOANSWER.

And I am getting the voice prompt: "The number is not answering". :shock:

How come, that the SIP response 402 gets translated into DIALSTATUS=NOANSWER?

And, does anybody know, which file do I need to edit (and what do I need to change) to don't get a DIALSTAUS=NOANSWER but instead something which I can trigger a prompt which says for example "No more funds at provider".

Ideally there would be a patch or add-on which says the correct reported cause and not something else.

rgds, devplan

stavros · **Posted:** Sun Jan 13, 2008 10:44 pm

Quote:

How come, that the SIP response 402 gets translated into DIALSTATUS=NOANSWER?

From voip-info on DIALSTATUS:

Quote:

CONGESTION "Congestion" is a bit misleading. Unfortunately at this time (2007-05-17), Dial() returns DIALSTATUS=CONGESTION for pretty much every call setup problem. The reasoning behind this is that Dial() can be used for multiple protocols (Zap,SIP,IAX etc.) and so it is limited to the lowest common denominator and is unable to return the protocol specific information (i.e. SIP 404 response) .

Also see Asterisk bug #9743 and RFC 3398.

Quote:

And, does anybody know, which file do I need to edit (and what do I need to change) to don't get a DIALSTAUS=NOANSWER but instead something which I can trigger a prompt which says for example "No more funds at provider".

You'd need to modify A2B to make good decisions based on the value of HANGUPCAUSE in addition to DIALSTATUS. I believe this would be quite an appreciable amount of work to do correctly.

xrg · **Posted:** Sun Jan 13, 2008 11:35 pm

stavros wrote:

You'd need to modify A2B to make good decisions based on the value of HANGUPCAUSE in addition to DIALSTATUS. I believe this would be quite an appreciable amount of work to do correctly.

Been there, done that..

devplan · **Joined:** Sat Apr 07, 2007 5:01 pm **Posts:** 48

Hi stavros & xrg

Yes, I had looked up DIALSTATUS at voip-info before posting and hoped that since that time, some fix had been made.
And I looked also into the HANGUPCAUSE para.
Looks like that this particulary issue hasn't been really addressed yet in *.

Another post reply I got concerning this point is:

Quote:

After googling a bit I see what you're saying now. * does a lossy translation of SIP codes and makes them available in the HANGUPCAUSE variable. That blows, and I see that this isn't the first person to run up against this particular issue.

The developers don't want to change it because they want the dial plan to be technology agnostic. Great.

Quote:

Been there, done that..

xrg, you say you've done a patch to use HANGUPCAUSE?
Can I implement this with YellowJacket?

rgds, devplan

xrg · **Posted:** Mon Jan 14, 2008 6:04 pm

First of all, I do get the reason DIALSTATUS is preferred: every channel uses a different set of responses. In fact, the most global ones are the ISDN codes, as in HANGUPCAUSE. Having said that, every provider (= softswitch of a provider) uses a different subset, too. So, it is a mess!

The code is not just a patch and certainly won't apply to 1.3, 1.4. It will be a brand new release of a2billing.

devplan · **Joined:** Sat Apr 07, 2007 5:01 pm **Posts:** 48

Isn't there a file I can modify so that a SIP response of, for example: 503
doesn't get "brushed off" as "All circuits are busy now", but congestion tone.

Code:

Jan 30 21:35:48 VERBOSE[2648] logger.c: -- Got SIP response 503 "Service Unavailable" back from 211.11.111.111
Jan 30 21:35:48 VERBOSE[3057] logger.c: -- SIP/Myprovider-08704c18 is circuit-busy
Jan 30 21:35:48 DEBUG[3057] chan_sip.c: update_call_counter(004533201000) - decrement call limit counter
Jan 30 21:35:48 VERBOSE[3057] logger.c: == Everyone is busy/congested at this time (1:0/1/0)
Jan 30 21:35:48 DEBUG[3057] app_dial.c: Exiting with DIALSTATUS=CONGESTION.
Jan 30 21:35:48 VERBOSE[3057] logger.c: -- Executing Goto("SIP/909-086a7320", "s-CONGESTION|1") in new stack
Jan 30 21:35:48 VERBOSE[3057] logger.c: -- Goto (macro-dialout-trunk,s-CONGESTION,1)

(I did the modification from middletn of the Class.RateEngine.php file, but I still get the "All circuits are busy now".)

The above SIP response 503 I got back from a provider when I called a vacant number.

rgds, devplan

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Inappropriate voice prompts

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